Revert "materialsystem/shaderapidx9: restore shaders from source-sdk-2013"

This reverts commit cca56d5a6f.

appframework/sdlmgr.cpp

@@ -63,10 +63,16 @@ static void *l_egl = NULL;
 static void *l_gles = NULL;
 
 typedef void *(*t_glGetProcAddress)( const char * );
-t_glGetProcAddress _glGetProcAddress;
-
 typedef EGLBoolean (*t_eglBindAPI)(EGLenum api);
+typedef EGLBoolean (*t_eglInitialize)(EGLDisplay display, EGLint *major, EGLint *minor);
+typedef EGLDisplay (*t_eglGetDisplay)(NativeDisplayType native_display);
+typedef char const *(*t_eglQueryString)(EGLDisplay display, EGLint name);
+
 t_eglBindAPI _eglBindAPI;
+t_glGetProcAddress _glGetProcAddress;
+t_eglInitialize _eglInitialize;
+t_eglGetDisplay _eglGetDisplay;
+t_eglQueryString _eglQueryString;
 #endif
 
 /*
@@ -594,11 +600,25 @@ InitReturnVal_t CSDLMgr::Init()
 	l_gles = dlopen("libGLESv3.so", RTLD_LAZY);
 
 	if( l_egl )
+	{
 		_glGetProcAddress = (t_glGetProcAddress)dlsym(l_egl, "eglGetProcAddress");
+	}
 
 	SET_GL_ATTR(SDL_GL_CONTEXT_PROFILE_MASK, SDL_GL_CONTEXT_PROFILE_ES);
 	SET_GL_ATTR(SDL_GL_CONTEXT_MAJOR_VERSION, 3);
 	SET_GL_ATTR(SDL_GL_CONTEXT_MINOR_VERSION, 0);
+
+	_eglInitialize = (t_eglInitialize)dlsym(l_egl, "eglInitialize");
+	_eglGetDisplay = (t_eglGetDisplay)dlsym(l_egl, "eglGetDisplay");
+	_eglQueryString = (t_eglQueryString)dlsym(l_egl, "eglQueryString");
+
+	if( _eglInitialize && _eglInitialize && _eglQueryString)
+	{
+		EGLDisplay display = _eglGetDisplay(EGL_DEFAULT_DISPLAY);
+		if( _eglInitialize(display, NULL, NULL) != -1 
+			&& strstr(_eglQueryString(display, EGL_EXTENSIONS) ,"EGL_KHR_gl_colorspace") )
+				SET_GL_ATTR(SDL_GL_FRAMEBUFFER_SRGB_CAPABLE, 1)
+	}
 #elif ANDROID
 	bool m_bOGL = false;
 

bitmap/imageformat.cpp

@@ -312,12 +312,17 @@ ImageFormat D3DFormatToImageFormat( D3DFORMAT format )
 
 	switch ( format )
 	{
-#if !defined( _X360 )
+#ifdef TOGLES
+	case D3DFMT_R8G8B8:
+		return IMAGE_FORMAT_RGB888;
+	case D3DFMT_A8R8G8B8:
+		return IMAGE_FORMAT_RGBA8888;
+#else
 	case D3DFMT_R8G8B8:
 		return IMAGE_FORMAT_BGR888;
-#endif
 	case D3DFMT_A8R8G8B8:
 		return IMAGE_FORMAT_BGRA8888;
+#endif
 	case D3DFMT_X8R8G8B8:
 		return IMAGE_FORMAT_BGRX8888;
 	case D3DFMT_R5G6B5:
@@ -426,6 +431,10 @@ D3DFORMAT ImageFormatToD3DFormat( ImageFormat format )
 #endif
 	case IMAGE_FORMAT_BGRA8888:
 		return D3DFMT_A8R8G8B8;
+	case IMAGE_FORMAT_RGB888:
+		return D3DFMT_R8G8B8;
+	case IMAGE_FORMAT_RGBA8888:
+		return D3DFMT_A8R8G8B8;
 	case IMAGE_FORMAT_BGRX8888:
 		return D3DFMT_X8R8G8B8;
 	case IMAGE_FORMAT_BGR565:

dedicated/wscript

@@ -13,7 +13,7 @@ def options(opt):
 
 def configure(conf):
 	conf.define('LAUNCHERONLY',1)
-#	conf.define('SUPPORT_PACKED_STORE',1)
+	conf.define('SUPPORT_PACKED_STORE',1)
 	conf.define('DEDICATED',1)
 
 def build(bld):
@@ -49,7 +49,7 @@ def build(bld):
 
 	defines = []
 
-	libs = ['tier0','tier1','tier2','tier3','vstdlib','steam_api','vpklib','appframework','mathlib', 'EDIT']
+	libs = ['tier0','vpklib','tier1','tier2','tier3','vstdlib','steam_api','appframework','mathlib', 'EDIT']
 
 	install_path = bld.env.LIBDIR
 

engine/audio/private/voice.cpp

@@ -37,7 +37,7 @@ void VoiceTweak_EndVoiceTweakMode();
 void EngineTool_OverrideSampleRate( int& rate );
 
 // A fallback codec that should be the most likely to work for local/offline use
-#define VOICE_FALLBACK_CODEC	"vaudio_celt"
+#define VOICE_FALLBACK_CODEC	"vaudio_opus"
 
 // Special entity index used for tweak mode.
 #define TWEAKMODE_ENTITYINDEX				-500
@@ -197,6 +197,9 @@ extern IVoiceRecord* CreateVoiceRecord_AudioQueue(int sampleRate);
 extern IVoiceRecord* CreateVoiceRecord_OpenAL(int sampleRate);
 #endif
 
+#ifdef USE_SDL
+extern IVoiceRecord *CreateVoiceRecord_SDL(int sampleRate);
+#endif
 
 static bool VoiceRecord_Start()
 {
@@ -583,12 +586,13 @@ bool Voice_Init( const char *pCodecName, int nSampleRate )
 
 	bool bSpeex = Q_stricmp( pCodecName, "vaudio_speex" ) == 0;
 	bool bCelt  = Q_stricmp( pCodecName, "vaudio_celt" )  == 0;
+	bool bOpus  = Q_stricmp( pCodecName, "vaudio_opus" )  == 0;
 	bool bSteam = Q_stricmp( pCodecName, "steam" )        == 0;
 	// Miles has not been in use for voice in a long long time.  Not worth the surface to support ancient demos that may
 	// use it (and probably do not work for other reasons)
 	// "vaudio_miles"
 
-	if ( !( bSpeex || bCelt || bSteam ) )
+	if ( !( bSpeex || bCelt || bOpus || bSteam ) )
 	{
 		Msg( "Voice_Init Failed: invalid voice codec %s.\n", pCodecName );
 		return false;
@@ -648,6 +652,8 @@ bool Voice_Init( const char *pCodecName, int nSampleRate )
 	}
 #elif defined( WIN32 )
 	g_pVoiceRecord = CreateVoiceRecord_DSound( Voice_SamplesPerSec() );
+#elif defined( USE_SDL )
+	g_pVoiceRecord = CreateVoiceRecord_SDL( Voice_SamplesPerSec() );
 #else
 	g_pVoiceRecord = CreateVoiceRecord_OpenAL( Voice_SamplesPerSec() );
 #endif
@@ -671,6 +677,12 @@ bool Voice_Init( const char *pCodecName, int nSampleRate )
 		CreateInterfaceFn createCodecFn = NULL;
 		g_hVoiceCodecDLL = FileSystem_LoadModule(pCodecName);
 
+		if( !g_hVoiceCodecDLL || (createCodecFn = Sys_GetFactory(g_hVoiceCodecDLL)) == NULL )
+		{
+			g_hVoiceCodecDLL = FileSystem_LoadModule( VOICE_FALLBACK_CODEC );
+			pCodecName = VOICE_FALLBACK_CODEC;
+		}
+
 		if ( !g_hVoiceCodecDLL || (createCodecFn = Sys_GetFactory(g_hVoiceCodecDLL)) == NULL ||
 		     (g_pEncodeCodec = (IVoiceCodec*)createCodecFn(pCodecName, NULL)) == NULL || !g_pEncodeCodec->Init( quality ) )
 		{
@@ -1075,7 +1087,7 @@ int Voice_GetCompressedData(char *pchDest, int nCount, bool bFinal)
 {
 	// Check g_bVoiceRecordStopping in case g_bUsingSteamVoice changes on us
 	// while waiting for the end of voice data.
-	if ( g_bUsingSteamVoice || g_bVoiceRecordStopping )
+	if ( g_bUsingSteamVoice && g_bVoiceRecordStopping )
 	{
 		uint32 cbCompressedWritten = 0;
 		uint32 cbUncompressedWritten = 0;

engine/audio/private/voice_record_sdl.cpp

@@ -0,0 +1,257 @@
+//========= Copyright 1996-2009, Valve Corporation, All rights reserved. ============//
+//
+// Purpose: 
+//
+// $NoKeywords: $
+//
+//=============================================================================//
+// This module implements the voice record and compression functions 
+
+//#include "audio_pch.h"
+//#include "voice.h"
+#include "tier0/platform.h"
+#include "ivoicerecord.h"
+#include "tier0/dbg.h"
+#include "tier0/threadtools.h"
+
+#include <assert.h>
+#include <SDL_audio.h>
+
+#define RECORDING_BUFFER_SECONDS 3
+#define SAMPLE_COUNT 2048
+
+// ------------------------------------------------------------------------------
+// VoiceRecord_SDL
+// ------------------------------------------------------------------------------
+
+struct AudioBuf
+{
+	int Read(char *out, int len)
+	{
+		int nAvalible = (size + (writePtr - readPtr)) % size;
+
+		if( nAvalible == 0 )
+			return 0;
+
+		if( len > nAvalible ) len = nAvalible;
+
+		int diff = (data + size) - readPtr;
+
+
+		if( len > diff )
+		{
+			memcpy(out, readPtr, diff );
+			memcpy(out+diff, data, len-diff );
+		} else memcpy(out, readPtr, len);
+
+		readPtr += len;
+
+		if( readPtr >= data + size )
+			readPtr -= size;
+
+		return len;
+	}
+
+	void Write(char *in, int len)
+	{
+		int diff = (data + size) - writePtr;
+
+		if( len > diff )
+		{
+			memcpy(writePtr, in, diff );
+			memcpy(data, in+diff, len-diff );
+		} else memcpy(writePtr, in, len);
+
+		writePtr += len;
+
+		if (writePtr >= (data + size))
+			writePtr -= size;
+	}
+
+	int size;
+	char *data;
+	char *readPtr;
+	char *writePtr;
+};
+
+class VoiceRecord_SDL : public IVoiceRecord
+{
+protected:
+	virtual				~VoiceRecord_SDL();
+public:
+	VoiceRecord_SDL();
+	virtual void		Release();
+	virtual bool		RecordStart();
+	virtual void		RecordStop();
+
+	// Initialize. The format of the data we expect from the provider is
+	// 8-bit signed mono at the specified sample rate.
+	virtual bool		Init(int sampleRate);
+	virtual void		Idle() {}; // Stub
+	void RenderBuffer( char *pszBuf, int size );
+
+	// Get the most recent N samples.
+	virtual int			GetRecordedData(short *pOut, int nSamplesWanted );
+
+	SDL_AudioSpec m_ReceivedRecordingSpec;
+	int m_BytesPerSample; // Да кому нужна эта ваша инкапсуляция?
+	int m_nSampleRate;
+private:
+	bool				InitalizeInterfaces();	// Initialize the openal capture buffers and other interfaces
+	void				ReleaseInterfaces();	// Release openal buffers and other interfaces
+	void				ClearInterfaces();				// Clear members.
+private:
+
+	SDL_AudioDeviceID m_Device;
+	AudioBuf m_AudioBuffer;
+};
+
+void audioRecordingCallback( void *userdata, uint8 *stream, int len )
+{
+	VoiceRecord_SDL *voice = (VoiceRecord_SDL*)userdata;
+	voice->RenderBuffer( stream, len );
+}
+
+VoiceRecord_SDL::VoiceRecord_SDL() :
+m_nSampleRate( 0 ) ,m_Device( 0 )
+{
+	m_AudioBuffer.data = NULL;
+	m_AudioBuffer.readPtr = NULL;
+	m_AudioBuffer.writePtr = NULL;
+
+	ClearInterfaces();
+}
+
+VoiceRecord_SDL::~VoiceRecord_SDL()
+{
+	ReleaseInterfaces();
+	ClearInterfaces();
+}
+
+void VoiceRecord_SDL::Release()
+{
+	ReleaseInterfaces();
+	ClearInterfaces();
+	delete this;
+}
+
+bool VoiceRecord_SDL::RecordStart()
+{
+	if ( !m_Device )
+		InitalizeInterfaces();
+
+	if ( !m_Device )
+		return false;
+
+	SDL_PauseAudioDevice( m_Device, SDL_FALSE );
+
+	return true;
+}
+
+
+void VoiceRecord_SDL::RecordStop()
+{
+	// Stop capturing.
+	if ( m_Device )
+		SDL_PauseAudioDevice( m_Device, SDL_TRUE );
+
+	// Release the capture buffer interface and any other resources that are no
+	// longer needed
+	ReleaseInterfaces();
+}
+
+bool VoiceRecord_SDL::InitalizeInterfaces()
+{
+	//Default audio spec
+	SDL_AudioSpec desiredRecordingSpec;
+	SDL_zero(desiredRecordingSpec);
+	desiredRecordingSpec.freq = m_nSampleRate;
+	desiredRecordingSpec.format = AUDIO_S16;
+	desiredRecordingSpec.channels = 1;
+	desiredRecordingSpec.samples = SAMPLE_COUNT;
+	desiredRecordingSpec.callback = audioRecordingCallback;
+	desiredRecordingSpec.userdata = (void*)this;
+
+	//Open recording device
+	m_Device = SDL_OpenAudioDevice( NULL, SDL_TRUE, &desiredRecordingSpec, &m_ReceivedRecordingSpec, 0 );
+
+	if( m_Device != 0 )
+	{
+		//Calculate per sample bytes
+		m_BytesPerSample = m_ReceivedRecordingSpec.channels * ( SDL_AUDIO_BITSIZE( m_ReceivedRecordingSpec.format ) / 8 );
+
+		//Calculate bytes per second
+		int bytesPerSecond = m_ReceivedRecordingSpec.freq * m_BytesPerSample;
+
+		//Allocate and initialize byte buffer
+		m_AudioBuffer.size = RECORDING_BUFFER_SECONDS * bytesPerSecond;
+
+		if( !m_AudioBuffer.data )
+			m_AudioBuffer.data = (char *)malloc( m_AudioBuffer.size );
+
+		m_AudioBuffer.readPtr = m_AudioBuffer.data;
+		m_AudioBuffer.writePtr = m_AudioBuffer.data + SAMPLE_COUNT*m_BytesPerSample*2;
+
+		memset( m_AudioBuffer.data, 0, m_AudioBuffer.size );
+
+		return true;
+	}
+	else
+		return false;
+}
+
+bool VoiceRecord_SDL::Init(int sampleRate)
+{
+	m_nSampleRate = sampleRate;
+	ReleaseInterfaces();
+
+	return true;
+}
+
+
+void VoiceRecord_SDL::ReleaseInterfaces()
+{
+	if( m_Device != 0 )
+		SDL_CloseAudioDevice( m_Device );
+
+	m_Device = 0;
+}
+
+
+void VoiceRecord_SDL::ClearInterfaces()
+{
+	if( m_AudioBuffer.data )
+	{
+		free( m_AudioBuffer.data );
+		m_AudioBuffer.data = NULL;
+		m_AudioBuffer.readPtr = NULL;
+		m_AudioBuffer.writePtr = NULL;
+	}
+	m_Device = 0;
+}
+
+void VoiceRecord_SDL::RenderBuffer( char *pszBuf, int size )
+{
+	m_AudioBuffer.Write( pszBuf, size );
+}
+
+int VoiceRecord_SDL::GetRecordedData(short *pOut, int nSamples )
+{
+	if ( !m_AudioBuffer.data || nSamples == 0 )
+		return 0;
+
+	int cbSamples = nSamples * m_BytesPerSample;
+
+	return m_AudioBuffer.Read( (char*)pOut, cbSamples )/m_BytesPerSample;
+}
+
+IVoiceRecord* CreateVoiceRecord_SDL(int sampleRate)
+{
+	VoiceRecord_SDL *pRecord = new VoiceRecord_SDL;
+	if ( pRecord && pRecord->Init(sampleRate) )
+		return pRecord;
+	else if( pRecord )
+		pRecord->Release();
+
+	return NULL;
+}

engine/baseclientstate.cpp

@@ -570,7 +570,7 @@ bool CBaseClientState::PrepareSteamConnectResponse( uint64 unGSSteamID, bool bGS
 		return true;
 	}
 
-#if !defined( NO_STEAM ) && !defined( SWDS  ) 
+#if 0 //!defined( NO_STEAM ) && !defined( SWDS  ) 
 	if ( !Steam3Client().SteamUser() )
 	{
 		COM_ExplainDisconnection( true, "#GameUI_ServerRequireSteam" );
@@ -578,14 +578,14 @@ bool CBaseClientState::PrepareSteamConnectResponse( uint64 unGSSteamID, bool bGS
 		return false;
 	}
 #endif
-	
+
 	netadr_t checkAdr = adr;
 	if ( adr.GetType() == NA_LOOPBACK || adr.IsLocalhost() )
 	{
 		checkAdr.SetIP( net_local_adr.GetIPHostByteOrder() );
 	}
 
-#ifndef SWDS
+#if 0 // #ifndef SWDS
 	// now append the steam3 cookie
 	char steam3Cookie[ STEAM_KEYSIZE ];
 	uint32 steam3CookieLen = 0;
@@ -936,6 +936,7 @@ bool CBaseClientState::ProcessConnectionlessPacket( netpacket_t *packet )
 								int authprotocol = msg.ReadLong();
 								uint64 unGSSteamID = 0;
 								bool bGSSecure = false;
+#if 0
 								if ( authprotocol == PROTOCOL_STEAM )
 								{
 									if ( msg.ReadShort() != 0 )
@@ -963,6 +964,7 @@ bool CBaseClientState::ProcessConnectionlessPacket( netpacket_t *packet )
 										return false;
 									}
 								}
+#endif
 								SendConnectPacket( challenge, authprotocol, unGSSteamID, bGSSecure );
 							}
 							break;

engine/baseserver.cpp

@@ -741,7 +741,7 @@ bool CBaseServer::ProcessConnectionlessPacket(netpacket_t * packet)
 //					break;
 // 				}
 
-				if ( authProtocol == PROTOCOL_STEAM )
+/*				if ( authProtocol == PROTOCOL_STEAM )
 				{
 					int keyLen = msg.ReadShort();
 					if ( keyLen < 0 || keyLen > sizeof(cdkey) )
@@ -753,7 +753,7 @@ bool CBaseServer::ProcessConnectionlessPacket(netpacket_t * packet)
 
 					ConnectClient( packet->from, protocol, challengeNr, clientChallenge, authProtocol, name, password, cdkey, keyLen );	// cd key is actually a raw encrypted key	
 				}
-				else
+				else*/
 				{
 					msg.ReadString( cdkey, sizeof(cdkey) );
 					ConnectClient( packet->from, protocol, challengeNr, clientChallenge, authProtocol, name, password, cdkey, strlen(cdkey) );
@@ -1434,11 +1434,13 @@ bool CBaseServer::CheckChallengeType( CBaseClient * client, int nNewUserID, neta
 		return false;
 	}
 
+#if 0
 	if ( ( nAuthProtocol == PROTOCOL_HASHEDCDKEY ) && (Q_strlen( pchLogonCookie ) <= 0 ||  Q_strlen(pchLogonCookie) != 32 ) )
 	{
 		RejectConnection( adr, clientChallenge, "#GameUI_ServerRejectInvalidCertLen" );
 		return false;
 	}
+#endif
 
 	Assert( !IsReplay() );
 
@@ -1470,23 +1472,24 @@ bool CBaseServer::CheckChallengeType( CBaseClient * client, int nNewUserID, neta
 		client->SetSteamID( CSteamID() ); // set an invalid SteamID
 
 		// Convert raw certificate back into data
-		if ( cbCookie <= 0 || cbCookie >= STEAM_KEYSIZE )
+/*		if ( cbCookie <= 0 || cbCookie >= STEAM_KEYSIZE )
 		{
 			RejectConnection( adr, clientChallenge, "#GameUI_ServerRejectInvalidSteamCertLen" );
 			return false;
-		}
+		}*/
 		netadr_t checkAdr = adr;
 		if ( adr.GetType() == NA_LOOPBACK || adr.IsLocalhost() )
 		{
 			checkAdr.SetIP( net_local_adr.GetIPHostByteOrder() );
 		}
-
+#if 0
 		if ( !Steam3Server().NotifyClientConnect( client, nNewUserID, checkAdr, pchLogonCookie, cbCookie ) 
 			&& !Steam3Server().BLanOnly() ) // the userID isn't alloc'd yet so we need to fill it in manually
 		{
 			RejectConnection( adr, clientChallenge, "#GameUI_ServerRejectSteam" );
 			return false;
 		}
+#endif
 
 		//
 		// Any rejections below this must call SendUserDisconnect

engine/buildcubemaps.cpp

@@ -1042,7 +1042,7 @@ void R_BuildCubemapSamples( int numIterations )
 			}
 		}
 
-		bool bSupportsHDR = g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE;
+		bool bSupportsHDR = true; //g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE;
 
 		for( i = 0; i < pWorldModel->brush.pShared->m_nCubemapSamples; i++ )
 		{
@@ -1080,7 +1080,7 @@ void R_BuildCubemapSamples( int numIterations )
 			return;
 		}
 
-		iBSPPack->SetHDRMode( g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE );
+		iBSPPack->SetHDRMode( true /*g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE*/ );
 
 		iBSPPack->LoadBSPFile( g_pFileSystem, cl.m_szLevelFileName );
 

engine/cl_ents_parse.cpp

@@ -451,7 +451,7 @@ void CL_PreserveExistingEntity( int nOldEntity )
 		return;
 	}
 
-	pEnt->OnDataUnchangedInPVS();
+//	pEnt->OnDataUnchangedInPVS();
 }
 
 void CL_CopyExistingEntity( CEntityReadInfo &u )

engine/cl_main.cpp

@@ -72,6 +72,7 @@
 #include "replay_internal.h"
 #endif
 
+#include "language.h"
 #include "igame.h"
 
 // memdbgon must be the last include file in a .cpp file!!!
@@ -2743,14 +2744,18 @@ void CL_InitLanguageCvar()
 	}
 	else
 	{
+		char *szLang = getenv("LANG");
+
 		if ( CommandLine()->CheckParm( "-language" ) )
-		{
 			cl_language.SetValue( CommandLine()->ParmValue( "-language", "english") );
+		else if( szLang )
+		{
+			ELanguage lang = PchLanguageICUCodeToELanguage(szLang, k_Lang_English);
+			char *szShortLang = GetLanguageShortName(lang);
+			cl_language.SetValue( szShortLang );
 		}
 		else
-		{
 			cl_language.SetValue( "english" );
-		}
 	}
 }
 

engine/cmd.cpp

@@ -579,7 +579,7 @@ void Cmd_Exec_f( const CCommand &args )
 
 	const char *szFile = args[1];
 
-	const char *pPathID = "MOD";
+	const char *pPathID = "*";
 
 	Q_snprintf( fileName, sizeof( fileName ), "//%s/cfg/%s", pPathID, szFile );
 	Q_DefaultExtension( fileName, ".cfg", sizeof( fileName ) );

engine/cmodel_bsp.cpp

@@ -1110,7 +1110,7 @@ void CollisionBSPData_LoadDispInfo( CCollisionBSPData *pBSPData )
     // get face data
     //
 	int face_lump_to_load = LUMP_FACES;
-	if ( g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE &&
+	if ( /*g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE &&*/
 		CMapLoadHelper::LumpSize( LUMP_FACES_HDR ) > 0 )
 	{
 		face_lump_to_load = LUMP_FACES_HDR;

engine/l_studio.cpp

@@ -3637,12 +3637,12 @@ void CModelRender::ValidateStaticPropColorData( ModelInstanceHandle_t handle )
 	// fetch the header
 	CUtlBuffer utlBuf;
 	char fileName[MAX_PATH];
-	if ( g_pMaterialSystemHardwareConfig->GetHDRType() == HDR_TYPE_NONE || g_bBakedPropLightingNoSeparateHDR )
+	if ( g_bBakedPropLightingNoSeparateHDR )
 	{
 		Q_snprintf( fileName, sizeof( fileName ), "sp_%d%s.vhv", StaticPropMgr()->GetStaticPropIndex( pProp ), GetPlatformExt() );
 	}
 	else
-	{	
+	{
 		Q_snprintf( fileName, sizeof( fileName ), "sp_hdr_%d%s.vhv", StaticPropMgr()->GetStaticPropIndex( pProp ), GetPlatformExt() );
 	}
 
@@ -3930,7 +3930,7 @@ bool CModelRender::LoadStaticPropColorData( IHandleEntity *pProp, DataCacheHandl
 
 	// each static prop has its own compiled color mesh
 	char fileName[MAX_PATH];
-	if ( g_pMaterialSystemHardwareConfig->GetHDRType() == HDR_TYPE_NONE || g_bBakedPropLightingNoSeparateHDR )
+	if ( g_bBakedPropLightingNoSeparateHDR )
 	{
         Q_snprintf( fileName, sizeof( fileName ), "sp_%d%s.vhv", StaticPropMgr()->GetStaticPropIndex( pProp ), GetPlatformExt() );
 	}

engine/modelloader.cpp

@@ -483,7 +483,8 @@ void CMapLoadHelper::Init( model_t *pMapModel, const char *loadname )
 
 	s_pMap = &g_ModelLoader.m_worldBrushData;
 
-#if 0
+	// nillerusr: Fuck you johns
+
 	// XXX(johns): There are security issues with this system currently. sv_pure doesn't handle unexpected/mismatched
 	//             lumps, so players can create lumps for maps not using them to wallhack/etc.. Currently unused,
 	//             disabling until we have time to make a proper security pass.
@@ -530,7 +531,6 @@ void CMapLoadHelper::Init( model_t *pMapModel, const char *loadname )
 			}
 		}
 	}
-#endif
 }
 
 //-----------------------------------------------------------------------------
@@ -1856,8 +1856,7 @@ void Mod_LoadFaces( void )
 	int			ti, di;
 
 	int face_lump_to_load = LUMP_FACES;
-	if ( g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE &&
-		CMapLoadHelper::LumpSize( LUMP_FACES_HDR ) > 0 )
+	if ( CMapLoadHelper::LumpSize( LUMP_FACES_HDR ) > 0 )
 	{
 		face_lump_to_load = LUMP_FACES_HDR;
 	}
@@ -2289,8 +2288,7 @@ void Mod_LoadLeafs( void )
 		Mod_LoadLeafs_Version_0( lh );
 		break;
 	case 1:
-		if( g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE &&
-	  		  CMapLoadHelper::LumpSize( LUMP_LEAF_AMBIENT_LIGHTING_HDR ) > 0 )
+		if( CMapLoadHelper::LumpSize( LUMP_LEAF_AMBIENT_LIGHTING_HDR ) > 0 )
 		{
 			CMapLoadHelper mlh( LUMP_LEAF_AMBIENT_LIGHTING_HDR );
 			CMapLoadHelper mlhTable( LUMP_LEAF_AMBIENT_INDEX_HDR );
@@ -2385,7 +2383,7 @@ void Mod_LoadCubemapSamples( void )
 	lh.GetMap()->m_pCubemapSamples = out;
 	lh.GetMap()->m_nCubemapSamples = count;
 
-	bool bHDR = g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE;
+	bool bHDR = true; //g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE;
 	int nCreateFlags = bHDR ? 0 : TEXTUREFLAGS_SRGB;
 
 	// We have separate HDR versions of the textures.  In order to deal with this,
@@ -4453,8 +4451,7 @@ void CModelLoader::Map_LoadModel( model_t *mod )
 
 	// Until BSP version 19, this must occur after loading texinfo
 	COM_TimestampedLog( "  Mod_LoadLighting" );
-	if ( g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE &&
-		CMapLoadHelper::LumpSize( LUMP_LIGHTING_HDR ) > 0 )
+	if ( CMapLoadHelper::LumpSize( LUMP_LIGHTING_HDR ) > 0 )
 	{
 		CMapLoadHelper mlh( LUMP_LIGHTING_HDR );
 		Mod_LoadLighting( mlh );
@@ -4546,8 +4543,7 @@ void CModelLoader::Map_LoadModel( model_t *mod )
 		&m_worldBrushData.m_nAreas );
 
 	COM_TimestampedLog( "  Mod_LoadWorldlights" );
-	if ( g_pMaterialSystemHardwareConfig->GetHDRType() != HDR_TYPE_NONE &&
-  		  CMapLoadHelper::LumpSize( LUMP_WORLDLIGHTS_HDR ) > 0 )
+	if ( CMapLoadHelper::LumpSize( LUMP_WORLDLIGHTS_HDR ) > 0 )
 	{
 		CMapLoadHelper mlh( LUMP_WORLDLIGHTS_HDR );
 		Mod_LoadWorldlights( mlh, true );

engine/voice_codecs/celt/opus/opus.h

@@ -0,0 +1,981 @@
+/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
+   Written by Jean-Marc Valin and Koen Vos */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus.h
+ * @brief Opus reference implementation API
+ */
+
+#ifndef OPUS_H
+#define OPUS_H
+
+#include "opus_types.h"
+#include "opus_defines.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/**
+ * @mainpage Opus
+ *
+ * The Opus codec is designed for interactive speech and audio transmission over the Internet.
+ * It is designed by the IETF Codec Working Group and incorporates technology from
+ * Skype's SILK codec and Xiph.Org's CELT codec.
+ *
+ * The Opus codec is designed to handle a wide range of interactive audio applications,
+ * including Voice over IP, videoconferencing, in-game chat, and even remote live music
+ * performances. It can scale from low bit-rate narrowband speech to very high quality
+ * stereo music. Its main features are:
+
+ * @li Sampling rates from 8 to 48 kHz
+ * @li Bit-rates from 6 kb/s to 510 kb/s
+ * @li Support for both constant bit-rate (CBR) and variable bit-rate (VBR)
+ * @li Audio bandwidth from narrowband to full-band
+ * @li Support for speech and music
+ * @li Support for mono and stereo
+ * @li Support for multichannel (up to 255 channels)
+ * @li Frame sizes from 2.5 ms to 60 ms
+ * @li Good loss robustness and packet loss concealment (PLC)
+ * @li Floating point and fixed-point implementation
+ *
+ * Documentation sections:
+ * @li @ref opus_encoder
+ * @li @ref opus_decoder
+ * @li @ref opus_repacketizer
+ * @li @ref opus_multistream
+ * @li @ref opus_libinfo
+ * @li @ref opus_custom
+ */
+
+/** @defgroup opus_encoder Opus Encoder
+  * @{
+  *
+  * @brief This page describes the process and functions used to encode Opus.
+  *
+  * Since Opus is a stateful codec, the encoding process starts with creating an encoder
+  * state. This can be done with:
+  *
+  * @code
+  * int          error;
+  * OpusEncoder *enc;
+  * enc = opus_encoder_create(Fs, channels, application, &error);
+  * @endcode
+  *
+  * From this point, @c enc can be used for encoding an audio stream. An encoder state
+  * @b must @b not be used for more than one stream at the same time. Similarly, the encoder
+  * state @b must @b not be re-initialized for each frame.
+  *
+  * While opus_encoder_create() allocates memory for the state, it's also possible
+  * to initialize pre-allocated memory:
+  *
+  * @code
+  * int          size;
+  * int          error;
+  * OpusEncoder *enc;
+  * size = opus_encoder_get_size(channels);
+  * enc = malloc(size);
+  * error = opus_encoder_init(enc, Fs, channels, application);
+  * @endcode
+  *
+  * where opus_encoder_get_size() returns the required size for the encoder state. Note that
+  * future versions of this code may change the size, so no assuptions should be made about it.
+  *
+  * The encoder state is always continuous in memory and only a shallow copy is sufficient
+  * to copy it (e.g. memcpy())
+  *
+  * It is possible to change some of the encoder's settings using the opus_encoder_ctl()
+  * interface. All these settings already default to the recommended value, so they should
+  * only be changed when necessary. The most common settings one may want to change are:
+  *
+  * @code
+  * opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate));
+  * opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(complexity));
+  * opus_encoder_ctl(enc, OPUS_SET_SIGNAL(signal_type));
+  * @endcode
+  *
+  * where
+  *
+  * @arg bitrate is in bits per second (b/s)
+  * @arg complexity is a value from 1 to 10, where 1 is the lowest complexity and 10 is the highest
+  * @arg signal_type is either OPUS_AUTO (default), OPUS_SIGNAL_VOICE, or OPUS_SIGNAL_MUSIC
+  *
+  * See @ref opus_encoderctls and @ref opus_genericctls for a complete list of parameters that can be set or queried. Most parameters can be set or changed at any time during a stream.
+  *
+  * To encode a frame, opus_encode() or opus_encode_float() must be called with exactly one frame (2.5, 5, 10, 20, 40 or 60 ms) of audio data:
+  * @code
+  * len = opus_encode(enc, audio_frame, frame_size, packet, max_packet);
+  * @endcode
+  *
+  * where
+  * <ul>
+  * <li>audio_frame is the audio data in opus_int16 (or float for opus_encode_float())</li>
+  * <li>frame_size is the duration of the frame in samples (per channel)</li>
+  * <li>packet is the byte array to which the compressed data is written</li>
+  * <li>max_packet is the maximum number of bytes that can be written in the packet (4000 bytes is recommended).
+  *     Do not use max_packet to control VBR target bitrate, instead use the #OPUS_SET_BITRATE CTL.</li>
+  * </ul>
+  *
+  * opus_encode() and opus_encode_float() return the number of bytes actually written to the packet.
+  * The return value <b>can be negative</b>, which indicates that an error has occurred. If the return value
+  * is 2 bytes or less, then the packet does not need to be transmitted (DTX).
+  *
+  * Once the encoder state if no longer needed, it can be destroyed with
+  *
+  * @code
+  * opus_encoder_destroy(enc);
+  * @endcode
+  *
+  * If the encoder was created with opus_encoder_init() rather than opus_encoder_create(),
+  * then no action is required aside from potentially freeing the memory that was manually
+  * allocated for it (calling free(enc) for the example above)
+  *
+  */
+
+/** Opus encoder state.
+  * This contains the complete state of an Opus encoder.
+  * It is position independent and can be freely copied.
+  * @see opus_encoder_create,opus_encoder_init
+  */
+typedef struct OpusEncoder OpusEncoder;
+
+/** Gets the size of an <code>OpusEncoder</code> structure.
+  * @param[in] channels <tt>int</tt>: Number of channels.
+  *                                   This must be 1 or 2.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_encoder_get_size(int channels);
+
+/**
+ */
+
+/** Allocates and initializes an encoder state.
+ * There are three coding modes:
+ *
+ * @ref OPUS_APPLICATION_VOIP gives best quality at a given bitrate for voice
+ *    signals. It enhances the  input signal by high-pass filtering and
+ *    emphasizing formants and harmonics. Optionally  it includes in-band
+ *    forward error correction to protect against packet loss. Use this
+ *    mode for typical VoIP applications. Because of the enhancement,
+ *    even at high bitrates the output may sound different from the input.
+ *
+ * @ref OPUS_APPLICATION_AUDIO gives best quality at a given bitrate for most
+ *    non-voice signals like music. Use this mode for music and mixed
+ *    (music/voice) content, broadcast, and applications requiring less
+ *    than 15 ms of coding delay.
+ *
+ * @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY configures low-delay mode that
+ *    disables the speech-optimized mode in exchange for slightly reduced delay.
+ *    This mode can only be set on an newly initialized or freshly reset encoder
+ *    because it changes the codec delay.
+ *
+ * This is useful when the caller knows that the speech-optimized modes will not be needed (use with caution).
+ * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
+ *                                     This must be one of 8000, 12000, 16000,
+ *                                     24000, or 48000.
+ * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
+ * @param [in] application <tt>int</tt>: Coding mode (@ref OPUS_APPLICATION_VOIP/@ref OPUS_APPLICATION_AUDIO/@ref OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+ * @param [out] error <tt>int*</tt>: @ref opus_errorcodes
+ * @note Regardless of the sampling rate and number channels selected, the Opus encoder
+ * can switch to a lower audio bandwidth or number of channels if the bitrate
+ * selected is too low. This also means that it is safe to always use 48 kHz stereo input
+ * and let the encoder optimize the encoding.
+ */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusEncoder *opus_encoder_create(
+    opus_int32 Fs,
+    int channels,
+    int application,
+    int *error
+);
+
+/** Initializes a previously allocated encoder state
+  * The memory pointed to by st must be at least the size returned by opus_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_encoder_create(),opus_encoder_get_size()
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
+ *                                      This must be one of 8000, 12000, 16000,
+ *                                      24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
+  * @param [in] application <tt>int</tt>: Coding mode (OPUS_APPLICATION_VOIP/OPUS_APPLICATION_AUDIO/OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+  * @retval #OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_EXPORT int opus_encoder_init(
+    OpusEncoder *st,
+    opus_int32 Fs,
+    int channels,
+    int application
+) OPUS_ARG_NONNULL(1);
+
+/** Encodes an Opus frame.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>opus_int16*</tt>: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
+  *                                      input signal.
+  *                                      This must be an Opus frame size for
+  *                                      the encoder's sampling rate.
+  *                                      For example, at 48 kHz the permitted
+  *                                      values are 120, 240, 480, 960, 1920,
+  *                                      and 2880.
+  *                                      Passing in a duration of less than
+  *                                      10 ms (480 samples at 48 kHz) will
+  *                                      prevent the encoder from using the LPC
+  *                                      or hybrid modes.
+  * @param [out] data <tt>unsigned char*</tt>: Output payload.
+  *                                            This must contain storage for at
+  *                                            least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode(
+    OpusEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes an Opus frame from floating point input.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>float*</tt>: Input in float format (interleaved if 2 channels), with a normal range of +/-1.0.
+  *          Samples with a range beyond +/-1.0 are supported but will
+  *          be clipped by decoders using the integer API and should
+  *          only be used if it is known that the far end supports
+  *          extended dynamic range.
+  *          length is frame_size*channels*sizeof(float)
+  * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
+  *                                      input signal.
+  *                                      This must be an Opus frame size for
+  *                                      the encoder's sampling rate.
+  *                                      For example, at 48 kHz the permitted
+  *                                      values are 120, 240, 480, 960, 1920,
+  *                                      and 2880.
+  *                                      Passing in a duration of less than
+  *                                      10 ms (480 samples at 48 kHz) will
+  *                                      prevent the encoder from using the LPC
+  *                                      or hybrid modes.
+  * @param [out] data <tt>unsigned char*</tt>: Output payload.
+  *                                            This must contain storage for at
+  *                                            least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode_float(
+    OpusEncoder *st,
+    const float *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Frees an <code>OpusEncoder</code> allocated by opus_encoder_create().
+  * @param[in] st <tt>OpusEncoder*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_encoder_destroy(OpusEncoder *st);
+
+/** Perform a CTL function on an Opus encoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @param st <tt>OpusEncoder*</tt>: Encoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls or
+  *                @ref opus_encoderctls.
+  * @see opus_genericctls
+  * @see opus_encoderctls
+  */
+OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+/**@}*/
+
+/** @defgroup opus_decoder Opus Decoder
+  * @{
+  *
+  * @brief This page describes the process and functions used to decode Opus.
+  *
+  * The decoding process also starts with creating a decoder
+  * state. This can be done with:
+  * @code
+  * int          error;
+  * OpusDecoder *dec;
+  * dec = opus_decoder_create(Fs, channels, &error);
+  * @endcode
+  * where
+  * @li Fs is the sampling rate and must be 8000, 12000, 16000, 24000, or 48000
+  * @li channels is the number of channels (1 or 2)
+  * @li error will hold the error code in case of failure (or #OPUS_OK on success)
+  * @li the return value is a newly created decoder state to be used for decoding
+  *
+  * While opus_decoder_create() allocates memory for the state, it's also possible
+  * to initialize pre-allocated memory:
+  * @code
+  * int          size;
+  * int          error;
+  * OpusDecoder *dec;
+  * size = opus_decoder_get_size(channels);
+  * dec = malloc(size);
+  * error = opus_decoder_init(dec, Fs, channels);
+  * @endcode
+  * where opus_decoder_get_size() returns the required size for the decoder state. Note that
+  * future versions of this code may change the size, so no assuptions should be made about it.
+  *
+  * The decoder state is always continuous in memory and only a shallow copy is sufficient
+  * to copy it (e.g. memcpy())
+  *
+  * To decode a frame, opus_decode() or opus_decode_float() must be called with a packet of compressed audio data:
+  * @code
+  * frame_size = opus_decode(dec, packet, len, decoded, max_size, 0);
+  * @endcode
+  * where
+  *
+  * @li packet is the byte array containing the compressed data
+  * @li len is the exact number of bytes contained in the packet
+  * @li decoded is the decoded audio data in opus_int16 (or float for opus_decode_float())
+  * @li max_size is the max duration of the frame in samples (per channel) that can fit into the decoded_frame array
+  *
+  * opus_decode() and opus_decode_float() return the number of samples (per channel) decoded from the packet.
+  * If that value is negative, then an error has occurred. This can occur if the packet is corrupted or if the audio
+  * buffer is too small to hold the decoded audio.
+  *
+  * Opus is a stateful codec with overlapping blocks and as a result Opus
+  * packets are not coded independently of each other. Packets must be
+  * passed into the decoder serially and in the correct order for a correct
+  * decode. Lost packets can be replaced with loss concealment by calling
+  * the decoder with a null pointer and zero length for the missing packet.
+  *
+  * A single codec state may only be accessed from a single thread at
+  * a time and any required locking must be performed by the caller. Separate
+  * streams must be decoded with separate decoder states and can be decoded
+  * in parallel unless the library was compiled with NONTHREADSAFE_PSEUDOSTACK
+  * defined.
+  *
+  */
+
+/** Opus decoder state.
+  * This contains the complete state of an Opus decoder.
+  * It is position independent and can be freely copied.
+  * @see opus_decoder_create,opus_decoder_init
+  */
+typedef struct OpusDecoder OpusDecoder;
+
+/** Gets the size of an <code>OpusDecoder</code> structure.
+  * @param [in] channels <tt>int</tt>: Number of channels.
+  *                                    This must be 1 or 2.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_size(int channels);
+
+/** Allocates and initializes a decoder state.
+  * @param [in] Fs <tt>opus_int32</tt>: Sample rate to decode at (Hz).
+  *                                     This must be one of 8000, 12000, 16000,
+  *                                     24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
+  * @param [out] error <tt>int*</tt>: #OPUS_OK Success or @ref opus_errorcodes
+  *
+  * Internally Opus stores data at 48000 Hz, so that should be the default
+  * value for Fs. However, the decoder can efficiently decode to buffers
+  * at 8, 12, 16, and 24 kHz so if for some reason the caller cannot use
+  * data at the full sample rate, or knows the compressed data doesn't
+  * use the full frequency range, it can request decoding at a reduced
+  * rate. Likewise, the decoder is capable of filling in either mono or
+  * interleaved stereo pcm buffers, at the caller's request.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusDecoder *opus_decoder_create(
+    opus_int32 Fs,
+    int channels,
+    int *error
+);
+
+/** Initializes a previously allocated decoder state.
+  * The state must be at least the size returned by opus_decoder_get_size().
+  * This is intended for applications which use their own allocator instead of malloc. @see opus_decoder_create,opus_decoder_get_size
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state.
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate to decode to (Hz).
+  *                                     This must be one of 8000, 12000, 16000,
+  *                                     24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
+  * @retval #OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_EXPORT int opus_decoder_init(
+    OpusDecoder *st,
+    opus_int32 Fs,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+/** Decode an Opus packet.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload*
+  * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size Number of samples per channel of available space in \a pcm.
+  *  If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
+  *  not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
+  *  then frame_size needs to be exactly the duration of audio that is missing, otherwise the
+  *  decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
+  *  FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
+  * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
+  *  decoded. If no such data is available, the frame is decoded as if it were lost.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode(
+    OpusDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int16 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode an Opus packet with floating point output.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(float)
+  * @param [in] frame_size Number of samples per channel of available space in \a pcm.
+  *  If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
+  *  not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
+  *  then frame_size needs to be exactly the duration of audio that is missing, otherwise the
+  *  decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
+  *  FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
+  * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
+  *  decoded. If no such data is available the frame is decoded as if it were lost.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode_float(
+    OpusDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    float *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus decoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @param st <tt>OpusDecoder*</tt>: Decoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls or
+  *                @ref opus_decoderctls.
+  * @see opus_genericctls
+  * @see opus_decoderctls
+  */
+OPUS_EXPORT int opus_decoder_ctl(OpusDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/** Frees an <code>OpusDecoder</code> allocated by opus_decoder_create().
+  * @param[in] st <tt>OpusDecoder*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_decoder_destroy(OpusDecoder *st);
+
+/** Parse an opus packet into one or more frames.
+  * Opus_decode will perform this operation internally so most applications do
+  * not need to use this function.
+  * This function does not copy the frames, the returned pointers are pointers into
+  * the input packet.
+  * @param [in] data <tt>char*</tt>: Opus packet to be parsed
+  * @param [in] len <tt>opus_int32</tt>: size of data
+  * @param [out] out_toc <tt>char*</tt>: TOC pointer
+  * @param [out] frames <tt>char*[48]</tt> encapsulated frames
+  * @param [out] size <tt>opus_int16[48]</tt> sizes of the encapsulated frames
+  * @param [out] payload_offset <tt>int*</tt>: returns the position of the payload within the packet (in bytes)
+  * @returns number of frames
+  */
+OPUS_EXPORT int opus_packet_parse(
+   const unsigned char *data,
+   opus_int32 len,
+   unsigned char *out_toc,
+   const unsigned char *frames[48],
+   opus_int16 size[48],
+   int *payload_offset
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5);
+
+/** Gets the bandwidth of an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet
+  * @retval OPUS_BANDWIDTH_NARROWBAND Narrowband (4kHz bandpass)
+  * @retval OPUS_BANDWIDTH_MEDIUMBAND Mediumband (6kHz bandpass)
+  * @retval OPUS_BANDWIDTH_WIDEBAND Wideband (8kHz bandpass)
+  * @retval OPUS_BANDWIDTH_SUPERWIDEBAND Superwideband (12kHz bandpass)
+  * @retval OPUS_BANDWIDTH_FULLBAND Fullband (20kHz bandpass)
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_bandwidth(const unsigned char *data) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples per frame from an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet.
+  *                                  This must contain at least one byte of
+  *                                  data.
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
+  *                                     This must be a multiple of 400, or
+  *                                     inaccurate results will be returned.
+  * @returns Number of samples per frame.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_samples_per_frame(const unsigned char *data, opus_int32 Fs) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of channels from an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet
+  * @returns Number of channels
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_channels(const unsigned char *data) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of frames in an Opus packet.
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @returns Number of frames
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples of an Opus packet.
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
+  *                                     This must be a multiple of 400, or
+  *                                     inaccurate results will be returned.
+  * @returns Number of samples
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_samples(const unsigned char packet[], opus_int32 len, opus_int32 Fs) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples of an Opus packet.
+  * @param [in] dec <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @returns Number of samples
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_nb_samples(const OpusDecoder *dec, const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+/** Applies soft-clipping to bring a float signal within the [-1,1] range. If
+  * the signal is already in that range, nothing is done. If there are values
+  * outside of [-1,1], then the signal is clipped as smoothly as possible to
+  * both fit in the range and avoid creating excessive distortion in the
+  * process.
+  * @param [in,out] pcm <tt>float*</tt>: Input PCM and modified PCM
+  * @param [in] frame_size <tt>int</tt> Number of samples per channel to process
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [in,out] softclip_mem <tt>float*</tt>: State memory for the soft clipping process (one float per channel, initialized to zero)
+  */
+OPUS_EXPORT void opus_pcm_soft_clip(float *pcm, int frame_size, int channels, float *softclip_mem);
+
+
+/**@}*/
+
+/** @defgroup opus_repacketizer Repacketizer
+  * @{
+  *
+  * The repacketizer can be used to merge multiple Opus packets into a single
+  * packet or alternatively to split Opus packets that have previously been
+  * merged. Splitting valid Opus packets is always guaranteed to succeed,
+  * whereas merging valid packets only succeeds if all frames have the same
+  * mode, bandwidth, and frame size, and when the total duration of the merged
+  * packet is no more than 120 ms. The 120 ms limit comes from the
+  * specification and limits decoder memory requirements at a point where
+  * framing overhead becomes negligible.
+  *
+  * The repacketizer currently only operates on elementary Opus
+  * streams. It will not manipualte multistream packets successfully, except in
+  * the degenerate case where they consist of data from a single stream.
+  *
+  * The repacketizing process starts with creating a repacketizer state, either
+  * by calling opus_repacketizer_create() or by allocating the memory yourself,
+  * e.g.,
+  * @code
+  * OpusRepacketizer *rp;
+  * rp = (OpusRepacketizer*)malloc(opus_repacketizer_get_size());
+  * if (rp != NULL)
+  *     opus_repacketizer_init(rp);
+  * @endcode
+  *
+  * Then the application should submit packets with opus_repacketizer_cat(),
+  * extract new packets with opus_repacketizer_out() or
+  * opus_repacketizer_out_range(), and then reset the state for the next set of
+  * input packets via opus_repacketizer_init().
+  *
+  * For example, to split a sequence of packets into individual frames:
+  * @code
+  * unsigned char *data;
+  * int len;
+  * while (get_next_packet(&data, &len))
+  * {
+  *   unsigned char out[1276];
+  *   opus_int32 out_len;
+  *   int nb_frames;
+  *   int err;
+  *   int i;
+  *   err = opus_repacketizer_cat(rp, data, len);
+  *   if (err != OPUS_OK)
+  *   {
+  *     release_packet(data);
+  *     return err;
+  *   }
+  *   nb_frames = opus_repacketizer_get_nb_frames(rp);
+  *   for (i = 0; i < nb_frames; i++)
+  *   {
+  *     out_len = opus_repacketizer_out_range(rp, i, i+1, out, sizeof(out));
+  *     if (out_len < 0)
+  *     {
+  *        release_packet(data);
+  *        return (int)out_len;
+  *     }
+  *     output_next_packet(out, out_len);
+  *   }
+  *   opus_repacketizer_init(rp);
+  *   release_packet(data);
+  * }
+  * @endcode
+  *
+  * Alternatively, to combine a sequence of frames into packets that each
+  * contain up to <code>TARGET_DURATION_MS</code> milliseconds of data:
+  * @code
+  * // The maximum number of packets with duration TARGET_DURATION_MS occurs
+  * // when the frame size is 2.5 ms, for a total of (TARGET_DURATION_MS*2/5)
+  * // packets.
+  * unsigned char *data[(TARGET_DURATION_MS*2/5)+1];
+  * opus_int32 len[(TARGET_DURATION_MS*2/5)+1];
+  * int nb_packets;
+  * unsigned char out[1277*(TARGET_DURATION_MS*2/2)];
+  * opus_int32 out_len;
+  * int prev_toc;
+  * nb_packets = 0;
+  * while (get_next_packet(data+nb_packets, len+nb_packets))
+  * {
+  *   int nb_frames;
+  *   int err;
+  *   nb_frames = opus_packet_get_nb_frames(data[nb_packets], len[nb_packets]);
+  *   if (nb_frames < 1)
+  *   {
+  *     release_packets(data, nb_packets+1);
+  *     return nb_frames;
+  *   }
+  *   nb_frames += opus_repacketizer_get_nb_frames(rp);
+  *   // If adding the next packet would exceed our target, or it has an
+  *   // incompatible TOC sequence, output the packets we already have before
+  *   // submitting it.
+  *   // N.B., The nb_packets > 0 check ensures we've submitted at least one
+  *   // packet since the last call to opus_repacketizer_init(). Otherwise a
+  *   // single packet longer than TARGET_DURATION_MS would cause us to try to
+  *   // output an (invalid) empty packet. It also ensures that prev_toc has
+  *   // been set to a valid value. Additionally, len[nb_packets] > 0 is
+  *   // guaranteed by the call to opus_packet_get_nb_frames() above, so the
+  *   // reference to data[nb_packets][0] should be valid.
+  *   if (nb_packets > 0 && (
+  *       ((prev_toc & 0xFC) != (data[nb_packets][0] & 0xFC)) ||
+  *       opus_packet_get_samples_per_frame(data[nb_packets], 48000)*nb_frames >
+  *       TARGET_DURATION_MS*48))
+  *   {
+  *     out_len = opus_repacketizer_out(rp, out, sizeof(out));
+  *     if (out_len < 0)
+  *     {
+  *        release_packets(data, nb_packets+1);
+  *        return (int)out_len;
+  *     }
+  *     output_next_packet(out, out_len);
+  *     opus_repacketizer_init(rp);
+  *     release_packets(data, nb_packets);
+  *     data[0] = data[nb_packets];
+  *     len[0] = len[nb_packets];
+  *     nb_packets = 0;
+  *   }
+  *   err = opus_repacketizer_cat(rp, data[nb_packets], len[nb_packets]);
+  *   if (err != OPUS_OK)
+  *   {
+  *     release_packets(data, nb_packets+1);
+  *     return err;
+  *   }
+  *   prev_toc = data[nb_packets][0];
+  *   nb_packets++;
+  * }
+  * // Output the final, partial packet.
+  * if (nb_packets > 0)
+  * {
+  *   out_len = opus_repacketizer_out(rp, out, sizeof(out));
+  *   release_packets(data, nb_packets);
+  *   if (out_len < 0)
+  *     return (int)out_len;
+  *   output_next_packet(out, out_len);
+  * }
+  * @endcode
+  *
+  * An alternate way of merging packets is to simply call opus_repacketizer_cat()
+  * unconditionally until it fails. At that point, the merged packet can be
+  * obtained with opus_repacketizer_out() and the input packet for which
+  * opus_repacketizer_cat() needs to be re-added to a newly reinitialized
+  * repacketizer state.
+  */
+
+typedef struct OpusRepacketizer OpusRepacketizer;
+
+/** Gets the size of an <code>OpusRepacketizer</code> structure.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_size(void);
+
+/** (Re)initializes a previously allocated repacketizer state.
+  * The state must be at least the size returned by opus_repacketizer_get_size().
+  * This can be used for applications which use their own allocator instead of
+  * malloc().
+  * It must also be called to reset the queue of packets waiting to be
+  * repacketized, which is necessary if the maximum packet duration of 120 ms
+  * is reached or if you wish to submit packets with a different Opus
+  * configuration (coding mode, audio bandwidth, frame size, or channel count).
+  * Failure to do so will prevent a new packet from being added with
+  * opus_repacketizer_cat().
+  * @see opus_repacketizer_create
+  * @see opus_repacketizer_get_size
+  * @see opus_repacketizer_cat
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to
+  *                                       (re)initialize.
+  * @returns A pointer to the same repacketizer state that was passed in.
+  */
+OPUS_EXPORT OpusRepacketizer *opus_repacketizer_init(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
+
+/** Allocates memory and initializes the new repacketizer with
+ * opus_repacketizer_init().
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusRepacketizer *opus_repacketizer_create(void);
+
+/** Frees an <code>OpusRepacketizer</code> allocated by
+  * opus_repacketizer_create().
+  * @param[in] rp <tt>OpusRepacketizer*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_repacketizer_destroy(OpusRepacketizer *rp);
+
+/** Add a packet to the current repacketizer state.
+  * This packet must match the configuration of any packets already submitted
+  * for repacketization since the last call to opus_repacketizer_init().
+  * This means that it must have the same coding mode, audio bandwidth, frame
+  * size, and channel count.
+  * This can be checked in advance by examining the top 6 bits of the first
+  * byte of the packet, and ensuring they match the top 6 bits of the first
+  * byte of any previously submitted packet.
+  * The total duration of audio in the repacketizer state also must not exceed
+  * 120 ms, the maximum duration of a single packet, after adding this packet.
+  *
+  * The contents of the current repacketizer state can be extracted into new
+  * packets using opus_repacketizer_out() or opus_repacketizer_out_range().
+  *
+  * In order to add a packet with a different configuration or to add more
+  * audio beyond 120 ms, you must clear the repacketizer state by calling
+  * opus_repacketizer_init().
+  * If a packet is too large to add to the current repacketizer state, no part
+  * of it is added, even if it contains multiple frames, some of which might
+  * fit.
+  * If you wish to be able to add parts of such packets, you should first use
+  * another repacketizer to split the packet into pieces and add them
+  * individually.
+  * @see opus_repacketizer_out_range
+  * @see opus_repacketizer_out
+  * @see opus_repacketizer_init
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to which to
+  *                                       add the packet.
+  * @param[in] data <tt>const unsigned char*</tt>: The packet data.
+  *                                                The application must ensure
+  *                                                this pointer remains valid
+  *                                                until the next call to
+  *                                                opus_repacketizer_init() or
+  *                                                opus_repacketizer_destroy().
+  * @param len <tt>opus_int32</tt>: The number of bytes in the packet data.
+  * @returns An error code indicating whether or not the operation succeeded.
+  * @retval #OPUS_OK The packet's contents have been added to the repacketizer
+  *                  state.
+  * @retval #OPUS_INVALID_PACKET The packet did not have a valid TOC sequence,
+  *                              the packet's TOC sequence was not compatible
+  *                              with previously submitted packets (because
+  *                              the coding mode, audio bandwidth, frame size,
+  *                              or channel count did not match), or adding
+  *                              this packet would increase the total amount of
+  *                              audio stored in the repacketizer state to more
+  *                              than 120 ms.
+  */
+OPUS_EXPORT int opus_repacketizer_cat(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+
+/** Construct a new packet from data previously submitted to the repacketizer
+  * state via opus_repacketizer_cat().
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
+  *                                       construct the new packet.
+  * @param begin <tt>int</tt>: The index of the first frame in the current
+  *                            repacketizer state to include in the output.
+  * @param end <tt>int</tt>: One past the index of the last frame in the
+  *                          current repacketizer state to include in the
+  *                          output.
+  * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
+  *                                                 store the output packet.
+  * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
+  *                                    the output buffer. In order to guarantee
+  *                                    success, this should be at least
+  *                                    <code>1276</code> for a single frame,
+  *                                    or for multiple frames,
+  *                                    <code>1277*(end-begin)</code>.
+  *                                    However, <code>1*(end-begin)</code> plus
+  *                                    the size of all packet data submitted to
+  *                                    the repacketizer since the last call to
+  *                                    opus_repacketizer_init() or
+  *                                    opus_repacketizer_create() is also
+  *                                    sufficient, and possibly much smaller.
+  * @returns The total size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG <code>[begin,end)</code> was an invalid range of
+  *                       frames (begin < 0, begin >= end, or end >
+  *                       opus_repacketizer_get_nb_frames()).
+  * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
+  *                                complete output packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out_range(OpusRepacketizer *rp, int begin, int end, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Return the total number of frames contained in packet data submitted to
+  * the repacketizer state so far via opus_repacketizer_cat() since the last
+  * call to opus_repacketizer_init() or opus_repacketizer_create().
+  * This defines the valid range of packets that can be extracted with
+  * opus_repacketizer_out_range() or opus_repacketizer_out().
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state containing the
+  *                                       frames.
+  * @returns The total number of frames contained in the packet data submitted
+  *          to the repacketizer state.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_nb_frames(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
+
+/** Construct a new packet from data previously submitted to the repacketizer
+  * state via opus_repacketizer_cat().
+  * This is a convenience routine that returns all the data submitted so far
+  * in a single packet.
+  * It is equivalent to calling
+  * @code
+  * opus_repacketizer_out_range(rp, 0, opus_repacketizer_get_nb_frames(rp),
+  *                             data, maxlen)
+  * @endcode
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
+  *                                       construct the new packet.
+  * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
+  *                                                 store the output packet.
+  * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
+  *                                    the output buffer. In order to guarantee
+  *                                    success, this should be at least
+  *                                    <code>1277*opus_repacketizer_get_nb_frames(rp)</code>.
+  *                                    However,
+  *                                    <code>1*opus_repacketizer_get_nb_frames(rp)</code>
+  *                                    plus the size of all packet data
+  *                                    submitted to the repacketizer since the
+  *                                    last call to opus_repacketizer_init() or
+  *                                    opus_repacketizer_create() is also
+  *                                    sufficient, and possibly much smaller.
+  * @returns The total size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
+  *                                complete output packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out(OpusRepacketizer *rp, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1);
+
+/** Pads a given Opus packet to a larger size (possibly changing the TOC sequence).
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to pad.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
+  *                                 This must be at least as large as len.
+  * @returns an error code
+  * @retval #OPUS_OK \a on success.
+  * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT int opus_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len);
+
+/** Remove all padding from a given Opus packet and rewrite the TOC sequence to
+  * minimize space usage.
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to strip.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @returns The new size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG \a len was less than 1.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_packet_unpad(unsigned char *data, opus_int32 len);
+
+/** Pads a given Opus multi-stream packet to a larger size (possibly changing the TOC sequence).
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to pad.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
+  *                                 This must be at least 1.
+  * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
+  *                                 This must be at least as large as len.
+  * @returns an error code
+  * @retval #OPUS_OK \a on success.
+  * @retval #OPUS_BAD_ARG \a len was less than 1.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT int opus_multistream_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len, int nb_streams);
+
+/** Remove all padding from a given Opus multi-stream packet and rewrite the TOC sequence to
+  * minimize space usage.
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to strip.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
+  *                                 This must be at least 1.
+  * @returns The new size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_packet_unpad(unsigned char *data, opus_int32 len, int nb_streams);
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_H */

engine/voice_codecs/celt/opus/opus_custom.h

@@ -0,0 +1,342 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Copyright (c) 2008-2012 Gregory Maxwell
+   Written by Jean-Marc Valin and Gregory Maxwell */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+  @file opus_custom.h
+  @brief Opus-Custom reference implementation API
+ */
+
+#ifndef OPUS_CUSTOM_H
+#define OPUS_CUSTOM_H
+
+#include "opus_defines.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#ifdef CUSTOM_MODES
+# define OPUS_CUSTOM_EXPORT OPUS_EXPORT
+# define OPUS_CUSTOM_EXPORT_STATIC OPUS_EXPORT
+#else
+# define OPUS_CUSTOM_EXPORT
+# ifdef OPUS_BUILD
+#  define OPUS_CUSTOM_EXPORT_STATIC static OPUS_INLINE
+# else
+#  define OPUS_CUSTOM_EXPORT_STATIC
+# endif
+#endif
+
+/** @defgroup opus_custom Opus Custom
+  * @{
+  *  Opus Custom is an optional part of the Opus specification and
+  * reference implementation which uses a distinct API from the regular
+  * API and supports frame sizes that are not normally supported.\ Use
+  * of Opus Custom is discouraged for all but very special applications
+  * for which a frame size different from 2.5, 5, 10, or 20 ms is needed
+  * (for either complexity or latency reasons) and where interoperability
+  * is less important.
+  *
+  * In addition to the interoperability limitations the use of Opus custom
+  * disables a substantial chunk of the codec and generally lowers the
+  * quality available at a given bitrate. Normally when an application needs
+  * a different frame size from the codec it should buffer to match the
+  * sizes but this adds a small amount of delay which may be important
+  * in some very low latency applications. Some transports (especially
+  * constant rate RF transports) may also work best with frames of
+  * particular durations.
+  *
+  * Libopus only supports custom modes if they are enabled at compile time.
+  *
+  * The Opus Custom API is similar to the regular API but the
+  * @ref opus_encoder_create and @ref opus_decoder_create calls take
+  * an additional mode parameter which is a structure produced by
+  * a call to @ref opus_custom_mode_create. Both the encoder and decoder
+  * must create a mode using the same sample rate (fs) and frame size
+  * (frame size) so these parameters must either be signaled out of band
+  * or fixed in a particular implementation.
+  *
+  * Similar to regular Opus the custom modes support on the fly frame size
+  * switching, but the sizes available depend on the particular frame size in
+  * use. For some initial frame sizes on a single on the fly size is available.
+  */
+
+/** Contains the state of an encoder. One encoder state is needed
+    for each stream. It is initialized once at the beginning of the
+    stream. Do *not* re-initialize the state for every frame.
+   @brief Encoder state
+ */
+typedef struct OpusCustomEncoder OpusCustomEncoder;
+
+/** State of the decoder. One decoder state is needed for each stream.
+    It is initialized once at the beginning of the stream. Do *not*
+    re-initialize the state for every frame.
+   @brief Decoder state
+ */
+typedef struct OpusCustomDecoder OpusCustomDecoder;
+
+/** The mode contains all the information necessary to create an
+    encoder. Both the encoder and decoder need to be initialized
+    with exactly the same mode, otherwise the output will be
+    corrupted.
+   @brief Mode configuration
+ */
+typedef struct OpusCustomMode OpusCustomMode;
+
+/** Creates a new mode struct. This will be passed to an encoder or
+  * decoder. The mode MUST NOT BE DESTROYED until the encoders and
+  * decoders that use it are destroyed as well.
+  * @param [in] Fs <tt>int</tt>: Sampling rate (8000 to 96000 Hz)
+  * @param [in] frame_size <tt>int</tt>: Number of samples (per channel) to encode in each
+  *        packet (64 - 1024, prime factorization must contain zero or more 2s, 3s, or 5s and no other primes)
+  * @param [out] error <tt>int*</tt>: Returned error code (if NULL, no error will be returned)
+  * @return A newly created mode
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomMode *opus_custom_mode_create(opus_int32 Fs, int frame_size, int *error);
+
+/** Destroys a mode struct. Only call this after all encoders and
+  * decoders using this mode are destroyed as well.
+  * @param [in] mode <tt>OpusCustomMode*</tt>: Mode to be freed.
+  */
+OPUS_CUSTOM_EXPORT void opus_custom_mode_destroy(OpusCustomMode *mode);
+
+
+#if !defined(OPUS_BUILD) || defined(CELT_ENCODER_C)
+
+/* Encoder */
+/** Gets the size of an OpusCustomEncoder structure.
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Mode configuration
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @returns size
+  */
+OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_encoder_get_size(
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+# ifdef CUSTOM_MODES
+/** Initializes a previously allocated encoder state
+  * The memory pointed to by st must be the size returned by opus_custom_encoder_get_size.
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_custom_encoder_create(),opus_custom_encoder_get_size()
+  * To reset a previously initialized state use the OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Contains all the information about the characteristics of
+  *  the stream (must be the same characteristics as used for the
+  *  decoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @return OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT int opus_custom_encoder_init(
+    OpusCustomEncoder *st,
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+# endif
+#endif
+
+
+/** Creates a new encoder state. Each stream needs its own encoder
+  * state (can't be shared across simultaneous streams).
+  * @param [in] mode <tt>OpusCustomMode*</tt>: Contains all the information about the characteristics of
+  *  the stream (must be the same characteristics as used for the
+  *  decoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [out] error <tt>int*</tt>: Returns an error code
+  * @return Newly created encoder state.
+*/
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomEncoder *opus_custom_encoder_create(
+    const OpusCustomMode *mode,
+    int channels,
+    int *error
+) OPUS_ARG_NONNULL(1);
+
+
+/** Destroys a an encoder state.
+  * @param[in] st <tt>OpusCustomEncoder*</tt>: State to be freed.
+  */
+OPUS_CUSTOM_EXPORT void opus_custom_encoder_destroy(OpusCustomEncoder *st);
+
+/** Encodes a frame of audio.
+  * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>float*</tt>: PCM audio in float format, with a normal range of +/-1.0.
+  *          Samples with a range beyond +/-1.0 are supported but will
+  *          be clipped by decoders using the integer API and should
+  *          only be used if it is known that the far end supports
+  *          extended dynamic range. There must be exactly
+  *          frame_size samples per channel.
+  * @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal
+  * @param [out] compressed <tt>char *</tt>: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long.
+  * @param [in] maxCompressedBytes <tt>int</tt>: Maximum number of bytes to use for compressing the frame
+  *          (can change from one frame to another)
+  * @return Number of bytes written to "compressed".
+  *       If negative, an error has occurred (see error codes). It is IMPORTANT that
+  *       the length returned be somehow transmitted to the decoder. Otherwise, no
+  *       decoding is possible.
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode_float(
+    OpusCustomEncoder *st,
+    const float *pcm,
+    int frame_size,
+    unsigned char *compressed,
+    int maxCompressedBytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes a frame of audio.
+  * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>opus_int16*</tt>: PCM audio in signed 16-bit format (native endian).
+  *          There must be exactly frame_size samples per channel.
+  * @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal
+  * @param [out] compressed <tt>char *</tt>: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long.
+  * @param [in] maxCompressedBytes <tt>int</tt>: Maximum number of bytes to use for compressing the frame
+  *          (can change from one frame to another)
+  * @return Number of bytes written to "compressed".
+  *       If negative, an error has occurred (see error codes). It is IMPORTANT that
+  *       the length returned be somehow transmitted to the decoder. Otherwise, no
+  *       decoding is possible.
+ */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode(
+    OpusCustomEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *compressed,
+    int maxCompressedBytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus custom encoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @see opus_encoderctls
+  */
+OPUS_CUSTOM_EXPORT int opus_custom_encoder_ctl(OpusCustomEncoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1);
+
+
+#if !defined(OPUS_BUILD) || defined(CELT_DECODER_C)
+/* Decoder */
+
+/** Gets the size of an OpusCustomDecoder structure.
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Mode configuration
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @returns size
+  */
+OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_decoder_get_size(
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+/** Initializes a previously allocated decoder state
+  * The memory pointed to by st must be the size returned by opus_custom_decoder_get_size.
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_custom_decoder_create(),opus_custom_decoder_get_size()
+  * To reset a previously initialized state use the OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Contains all the information about the characteristics of
+  *  the stream (must be the same characteristics as used for the
+  *  encoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @return OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT_STATIC int opus_custom_decoder_init(
+    OpusCustomDecoder *st,
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+#endif
+
+
+/** Creates a new decoder state. Each stream needs its own decoder state (can't
+  * be shared across simultaneous streams).
+  * @param [in] mode <tt>OpusCustomMode</tt>: Contains all the information about the characteristics of the
+  *          stream (must be the same characteristics as used for the encoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [out] error <tt>int*</tt>: Returns an error code
+  * @return Newly created decoder state.
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomDecoder *opus_custom_decoder_create(
+    const OpusCustomMode *mode,
+    int channels,
+    int *error
+) OPUS_ARG_NONNULL(1);
+
+/** Destroys a an decoder state.
+  * @param[in] st <tt>OpusCustomDecoder*</tt>: State to be freed.
+  */
+OPUS_CUSTOM_EXPORT void opus_custom_decoder_destroy(OpusCustomDecoder *st);
+
+/** Decode an opus custom frame with floating point output
+  * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>int</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(float)
+  * @param [in] frame_size Number of samples per channel of available space in *pcm.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode_float(
+    OpusCustomDecoder *st,
+    const unsigned char *data,
+    int len,
+    float *pcm,
+    int frame_size
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode an opus custom frame
+  * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>int</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size Number of samples per channel of available space in *pcm.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode(
+    OpusCustomDecoder *st,
+    const unsigned char *data,
+    int len,
+    opus_int16 *pcm,
+    int frame_size
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus custom decoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @see opus_genericctls
+  */
+OPUS_CUSTOM_EXPORT int opus_custom_decoder_ctl(OpusCustomDecoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_CUSTOM_H */

engine/voice_codecs/celt/opus/opus_defines.h

@@ -0,0 +1,799 @@
+/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
+   Written by Jean-Marc Valin and Koen Vos */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus_defines.h
+ * @brief Opus reference implementation constants
+ */
+
+#ifndef OPUS_DEFINES_H
+#define OPUS_DEFINES_H
+
+#include "opus_types.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** @defgroup opus_errorcodes Error codes
+ * @{
+ */
+/** No error @hideinitializer*/
+#define OPUS_OK                0
+/** One or more invalid/out of range arguments @hideinitializer*/
+#define OPUS_BAD_ARG          -1
+/** Not enough bytes allocated in the buffer @hideinitializer*/
+#define OPUS_BUFFER_TOO_SMALL -2
+/** An internal error was detected @hideinitializer*/
+#define OPUS_INTERNAL_ERROR   -3
+/** The compressed data passed is corrupted @hideinitializer*/
+#define OPUS_INVALID_PACKET   -4
+/** Invalid/unsupported request number @hideinitializer*/
+#define OPUS_UNIMPLEMENTED    -5
+/** An encoder or decoder structure is invalid or already freed @hideinitializer*/
+#define OPUS_INVALID_STATE    -6
+/** Memory allocation has failed @hideinitializer*/
+#define OPUS_ALLOC_FAIL       -7
+/**@}*/
+
+/** @cond OPUS_INTERNAL_DOC */
+/**Export control for opus functions */
+
+#ifndef OPUS_EXPORT
+# if defined(WIN32)
+#  if defined(OPUS_BUILD) && defined(DLL_EXPORT)
+#   define OPUS_EXPORT __declspec(dllexport)
+#  else
+#   define OPUS_EXPORT
+#  endif
+# elif defined(__GNUC__) && defined(OPUS_BUILD)
+#  define OPUS_EXPORT __attribute__ ((visibility ("default")))
+# else
+#  define OPUS_EXPORT
+# endif
+#endif
+
+# if !defined(OPUS_GNUC_PREREQ)
+#  if defined(__GNUC__)&&defined(__GNUC_MINOR__)
+#   define OPUS_GNUC_PREREQ(_maj,_min) \
+ ((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min))
+#  else
+#   define OPUS_GNUC_PREREQ(_maj,_min) 0
+#  endif
+# endif
+
+#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
+# if OPUS_GNUC_PREREQ(3,0)
+#  define OPUS_RESTRICT __restrict__
+# elif (defined(_MSC_VER) && _MSC_VER >= 1400)
+#  define OPUS_RESTRICT __restrict
+# else
+#  define OPUS_RESTRICT
+# endif
+#else
+# define OPUS_RESTRICT restrict
+#endif
+
+#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
+# if OPUS_GNUC_PREREQ(2,7)
+#  define OPUS_INLINE __inline__
+# elif (defined(_MSC_VER))
+#  define OPUS_INLINE __inline
+# else
+#  define OPUS_INLINE
+# endif
+#else
+# define OPUS_INLINE inline
+#endif
+
+/**Warning attributes for opus functions
+  * NONNULL is not used in OPUS_BUILD to avoid the compiler optimizing out
+  * some paranoid null checks. */
+#if defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
+# define OPUS_WARN_UNUSED_RESULT __attribute__ ((__warn_unused_result__))
+#else
+# define OPUS_WARN_UNUSED_RESULT
+#endif
+#if !defined(OPUS_BUILD) && defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
+# define OPUS_ARG_NONNULL(_x)  __attribute__ ((__nonnull__(_x)))
+#else
+# define OPUS_ARG_NONNULL(_x)
+#endif
+
+/** These are the actual Encoder CTL ID numbers.
+  * They should not be used directly by applications.
+  * In general, SETs should be even and GETs should be odd.*/
+#define OPUS_SET_APPLICATION_REQUEST         4000
+#define OPUS_GET_APPLICATION_REQUEST         4001
+#define OPUS_SET_BITRATE_REQUEST             4002
+#define OPUS_GET_BITRATE_REQUEST             4003
+#define OPUS_SET_MAX_BANDWIDTH_REQUEST       4004
+#define OPUS_GET_MAX_BANDWIDTH_REQUEST       4005
+#define OPUS_SET_VBR_REQUEST                 4006
+#define OPUS_GET_VBR_REQUEST                 4007
+#define OPUS_SET_BANDWIDTH_REQUEST           4008
+#define OPUS_GET_BANDWIDTH_REQUEST           4009
+#define OPUS_SET_COMPLEXITY_REQUEST          4010
+#define OPUS_GET_COMPLEXITY_REQUEST          4011
+#define OPUS_SET_INBAND_FEC_REQUEST          4012
+#define OPUS_GET_INBAND_FEC_REQUEST          4013
+#define OPUS_SET_PACKET_LOSS_PERC_REQUEST    4014
+#define OPUS_GET_PACKET_LOSS_PERC_REQUEST    4015
+#define OPUS_SET_DTX_REQUEST                 4016
+#define OPUS_GET_DTX_REQUEST                 4017
+#define OPUS_SET_VBR_CONSTRAINT_REQUEST      4020
+#define OPUS_GET_VBR_CONSTRAINT_REQUEST      4021
+#define OPUS_SET_FORCE_CHANNELS_REQUEST      4022
+#define OPUS_GET_FORCE_CHANNELS_REQUEST      4023
+#define OPUS_SET_SIGNAL_REQUEST              4024
+#define OPUS_GET_SIGNAL_REQUEST              4025
+#define OPUS_GET_LOOKAHEAD_REQUEST           4027
+/* #define OPUS_RESET_STATE 4028 */
+#define OPUS_GET_SAMPLE_RATE_REQUEST         4029
+#define OPUS_GET_FINAL_RANGE_REQUEST         4031
+#define OPUS_GET_PITCH_REQUEST               4033
+#define OPUS_SET_GAIN_REQUEST                4034
+#define OPUS_GET_GAIN_REQUEST                4045 /* Should have been 4035 */
+#define OPUS_SET_LSB_DEPTH_REQUEST           4036
+#define OPUS_GET_LSB_DEPTH_REQUEST           4037
+#define OPUS_GET_LAST_PACKET_DURATION_REQUEST 4039
+#define OPUS_SET_EXPERT_FRAME_DURATION_REQUEST 4040
+#define OPUS_GET_EXPERT_FRAME_DURATION_REQUEST 4041
+#define OPUS_SET_PREDICTION_DISABLED_REQUEST 4042
+#define OPUS_GET_PREDICTION_DISABLED_REQUEST 4043
+/* Don't use 4045, it's already taken by OPUS_GET_GAIN_REQUEST */
+#define OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST 4046
+#define OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST 4047
+#define OPUS_GET_IN_DTX_REQUEST              4049
+
+/** Defines for the presence of extended APIs. */
+#define OPUS_HAVE_OPUS_PROJECTION_H
+
+/* Macros to trigger compilation errors when the wrong types are provided to a CTL */
+#define __opus_check_int(x) (((void)((x) == (opus_int32)0)), (opus_int32)(x))
+#define __opus_check_int_ptr(ptr) ((ptr) + ((ptr) - (opus_int32*)(ptr)))
+#define __opus_check_uint_ptr(ptr) ((ptr) + ((ptr) - (opus_uint32*)(ptr)))
+#define __opus_check_val16_ptr(ptr) ((ptr) + ((ptr) - (opus_val16*)(ptr)))
+/** @endcond */
+
+/** @defgroup opus_ctlvalues Pre-defined values for CTL interface
+  * @see opus_genericctls, opus_encoderctls
+  * @{
+  */
+/* Values for the various encoder CTLs */
+#define OPUS_AUTO                           -1000 /**<Auto/default setting @hideinitializer*/
+#define OPUS_BITRATE_MAX                       -1 /**<Maximum bitrate @hideinitializer*/
+
+/** Best for most VoIP/videoconference applications where listening quality and intelligibility matter most
+ * @hideinitializer */
+#define OPUS_APPLICATION_VOIP                2048
+/** Best for broadcast/high-fidelity application where the decoded audio should be as close as possible to the input
+ * @hideinitializer */
+#define OPUS_APPLICATION_AUDIO               2049
+/** Only use when lowest-achievable latency is what matters most. Voice-optimized modes cannot be used.
+ * @hideinitializer */
+#define OPUS_APPLICATION_RESTRICTED_LOWDELAY 2051
+
+#define OPUS_SIGNAL_VOICE                    3001 /**< Signal being encoded is voice */
+#define OPUS_SIGNAL_MUSIC                    3002 /**< Signal being encoded is music */
+#define OPUS_BANDWIDTH_NARROWBAND            1101 /**< 4 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_MEDIUMBAND            1102 /**< 6 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_WIDEBAND              1103 /**< 8 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_SUPERWIDEBAND         1104 /**<12 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_FULLBAND              1105 /**<20 kHz bandpass @hideinitializer*/
+
+#define OPUS_FRAMESIZE_ARG                   5000 /**< Select frame size from the argument (default) */
+#define OPUS_FRAMESIZE_2_5_MS                5001 /**< Use 2.5 ms frames */
+#define OPUS_FRAMESIZE_5_MS                  5002 /**< Use 5 ms frames */
+#define OPUS_FRAMESIZE_10_MS                 5003 /**< Use 10 ms frames */
+#define OPUS_FRAMESIZE_20_MS                 5004 /**< Use 20 ms frames */
+#define OPUS_FRAMESIZE_40_MS                 5005 /**< Use 40 ms frames */
+#define OPUS_FRAMESIZE_60_MS                 5006 /**< Use 60 ms frames */
+#define OPUS_FRAMESIZE_80_MS                 5007 /**< Use 80 ms frames */
+#define OPUS_FRAMESIZE_100_MS                5008 /**< Use 100 ms frames */
+#define OPUS_FRAMESIZE_120_MS                5009 /**< Use 120 ms frames */
+
+/**@}*/
+
+
+/** @defgroup opus_encoderctls Encoder related CTLs
+  *
+  * These are convenience macros for use with the \c opus_encode_ctl
+  * interface. They are used to generate the appropriate series of
+  * arguments for that call, passing the correct type, size and so
+  * on as expected for each particular request.
+  *
+  * Some usage examples:
+  *
+  * @code
+  * int ret;
+  * ret = opus_encoder_ctl(enc_ctx, OPUS_SET_BANDWIDTH(OPUS_AUTO));
+  * if (ret != OPUS_OK) return ret;
+  *
+  * opus_int32 rate;
+  * opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&rate));
+  *
+  * opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
+  * @endcode
+  *
+  * @see opus_genericctls, opus_encoder
+  * @{
+  */
+
+/** Configures the encoder's computational complexity.
+  * The supported range is 0-10 inclusive with 10 representing the highest complexity.
+  * @see OPUS_GET_COMPLEXITY
+  * @param[in] x <tt>opus_int32</tt>: Allowed values: 0-10, inclusive.
+  *
+  * @hideinitializer */
+#define OPUS_SET_COMPLEXITY(x) OPUS_SET_COMPLEXITY_REQUEST, __opus_check_int(x)
+/** Gets the encoder's complexity configuration.
+  * @see OPUS_SET_COMPLEXITY
+  * @param[out] x <tt>opus_int32 *</tt>: Returns a value in the range 0-10,
+  *                                      inclusive.
+  * @hideinitializer */
+#define OPUS_GET_COMPLEXITY(x) OPUS_GET_COMPLEXITY_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the bitrate in the encoder.
+  * Rates from 500 to 512000 bits per second are meaningful, as well as the
+  * special values #OPUS_AUTO and #OPUS_BITRATE_MAX.
+  * The value #OPUS_BITRATE_MAX can be used to cause the codec to use as much
+  * rate as it can, which is useful for controlling the rate by adjusting the
+  * output buffer size.
+  * @see OPUS_GET_BITRATE
+  * @param[in] x <tt>opus_int32</tt>: Bitrate in bits per second. The default
+  *                                   is determined based on the number of
+  *                                   channels and the input sampling rate.
+  * @hideinitializer */
+#define OPUS_SET_BITRATE(x) OPUS_SET_BITRATE_REQUEST, __opus_check_int(x)
+/** Gets the encoder's bitrate configuration.
+  * @see OPUS_SET_BITRATE
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the bitrate in bits per second.
+  *                                      The default is determined based on the
+  *                                      number of channels and the input
+  *                                      sampling rate.
+  * @hideinitializer */
+#define OPUS_GET_BITRATE(x) OPUS_GET_BITRATE_REQUEST, __opus_check_int_ptr(x)
+
+/** Enables or disables variable bitrate (VBR) in the encoder.
+  * The configured bitrate may not be met exactly because frames must
+  * be an integer number of bytes in length.
+  * @see OPUS_GET_VBR
+  * @see OPUS_SET_VBR_CONSTRAINT
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Hard CBR. For LPC/hybrid modes at very low bit-rate, this can
+  *               cause noticeable quality degradation.</dd>
+  * <dt>1</dt><dd>VBR (default). The exact type of VBR is controlled by
+  *               #OPUS_SET_VBR_CONSTRAINT.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_VBR(x) OPUS_SET_VBR_REQUEST, __opus_check_int(x)
+/** Determine if variable bitrate (VBR) is enabled in the encoder.
+  * @see OPUS_SET_VBR
+  * @see OPUS_GET_VBR_CONSTRAINT
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Hard CBR.</dd>
+  * <dt>1</dt><dd>VBR (default). The exact type of VBR may be retrieved via
+  *               #OPUS_GET_VBR_CONSTRAINT.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_VBR(x) OPUS_GET_VBR_REQUEST, __opus_check_int_ptr(x)
+
+/** Enables or disables constrained VBR in the encoder.
+  * This setting is ignored when the encoder is in CBR mode.
+  * @warning Only the MDCT mode of Opus currently heeds the constraint.
+  *  Speech mode ignores it completely, hybrid mode may fail to obey it
+  *  if the LPC layer uses more bitrate than the constraint would have
+  *  permitted.
+  * @see OPUS_GET_VBR_CONSTRAINT
+  * @see OPUS_SET_VBR
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Unconstrained VBR.</dd>
+  * <dt>1</dt><dd>Constrained VBR (default). This creates a maximum of one
+  *               frame of buffering delay assuming a transport with a
+  *               serialization speed of the nominal bitrate.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_VBR_CONSTRAINT(x) OPUS_SET_VBR_CONSTRAINT_REQUEST, __opus_check_int(x)
+/** Determine if constrained VBR is enabled in the encoder.
+  * @see OPUS_SET_VBR_CONSTRAINT
+  * @see OPUS_GET_VBR
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Unconstrained VBR.</dd>
+  * <dt>1</dt><dd>Constrained VBR (default).</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_VBR_CONSTRAINT(x) OPUS_GET_VBR_CONSTRAINT_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures mono/stereo forcing in the encoder.
+  * This can force the encoder to produce packets encoded as either mono or
+  * stereo, regardless of the format of the input audio. This is useful when
+  * the caller knows that the input signal is currently a mono source embedded
+  * in a stereo stream.
+  * @see OPUS_GET_FORCE_CHANNELS
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
+  * <dt>1</dt>         <dd>Forced mono</dd>
+  * <dt>2</dt>         <dd>Forced stereo</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_FORCE_CHANNELS(x) OPUS_SET_FORCE_CHANNELS_REQUEST, __opus_check_int(x)
+/** Gets the encoder's forced channel configuration.
+  * @see OPUS_SET_FORCE_CHANNELS
+  * @param[out] x <tt>opus_int32 *</tt>:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
+  * <dt>1</dt>         <dd>Forced mono</dd>
+  * <dt>2</dt>         <dd>Forced stereo</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_FORCE_CHANNELS(x) OPUS_GET_FORCE_CHANNELS_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the maximum bandpass that the encoder will select automatically.
+  * Applications should normally use this instead of #OPUS_SET_BANDWIDTH
+  * (leaving that set to the default, #OPUS_AUTO). This allows the
+  * application to set an upper bound based on the type of input it is
+  * providing, but still gives the encoder the freedom to reduce the bandpass
+  * when the bitrate becomes too low, for better overall quality.
+  * @see OPUS_GET_MAX_BANDWIDTH
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband (default)</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_MAX_BANDWIDTH(x) OPUS_SET_MAX_BANDWIDTH_REQUEST, __opus_check_int(x)
+
+/** Gets the encoder's configured maximum allowed bandpass.
+  * @see OPUS_SET_MAX_BANDWIDTH
+  * @param[out] x <tt>opus_int32 *</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband (default)</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_MAX_BANDWIDTH(x) OPUS_GET_MAX_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Sets the encoder's bandpass to a specific value.
+  * This prevents the encoder from automatically selecting the bandpass based
+  * on the available bitrate. If an application knows the bandpass of the input
+  * audio it is providing, it should normally use #OPUS_SET_MAX_BANDWIDTH
+  * instead, which still gives the encoder the freedom to reduce the bandpass
+  * when the bitrate becomes too low, for better overall quality.
+  * @see OPUS_GET_BANDWIDTH
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>                    <dd>(default)</dd>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_BANDWIDTH(x) OPUS_SET_BANDWIDTH_REQUEST, __opus_check_int(x)
+
+/** Configures the type of signal being encoded.
+  * This is a hint which helps the encoder's mode selection.
+  * @see OPUS_GET_SIGNAL
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>        <dd>(default)</dd>
+  * <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
+  * <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_SIGNAL(x) OPUS_SET_SIGNAL_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured signal type.
+  * @see OPUS_SET_SIGNAL
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>        <dd>(default)</dd>
+  * <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
+  * <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_SIGNAL(x) OPUS_GET_SIGNAL_REQUEST, __opus_check_int_ptr(x)
+
+
+/** Configures the encoder's intended application.
+  * The initial value is a mandatory argument to the encoder_create function.
+  * @see OPUS_GET_APPLICATION
+  * @param[in] x <tt>opus_int32</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_APPLICATION(x) OPUS_SET_APPLICATION_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured application.
+  * @see OPUS_SET_APPLICATION
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_APPLICATION(x) OPUS_GET_APPLICATION_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the total samples of delay added by the entire codec.
+  * This can be queried by the encoder and then the provided number of samples can be
+  * skipped on from the start of the decoder's output to provide time aligned input
+  * and output. From the perspective of a decoding application the real data begins this many
+  * samples late.
+  *
+  * The decoder contribution to this delay is identical for all decoders, but the
+  * encoder portion of the delay may vary from implementation to implementation,
+  * version to version, or even depend on the encoder's initial configuration.
+  * Applications needing delay compensation should call this CTL rather than
+  * hard-coding a value.
+  * @param[out] x <tt>opus_int32 *</tt>:   Number of lookahead samples
+  * @hideinitializer */
+#define OPUS_GET_LOOKAHEAD(x) OPUS_GET_LOOKAHEAD_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of inband forward error correction (FEC).
+  * @note This is only applicable to the LPC layer
+  * @see OPUS_GET_INBAND_FEC
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Disable inband FEC (default).</dd>
+  * <dt>1</dt><dd>Enable inband FEC.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_INBAND_FEC(x) OPUS_SET_INBAND_FEC_REQUEST, __opus_check_int(x)
+/** Gets encoder's configured use of inband forward error correction.
+  * @see OPUS_SET_INBAND_FEC
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Inband FEC disabled (default).</dd>
+  * <dt>1</dt><dd>Inband FEC enabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_INBAND_FEC(x) OPUS_GET_INBAND_FEC_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's expected packet loss percentage.
+  * Higher values trigger progressively more loss resistant behavior in the encoder
+  * at the expense of quality at a given bitrate in the absence of packet loss, but
+  * greater quality under loss.
+  * @see OPUS_GET_PACKET_LOSS_PERC
+  * @param[in] x <tt>opus_int32</tt>:   Loss percentage in the range 0-100, inclusive (default: 0).
+  * @hideinitializer */
+#define OPUS_SET_PACKET_LOSS_PERC(x) OPUS_SET_PACKET_LOSS_PERC_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured packet loss percentage.
+  * @see OPUS_SET_PACKET_LOSS_PERC
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the configured loss percentage
+  *                                      in the range 0-100, inclusive (default: 0).
+  * @hideinitializer */
+#define OPUS_GET_PACKET_LOSS_PERC(x) OPUS_GET_PACKET_LOSS_PERC_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of discontinuous transmission (DTX).
+  * @note This is only applicable to the LPC layer
+  * @see OPUS_GET_DTX
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Disable DTX (default).</dd>
+  * <dt>1</dt><dd>Enabled DTX.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_DTX(x) OPUS_SET_DTX_REQUEST, __opus_check_int(x)
+/** Gets encoder's configured use of discontinuous transmission.
+  * @see OPUS_SET_DTX
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>DTX disabled (default).</dd>
+  * <dt>1</dt><dd>DTX enabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_DTX(x) OPUS_GET_DTX_REQUEST, __opus_check_int_ptr(x)
+/** Configures the depth of signal being encoded.
+  *
+  * This is a hint which helps the encoder identify silence and near-silence.
+  * It represents the number of significant bits of linear intensity below
+  * which the signal contains ignorable quantization or other noise.
+  *
+  * For example, OPUS_SET_LSB_DEPTH(14) would be an appropriate setting
+  * for G.711 u-law input. OPUS_SET_LSB_DEPTH(16) would be appropriate
+  * for 16-bit linear pcm input with opus_encode_float().
+  *
+  * When using opus_encode() instead of opus_encode_float(), or when libopus
+  * is compiled for fixed-point, the encoder uses the minimum of the value
+  * set here and the value 16.
+  *
+  * @see OPUS_GET_LSB_DEPTH
+  * @param[in] x <tt>opus_int32</tt>: Input precision in bits, between 8 and 24
+  *                                   (default: 24).
+  * @hideinitializer */
+#define OPUS_SET_LSB_DEPTH(x) OPUS_SET_LSB_DEPTH_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured signal depth.
+  * @see OPUS_SET_LSB_DEPTH
+  * @param[out] x <tt>opus_int32 *</tt>: Input precision in bits, between 8 and
+  *                                      24 (default: 24).
+  * @hideinitializer */
+#define OPUS_GET_LSB_DEPTH(x) OPUS_GET_LSB_DEPTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of variable duration frames.
+  * When variable duration is enabled, the encoder is free to use a shorter frame
+  * size than the one requested in the opus_encode*() call.
+  * It is then the user's responsibility
+  * to verify how much audio was encoded by checking the ToC byte of the encoded
+  * packet. The part of the audio that was not encoded needs to be resent to the
+  * encoder for the next call. Do not use this option unless you <b>really</b>
+  * know what you are doing.
+  * @see OPUS_GET_EXPERT_FRAME_DURATION
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
+  * <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_80_MS</dt><dd>Use 80 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_100_MS</dt><dd>Use 100 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_120_MS</dt><dd>Use 120 ms frames.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_EXPERT_FRAME_DURATION(x) OPUS_SET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured use of variable duration frames.
+  * @see OPUS_SET_EXPERT_FRAME_DURATION
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
+  * <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_80_MS</dt><dd>Use 80 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_100_MS</dt><dd>Use 100 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_120_MS</dt><dd>Use 120 ms frames.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_EXPERT_FRAME_DURATION(x) OPUS_GET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int_ptr(x)
+
+/** If set to 1, disables almost all use of prediction, making frames almost
+  * completely independent. This reduces quality.
+  * @see OPUS_GET_PREDICTION_DISABLED
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Enable prediction (default).</dd>
+  * <dt>1</dt><dd>Disable prediction.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_PREDICTION_DISABLED(x) OPUS_SET_PREDICTION_DISABLED_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured prediction status.
+  * @see OPUS_SET_PREDICTION_DISABLED
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Prediction enabled (default).</dd>
+  * <dt>1</dt><dd>Prediction disabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_PREDICTION_DISABLED(x) OPUS_GET_PREDICTION_DISABLED_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_genericctls Generic CTLs
+  *
+  * These macros are used with the \c opus_decoder_ctl and
+  * \c opus_encoder_ctl calls to generate a particular
+  * request.
+  *
+  * When called on an \c OpusDecoder they apply to that
+  * particular decoder instance. When called on an
+  * \c OpusEncoder they apply to the corresponding setting
+  * on that encoder instance, if present.
+  *
+  * Some usage examples:
+  *
+  * @code
+  * int ret;
+  * opus_int32 pitch;
+  * ret = opus_decoder_ctl(dec_ctx, OPUS_GET_PITCH(&pitch));
+  * if (ret == OPUS_OK) return ret;
+  *
+  * opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
+  * opus_decoder_ctl(dec_ctx, OPUS_RESET_STATE);
+  *
+  * opus_int32 enc_bw, dec_bw;
+  * opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&enc_bw));
+  * opus_decoder_ctl(dec_ctx, OPUS_GET_BANDWIDTH(&dec_bw));
+  * if (enc_bw != dec_bw) {
+  *   printf("packet bandwidth mismatch!\n");
+  * }
+  * @endcode
+  *
+  * @see opus_encoder, opus_decoder_ctl, opus_encoder_ctl, opus_decoderctls, opus_encoderctls
+  * @{
+  */
+
+/** Resets the codec state to be equivalent to a freshly initialized state.
+  * This should be called when switching streams in order to prevent
+  * the back to back decoding from giving different results from
+  * one at a time decoding.
+  * @hideinitializer */
+#define OPUS_RESET_STATE 4028
+
+/** Gets the final state of the codec's entropy coder.
+  * This is used for testing purposes,
+  * The encoder and decoder state should be identical after coding a payload
+  * (assuming no data corruption or software bugs)
+  *
+  * @param[out] x <tt>opus_uint32 *</tt>: Entropy coder state
+  *
+  * @hideinitializer */
+#define OPUS_GET_FINAL_RANGE(x) OPUS_GET_FINAL_RANGE_REQUEST, __opus_check_uint_ptr(x)
+
+/** Gets the encoder's configured bandpass or the decoder's last bandpass.
+  * @see OPUS_SET_BANDWIDTH
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>                    <dd>(default)</dd>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_BANDWIDTH(x) OPUS_GET_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the sampling rate the encoder or decoder was initialized with.
+  * This simply returns the <code>Fs</code> value passed to opus_encoder_init()
+  * or opus_decoder_init().
+  * @param[out] x <tt>opus_int32 *</tt>: Sampling rate of encoder or decoder.
+  * @hideinitializer
+  */
+#define OPUS_GET_SAMPLE_RATE(x) OPUS_GET_SAMPLE_RATE_REQUEST, __opus_check_int_ptr(x)
+
+/** If set to 1, disables the use of phase inversion for intensity stereo,
+  * improving the quality of mono downmixes, but slightly reducing normal
+  * stereo quality. Disabling phase inversion in the decoder does not comply
+  * with RFC 6716, although it does not cause any interoperability issue and
+  * is expected to become part of the Opus standard once RFC 6716 is updated
+  * by draft-ietf-codec-opus-update.
+  * @see OPUS_GET_PHASE_INVERSION_DISABLED
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Enable phase inversion (default).</dd>
+  * <dt>1</dt><dd>Disable phase inversion.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_PHASE_INVERSION_DISABLED(x) OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured phase inversion status.
+  * @see OPUS_SET_PHASE_INVERSION_DISABLED
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Stereo phase inversion enabled (default).</dd>
+  * <dt>1</dt><dd>Stereo phase inversion disabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_PHASE_INVERSION_DISABLED(x) OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST, __opus_check_int_ptr(x)
+/** Gets the DTX state of the encoder.
+  * Returns whether the last encoded frame was either a comfort noise update
+  * during DTX or not encoded because of DTX.
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>The encoder is not in DTX.</dd>
+  * <dt>1</dt><dd>The encoder is in DTX.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_IN_DTX(x) OPUS_GET_IN_DTX_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_decoderctls Decoder related CTLs
+  * @see opus_genericctls, opus_encoderctls, opus_decoder
+  * @{
+  */
+
+/** Configures decoder gain adjustment.
+  * Scales the decoded output by a factor specified in Q8 dB units.
+  * This has a maximum range of -32768 to 32767 inclusive, and returns
+  * OPUS_BAD_ARG otherwise. The default is zero indicating no adjustment.
+  * This setting survives decoder reset.
+  *
+  * gain = pow(10, x/(20.0*256))
+  *
+  * @param[in] x <tt>opus_int32</tt>:   Amount to scale PCM signal by in Q8 dB units.
+  * @hideinitializer */
+#define OPUS_SET_GAIN(x) OPUS_SET_GAIN_REQUEST, __opus_check_int(x)
+/** Gets the decoder's configured gain adjustment. @see OPUS_SET_GAIN
+  *
+  * @param[out] x <tt>opus_int32 *</tt>: Amount to scale PCM signal by in Q8 dB units.
+  * @hideinitializer */
+#define OPUS_GET_GAIN(x) OPUS_GET_GAIN_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the duration (in samples) of the last packet successfully decoded or concealed.
+  * @param[out] x <tt>opus_int32 *</tt>: Number of samples (at current sampling rate).
+  * @hideinitializer */
+#define OPUS_GET_LAST_PACKET_DURATION(x) OPUS_GET_LAST_PACKET_DURATION_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the pitch of the last decoded frame, if available.
+  * This can be used for any post-processing algorithm requiring the use of pitch,
+  * e.g. time stretching/shortening. If the last frame was not voiced, or if the
+  * pitch was not coded in the frame, then zero is returned.
+  *
+  * This CTL is only implemented for decoder instances.
+  *
+  * @param[out] x <tt>opus_int32 *</tt>: pitch period at 48 kHz (or 0 if not available)
+  *
+  * @hideinitializer */
+#define OPUS_GET_PITCH(x) OPUS_GET_PITCH_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_libinfo Opus library information functions
+  * @{
+  */
+
+/** Converts an opus error code into a human readable string.
+  *
+  * @param[in] error <tt>int</tt>: Error number
+  * @returns Error string
+  */
+OPUS_EXPORT const char *opus_strerror(int error);
+
+/** Gets the libopus version string.
+  *
+  * Applications may look for the substring "-fixed" in the version string to
+  * determine whether they have a fixed-point or floating-point build at
+  * runtime.
+  *
+  * @returns Version string
+  */
+OPUS_EXPORT const char *opus_get_version_string(void);
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_DEFINES_H */

engine/voice_codecs/celt/opus/opus_multistream.h

@@ -0,0 +1,660 @@
+/* Copyright (c) 2011 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus_multistream.h
+ * @brief Opus reference implementation multistream API
+ */
+
+#ifndef OPUS_MULTISTREAM_H
+#define OPUS_MULTISTREAM_H
+
+#include "opus.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** @cond OPUS_INTERNAL_DOC */
+
+/** Macros to trigger compilation errors when the wrong types are provided to a
+  * CTL. */
+/**@{*/
+#define __opus_check_encstate_ptr(ptr) ((ptr) + ((ptr) - (OpusEncoder**)(ptr)))
+#define __opus_check_decstate_ptr(ptr) ((ptr) + ((ptr) - (OpusDecoder**)(ptr)))
+/**@}*/
+
+/** These are the actual encoder and decoder CTL ID numbers.
+  * They should not be used directly by applications.
+  * In general, SETs should be even and GETs should be odd.*/
+/**@{*/
+#define OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST 5120
+#define OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST 5122
+/**@}*/
+
+/** @endcond */
+
+/** @defgroup opus_multistream_ctls Multistream specific encoder and decoder CTLs
+  *
+  * These are convenience macros that are specific to the
+  * opus_multistream_encoder_ctl() and opus_multistream_decoder_ctl()
+  * interface.
+  * The CTLs from @ref opus_genericctls, @ref opus_encoderctls, and
+  * @ref opus_decoderctls may be applied to a multistream encoder or decoder as
+  * well.
+  * In addition, you may retrieve the encoder or decoder state for an specific
+  * stream via #OPUS_MULTISTREAM_GET_ENCODER_STATE or
+  * #OPUS_MULTISTREAM_GET_DECODER_STATE and apply CTLs to it individually.
+  */
+/**@{*/
+
+/** Gets the encoder state for an individual stream of a multistream encoder.
+  * @param[in] x <tt>opus_int32</tt>: The index of the stream whose encoder you
+  *                                   wish to retrieve.
+  *                                   This must be non-negative and less than
+  *                                   the <code>streams</code> parameter used
+  *                                   to initialize the encoder.
+  * @param[out] y <tt>OpusEncoder**</tt>: Returns a pointer to the given
+  *                                       encoder state.
+  * @retval OPUS_BAD_ARG The index of the requested stream was out of range.
+  * @hideinitializer
+  */
+#define OPUS_MULTISTREAM_GET_ENCODER_STATE(x,y) OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST, __opus_check_int(x), __opus_check_encstate_ptr(y)
+
+/** Gets the decoder state for an individual stream of a multistream decoder.
+  * @param[in] x <tt>opus_int32</tt>: The index of the stream whose decoder you
+  *                                   wish to retrieve.
+  *                                   This must be non-negative and less than
+  *                                   the <code>streams</code> parameter used
+  *                                   to initialize the decoder.
+  * @param[out] y <tt>OpusDecoder**</tt>: Returns a pointer to the given
+  *                                       decoder state.
+  * @retval OPUS_BAD_ARG The index of the requested stream was out of range.
+  * @hideinitializer
+  */
+#define OPUS_MULTISTREAM_GET_DECODER_STATE(x,y) OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST, __opus_check_int(x), __opus_check_decstate_ptr(y)
+
+/**@}*/
+
+/** @defgroup opus_multistream Opus Multistream API
+  * @{
+  *
+  * The multistream API allows individual Opus streams to be combined into a
+  * single packet, enabling support for up to 255 channels. Unlike an
+  * elementary Opus stream, the encoder and decoder must negotiate the channel
+  * configuration before the decoder can successfully interpret the data in the
+  * packets produced by the encoder. Some basic information, such as packet
+  * duration, can be computed without any special negotiation.
+  *
+  * The format for multistream Opus packets is defined in
+  * <a href="https://tools.ietf.org/html/rfc7845">RFC 7845</a>
+  * and is based on the self-delimited Opus framing described in Appendix B of
+  * <a href="https://tools.ietf.org/html/rfc6716">RFC 6716</a>.
+  * Normal Opus packets are just a degenerate case of multistream Opus packets,
+  * and can be encoded or decoded with the multistream API by setting
+  * <code>streams</code> to <code>1</code> when initializing the encoder or
+  * decoder.
+  *
+  * Multistream Opus streams can contain up to 255 elementary Opus streams.
+  * These may be either "uncoupled" or "coupled", indicating that the decoder
+  * is configured to decode them to either 1 or 2 channels, respectively.
+  * The streams are ordered so that all coupled streams appear at the
+  * beginning.
+  *
+  * A <code>mapping</code> table defines which decoded channel <code>i</code>
+  * should be used for each input/output (I/O) channel <code>j</code>. This table is
+  * typically provided as an unsigned char array.
+  * Let <code>i = mapping[j]</code> be the index for I/O channel <code>j</code>.
+  * If <code>i < 2*coupled_streams</code>, then I/O channel <code>j</code> is
+  * encoded as the left channel of stream <code>(i/2)</code> if <code>i</code>
+  * is even, or  as the right channel of stream <code>(i/2)</code> if
+  * <code>i</code> is odd. Otherwise, I/O channel <code>j</code> is encoded as
+  * mono in stream <code>(i - coupled_streams)</code>, unless it has the special
+  * value 255, in which case it is omitted from the encoding entirely (the
+  * decoder will reproduce it as silence). Each value <code>i</code> must either
+  * be the special value 255 or be less than <code>streams + coupled_streams</code>.
+  *
+  * The output channels specified by the encoder
+  * should use the
+  * <a href="https://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-810004.3.9">Vorbis
+  * channel ordering</a>. A decoder may wish to apply an additional permutation
+  * to the mapping the encoder used to achieve a different output channel
+  * order (e.g. for outputing in WAV order).
+  *
+  * Each multistream packet contains an Opus packet for each stream, and all of
+  * the Opus packets in a single multistream packet must have the same
+  * duration. Therefore the duration of a multistream packet can be extracted
+  * from the TOC sequence of the first stream, which is located at the
+  * beginning of the packet, just like an elementary Opus stream:
+  *
+  * @code
+  * int nb_samples;
+  * int nb_frames;
+  * nb_frames = opus_packet_get_nb_frames(data, len);
+  * if (nb_frames < 1)
+  *   return nb_frames;
+  * nb_samples = opus_packet_get_samples_per_frame(data, 48000) * nb_frames;
+  * @endcode
+  *
+  * The general encoding and decoding process proceeds exactly the same as in
+  * the normal @ref opus_encoder and @ref opus_decoder APIs.
+  * See their documentation for an overview of how to use the corresponding
+  * multistream functions.
+  */
+
+/** Opus multistream encoder state.
+  * This contains the complete state of a multistream Opus encoder.
+  * It is position independent and can be freely copied.
+  * @see opus_multistream_encoder_create
+  * @see opus_multistream_encoder_init
+  */
+typedef struct OpusMSEncoder OpusMSEncoder;
+
+/** Opus multistream decoder state.
+  * This contains the complete state of a multistream Opus decoder.
+  * It is position independent and can be freely copied.
+  * @see opus_multistream_decoder_create
+  * @see opus_multistream_decoder_init
+  */
+typedef struct OpusMSDecoder OpusMSDecoder;
+
+/**\name Multistream encoder functions */
+/**@{*/
+
+/** Gets the size of an OpusMSEncoder structure.
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_encoder_get_size(
+      int streams,
+      int coupled_streams
+);
+
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_surround_encoder_get_size(
+      int channels,
+      int mapping_family
+);
+
+
+/** Allocates and initializes a multistream encoder state.
+  * Call opus_multistream_encoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than the number of channels.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than the number of input channels.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    encoded channels to input channels, as described in
+  *                    @ref opus_multistream. As an extra constraint, the
+  *                    multistream encoder does not allow encoding coupled
+  *                    streams for which one channel is unused since this
+  *                    is never a good idea.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_encoder_create(
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int application,
+      int *error
+) OPUS_ARG_NONNULL(5);
+
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_surround_encoder_create(
+      opus_int32 Fs,
+      int channels,
+      int mapping_family,
+      int *streams,
+      int *coupled_streams,
+      unsigned char *mapping,
+      int application,
+      int *error
+) OPUS_ARG_NONNULL(4) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6);
+
+/** Initialize a previously allocated multistream encoder state.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_multistream_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_multistream_encoder_create
+  * @see opus_multistream_encoder_get_size
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than the number of channels.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than the number of input channels.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    encoded channels to input channels, as described in
+  *                    @ref opus_multistream. As an extra constraint, the
+  *                    multistream encoder does not allow encoding coupled
+  *                    streams for which one channel is unused since this
+  *                    is never a good idea.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_multistream_encoder_init(
+      OpusMSEncoder *st,
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int application
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
+
+OPUS_EXPORT int opus_multistream_surround_encoder_init(
+      OpusMSEncoder *st,
+      opus_int32 Fs,
+      int channels,
+      int mapping_family,
+      int *streams,
+      int *coupled_streams,
+      unsigned char *mapping,
+      int application
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6) OPUS_ARG_NONNULL(7);
+
+/** Encodes a multistream Opus frame.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param[in] pcm <tt>const opus_int16*</tt>: The input signal as interleaved
+  *                                            samples.
+  *                                            This must contain
+  *                                            <code>frame_size*channels</code>
+  *                                            samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode(
+    OpusMSEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes a multistream Opus frame from floating point input.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param[in] pcm <tt>const float*</tt>: The input signal as interleaved
+  *                                       samples with a normal range of
+  *                                       +/-1.0.
+  *                                       Samples with a range beyond +/-1.0
+  *                                       are supported but will be clipped by
+  *                                       decoders using the integer API and
+  *                                       should only be used if it is known
+  *                                       that the far end supports extended
+  *                                       dynamic range.
+  *                                       This must contain
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode_float(
+      OpusMSEncoder *st,
+      const float *pcm,
+      int frame_size,
+      unsigned char *data,
+      opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Frees an <code>OpusMSEncoder</code> allocated by
+  * opus_multistream_encoder_create().
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to be freed.
+  */
+OPUS_EXPORT void opus_multistream_encoder_destroy(OpusMSEncoder *st);
+
+/** Perform a CTL function on a multistream Opus encoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_encoderctls, or @ref opus_multistream_ctls.
+  * @see opus_genericctls
+  * @see opus_encoderctls
+  * @see opus_multistream_ctls
+  */
+OPUS_EXPORT int opus_multistream_encoder_ctl(OpusMSEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/**@}*/
+
+/**\name Multistream decoder functions */
+/**@{*/
+
+/** Gets the size of an <code>OpusMSDecoder</code> structure.
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_decoder_get_size(
+      int streams,
+      int coupled_streams
+);
+
+/** Allocates and initializes a multistream decoder state.
+  * Call opus_multistream_decoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    coded channels to output channels, as described in
+  *                    @ref opus_multistream.
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSDecoder *opus_multistream_decoder_create(
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int *error
+) OPUS_ARG_NONNULL(5);
+
+/** Intialize a previously allocated decoder state object.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_multistream_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_multistream_decoder_create
+  * @see opus_multistream_deocder_get_size
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    coded channels to output channels, as described in
+  *                    @ref opus_multistream.
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_multistream_decoder_init(
+      OpusMSDecoder *st,
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
+
+/** Decode a multistream Opus packet.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode(
+    OpusMSDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int16 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode a multistream Opus packet with floating point output.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode_float(
+    OpusMSDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    float *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on a multistream Opus decoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_decoderctls, or @ref opus_multistream_ctls.
+  * @see opus_genericctls
+  * @see opus_decoderctls
+  * @see opus_multistream_ctls
+  */
+OPUS_EXPORT int opus_multistream_decoder_ctl(OpusMSDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/** Frees an <code>OpusMSDecoder</code> allocated by
+  * opus_multistream_decoder_create().
+  * @param st <tt>OpusMSDecoder</tt>: Multistream decoder state to be freed.
+  */
+OPUS_EXPORT void opus_multistream_decoder_destroy(OpusMSDecoder *st);
+
+/**@}*/
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_MULTISTREAM_H */

engine/voice_codecs/celt/opus/opus_projection.h

@@ -0,0 +1,568 @@
+/* Copyright (c) 2017 Google Inc.
+   Written by Andrew Allen */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus_projection.h
+ * @brief Opus projection reference API
+ */
+
+#ifndef OPUS_PROJECTION_H
+#define OPUS_PROJECTION_H
+
+#include "opus_multistream.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** @cond OPUS_INTERNAL_DOC */
+
+/** These are the actual encoder and decoder CTL ID numbers.
+  * They should not be used directly by applications.c
+  * In general, SETs should be even and GETs should be odd.*/
+/**@{*/
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN_REQUEST    6001
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE_REQUEST    6003
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_REQUEST         6005
+/**@}*/
+
+
+/** @endcond */
+
+/** @defgroup opus_projection_ctls Projection specific encoder and decoder CTLs
+  *
+  * These are convenience macros that are specific to the
+  * opus_projection_encoder_ctl() and opus_projection_decoder_ctl()
+  * interface.
+  * The CTLs from @ref opus_genericctls, @ref opus_encoderctls,
+  * @ref opus_decoderctls, and @ref opus_multistream_ctls may be applied to a
+  * projection encoder or decoder as well.
+  */
+/**@{*/
+
+/** Gets the gain (in dB. S7.8-format) of the demixing matrix from the encoder.
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the gain (in dB. S7.8-format)
+  *                                      of the demixing matrix.
+  * @hideinitializer
+  */
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN(x) OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN_REQUEST, __opus_check_int_ptr(x)
+
+
+/** Gets the size in bytes of the demixing matrix from the encoder.
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the size in bytes of the
+  *                                      demixing matrix.
+  * @hideinitializer
+  */
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE(x) OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE_REQUEST, __opus_check_int_ptr(x)
+
+
+/** Copies the demixing matrix to the supplied pointer location.
+  * @param[out] x <tt>unsigned char *</tt>: Returns the demixing matrix to the
+  *                                         supplied pointer location.
+  * @param y <tt>opus_int32</tt>: The size in bytes of the reserved memory at the
+  *                              pointer location.
+  * @hideinitializer
+  */
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX(x,y) OPUS_PROJECTION_GET_DEMIXING_MATRIX_REQUEST, x, __opus_check_int(y)
+
+
+/**@}*/
+
+/** Opus projection encoder state.
+ * This contains the complete state of a projection Opus encoder.
+ * It is position independent and can be freely copied.
+ * @see opus_projection_ambisonics_encoder_create
+ */
+typedef struct OpusProjectionEncoder OpusProjectionEncoder;
+
+
+/** Opus projection decoder state.
+  * This contains the complete state of a projection Opus decoder.
+  * It is position independent and can be freely copied.
+  * @see opus_projection_decoder_create
+  * @see opus_projection_decoder_init
+  */
+typedef struct OpusProjectionDecoder OpusProjectionDecoder;
+
+
+/**\name Projection encoder functions */
+/**@{*/
+
+/** Gets the size of an OpusProjectionEncoder structure.
+  * @param channels <tt>int</tt>: The total number of input channels to encode.
+  *                               This must be no more than 255.
+  * @param mapping_family <tt>int</tt>: The mapping family to use for selecting
+  *                                     the appropriate projection.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_projection_ambisonics_encoder_get_size(
+    int channels,
+    int mapping_family
+);
+
+
+/** Allocates and initializes a projection encoder state.
+  * Call opus_projection_encoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param mapping_family <tt>int</tt>: The mapping family to use for selecting
+  *                                     the appropriate projection.
+  * @param[out] streams <tt>int *</tt>: The total number of streams that will
+  *                                     be encoded from the input.
+  * @param[out] coupled_streams <tt>int *</tt>: Number of coupled (2 channel)
+  *                                 streams that will be encoded from the input.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusProjectionEncoder *opus_projection_ambisonics_encoder_create(
+    opus_int32 Fs,
+    int channels,
+    int mapping_family,
+    int *streams,
+    int *coupled_streams,
+    int application,
+    int *error
+) OPUS_ARG_NONNULL(4) OPUS_ARG_NONNULL(5);
+
+
+/** Initialize a previously allocated projection encoder state.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_projection_ambisonics_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_projection_ambisonics_encoder_create
+  * @see opus_projection_ambisonics_encoder_get_size
+  * @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than the number of channels.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than the number of input channels.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_projection_ambisonics_encoder_init(
+    OpusProjectionEncoder *st,
+    opus_int32 Fs,
+    int channels,
+    int mapping_family,
+    int *streams,
+    int *coupled_streams,
+    int application
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6);
+
+
+/** Encodes a projection Opus frame.
+  * @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
+  * @param[in] pcm <tt>const opus_int16*</tt>: The input signal as interleaved
+  *                                            samples.
+  *                                            This must contain
+  *                                            <code>frame_size*channels</code>
+  *                                            samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_encode(
+    OpusProjectionEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+
+/** Encodes a projection Opus frame from floating point input.
+  * @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
+  * @param[in] pcm <tt>const float*</tt>: The input signal as interleaved
+  *                                       samples with a normal range of
+  *                                       +/-1.0.
+  *                                       Samples with a range beyond +/-1.0
+  *                                       are supported but will be clipped by
+  *                                       decoders using the integer API and
+  *                                       should only be used if it is known
+  *                                       that the far end supports extended
+  *                                       dynamic range.
+  *                                       This must contain
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_encode_float(
+    OpusProjectionEncoder *st,
+    const float *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+
+/** Frees an <code>OpusProjectionEncoder</code> allocated by
+  * opus_projection_ambisonics_encoder_create().
+  * @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state to be freed.
+  */
+OPUS_EXPORT void opus_projection_encoder_destroy(OpusProjectionEncoder *st);
+
+
+/** Perform a CTL function on a projection Opus encoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_encoderctls, @ref opus_multistream_ctls, or
+  *                @ref opus_projection_ctls
+  * @see opus_genericctls
+  * @see opus_encoderctls
+  * @see opus_multistream_ctls
+  * @see opus_projection_ctls
+  */
+OPUS_EXPORT int opus_projection_encoder_ctl(OpusProjectionEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+
+/**@}*/
+
+/**\name Projection decoder functions */
+/**@{*/
+
+/** Gets the size of an <code>OpusProjectionDecoder</code> structure.
+  * @param channels <tt>int</tt>: The total number of output channels.
+  *                               This must be no more than 255.
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_projection_decoder_get_size(
+    int channels,
+    int streams,
+    int coupled_streams
+);
+
+
+/** Allocates and initializes a projection decoder state.
+  * Call opus_projection_decoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] demixing_matrix <tt>const unsigned char[demixing_matrix_size]</tt>: Demixing matrix
+  *                         that mapping from coded channels to output channels,
+  *                         as described in @ref opus_projection and
+  *                         @ref opus_projection_ctls.
+  * @param demixing_matrix_size <tt>opus_int32</tt>: The size in bytes of the
+  *                                                  demixing matrix, as
+  *                                                  described in @ref
+  *                                                  opus_projection_ctls.
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusProjectionDecoder *opus_projection_decoder_create(
+    opus_int32 Fs,
+    int channels,
+    int streams,
+    int coupled_streams,
+    unsigned char *demixing_matrix,
+    opus_int32 demixing_matrix_size,
+    int *error
+) OPUS_ARG_NONNULL(5);
+
+
+/** Intialize a previously allocated projection decoder state object.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_projection_decoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_projection_decoder_create
+  * @see opus_projection_deocder_get_size
+  * @param st <tt>OpusProjectionDecoder*</tt>: Projection encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] demixing_matrix <tt>const unsigned char[demixing_matrix_size]</tt>: Demixing matrix
+  *                         that mapping from coded channels to output channels,
+  *                         as described in @ref opus_projection and
+  *                         @ref opus_projection_ctls.
+  * @param demixing_matrix_size <tt>opus_int32</tt>: The size in bytes of the
+  *                                                  demixing matrix, as
+  *                                                  described in @ref
+  *                                                  opus_projection_ctls.
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_projection_decoder_init(
+    OpusProjectionDecoder *st,
+    opus_int32 Fs,
+    int channels,
+    int streams,
+    int coupled_streams,
+    unsigned char *demixing_matrix,
+    opus_int32 demixing_matrix_size
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
+
+
+/** Decode a projection Opus packet.
+  * @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_decode(
+    OpusProjectionDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int16 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+
+/** Decode a projection Opus packet with floating point output.
+  * @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_decode_float(
+    OpusProjectionDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    float *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+
+/** Perform a CTL function on a projection Opus decoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_decoderctls, @ref opus_multistream_ctls, or
+  *                @ref opus_projection_ctls.
+  * @see opus_genericctls
+  * @see opus_decoderctls
+  * @see opus_multistream_ctls
+  * @see opus_projection_ctls
+  */
+OPUS_EXPORT int opus_projection_decoder_ctl(OpusProjectionDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+
+/** Frees an <code>OpusProjectionDecoder</code> allocated by
+  * opus_projection_decoder_create().
+  * @param st <tt>OpusProjectionDecoder</tt>: Projection decoder state to be freed.
+  */
+OPUS_EXPORT void opus_projection_decoder_destroy(OpusProjectionDecoder *st);
+
+
+/**@}*/
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_PROJECTION_H */

engine/voice_codecs/celt/opus/opus_types.h

@@ -0,0 +1,166 @@
+/* (C) COPYRIGHT 1994-2002 Xiph.Org Foundation */
+/* Modified by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+/* opus_types.h based on ogg_types.h from libogg */
+
+/**
+   @file opus_types.h
+   @brief Opus reference implementation types
+*/
+#ifndef OPUS_TYPES_H
+#define OPUS_TYPES_H
+
+#define opus_int         int                     /* used for counters etc; at least 16 bits */
+#define opus_int64       long long
+#define opus_int8        signed char
+
+#define opus_uint        unsigned int            /* used for counters etc; at least 16 bits */
+#define opus_uint64      unsigned long long
+#define opus_uint8       unsigned char
+
+/* Use the real stdint.h if it's there (taken from Paul Hsieh's pstdint.h) */
+#if (defined(__STDC__) && __STDC__ && defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L) || (defined(__GNUC__) && (defined(_STDINT_H) || defined(_STDINT_H_)) || defined (HAVE_STDINT_H))
+#include <stdint.h>
+#  undef opus_int64
+#  undef opus_int8
+#  undef opus_uint64
+#  undef opus_uint8
+   typedef int8_t opus_int8;
+   typedef uint8_t opus_uint8;
+   typedef int16_t opus_int16;
+   typedef uint16_t opus_uint16;
+   typedef int32_t opus_int32;
+   typedef uint32_t opus_uint32;
+   typedef int64_t opus_int64;
+   typedef uint64_t opus_uint64;
+#elif defined(_WIN32)
+
+#  if defined(__CYGWIN__)
+#    include <_G_config.h>
+     typedef _G_int32_t opus_int32;
+     typedef _G_uint32_t opus_uint32;
+     typedef _G_int16 opus_int16;
+     typedef _G_uint16 opus_uint16;
+#  elif defined(__MINGW32__)
+     typedef short opus_int16;
+     typedef unsigned short opus_uint16;
+     typedef int opus_int32;
+     typedef unsigned int opus_uint32;
+#  elif defined(__MWERKS__)
+     typedef int opus_int32;
+     typedef unsigned int opus_uint32;
+     typedef short opus_int16;
+     typedef unsigned short opus_uint16;
+#  else
+     /* MSVC/Borland */
+     typedef __int32 opus_int32;
+     typedef unsigned __int32 opus_uint32;
+     typedef __int16 opus_int16;
+     typedef unsigned __int16 opus_uint16;
+#  endif
+
+#elif defined(__MACOS__)
+
+#  include <sys/types.h>
+   typedef SInt16 opus_int16;
+   typedef UInt16 opus_uint16;
+   typedef SInt32 opus_int32;
+   typedef UInt32 opus_uint32;
+
+#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */
+
+#  include <sys/types.h>
+   typedef int16_t opus_int16;
+   typedef u_int16_t opus_uint16;
+   typedef int32_t opus_int32;
+   typedef u_int32_t opus_uint32;
+
+#elif defined(__BEOS__)
+
+   /* Be */
+#  include <inttypes.h>
+   typedef int16 opus_int16;
+   typedef u_int16 opus_uint16;
+   typedef int32_t opus_int32;
+   typedef u_int32_t opus_uint32;
+
+#elif defined (__EMX__)
+
+   /* OS/2 GCC */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined (DJGPP)
+
+   /* DJGPP */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined(R5900)
+
+   /* PS2 EE */
+   typedef int opus_int32;
+   typedef unsigned opus_uint32;
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+
+#elif defined(__SYMBIAN32__)
+
+   /* Symbian GCC */
+   typedef signed short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef signed int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined(CONFIG_TI_C54X) || defined (CONFIG_TI_C55X)
+
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef long opus_int32;
+   typedef unsigned long opus_uint32;
+
+#elif defined(CONFIG_TI_C6X)
+
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#else
+
+   /* Give up, take a reasonable guess */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#endif
+
+#endif  /* OPUS_TYPES_H */

engine/voice_codecs/celt/voiceencoder_celt.cpp

@@ -41,13 +41,13 @@ damage. */
 #include "ivoicecodec.h"
 #include "iframeencoder.h"
 
-#ifdef POSIX
+/*#ifdef POSIX
 #include "source/osx/config.h"
 #else
 #include "source/msvc/config.h"
-#endif
+#endif*/
 #include <stdio.h>
-#include "celt.h"
+#include <celt/celt.h>
 
 
 // NOTE: This has to be the last file included!
@@ -67,21 +67,10 @@ struct celt_versions
 
 celt_versions g_CeltVersion[CELT_VERSION] = 
 {
-	{
-		44100, 256, 120 
-	},
-
-	{
-		22050, 120, 60
-	},
-
-	{
-		22050, 256, 60
-	},
-
-	{
-		22050, 512, 64
-	},
+	{44100, 256, 120},
+	{22050, 120, 60},
+	{22050, 256, 60},
+	{22050, 512, 64},
 };
 
 class VoiceEncoder_Celt : public IFrameEncoder

engine/voice_codecs/celt/wscript

@@ -0,0 +1,54 @@
+#! /usr/bin/env python
+# encoding: utf-8
+
+from waflib import Utils
+import os
+
+top = '.'
+PROJECT_NAME = 'vaudio_celt'
+
+def options(opt):
+	# stub
+	return
+
+def configure(conf):
+	conf.define('SPEEX_EXPORTS',1)
+#	conf.define('NO_HOOK_MALLOC',1)
+
+def build(bld):
+	source = [
+		'voiceencoder_celt.cpp',
+		'../frame_encoder/voice_codec_frame.cpp',
+		'../../../tier1/interface.cpp'
+	]
+
+	includes = [
+		'.',
+		'../frame_encoder',
+		'../../../public',
+		'../../../public/tier1',
+		'../../../public/tier0',
+		'../../',
+		'../../../common',
+		'../../audio/public'
+	]
+
+	defines = []
+
+	libs = ['tier0','tier1','vstdlib','CELT']
+
+	install_path = bld.env.LIBDIR
+
+	bld.shlib(
+		source   = source,
+		target   = PROJECT_NAME,
+		name     = PROJECT_NAME,
+		features = 'c cxx',
+		includes = includes,
+		defines  = defines,
+		use      = libs,
+		install_path = install_path,
+		subsystem = bld.env.MSVC_SUBSYSTEM,
+		idx      = bld.get_taskgen_count()
+	)
+

engine/voice_codecs/opus/opus/opus.h

@@ -0,0 +1,981 @@
+/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
+   Written by Jean-Marc Valin and Koen Vos */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus.h
+ * @brief Opus reference implementation API
+ */
+
+#ifndef OPUS_H
+#define OPUS_H
+
+#include "opus_types.h"
+#include "opus_defines.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/**
+ * @mainpage Opus
+ *
+ * The Opus codec is designed for interactive speech and audio transmission over the Internet.
+ * It is designed by the IETF Codec Working Group and incorporates technology from
+ * Skype's SILK codec and Xiph.Org's CELT codec.
+ *
+ * The Opus codec is designed to handle a wide range of interactive audio applications,
+ * including Voice over IP, videoconferencing, in-game chat, and even remote live music
+ * performances. It can scale from low bit-rate narrowband speech to very high quality
+ * stereo music. Its main features are:
+
+ * @li Sampling rates from 8 to 48 kHz
+ * @li Bit-rates from 6 kb/s to 510 kb/s
+ * @li Support for both constant bit-rate (CBR) and variable bit-rate (VBR)
+ * @li Audio bandwidth from narrowband to full-band
+ * @li Support for speech and music
+ * @li Support for mono and stereo
+ * @li Support for multichannel (up to 255 channels)
+ * @li Frame sizes from 2.5 ms to 60 ms
+ * @li Good loss robustness and packet loss concealment (PLC)
+ * @li Floating point and fixed-point implementation
+ *
+ * Documentation sections:
+ * @li @ref opus_encoder
+ * @li @ref opus_decoder
+ * @li @ref opus_repacketizer
+ * @li @ref opus_multistream
+ * @li @ref opus_libinfo
+ * @li @ref opus_custom
+ */
+
+/** @defgroup opus_encoder Opus Encoder
+  * @{
+  *
+  * @brief This page describes the process and functions used to encode Opus.
+  *
+  * Since Opus is a stateful codec, the encoding process starts with creating an encoder
+  * state. This can be done with:
+  *
+  * @code
+  * int          error;
+  * OpusEncoder *enc;
+  * enc = opus_encoder_create(Fs, channels, application, &error);
+  * @endcode
+  *
+  * From this point, @c enc can be used for encoding an audio stream. An encoder state
+  * @b must @b not be used for more than one stream at the same time. Similarly, the encoder
+  * state @b must @b not be re-initialized for each frame.
+  *
+  * While opus_encoder_create() allocates memory for the state, it's also possible
+  * to initialize pre-allocated memory:
+  *
+  * @code
+  * int          size;
+  * int          error;
+  * OpusEncoder *enc;
+  * size = opus_encoder_get_size(channels);
+  * enc = malloc(size);
+  * error = opus_encoder_init(enc, Fs, channels, application);
+  * @endcode
+  *
+  * where opus_encoder_get_size() returns the required size for the encoder state. Note that
+  * future versions of this code may change the size, so no assuptions should be made about it.
+  *
+  * The encoder state is always continuous in memory and only a shallow copy is sufficient
+  * to copy it (e.g. memcpy())
+  *
+  * It is possible to change some of the encoder's settings using the opus_encoder_ctl()
+  * interface. All these settings already default to the recommended value, so they should
+  * only be changed when necessary. The most common settings one may want to change are:
+  *
+  * @code
+  * opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate));
+  * opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(complexity));
+  * opus_encoder_ctl(enc, OPUS_SET_SIGNAL(signal_type));
+  * @endcode
+  *
+  * where
+  *
+  * @arg bitrate is in bits per second (b/s)
+  * @arg complexity is a value from 1 to 10, where 1 is the lowest complexity and 10 is the highest
+  * @arg signal_type is either OPUS_AUTO (default), OPUS_SIGNAL_VOICE, or OPUS_SIGNAL_MUSIC
+  *
+  * See @ref opus_encoderctls and @ref opus_genericctls for a complete list of parameters that can be set or queried. Most parameters can be set or changed at any time during a stream.
+  *
+  * To encode a frame, opus_encode() or opus_encode_float() must be called with exactly one frame (2.5, 5, 10, 20, 40 or 60 ms) of audio data:
+  * @code
+  * len = opus_encode(enc, audio_frame, frame_size, packet, max_packet);
+  * @endcode
+  *
+  * where
+  * <ul>
+  * <li>audio_frame is the audio data in opus_int16 (or float for opus_encode_float())</li>
+  * <li>frame_size is the duration of the frame in samples (per channel)</li>
+  * <li>packet is the byte array to which the compressed data is written</li>
+  * <li>max_packet is the maximum number of bytes that can be written in the packet (4000 bytes is recommended).
+  *     Do not use max_packet to control VBR target bitrate, instead use the #OPUS_SET_BITRATE CTL.</li>
+  * </ul>
+  *
+  * opus_encode() and opus_encode_float() return the number of bytes actually written to the packet.
+  * The return value <b>can be negative</b>, which indicates that an error has occurred. If the return value
+  * is 2 bytes or less, then the packet does not need to be transmitted (DTX).
+  *
+  * Once the encoder state if no longer needed, it can be destroyed with
+  *
+  * @code
+  * opus_encoder_destroy(enc);
+  * @endcode
+  *
+  * If the encoder was created with opus_encoder_init() rather than opus_encoder_create(),
+  * then no action is required aside from potentially freeing the memory that was manually
+  * allocated for it (calling free(enc) for the example above)
+  *
+  */
+
+/** Opus encoder state.
+  * This contains the complete state of an Opus encoder.
+  * It is position independent and can be freely copied.
+  * @see opus_encoder_create,opus_encoder_init
+  */
+typedef struct OpusEncoder OpusEncoder;
+
+/** Gets the size of an <code>OpusEncoder</code> structure.
+  * @param[in] channels <tt>int</tt>: Number of channels.
+  *                                   This must be 1 or 2.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_encoder_get_size(int channels);
+
+/**
+ */
+
+/** Allocates and initializes an encoder state.
+ * There are three coding modes:
+ *
+ * @ref OPUS_APPLICATION_VOIP gives best quality at a given bitrate for voice
+ *    signals. It enhances the  input signal by high-pass filtering and
+ *    emphasizing formants and harmonics. Optionally  it includes in-band
+ *    forward error correction to protect against packet loss. Use this
+ *    mode for typical VoIP applications. Because of the enhancement,
+ *    even at high bitrates the output may sound different from the input.
+ *
+ * @ref OPUS_APPLICATION_AUDIO gives best quality at a given bitrate for most
+ *    non-voice signals like music. Use this mode for music and mixed
+ *    (music/voice) content, broadcast, and applications requiring less
+ *    than 15 ms of coding delay.
+ *
+ * @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY configures low-delay mode that
+ *    disables the speech-optimized mode in exchange for slightly reduced delay.
+ *    This mode can only be set on an newly initialized or freshly reset encoder
+ *    because it changes the codec delay.
+ *
+ * This is useful when the caller knows that the speech-optimized modes will not be needed (use with caution).
+ * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
+ *                                     This must be one of 8000, 12000, 16000,
+ *                                     24000, or 48000.
+ * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
+ * @param [in] application <tt>int</tt>: Coding mode (@ref OPUS_APPLICATION_VOIP/@ref OPUS_APPLICATION_AUDIO/@ref OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+ * @param [out] error <tt>int*</tt>: @ref opus_errorcodes
+ * @note Regardless of the sampling rate and number channels selected, the Opus encoder
+ * can switch to a lower audio bandwidth or number of channels if the bitrate
+ * selected is too low. This also means that it is safe to always use 48 kHz stereo input
+ * and let the encoder optimize the encoding.
+ */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusEncoder *opus_encoder_create(
+    opus_int32 Fs,
+    int channels,
+    int application,
+    int *error
+);
+
+/** Initializes a previously allocated encoder state
+  * The memory pointed to by st must be at least the size returned by opus_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_encoder_create(),opus_encoder_get_size()
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
+ *                                      This must be one of 8000, 12000, 16000,
+ *                                      24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
+  * @param [in] application <tt>int</tt>: Coding mode (OPUS_APPLICATION_VOIP/OPUS_APPLICATION_AUDIO/OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+  * @retval #OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_EXPORT int opus_encoder_init(
+    OpusEncoder *st,
+    opus_int32 Fs,
+    int channels,
+    int application
+) OPUS_ARG_NONNULL(1);
+
+/** Encodes an Opus frame.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>opus_int16*</tt>: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
+  *                                      input signal.
+  *                                      This must be an Opus frame size for
+  *                                      the encoder's sampling rate.
+  *                                      For example, at 48 kHz the permitted
+  *                                      values are 120, 240, 480, 960, 1920,
+  *                                      and 2880.
+  *                                      Passing in a duration of less than
+  *                                      10 ms (480 samples at 48 kHz) will
+  *                                      prevent the encoder from using the LPC
+  *                                      or hybrid modes.
+  * @param [out] data <tt>unsigned char*</tt>: Output payload.
+  *                                            This must contain storage for at
+  *                                            least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode(
+    OpusEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes an Opus frame from floating point input.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>float*</tt>: Input in float format (interleaved if 2 channels), with a normal range of +/-1.0.
+  *          Samples with a range beyond +/-1.0 are supported but will
+  *          be clipped by decoders using the integer API and should
+  *          only be used if it is known that the far end supports
+  *          extended dynamic range.
+  *          length is frame_size*channels*sizeof(float)
+  * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
+  *                                      input signal.
+  *                                      This must be an Opus frame size for
+  *                                      the encoder's sampling rate.
+  *                                      For example, at 48 kHz the permitted
+  *                                      values are 120, 240, 480, 960, 1920,
+  *                                      and 2880.
+  *                                      Passing in a duration of less than
+  *                                      10 ms (480 samples at 48 kHz) will
+  *                                      prevent the encoder from using the LPC
+  *                                      or hybrid modes.
+  * @param [out] data <tt>unsigned char*</tt>: Output payload.
+  *                                            This must contain storage for at
+  *                                            least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode_float(
+    OpusEncoder *st,
+    const float *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Frees an <code>OpusEncoder</code> allocated by opus_encoder_create().
+  * @param[in] st <tt>OpusEncoder*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_encoder_destroy(OpusEncoder *st);
+
+/** Perform a CTL function on an Opus encoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @param st <tt>OpusEncoder*</tt>: Encoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls or
+  *                @ref opus_encoderctls.
+  * @see opus_genericctls
+  * @see opus_encoderctls
+  */
+OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+/**@}*/
+
+/** @defgroup opus_decoder Opus Decoder
+  * @{
+  *
+  * @brief This page describes the process and functions used to decode Opus.
+  *
+  * The decoding process also starts with creating a decoder
+  * state. This can be done with:
+  * @code
+  * int          error;
+  * OpusDecoder *dec;
+  * dec = opus_decoder_create(Fs, channels, &error);
+  * @endcode
+  * where
+  * @li Fs is the sampling rate and must be 8000, 12000, 16000, 24000, or 48000
+  * @li channels is the number of channels (1 or 2)
+  * @li error will hold the error code in case of failure (or #OPUS_OK on success)
+  * @li the return value is a newly created decoder state to be used for decoding
+  *
+  * While opus_decoder_create() allocates memory for the state, it's also possible
+  * to initialize pre-allocated memory:
+  * @code
+  * int          size;
+  * int          error;
+  * OpusDecoder *dec;
+  * size = opus_decoder_get_size(channels);
+  * dec = malloc(size);
+  * error = opus_decoder_init(dec, Fs, channels);
+  * @endcode
+  * where opus_decoder_get_size() returns the required size for the decoder state. Note that
+  * future versions of this code may change the size, so no assuptions should be made about it.
+  *
+  * The decoder state is always continuous in memory and only a shallow copy is sufficient
+  * to copy it (e.g. memcpy())
+  *
+  * To decode a frame, opus_decode() or opus_decode_float() must be called with a packet of compressed audio data:
+  * @code
+  * frame_size = opus_decode(dec, packet, len, decoded, max_size, 0);
+  * @endcode
+  * where
+  *
+  * @li packet is the byte array containing the compressed data
+  * @li len is the exact number of bytes contained in the packet
+  * @li decoded is the decoded audio data in opus_int16 (or float for opus_decode_float())
+  * @li max_size is the max duration of the frame in samples (per channel) that can fit into the decoded_frame array
+  *
+  * opus_decode() and opus_decode_float() return the number of samples (per channel) decoded from the packet.
+  * If that value is negative, then an error has occurred. This can occur if the packet is corrupted or if the audio
+  * buffer is too small to hold the decoded audio.
+  *
+  * Opus is a stateful codec with overlapping blocks and as a result Opus
+  * packets are not coded independently of each other. Packets must be
+  * passed into the decoder serially and in the correct order for a correct
+  * decode. Lost packets can be replaced with loss concealment by calling
+  * the decoder with a null pointer and zero length for the missing packet.
+  *
+  * A single codec state may only be accessed from a single thread at
+  * a time and any required locking must be performed by the caller. Separate
+  * streams must be decoded with separate decoder states and can be decoded
+  * in parallel unless the library was compiled with NONTHREADSAFE_PSEUDOSTACK
+  * defined.
+  *
+  */
+
+/** Opus decoder state.
+  * This contains the complete state of an Opus decoder.
+  * It is position independent and can be freely copied.
+  * @see opus_decoder_create,opus_decoder_init
+  */
+typedef struct OpusDecoder OpusDecoder;
+
+/** Gets the size of an <code>OpusDecoder</code> structure.
+  * @param [in] channels <tt>int</tt>: Number of channels.
+  *                                    This must be 1 or 2.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_size(int channels);
+
+/** Allocates and initializes a decoder state.
+  * @param [in] Fs <tt>opus_int32</tt>: Sample rate to decode at (Hz).
+  *                                     This must be one of 8000, 12000, 16000,
+  *                                     24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
+  * @param [out] error <tt>int*</tt>: #OPUS_OK Success or @ref opus_errorcodes
+  *
+  * Internally Opus stores data at 48000 Hz, so that should be the default
+  * value for Fs. However, the decoder can efficiently decode to buffers
+  * at 8, 12, 16, and 24 kHz so if for some reason the caller cannot use
+  * data at the full sample rate, or knows the compressed data doesn't
+  * use the full frequency range, it can request decoding at a reduced
+  * rate. Likewise, the decoder is capable of filling in either mono or
+  * interleaved stereo pcm buffers, at the caller's request.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusDecoder *opus_decoder_create(
+    opus_int32 Fs,
+    int channels,
+    int *error
+);
+
+/** Initializes a previously allocated decoder state.
+  * The state must be at least the size returned by opus_decoder_get_size().
+  * This is intended for applications which use their own allocator instead of malloc. @see opus_decoder_create,opus_decoder_get_size
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state.
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate to decode to (Hz).
+  *                                     This must be one of 8000, 12000, 16000,
+  *                                     24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
+  * @retval #OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_EXPORT int opus_decoder_init(
+    OpusDecoder *st,
+    opus_int32 Fs,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+/** Decode an Opus packet.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload*
+  * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size Number of samples per channel of available space in \a pcm.
+  *  If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
+  *  not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
+  *  then frame_size needs to be exactly the duration of audio that is missing, otherwise the
+  *  decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
+  *  FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
+  * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
+  *  decoded. If no such data is available, the frame is decoded as if it were lost.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode(
+    OpusDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int16 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode an Opus packet with floating point output.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(float)
+  * @param [in] frame_size Number of samples per channel of available space in \a pcm.
+  *  If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
+  *  not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
+  *  then frame_size needs to be exactly the duration of audio that is missing, otherwise the
+  *  decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
+  *  FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
+  * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
+  *  decoded. If no such data is available the frame is decoded as if it were lost.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode_float(
+    OpusDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    float *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus decoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @param st <tt>OpusDecoder*</tt>: Decoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls or
+  *                @ref opus_decoderctls.
+  * @see opus_genericctls
+  * @see opus_decoderctls
+  */
+OPUS_EXPORT int opus_decoder_ctl(OpusDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/** Frees an <code>OpusDecoder</code> allocated by opus_decoder_create().
+  * @param[in] st <tt>OpusDecoder*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_decoder_destroy(OpusDecoder *st);
+
+/** Parse an opus packet into one or more frames.
+  * Opus_decode will perform this operation internally so most applications do
+  * not need to use this function.
+  * This function does not copy the frames, the returned pointers are pointers into
+  * the input packet.
+  * @param [in] data <tt>char*</tt>: Opus packet to be parsed
+  * @param [in] len <tt>opus_int32</tt>: size of data
+  * @param [out] out_toc <tt>char*</tt>: TOC pointer
+  * @param [out] frames <tt>char*[48]</tt> encapsulated frames
+  * @param [out] size <tt>opus_int16[48]</tt> sizes of the encapsulated frames
+  * @param [out] payload_offset <tt>int*</tt>: returns the position of the payload within the packet (in bytes)
+  * @returns number of frames
+  */
+OPUS_EXPORT int opus_packet_parse(
+   const unsigned char *data,
+   opus_int32 len,
+   unsigned char *out_toc,
+   const unsigned char *frames[48],
+   opus_int16 size[48],
+   int *payload_offset
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5);
+
+/** Gets the bandwidth of an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet
+  * @retval OPUS_BANDWIDTH_NARROWBAND Narrowband (4kHz bandpass)
+  * @retval OPUS_BANDWIDTH_MEDIUMBAND Mediumband (6kHz bandpass)
+  * @retval OPUS_BANDWIDTH_WIDEBAND Wideband (8kHz bandpass)
+  * @retval OPUS_BANDWIDTH_SUPERWIDEBAND Superwideband (12kHz bandpass)
+  * @retval OPUS_BANDWIDTH_FULLBAND Fullband (20kHz bandpass)
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_bandwidth(const unsigned char *data) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples per frame from an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet.
+  *                                  This must contain at least one byte of
+  *                                  data.
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
+  *                                     This must be a multiple of 400, or
+  *                                     inaccurate results will be returned.
+  * @returns Number of samples per frame.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_samples_per_frame(const unsigned char *data, opus_int32 Fs) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of channels from an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet
+  * @returns Number of channels
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_channels(const unsigned char *data) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of frames in an Opus packet.
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @returns Number of frames
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples of an Opus packet.
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
+  *                                     This must be a multiple of 400, or
+  *                                     inaccurate results will be returned.
+  * @returns Number of samples
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_samples(const unsigned char packet[], opus_int32 len, opus_int32 Fs) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples of an Opus packet.
+  * @param [in] dec <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @returns Number of samples
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_nb_samples(const OpusDecoder *dec, const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+/** Applies soft-clipping to bring a float signal within the [-1,1] range. If
+  * the signal is already in that range, nothing is done. If there are values
+  * outside of [-1,1], then the signal is clipped as smoothly as possible to
+  * both fit in the range and avoid creating excessive distortion in the
+  * process.
+  * @param [in,out] pcm <tt>float*</tt>: Input PCM and modified PCM
+  * @param [in] frame_size <tt>int</tt> Number of samples per channel to process
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [in,out] softclip_mem <tt>float*</tt>: State memory for the soft clipping process (one float per channel, initialized to zero)
+  */
+OPUS_EXPORT void opus_pcm_soft_clip(float *pcm, int frame_size, int channels, float *softclip_mem);
+
+
+/**@}*/
+
+/** @defgroup opus_repacketizer Repacketizer
+  * @{
+  *
+  * The repacketizer can be used to merge multiple Opus packets into a single
+  * packet or alternatively to split Opus packets that have previously been
+  * merged. Splitting valid Opus packets is always guaranteed to succeed,
+  * whereas merging valid packets only succeeds if all frames have the same
+  * mode, bandwidth, and frame size, and when the total duration of the merged
+  * packet is no more than 120 ms. The 120 ms limit comes from the
+  * specification and limits decoder memory requirements at a point where
+  * framing overhead becomes negligible.
+  *
+  * The repacketizer currently only operates on elementary Opus
+  * streams. It will not manipualte multistream packets successfully, except in
+  * the degenerate case where they consist of data from a single stream.
+  *
+  * The repacketizing process starts with creating a repacketizer state, either
+  * by calling opus_repacketizer_create() or by allocating the memory yourself,
+  * e.g.,
+  * @code
+  * OpusRepacketizer *rp;
+  * rp = (OpusRepacketizer*)malloc(opus_repacketizer_get_size());
+  * if (rp != NULL)
+  *     opus_repacketizer_init(rp);
+  * @endcode
+  *
+  * Then the application should submit packets with opus_repacketizer_cat(),
+  * extract new packets with opus_repacketizer_out() or
+  * opus_repacketizer_out_range(), and then reset the state for the next set of
+  * input packets via opus_repacketizer_init().
+  *
+  * For example, to split a sequence of packets into individual frames:
+  * @code
+  * unsigned char *data;
+  * int len;
+  * while (get_next_packet(&data, &len))
+  * {
+  *   unsigned char out[1276];
+  *   opus_int32 out_len;
+  *   int nb_frames;
+  *   int err;
+  *   int i;
+  *   err = opus_repacketizer_cat(rp, data, len);
+  *   if (err != OPUS_OK)
+  *   {
+  *     release_packet(data);
+  *     return err;
+  *   }
+  *   nb_frames = opus_repacketizer_get_nb_frames(rp);
+  *   for (i = 0; i < nb_frames; i++)
+  *   {
+  *     out_len = opus_repacketizer_out_range(rp, i, i+1, out, sizeof(out));
+  *     if (out_len < 0)
+  *     {
+  *        release_packet(data);
+  *        return (int)out_len;
+  *     }
+  *     output_next_packet(out, out_len);
+  *   }
+  *   opus_repacketizer_init(rp);
+  *   release_packet(data);
+  * }
+  * @endcode
+  *
+  * Alternatively, to combine a sequence of frames into packets that each
+  * contain up to <code>TARGET_DURATION_MS</code> milliseconds of data:
+  * @code
+  * // The maximum number of packets with duration TARGET_DURATION_MS occurs
+  * // when the frame size is 2.5 ms, for a total of (TARGET_DURATION_MS*2/5)
+  * // packets.
+  * unsigned char *data[(TARGET_DURATION_MS*2/5)+1];
+  * opus_int32 len[(TARGET_DURATION_MS*2/5)+1];
+  * int nb_packets;
+  * unsigned char out[1277*(TARGET_DURATION_MS*2/2)];
+  * opus_int32 out_len;
+  * int prev_toc;
+  * nb_packets = 0;
+  * while (get_next_packet(data+nb_packets, len+nb_packets))
+  * {
+  *   int nb_frames;
+  *   int err;
+  *   nb_frames = opus_packet_get_nb_frames(data[nb_packets], len[nb_packets]);
+  *   if (nb_frames < 1)
+  *   {
+  *     release_packets(data, nb_packets+1);
+  *     return nb_frames;
+  *   }
+  *   nb_frames += opus_repacketizer_get_nb_frames(rp);
+  *   // If adding the next packet would exceed our target, or it has an
+  *   // incompatible TOC sequence, output the packets we already have before
+  *   // submitting it.
+  *   // N.B., The nb_packets > 0 check ensures we've submitted at least one
+  *   // packet since the last call to opus_repacketizer_init(). Otherwise a
+  *   // single packet longer than TARGET_DURATION_MS would cause us to try to
+  *   // output an (invalid) empty packet. It also ensures that prev_toc has
+  *   // been set to a valid value. Additionally, len[nb_packets] > 0 is
+  *   // guaranteed by the call to opus_packet_get_nb_frames() above, so the
+  *   // reference to data[nb_packets][0] should be valid.
+  *   if (nb_packets > 0 && (
+  *       ((prev_toc & 0xFC) != (data[nb_packets][0] & 0xFC)) ||
+  *       opus_packet_get_samples_per_frame(data[nb_packets], 48000)*nb_frames >
+  *       TARGET_DURATION_MS*48))
+  *   {
+  *     out_len = opus_repacketizer_out(rp, out, sizeof(out));
+  *     if (out_len < 0)
+  *     {
+  *        release_packets(data, nb_packets+1);
+  *        return (int)out_len;
+  *     }
+  *     output_next_packet(out, out_len);
+  *     opus_repacketizer_init(rp);
+  *     release_packets(data, nb_packets);
+  *     data[0] = data[nb_packets];
+  *     len[0] = len[nb_packets];
+  *     nb_packets = 0;
+  *   }
+  *   err = opus_repacketizer_cat(rp, data[nb_packets], len[nb_packets]);
+  *   if (err != OPUS_OK)
+  *   {
+  *     release_packets(data, nb_packets+1);
+  *     return err;
+  *   }
+  *   prev_toc = data[nb_packets][0];
+  *   nb_packets++;
+  * }
+  * // Output the final, partial packet.
+  * if (nb_packets > 0)
+  * {
+  *   out_len = opus_repacketizer_out(rp, out, sizeof(out));
+  *   release_packets(data, nb_packets);
+  *   if (out_len < 0)
+  *     return (int)out_len;
+  *   output_next_packet(out, out_len);
+  * }
+  * @endcode
+  *
+  * An alternate way of merging packets is to simply call opus_repacketizer_cat()
+  * unconditionally until it fails. At that point, the merged packet can be
+  * obtained with opus_repacketizer_out() and the input packet for which
+  * opus_repacketizer_cat() needs to be re-added to a newly reinitialized
+  * repacketizer state.
+  */
+
+typedef struct OpusRepacketizer OpusRepacketizer;
+
+/** Gets the size of an <code>OpusRepacketizer</code> structure.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_size(void);
+
+/** (Re)initializes a previously allocated repacketizer state.
+  * The state must be at least the size returned by opus_repacketizer_get_size().
+  * This can be used for applications which use their own allocator instead of
+  * malloc().
+  * It must also be called to reset the queue of packets waiting to be
+  * repacketized, which is necessary if the maximum packet duration of 120 ms
+  * is reached or if you wish to submit packets with a different Opus
+  * configuration (coding mode, audio bandwidth, frame size, or channel count).
+  * Failure to do so will prevent a new packet from being added with
+  * opus_repacketizer_cat().
+  * @see opus_repacketizer_create
+  * @see opus_repacketizer_get_size
+  * @see opus_repacketizer_cat
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to
+  *                                       (re)initialize.
+  * @returns A pointer to the same repacketizer state that was passed in.
+  */
+OPUS_EXPORT OpusRepacketizer *opus_repacketizer_init(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
+
+/** Allocates memory and initializes the new repacketizer with
+ * opus_repacketizer_init().
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusRepacketizer *opus_repacketizer_create(void);
+
+/** Frees an <code>OpusRepacketizer</code> allocated by
+  * opus_repacketizer_create().
+  * @param[in] rp <tt>OpusRepacketizer*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_repacketizer_destroy(OpusRepacketizer *rp);
+
+/** Add a packet to the current repacketizer state.
+  * This packet must match the configuration of any packets already submitted
+  * for repacketization since the last call to opus_repacketizer_init().
+  * This means that it must have the same coding mode, audio bandwidth, frame
+  * size, and channel count.
+  * This can be checked in advance by examining the top 6 bits of the first
+  * byte of the packet, and ensuring they match the top 6 bits of the first
+  * byte of any previously submitted packet.
+  * The total duration of audio in the repacketizer state also must not exceed
+  * 120 ms, the maximum duration of a single packet, after adding this packet.
+  *
+  * The contents of the current repacketizer state can be extracted into new
+  * packets using opus_repacketizer_out() or opus_repacketizer_out_range().
+  *
+  * In order to add a packet with a different configuration or to add more
+  * audio beyond 120 ms, you must clear the repacketizer state by calling
+  * opus_repacketizer_init().
+  * If a packet is too large to add to the current repacketizer state, no part
+  * of it is added, even if it contains multiple frames, some of which might
+  * fit.
+  * If you wish to be able to add parts of such packets, you should first use
+  * another repacketizer to split the packet into pieces and add them
+  * individually.
+  * @see opus_repacketizer_out_range
+  * @see opus_repacketizer_out
+  * @see opus_repacketizer_init
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to which to
+  *                                       add the packet.
+  * @param[in] data <tt>const unsigned char*</tt>: The packet data.
+  *                                                The application must ensure
+  *                                                this pointer remains valid
+  *                                                until the next call to
+  *                                                opus_repacketizer_init() or
+  *                                                opus_repacketizer_destroy().
+  * @param len <tt>opus_int32</tt>: The number of bytes in the packet data.
+  * @returns An error code indicating whether or not the operation succeeded.
+  * @retval #OPUS_OK The packet's contents have been added to the repacketizer
+  *                  state.
+  * @retval #OPUS_INVALID_PACKET The packet did not have a valid TOC sequence,
+  *                              the packet's TOC sequence was not compatible
+  *                              with previously submitted packets (because
+  *                              the coding mode, audio bandwidth, frame size,
+  *                              or channel count did not match), or adding
+  *                              this packet would increase the total amount of
+  *                              audio stored in the repacketizer state to more
+  *                              than 120 ms.
+  */
+OPUS_EXPORT int opus_repacketizer_cat(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+
+/** Construct a new packet from data previously submitted to the repacketizer
+  * state via opus_repacketizer_cat().
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
+  *                                       construct the new packet.
+  * @param begin <tt>int</tt>: The index of the first frame in the current
+  *                            repacketizer state to include in the output.
+  * @param end <tt>int</tt>: One past the index of the last frame in the
+  *                          current repacketizer state to include in the
+  *                          output.
+  * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
+  *                                                 store the output packet.
+  * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
+  *                                    the output buffer. In order to guarantee
+  *                                    success, this should be at least
+  *                                    <code>1276</code> for a single frame,
+  *                                    or for multiple frames,
+  *                                    <code>1277*(end-begin)</code>.
+  *                                    However, <code>1*(end-begin)</code> plus
+  *                                    the size of all packet data submitted to
+  *                                    the repacketizer since the last call to
+  *                                    opus_repacketizer_init() or
+  *                                    opus_repacketizer_create() is also
+  *                                    sufficient, and possibly much smaller.
+  * @returns The total size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG <code>[begin,end)</code> was an invalid range of
+  *                       frames (begin < 0, begin >= end, or end >
+  *                       opus_repacketizer_get_nb_frames()).
+  * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
+  *                                complete output packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out_range(OpusRepacketizer *rp, int begin, int end, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Return the total number of frames contained in packet data submitted to
+  * the repacketizer state so far via opus_repacketizer_cat() since the last
+  * call to opus_repacketizer_init() or opus_repacketizer_create().
+  * This defines the valid range of packets that can be extracted with
+  * opus_repacketizer_out_range() or opus_repacketizer_out().
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state containing the
+  *                                       frames.
+  * @returns The total number of frames contained in the packet data submitted
+  *          to the repacketizer state.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_nb_frames(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
+
+/** Construct a new packet from data previously submitted to the repacketizer
+  * state via opus_repacketizer_cat().
+  * This is a convenience routine that returns all the data submitted so far
+  * in a single packet.
+  * It is equivalent to calling
+  * @code
+  * opus_repacketizer_out_range(rp, 0, opus_repacketizer_get_nb_frames(rp),
+  *                             data, maxlen)
+  * @endcode
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
+  *                                       construct the new packet.
+  * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
+  *                                                 store the output packet.
+  * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
+  *                                    the output buffer. In order to guarantee
+  *                                    success, this should be at least
+  *                                    <code>1277*opus_repacketizer_get_nb_frames(rp)</code>.
+  *                                    However,
+  *                                    <code>1*opus_repacketizer_get_nb_frames(rp)</code>
+  *                                    plus the size of all packet data
+  *                                    submitted to the repacketizer since the
+  *                                    last call to opus_repacketizer_init() or
+  *                                    opus_repacketizer_create() is also
+  *                                    sufficient, and possibly much smaller.
+  * @returns The total size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
+  *                                complete output packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out(OpusRepacketizer *rp, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1);
+
+/** Pads a given Opus packet to a larger size (possibly changing the TOC sequence).
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to pad.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
+  *                                 This must be at least as large as len.
+  * @returns an error code
+  * @retval #OPUS_OK \a on success.
+  * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT int opus_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len);
+
+/** Remove all padding from a given Opus packet and rewrite the TOC sequence to
+  * minimize space usage.
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to strip.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @returns The new size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG \a len was less than 1.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_packet_unpad(unsigned char *data, opus_int32 len);
+
+/** Pads a given Opus multi-stream packet to a larger size (possibly changing the TOC sequence).
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to pad.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
+  *                                 This must be at least 1.
+  * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
+  *                                 This must be at least as large as len.
+  * @returns an error code
+  * @retval #OPUS_OK \a on success.
+  * @retval #OPUS_BAD_ARG \a len was less than 1.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT int opus_multistream_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len, int nb_streams);
+
+/** Remove all padding from a given Opus multi-stream packet and rewrite the TOC sequence to
+  * minimize space usage.
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to strip.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
+  *                                 This must be at least 1.
+  * @returns The new size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_packet_unpad(unsigned char *data, opus_int32 len, int nb_streams);
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_H */

engine/voice_codecs/opus/opus/opus_custom.h

@@ -0,0 +1,342 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Copyright (c) 2008-2012 Gregory Maxwell
+   Written by Jean-Marc Valin and Gregory Maxwell */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+  @file opus_custom.h
+  @brief Opus-Custom reference implementation API
+ */
+
+#ifndef OPUS_CUSTOM_H
+#define OPUS_CUSTOM_H
+
+#include "opus_defines.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#ifdef CUSTOM_MODES
+# define OPUS_CUSTOM_EXPORT OPUS_EXPORT
+# define OPUS_CUSTOM_EXPORT_STATIC OPUS_EXPORT
+#else
+# define OPUS_CUSTOM_EXPORT
+# ifdef OPUS_BUILD
+#  define OPUS_CUSTOM_EXPORT_STATIC static OPUS_INLINE
+# else
+#  define OPUS_CUSTOM_EXPORT_STATIC
+# endif
+#endif
+
+/** @defgroup opus_custom Opus Custom
+  * @{
+  *  Opus Custom is an optional part of the Opus specification and
+  * reference implementation which uses a distinct API from the regular
+  * API and supports frame sizes that are not normally supported.\ Use
+  * of Opus Custom is discouraged for all but very special applications
+  * for which a frame size different from 2.5, 5, 10, or 20 ms is needed
+  * (for either complexity or latency reasons) and where interoperability
+  * is less important.
+  *
+  * In addition to the interoperability limitations the use of Opus custom
+  * disables a substantial chunk of the codec and generally lowers the
+  * quality available at a given bitrate. Normally when an application needs
+  * a different frame size from the codec it should buffer to match the
+  * sizes but this adds a small amount of delay which may be important
+  * in some very low latency applications. Some transports (especially
+  * constant rate RF transports) may also work best with frames of
+  * particular durations.
+  *
+  * Libopus only supports custom modes if they are enabled at compile time.
+  *
+  * The Opus Custom API is similar to the regular API but the
+  * @ref opus_encoder_create and @ref opus_decoder_create calls take
+  * an additional mode parameter which is a structure produced by
+  * a call to @ref opus_custom_mode_create. Both the encoder and decoder
+  * must create a mode using the same sample rate (fs) and frame size
+  * (frame size) so these parameters must either be signaled out of band
+  * or fixed in a particular implementation.
+  *
+  * Similar to regular Opus the custom modes support on the fly frame size
+  * switching, but the sizes available depend on the particular frame size in
+  * use. For some initial frame sizes on a single on the fly size is available.
+  */
+
+/** Contains the state of an encoder. One encoder state is needed
+    for each stream. It is initialized once at the beginning of the
+    stream. Do *not* re-initialize the state for every frame.
+   @brief Encoder state
+ */
+typedef struct OpusCustomEncoder OpusCustomEncoder;
+
+/** State of the decoder. One decoder state is needed for each stream.
+    It is initialized once at the beginning of the stream. Do *not*
+    re-initialize the state for every frame.
+   @brief Decoder state
+ */
+typedef struct OpusCustomDecoder OpusCustomDecoder;
+
+/** The mode contains all the information necessary to create an
+    encoder. Both the encoder and decoder need to be initialized
+    with exactly the same mode, otherwise the output will be
+    corrupted.
+   @brief Mode configuration
+ */
+typedef struct OpusCustomMode OpusCustomMode;
+
+/** Creates a new mode struct. This will be passed to an encoder or
+  * decoder. The mode MUST NOT BE DESTROYED until the encoders and
+  * decoders that use it are destroyed as well.
+  * @param [in] Fs <tt>int</tt>: Sampling rate (8000 to 96000 Hz)
+  * @param [in] frame_size <tt>int</tt>: Number of samples (per channel) to encode in each
+  *        packet (64 - 1024, prime factorization must contain zero or more 2s, 3s, or 5s and no other primes)
+  * @param [out] error <tt>int*</tt>: Returned error code (if NULL, no error will be returned)
+  * @return A newly created mode
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomMode *opus_custom_mode_create(opus_int32 Fs, int frame_size, int *error);
+
+/** Destroys a mode struct. Only call this after all encoders and
+  * decoders using this mode are destroyed as well.
+  * @param [in] mode <tt>OpusCustomMode*</tt>: Mode to be freed.
+  */
+OPUS_CUSTOM_EXPORT void opus_custom_mode_destroy(OpusCustomMode *mode);
+
+
+#if !defined(OPUS_BUILD) || defined(CELT_ENCODER_C)
+
+/* Encoder */
+/** Gets the size of an OpusCustomEncoder structure.
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Mode configuration
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @returns size
+  */
+OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_encoder_get_size(
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+# ifdef CUSTOM_MODES
+/** Initializes a previously allocated encoder state
+  * The memory pointed to by st must be the size returned by opus_custom_encoder_get_size.
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_custom_encoder_create(),opus_custom_encoder_get_size()
+  * To reset a previously initialized state use the OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Contains all the information about the characteristics of
+  *  the stream (must be the same characteristics as used for the
+  *  decoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @return OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT int opus_custom_encoder_init(
+    OpusCustomEncoder *st,
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+# endif
+#endif
+
+
+/** Creates a new encoder state. Each stream needs its own encoder
+  * state (can't be shared across simultaneous streams).
+  * @param [in] mode <tt>OpusCustomMode*</tt>: Contains all the information about the characteristics of
+  *  the stream (must be the same characteristics as used for the
+  *  decoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [out] error <tt>int*</tt>: Returns an error code
+  * @return Newly created encoder state.
+*/
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomEncoder *opus_custom_encoder_create(
+    const OpusCustomMode *mode,
+    int channels,
+    int *error
+) OPUS_ARG_NONNULL(1);
+
+
+/** Destroys a an encoder state.
+  * @param[in] st <tt>OpusCustomEncoder*</tt>: State to be freed.
+  */
+OPUS_CUSTOM_EXPORT void opus_custom_encoder_destroy(OpusCustomEncoder *st);
+
+/** Encodes a frame of audio.
+  * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>float*</tt>: PCM audio in float format, with a normal range of +/-1.0.
+  *          Samples with a range beyond +/-1.0 are supported but will
+  *          be clipped by decoders using the integer API and should
+  *          only be used if it is known that the far end supports
+  *          extended dynamic range. There must be exactly
+  *          frame_size samples per channel.
+  * @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal
+  * @param [out] compressed <tt>char *</tt>: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long.
+  * @param [in] maxCompressedBytes <tt>int</tt>: Maximum number of bytes to use for compressing the frame
+  *          (can change from one frame to another)
+  * @return Number of bytes written to "compressed".
+  *       If negative, an error has occurred (see error codes). It is IMPORTANT that
+  *       the length returned be somehow transmitted to the decoder. Otherwise, no
+  *       decoding is possible.
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode_float(
+    OpusCustomEncoder *st,
+    const float *pcm,
+    int frame_size,
+    unsigned char *compressed,
+    int maxCompressedBytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes a frame of audio.
+  * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>opus_int16*</tt>: PCM audio in signed 16-bit format (native endian).
+  *          There must be exactly frame_size samples per channel.
+  * @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal
+  * @param [out] compressed <tt>char *</tt>: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long.
+  * @param [in] maxCompressedBytes <tt>int</tt>: Maximum number of bytes to use for compressing the frame
+  *          (can change from one frame to another)
+  * @return Number of bytes written to "compressed".
+  *       If negative, an error has occurred (see error codes). It is IMPORTANT that
+  *       the length returned be somehow transmitted to the decoder. Otherwise, no
+  *       decoding is possible.
+ */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode(
+    OpusCustomEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *compressed,
+    int maxCompressedBytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus custom encoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @see opus_encoderctls
+  */
+OPUS_CUSTOM_EXPORT int opus_custom_encoder_ctl(OpusCustomEncoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1);
+
+
+#if !defined(OPUS_BUILD) || defined(CELT_DECODER_C)
+/* Decoder */
+
+/** Gets the size of an OpusCustomDecoder structure.
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Mode configuration
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @returns size
+  */
+OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_decoder_get_size(
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+/** Initializes a previously allocated decoder state
+  * The memory pointed to by st must be the size returned by opus_custom_decoder_get_size.
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_custom_decoder_create(),opus_custom_decoder_get_size()
+  * To reset a previously initialized state use the OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Contains all the information about the characteristics of
+  *  the stream (must be the same characteristics as used for the
+  *  encoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @return OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT_STATIC int opus_custom_decoder_init(
+    OpusCustomDecoder *st,
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+#endif
+
+
+/** Creates a new decoder state. Each stream needs its own decoder state (can't
+  * be shared across simultaneous streams).
+  * @param [in] mode <tt>OpusCustomMode</tt>: Contains all the information about the characteristics of the
+  *          stream (must be the same characteristics as used for the encoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [out] error <tt>int*</tt>: Returns an error code
+  * @return Newly created decoder state.
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomDecoder *opus_custom_decoder_create(
+    const OpusCustomMode *mode,
+    int channels,
+    int *error
+) OPUS_ARG_NONNULL(1);
+
+/** Destroys a an decoder state.
+  * @param[in] st <tt>OpusCustomDecoder*</tt>: State to be freed.
+  */
+OPUS_CUSTOM_EXPORT void opus_custom_decoder_destroy(OpusCustomDecoder *st);
+
+/** Decode an opus custom frame with floating point output
+  * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>int</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(float)
+  * @param [in] frame_size Number of samples per channel of available space in *pcm.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode_float(
+    OpusCustomDecoder *st,
+    const unsigned char *data,
+    int len,
+    float *pcm,
+    int frame_size
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode an opus custom frame
+  * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>int</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size Number of samples per channel of available space in *pcm.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode(
+    OpusCustomDecoder *st,
+    const unsigned char *data,
+    int len,
+    opus_int16 *pcm,
+    int frame_size
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus custom decoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @see opus_genericctls
+  */
+OPUS_CUSTOM_EXPORT int opus_custom_decoder_ctl(OpusCustomDecoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_CUSTOM_H */

engine/voice_codecs/opus/opus/opus_defines.h

@@ -0,0 +1,799 @@
+/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
+   Written by Jean-Marc Valin and Koen Vos */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus_defines.h
+ * @brief Opus reference implementation constants
+ */
+
+#ifndef OPUS_DEFINES_H
+#define OPUS_DEFINES_H
+
+#include "opus_types.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** @defgroup opus_errorcodes Error codes
+ * @{
+ */
+/** No error @hideinitializer*/
+#define OPUS_OK                0
+/** One or more invalid/out of range arguments @hideinitializer*/
+#define OPUS_BAD_ARG          -1
+/** Not enough bytes allocated in the buffer @hideinitializer*/
+#define OPUS_BUFFER_TOO_SMALL -2
+/** An internal error was detected @hideinitializer*/
+#define OPUS_INTERNAL_ERROR   -3
+/** The compressed data passed is corrupted @hideinitializer*/
+#define OPUS_INVALID_PACKET   -4
+/** Invalid/unsupported request number @hideinitializer*/
+#define OPUS_UNIMPLEMENTED    -5
+/** An encoder or decoder structure is invalid or already freed @hideinitializer*/
+#define OPUS_INVALID_STATE    -6
+/** Memory allocation has failed @hideinitializer*/
+#define OPUS_ALLOC_FAIL       -7
+/**@}*/
+
+/** @cond OPUS_INTERNAL_DOC */
+/**Export control for opus functions */
+
+#ifndef OPUS_EXPORT
+# if defined(WIN32)
+#  if defined(OPUS_BUILD) && defined(DLL_EXPORT)
+#   define OPUS_EXPORT __declspec(dllexport)
+#  else
+#   define OPUS_EXPORT
+#  endif
+# elif defined(__GNUC__) && defined(OPUS_BUILD)
+#  define OPUS_EXPORT __attribute__ ((visibility ("default")))
+# else
+#  define OPUS_EXPORT
+# endif
+#endif
+
+# if !defined(OPUS_GNUC_PREREQ)
+#  if defined(__GNUC__)&&defined(__GNUC_MINOR__)
+#   define OPUS_GNUC_PREREQ(_maj,_min) \
+ ((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min))
+#  else
+#   define OPUS_GNUC_PREREQ(_maj,_min) 0
+#  endif
+# endif
+
+#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
+# if OPUS_GNUC_PREREQ(3,0)
+#  define OPUS_RESTRICT __restrict__
+# elif (defined(_MSC_VER) && _MSC_VER >= 1400)
+#  define OPUS_RESTRICT __restrict
+# else
+#  define OPUS_RESTRICT
+# endif
+#else
+# define OPUS_RESTRICT restrict
+#endif
+
+#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
+# if OPUS_GNUC_PREREQ(2,7)
+#  define OPUS_INLINE __inline__
+# elif (defined(_MSC_VER))
+#  define OPUS_INLINE __inline
+# else
+#  define OPUS_INLINE
+# endif
+#else
+# define OPUS_INLINE inline
+#endif
+
+/**Warning attributes for opus functions
+  * NONNULL is not used in OPUS_BUILD to avoid the compiler optimizing out
+  * some paranoid null checks. */
+#if defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
+# define OPUS_WARN_UNUSED_RESULT __attribute__ ((__warn_unused_result__))
+#else
+# define OPUS_WARN_UNUSED_RESULT
+#endif
+#if !defined(OPUS_BUILD) && defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
+# define OPUS_ARG_NONNULL(_x)  __attribute__ ((__nonnull__(_x)))
+#else
+# define OPUS_ARG_NONNULL(_x)
+#endif
+
+/** These are the actual Encoder CTL ID numbers.
+  * They should not be used directly by applications.
+  * In general, SETs should be even and GETs should be odd.*/
+#define OPUS_SET_APPLICATION_REQUEST         4000
+#define OPUS_GET_APPLICATION_REQUEST         4001
+#define OPUS_SET_BITRATE_REQUEST             4002
+#define OPUS_GET_BITRATE_REQUEST             4003
+#define OPUS_SET_MAX_BANDWIDTH_REQUEST       4004
+#define OPUS_GET_MAX_BANDWIDTH_REQUEST       4005
+#define OPUS_SET_VBR_REQUEST                 4006
+#define OPUS_GET_VBR_REQUEST                 4007
+#define OPUS_SET_BANDWIDTH_REQUEST           4008
+#define OPUS_GET_BANDWIDTH_REQUEST           4009
+#define OPUS_SET_COMPLEXITY_REQUEST          4010
+#define OPUS_GET_COMPLEXITY_REQUEST          4011
+#define OPUS_SET_INBAND_FEC_REQUEST          4012
+#define OPUS_GET_INBAND_FEC_REQUEST          4013
+#define OPUS_SET_PACKET_LOSS_PERC_REQUEST    4014
+#define OPUS_GET_PACKET_LOSS_PERC_REQUEST    4015
+#define OPUS_SET_DTX_REQUEST                 4016
+#define OPUS_GET_DTX_REQUEST                 4017
+#define OPUS_SET_VBR_CONSTRAINT_REQUEST      4020
+#define OPUS_GET_VBR_CONSTRAINT_REQUEST      4021
+#define OPUS_SET_FORCE_CHANNELS_REQUEST      4022
+#define OPUS_GET_FORCE_CHANNELS_REQUEST      4023
+#define OPUS_SET_SIGNAL_REQUEST              4024
+#define OPUS_GET_SIGNAL_REQUEST              4025
+#define OPUS_GET_LOOKAHEAD_REQUEST           4027
+/* #define OPUS_RESET_STATE 4028 */
+#define OPUS_GET_SAMPLE_RATE_REQUEST         4029
+#define OPUS_GET_FINAL_RANGE_REQUEST         4031
+#define OPUS_GET_PITCH_REQUEST               4033
+#define OPUS_SET_GAIN_REQUEST                4034
+#define OPUS_GET_GAIN_REQUEST                4045 /* Should have been 4035 */
+#define OPUS_SET_LSB_DEPTH_REQUEST           4036
+#define OPUS_GET_LSB_DEPTH_REQUEST           4037
+#define OPUS_GET_LAST_PACKET_DURATION_REQUEST 4039
+#define OPUS_SET_EXPERT_FRAME_DURATION_REQUEST 4040
+#define OPUS_GET_EXPERT_FRAME_DURATION_REQUEST 4041
+#define OPUS_SET_PREDICTION_DISABLED_REQUEST 4042
+#define OPUS_GET_PREDICTION_DISABLED_REQUEST 4043
+/* Don't use 4045, it's already taken by OPUS_GET_GAIN_REQUEST */
+#define OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST 4046
+#define OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST 4047
+#define OPUS_GET_IN_DTX_REQUEST              4049
+
+/** Defines for the presence of extended APIs. */
+#define OPUS_HAVE_OPUS_PROJECTION_H
+
+/* Macros to trigger compilation errors when the wrong types are provided to a CTL */
+#define __opus_check_int(x) (((void)((x) == (opus_int32)0)), (opus_int32)(x))
+#define __opus_check_int_ptr(ptr) ((ptr) + ((ptr) - (opus_int32*)(ptr)))
+#define __opus_check_uint_ptr(ptr) ((ptr) + ((ptr) - (opus_uint32*)(ptr)))
+#define __opus_check_val16_ptr(ptr) ((ptr) + ((ptr) - (opus_val16*)(ptr)))
+/** @endcond */
+
+/** @defgroup opus_ctlvalues Pre-defined values for CTL interface
+  * @see opus_genericctls, opus_encoderctls
+  * @{
+  */
+/* Values for the various encoder CTLs */
+#define OPUS_AUTO                           -1000 /**<Auto/default setting @hideinitializer*/
+#define OPUS_BITRATE_MAX                       -1 /**<Maximum bitrate @hideinitializer*/
+
+/** Best for most VoIP/videoconference applications where listening quality and intelligibility matter most
+ * @hideinitializer */
+#define OPUS_APPLICATION_VOIP                2048
+/** Best for broadcast/high-fidelity application where the decoded audio should be as close as possible to the input
+ * @hideinitializer */
+#define OPUS_APPLICATION_AUDIO               2049
+/** Only use when lowest-achievable latency is what matters most. Voice-optimized modes cannot be used.
+ * @hideinitializer */
+#define OPUS_APPLICATION_RESTRICTED_LOWDELAY 2051
+
+#define OPUS_SIGNAL_VOICE                    3001 /**< Signal being encoded is voice */
+#define OPUS_SIGNAL_MUSIC                    3002 /**< Signal being encoded is music */
+#define OPUS_BANDWIDTH_NARROWBAND            1101 /**< 4 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_MEDIUMBAND            1102 /**< 6 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_WIDEBAND              1103 /**< 8 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_SUPERWIDEBAND         1104 /**<12 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_FULLBAND              1105 /**<20 kHz bandpass @hideinitializer*/
+
+#define OPUS_FRAMESIZE_ARG                   5000 /**< Select frame size from the argument (default) */
+#define OPUS_FRAMESIZE_2_5_MS                5001 /**< Use 2.5 ms frames */
+#define OPUS_FRAMESIZE_5_MS                  5002 /**< Use 5 ms frames */
+#define OPUS_FRAMESIZE_10_MS                 5003 /**< Use 10 ms frames */
+#define OPUS_FRAMESIZE_20_MS                 5004 /**< Use 20 ms frames */
+#define OPUS_FRAMESIZE_40_MS                 5005 /**< Use 40 ms frames */
+#define OPUS_FRAMESIZE_60_MS                 5006 /**< Use 60 ms frames */
+#define OPUS_FRAMESIZE_80_MS                 5007 /**< Use 80 ms frames */
+#define OPUS_FRAMESIZE_100_MS                5008 /**< Use 100 ms frames */
+#define OPUS_FRAMESIZE_120_MS                5009 /**< Use 120 ms frames */
+
+/**@}*/
+
+
+/** @defgroup opus_encoderctls Encoder related CTLs
+  *
+  * These are convenience macros for use with the \c opus_encode_ctl
+  * interface. They are used to generate the appropriate series of
+  * arguments for that call, passing the correct type, size and so
+  * on as expected for each particular request.
+  *
+  * Some usage examples:
+  *
+  * @code
+  * int ret;
+  * ret = opus_encoder_ctl(enc_ctx, OPUS_SET_BANDWIDTH(OPUS_AUTO));
+  * if (ret != OPUS_OK) return ret;
+  *
+  * opus_int32 rate;
+  * opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&rate));
+  *
+  * opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
+  * @endcode
+  *
+  * @see opus_genericctls, opus_encoder
+  * @{
+  */
+
+/** Configures the encoder's computational complexity.
+  * The supported range is 0-10 inclusive with 10 representing the highest complexity.
+  * @see OPUS_GET_COMPLEXITY
+  * @param[in] x <tt>opus_int32</tt>: Allowed values: 0-10, inclusive.
+  *
+  * @hideinitializer */
+#define OPUS_SET_COMPLEXITY(x) OPUS_SET_COMPLEXITY_REQUEST, __opus_check_int(x)
+/** Gets the encoder's complexity configuration.
+  * @see OPUS_SET_COMPLEXITY
+  * @param[out] x <tt>opus_int32 *</tt>: Returns a value in the range 0-10,
+  *                                      inclusive.
+  * @hideinitializer */
+#define OPUS_GET_COMPLEXITY(x) OPUS_GET_COMPLEXITY_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the bitrate in the encoder.
+  * Rates from 500 to 512000 bits per second are meaningful, as well as the
+  * special values #OPUS_AUTO and #OPUS_BITRATE_MAX.
+  * The value #OPUS_BITRATE_MAX can be used to cause the codec to use as much
+  * rate as it can, which is useful for controlling the rate by adjusting the
+  * output buffer size.
+  * @see OPUS_GET_BITRATE
+  * @param[in] x <tt>opus_int32</tt>: Bitrate in bits per second. The default
+  *                                   is determined based on the number of
+  *                                   channels and the input sampling rate.
+  * @hideinitializer */
+#define OPUS_SET_BITRATE(x) OPUS_SET_BITRATE_REQUEST, __opus_check_int(x)
+/** Gets the encoder's bitrate configuration.
+  * @see OPUS_SET_BITRATE
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the bitrate in bits per second.
+  *                                      The default is determined based on the
+  *                                      number of channels and the input
+  *                                      sampling rate.
+  * @hideinitializer */
+#define OPUS_GET_BITRATE(x) OPUS_GET_BITRATE_REQUEST, __opus_check_int_ptr(x)
+
+/** Enables or disables variable bitrate (VBR) in the encoder.
+  * The configured bitrate may not be met exactly because frames must
+  * be an integer number of bytes in length.
+  * @see OPUS_GET_VBR
+  * @see OPUS_SET_VBR_CONSTRAINT
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Hard CBR. For LPC/hybrid modes at very low bit-rate, this can
+  *               cause noticeable quality degradation.</dd>
+  * <dt>1</dt><dd>VBR (default). The exact type of VBR is controlled by
+  *               #OPUS_SET_VBR_CONSTRAINT.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_VBR(x) OPUS_SET_VBR_REQUEST, __opus_check_int(x)
+/** Determine if variable bitrate (VBR) is enabled in the encoder.
+  * @see OPUS_SET_VBR
+  * @see OPUS_GET_VBR_CONSTRAINT
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Hard CBR.</dd>
+  * <dt>1</dt><dd>VBR (default). The exact type of VBR may be retrieved via
+  *               #OPUS_GET_VBR_CONSTRAINT.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_VBR(x) OPUS_GET_VBR_REQUEST, __opus_check_int_ptr(x)
+
+/** Enables or disables constrained VBR in the encoder.
+  * This setting is ignored when the encoder is in CBR mode.
+  * @warning Only the MDCT mode of Opus currently heeds the constraint.
+  *  Speech mode ignores it completely, hybrid mode may fail to obey it
+  *  if the LPC layer uses more bitrate than the constraint would have
+  *  permitted.
+  * @see OPUS_GET_VBR_CONSTRAINT
+  * @see OPUS_SET_VBR
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Unconstrained VBR.</dd>
+  * <dt>1</dt><dd>Constrained VBR (default). This creates a maximum of one
+  *               frame of buffering delay assuming a transport with a
+  *               serialization speed of the nominal bitrate.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_VBR_CONSTRAINT(x) OPUS_SET_VBR_CONSTRAINT_REQUEST, __opus_check_int(x)
+/** Determine if constrained VBR is enabled in the encoder.
+  * @see OPUS_SET_VBR_CONSTRAINT
+  * @see OPUS_GET_VBR
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Unconstrained VBR.</dd>
+  * <dt>1</dt><dd>Constrained VBR (default).</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_VBR_CONSTRAINT(x) OPUS_GET_VBR_CONSTRAINT_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures mono/stereo forcing in the encoder.
+  * This can force the encoder to produce packets encoded as either mono or
+  * stereo, regardless of the format of the input audio. This is useful when
+  * the caller knows that the input signal is currently a mono source embedded
+  * in a stereo stream.
+  * @see OPUS_GET_FORCE_CHANNELS
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
+  * <dt>1</dt>         <dd>Forced mono</dd>
+  * <dt>2</dt>         <dd>Forced stereo</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_FORCE_CHANNELS(x) OPUS_SET_FORCE_CHANNELS_REQUEST, __opus_check_int(x)
+/** Gets the encoder's forced channel configuration.
+  * @see OPUS_SET_FORCE_CHANNELS
+  * @param[out] x <tt>opus_int32 *</tt>:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
+  * <dt>1</dt>         <dd>Forced mono</dd>
+  * <dt>2</dt>         <dd>Forced stereo</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_FORCE_CHANNELS(x) OPUS_GET_FORCE_CHANNELS_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the maximum bandpass that the encoder will select automatically.
+  * Applications should normally use this instead of #OPUS_SET_BANDWIDTH
+  * (leaving that set to the default, #OPUS_AUTO). This allows the
+  * application to set an upper bound based on the type of input it is
+  * providing, but still gives the encoder the freedom to reduce the bandpass
+  * when the bitrate becomes too low, for better overall quality.
+  * @see OPUS_GET_MAX_BANDWIDTH
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband (default)</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_MAX_BANDWIDTH(x) OPUS_SET_MAX_BANDWIDTH_REQUEST, __opus_check_int(x)
+
+/** Gets the encoder's configured maximum allowed bandpass.
+  * @see OPUS_SET_MAX_BANDWIDTH
+  * @param[out] x <tt>opus_int32 *</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband (default)</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_MAX_BANDWIDTH(x) OPUS_GET_MAX_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Sets the encoder's bandpass to a specific value.
+  * This prevents the encoder from automatically selecting the bandpass based
+  * on the available bitrate. If an application knows the bandpass of the input
+  * audio it is providing, it should normally use #OPUS_SET_MAX_BANDWIDTH
+  * instead, which still gives the encoder the freedom to reduce the bandpass
+  * when the bitrate becomes too low, for better overall quality.
+  * @see OPUS_GET_BANDWIDTH
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>                    <dd>(default)</dd>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_BANDWIDTH(x) OPUS_SET_BANDWIDTH_REQUEST, __opus_check_int(x)
+
+/** Configures the type of signal being encoded.
+  * This is a hint which helps the encoder's mode selection.
+  * @see OPUS_GET_SIGNAL
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>        <dd>(default)</dd>
+  * <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
+  * <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_SIGNAL(x) OPUS_SET_SIGNAL_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured signal type.
+  * @see OPUS_SET_SIGNAL
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>        <dd>(default)</dd>
+  * <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
+  * <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_SIGNAL(x) OPUS_GET_SIGNAL_REQUEST, __opus_check_int_ptr(x)
+
+
+/** Configures the encoder's intended application.
+  * The initial value is a mandatory argument to the encoder_create function.
+  * @see OPUS_GET_APPLICATION
+  * @param[in] x <tt>opus_int32</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_APPLICATION(x) OPUS_SET_APPLICATION_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured application.
+  * @see OPUS_SET_APPLICATION
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_APPLICATION(x) OPUS_GET_APPLICATION_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the total samples of delay added by the entire codec.
+  * This can be queried by the encoder and then the provided number of samples can be
+  * skipped on from the start of the decoder's output to provide time aligned input
+  * and output. From the perspective of a decoding application the real data begins this many
+  * samples late.
+  *
+  * The decoder contribution to this delay is identical for all decoders, but the
+  * encoder portion of the delay may vary from implementation to implementation,
+  * version to version, or even depend on the encoder's initial configuration.
+  * Applications needing delay compensation should call this CTL rather than
+  * hard-coding a value.
+  * @param[out] x <tt>opus_int32 *</tt>:   Number of lookahead samples
+  * @hideinitializer */
+#define OPUS_GET_LOOKAHEAD(x) OPUS_GET_LOOKAHEAD_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of inband forward error correction (FEC).
+  * @note This is only applicable to the LPC layer
+  * @see OPUS_GET_INBAND_FEC
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Disable inband FEC (default).</dd>
+  * <dt>1</dt><dd>Enable inband FEC.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_INBAND_FEC(x) OPUS_SET_INBAND_FEC_REQUEST, __opus_check_int(x)
+/** Gets encoder's configured use of inband forward error correction.
+  * @see OPUS_SET_INBAND_FEC
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Inband FEC disabled (default).</dd>
+  * <dt>1</dt><dd>Inband FEC enabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_INBAND_FEC(x) OPUS_GET_INBAND_FEC_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's expected packet loss percentage.
+  * Higher values trigger progressively more loss resistant behavior in the encoder
+  * at the expense of quality at a given bitrate in the absence of packet loss, but
+  * greater quality under loss.
+  * @see OPUS_GET_PACKET_LOSS_PERC
+  * @param[in] x <tt>opus_int32</tt>:   Loss percentage in the range 0-100, inclusive (default: 0).
+  * @hideinitializer */
+#define OPUS_SET_PACKET_LOSS_PERC(x) OPUS_SET_PACKET_LOSS_PERC_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured packet loss percentage.
+  * @see OPUS_SET_PACKET_LOSS_PERC
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the configured loss percentage
+  *                                      in the range 0-100, inclusive (default: 0).
+  * @hideinitializer */
+#define OPUS_GET_PACKET_LOSS_PERC(x) OPUS_GET_PACKET_LOSS_PERC_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of discontinuous transmission (DTX).
+  * @note This is only applicable to the LPC layer
+  * @see OPUS_GET_DTX
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Disable DTX (default).</dd>
+  * <dt>1</dt><dd>Enabled DTX.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_DTX(x) OPUS_SET_DTX_REQUEST, __opus_check_int(x)
+/** Gets encoder's configured use of discontinuous transmission.
+  * @see OPUS_SET_DTX
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>DTX disabled (default).</dd>
+  * <dt>1</dt><dd>DTX enabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_DTX(x) OPUS_GET_DTX_REQUEST, __opus_check_int_ptr(x)
+/** Configures the depth of signal being encoded.
+  *
+  * This is a hint which helps the encoder identify silence and near-silence.
+  * It represents the number of significant bits of linear intensity below
+  * which the signal contains ignorable quantization or other noise.
+  *
+  * For example, OPUS_SET_LSB_DEPTH(14) would be an appropriate setting
+  * for G.711 u-law input. OPUS_SET_LSB_DEPTH(16) would be appropriate
+  * for 16-bit linear pcm input with opus_encode_float().
+  *
+  * When using opus_encode() instead of opus_encode_float(), or when libopus
+  * is compiled for fixed-point, the encoder uses the minimum of the value
+  * set here and the value 16.
+  *
+  * @see OPUS_GET_LSB_DEPTH
+  * @param[in] x <tt>opus_int32</tt>: Input precision in bits, between 8 and 24
+  *                                   (default: 24).
+  * @hideinitializer */
+#define OPUS_SET_LSB_DEPTH(x) OPUS_SET_LSB_DEPTH_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured signal depth.
+  * @see OPUS_SET_LSB_DEPTH
+  * @param[out] x <tt>opus_int32 *</tt>: Input precision in bits, between 8 and
+  *                                      24 (default: 24).
+  * @hideinitializer */
+#define OPUS_GET_LSB_DEPTH(x) OPUS_GET_LSB_DEPTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of variable duration frames.
+  * When variable duration is enabled, the encoder is free to use a shorter frame
+  * size than the one requested in the opus_encode*() call.
+  * It is then the user's responsibility
+  * to verify how much audio was encoded by checking the ToC byte of the encoded
+  * packet. The part of the audio that was not encoded needs to be resent to the
+  * encoder for the next call. Do not use this option unless you <b>really</b>
+  * know what you are doing.
+  * @see OPUS_GET_EXPERT_FRAME_DURATION
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
+  * <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_80_MS</dt><dd>Use 80 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_100_MS</dt><dd>Use 100 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_120_MS</dt><dd>Use 120 ms frames.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_EXPERT_FRAME_DURATION(x) OPUS_SET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured use of variable duration frames.
+  * @see OPUS_SET_EXPERT_FRAME_DURATION
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
+  * <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_80_MS</dt><dd>Use 80 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_100_MS</dt><dd>Use 100 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_120_MS</dt><dd>Use 120 ms frames.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_EXPERT_FRAME_DURATION(x) OPUS_GET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int_ptr(x)
+
+/** If set to 1, disables almost all use of prediction, making frames almost
+  * completely independent. This reduces quality.
+  * @see OPUS_GET_PREDICTION_DISABLED
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Enable prediction (default).</dd>
+  * <dt>1</dt><dd>Disable prediction.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_PREDICTION_DISABLED(x) OPUS_SET_PREDICTION_DISABLED_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured prediction status.
+  * @see OPUS_SET_PREDICTION_DISABLED
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Prediction enabled (default).</dd>
+  * <dt>1</dt><dd>Prediction disabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_PREDICTION_DISABLED(x) OPUS_GET_PREDICTION_DISABLED_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_genericctls Generic CTLs
+  *
+  * These macros are used with the \c opus_decoder_ctl and
+  * \c opus_encoder_ctl calls to generate a particular
+  * request.
+  *
+  * When called on an \c OpusDecoder they apply to that
+  * particular decoder instance. When called on an
+  * \c OpusEncoder they apply to the corresponding setting
+  * on that encoder instance, if present.
+  *
+  * Some usage examples:
+  *
+  * @code
+  * int ret;
+  * opus_int32 pitch;
+  * ret = opus_decoder_ctl(dec_ctx, OPUS_GET_PITCH(&pitch));
+  * if (ret == OPUS_OK) return ret;
+  *
+  * opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
+  * opus_decoder_ctl(dec_ctx, OPUS_RESET_STATE);
+  *
+  * opus_int32 enc_bw, dec_bw;
+  * opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&enc_bw));
+  * opus_decoder_ctl(dec_ctx, OPUS_GET_BANDWIDTH(&dec_bw));
+  * if (enc_bw != dec_bw) {
+  *   printf("packet bandwidth mismatch!\n");
+  * }
+  * @endcode
+  *
+  * @see opus_encoder, opus_decoder_ctl, opus_encoder_ctl, opus_decoderctls, opus_encoderctls
+  * @{
+  */
+
+/** Resets the codec state to be equivalent to a freshly initialized state.
+  * This should be called when switching streams in order to prevent
+  * the back to back decoding from giving different results from
+  * one at a time decoding.
+  * @hideinitializer */
+#define OPUS_RESET_STATE 4028
+
+/** Gets the final state of the codec's entropy coder.
+  * This is used for testing purposes,
+  * The encoder and decoder state should be identical after coding a payload
+  * (assuming no data corruption or software bugs)
+  *
+  * @param[out] x <tt>opus_uint32 *</tt>: Entropy coder state
+  *
+  * @hideinitializer */
+#define OPUS_GET_FINAL_RANGE(x) OPUS_GET_FINAL_RANGE_REQUEST, __opus_check_uint_ptr(x)
+
+/** Gets the encoder's configured bandpass or the decoder's last bandpass.
+  * @see OPUS_SET_BANDWIDTH
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>                    <dd>(default)</dd>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_BANDWIDTH(x) OPUS_GET_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the sampling rate the encoder or decoder was initialized with.
+  * This simply returns the <code>Fs</code> value passed to opus_encoder_init()
+  * or opus_decoder_init().
+  * @param[out] x <tt>opus_int32 *</tt>: Sampling rate of encoder or decoder.
+  * @hideinitializer
+  */
+#define OPUS_GET_SAMPLE_RATE(x) OPUS_GET_SAMPLE_RATE_REQUEST, __opus_check_int_ptr(x)
+
+/** If set to 1, disables the use of phase inversion for intensity stereo,
+  * improving the quality of mono downmixes, but slightly reducing normal
+  * stereo quality. Disabling phase inversion in the decoder does not comply
+  * with RFC 6716, although it does not cause any interoperability issue and
+  * is expected to become part of the Opus standard once RFC 6716 is updated
+  * by draft-ietf-codec-opus-update.
+  * @see OPUS_GET_PHASE_INVERSION_DISABLED
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Enable phase inversion (default).</dd>
+  * <dt>1</dt><dd>Disable phase inversion.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_PHASE_INVERSION_DISABLED(x) OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured phase inversion status.
+  * @see OPUS_SET_PHASE_INVERSION_DISABLED
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Stereo phase inversion enabled (default).</dd>
+  * <dt>1</dt><dd>Stereo phase inversion disabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_PHASE_INVERSION_DISABLED(x) OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST, __opus_check_int_ptr(x)
+/** Gets the DTX state of the encoder.
+  * Returns whether the last encoded frame was either a comfort noise update
+  * during DTX or not encoded because of DTX.
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>The encoder is not in DTX.</dd>
+  * <dt>1</dt><dd>The encoder is in DTX.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_IN_DTX(x) OPUS_GET_IN_DTX_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_decoderctls Decoder related CTLs
+  * @see opus_genericctls, opus_encoderctls, opus_decoder
+  * @{
+  */
+
+/** Configures decoder gain adjustment.
+  * Scales the decoded output by a factor specified in Q8 dB units.
+  * This has a maximum range of -32768 to 32767 inclusive, and returns
+  * OPUS_BAD_ARG otherwise. The default is zero indicating no adjustment.
+  * This setting survives decoder reset.
+  *
+  * gain = pow(10, x/(20.0*256))
+  *
+  * @param[in] x <tt>opus_int32</tt>:   Amount to scale PCM signal by in Q8 dB units.
+  * @hideinitializer */
+#define OPUS_SET_GAIN(x) OPUS_SET_GAIN_REQUEST, __opus_check_int(x)
+/** Gets the decoder's configured gain adjustment. @see OPUS_SET_GAIN
+  *
+  * @param[out] x <tt>opus_int32 *</tt>: Amount to scale PCM signal by in Q8 dB units.
+  * @hideinitializer */
+#define OPUS_GET_GAIN(x) OPUS_GET_GAIN_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the duration (in samples) of the last packet successfully decoded or concealed.
+  * @param[out] x <tt>opus_int32 *</tt>: Number of samples (at current sampling rate).
+  * @hideinitializer */
+#define OPUS_GET_LAST_PACKET_DURATION(x) OPUS_GET_LAST_PACKET_DURATION_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the pitch of the last decoded frame, if available.
+  * This can be used for any post-processing algorithm requiring the use of pitch,
+  * e.g. time stretching/shortening. If the last frame was not voiced, or if the
+  * pitch was not coded in the frame, then zero is returned.
+  *
+  * This CTL is only implemented for decoder instances.
+  *
+  * @param[out] x <tt>opus_int32 *</tt>: pitch period at 48 kHz (or 0 if not available)
+  *
+  * @hideinitializer */
+#define OPUS_GET_PITCH(x) OPUS_GET_PITCH_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_libinfo Opus library information functions
+  * @{
+  */
+
+/** Converts an opus error code into a human readable string.
+  *
+  * @param[in] error <tt>int</tt>: Error number
+  * @returns Error string
+  */
+OPUS_EXPORT const char *opus_strerror(int error);
+
+/** Gets the libopus version string.
+  *
+  * Applications may look for the substring "-fixed" in the version string to
+  * determine whether they have a fixed-point or floating-point build at
+  * runtime.
+  *
+  * @returns Version string
+  */
+OPUS_EXPORT const char *opus_get_version_string(void);
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_DEFINES_H */

engine/voice_codecs/opus/opus/opus_multistream.h

@@ -0,0 +1,660 @@
+/* Copyright (c) 2011 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus_multistream.h
+ * @brief Opus reference implementation multistream API
+ */
+
+#ifndef OPUS_MULTISTREAM_H
+#define OPUS_MULTISTREAM_H
+
+#include "opus.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** @cond OPUS_INTERNAL_DOC */
+
+/** Macros to trigger compilation errors when the wrong types are provided to a
+  * CTL. */
+/**@{*/
+#define __opus_check_encstate_ptr(ptr) ((ptr) + ((ptr) - (OpusEncoder**)(ptr)))
+#define __opus_check_decstate_ptr(ptr) ((ptr) + ((ptr) - (OpusDecoder**)(ptr)))
+/**@}*/
+
+/** These are the actual encoder and decoder CTL ID numbers.
+  * They should not be used directly by applications.
+  * In general, SETs should be even and GETs should be odd.*/
+/**@{*/
+#define OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST 5120
+#define OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST 5122
+/**@}*/
+
+/** @endcond */
+
+/** @defgroup opus_multistream_ctls Multistream specific encoder and decoder CTLs
+  *
+  * These are convenience macros that are specific to the
+  * opus_multistream_encoder_ctl() and opus_multistream_decoder_ctl()
+  * interface.
+  * The CTLs from @ref opus_genericctls, @ref opus_encoderctls, and
+  * @ref opus_decoderctls may be applied to a multistream encoder or decoder as
+  * well.
+  * In addition, you may retrieve the encoder or decoder state for an specific
+  * stream via #OPUS_MULTISTREAM_GET_ENCODER_STATE or
+  * #OPUS_MULTISTREAM_GET_DECODER_STATE and apply CTLs to it individually.
+  */
+/**@{*/
+
+/** Gets the encoder state for an individual stream of a multistream encoder.
+  * @param[in] x <tt>opus_int32</tt>: The index of the stream whose encoder you
+  *                                   wish to retrieve.
+  *                                   This must be non-negative and less than
+  *                                   the <code>streams</code> parameter used
+  *                                   to initialize the encoder.
+  * @param[out] y <tt>OpusEncoder**</tt>: Returns a pointer to the given
+  *                                       encoder state.
+  * @retval OPUS_BAD_ARG The index of the requested stream was out of range.
+  * @hideinitializer
+  */
+#define OPUS_MULTISTREAM_GET_ENCODER_STATE(x,y) OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST, __opus_check_int(x), __opus_check_encstate_ptr(y)
+
+/** Gets the decoder state for an individual stream of a multistream decoder.
+  * @param[in] x <tt>opus_int32</tt>: The index of the stream whose decoder you
+  *                                   wish to retrieve.
+  *                                   This must be non-negative and less than
+  *                                   the <code>streams</code> parameter used
+  *                                   to initialize the decoder.
+  * @param[out] y <tt>OpusDecoder**</tt>: Returns a pointer to the given
+  *                                       decoder state.
+  * @retval OPUS_BAD_ARG The index of the requested stream was out of range.
+  * @hideinitializer
+  */
+#define OPUS_MULTISTREAM_GET_DECODER_STATE(x,y) OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST, __opus_check_int(x), __opus_check_decstate_ptr(y)
+
+/**@}*/
+
+/** @defgroup opus_multistream Opus Multistream API
+  * @{
+  *
+  * The multistream API allows individual Opus streams to be combined into a
+  * single packet, enabling support for up to 255 channels. Unlike an
+  * elementary Opus stream, the encoder and decoder must negotiate the channel
+  * configuration before the decoder can successfully interpret the data in the
+  * packets produced by the encoder. Some basic information, such as packet
+  * duration, can be computed without any special negotiation.
+  *
+  * The format for multistream Opus packets is defined in
+  * <a href="https://tools.ietf.org/html/rfc7845">RFC 7845</a>
+  * and is based on the self-delimited Opus framing described in Appendix B of
+  * <a href="https://tools.ietf.org/html/rfc6716">RFC 6716</a>.
+  * Normal Opus packets are just a degenerate case of multistream Opus packets,
+  * and can be encoded or decoded with the multistream API by setting
+  * <code>streams</code> to <code>1</code> when initializing the encoder or
+  * decoder.
+  *
+  * Multistream Opus streams can contain up to 255 elementary Opus streams.
+  * These may be either "uncoupled" or "coupled", indicating that the decoder
+  * is configured to decode them to either 1 or 2 channels, respectively.
+  * The streams are ordered so that all coupled streams appear at the
+  * beginning.
+  *
+  * A <code>mapping</code> table defines which decoded channel <code>i</code>
+  * should be used for each input/output (I/O) channel <code>j</code>. This table is
+  * typically provided as an unsigned char array.
+  * Let <code>i = mapping[j]</code> be the index for I/O channel <code>j</code>.
+  * If <code>i < 2*coupled_streams</code>, then I/O channel <code>j</code> is
+  * encoded as the left channel of stream <code>(i/2)</code> if <code>i</code>
+  * is even, or  as the right channel of stream <code>(i/2)</code> if
+  * <code>i</code> is odd. Otherwise, I/O channel <code>j</code> is encoded as
+  * mono in stream <code>(i - coupled_streams)</code>, unless it has the special
+  * value 255, in which case it is omitted from the encoding entirely (the
+  * decoder will reproduce it as silence). Each value <code>i</code> must either
+  * be the special value 255 or be less than <code>streams + coupled_streams</code>.
+  *
+  * The output channels specified by the encoder
+  * should use the
+  * <a href="https://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-810004.3.9">Vorbis
+  * channel ordering</a>. A decoder may wish to apply an additional permutation
+  * to the mapping the encoder used to achieve a different output channel
+  * order (e.g. for outputing in WAV order).
+  *
+  * Each multistream packet contains an Opus packet for each stream, and all of
+  * the Opus packets in a single multistream packet must have the same
+  * duration. Therefore the duration of a multistream packet can be extracted
+  * from the TOC sequence of the first stream, which is located at the
+  * beginning of the packet, just like an elementary Opus stream:
+  *
+  * @code
+  * int nb_samples;
+  * int nb_frames;
+  * nb_frames = opus_packet_get_nb_frames(data, len);
+  * if (nb_frames < 1)
+  *   return nb_frames;
+  * nb_samples = opus_packet_get_samples_per_frame(data, 48000) * nb_frames;
+  * @endcode
+  *
+  * The general encoding and decoding process proceeds exactly the same as in
+  * the normal @ref opus_encoder and @ref opus_decoder APIs.
+  * See their documentation for an overview of how to use the corresponding
+  * multistream functions.
+  */
+
+/** Opus multistream encoder state.
+  * This contains the complete state of a multistream Opus encoder.
+  * It is position independent and can be freely copied.
+  * @see opus_multistream_encoder_create
+  * @see opus_multistream_encoder_init
+  */
+typedef struct OpusMSEncoder OpusMSEncoder;
+
+/** Opus multistream decoder state.
+  * This contains the complete state of a multistream Opus decoder.
+  * It is position independent and can be freely copied.
+  * @see opus_multistream_decoder_create
+  * @see opus_multistream_decoder_init
+  */
+typedef struct OpusMSDecoder OpusMSDecoder;
+
+/**\name Multistream encoder functions */
+/**@{*/
+
+/** Gets the size of an OpusMSEncoder structure.
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_encoder_get_size(
+      int streams,
+      int coupled_streams
+);
+
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_surround_encoder_get_size(
+      int channels,
+      int mapping_family
+);
+
+
+/** Allocates and initializes a multistream encoder state.
+  * Call opus_multistream_encoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than the number of channels.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than the number of input channels.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    encoded channels to input channels, as described in
+  *                    @ref opus_multistream. As an extra constraint, the
+  *                    multistream encoder does not allow encoding coupled
+  *                    streams for which one channel is unused since this
+  *                    is never a good idea.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_encoder_create(
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int application,
+      int *error
+) OPUS_ARG_NONNULL(5);
+
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_surround_encoder_create(
+      opus_int32 Fs,
+      int channels,
+      int mapping_family,
+      int *streams,
+      int *coupled_streams,
+      unsigned char *mapping,
+      int application,
+      int *error
+) OPUS_ARG_NONNULL(4) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6);
+
+/** Initialize a previously allocated multistream encoder state.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_multistream_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_multistream_encoder_create
+  * @see opus_multistream_encoder_get_size
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than the number of channels.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than the number of input channels.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    encoded channels to input channels, as described in
+  *                    @ref opus_multistream. As an extra constraint, the
+  *                    multistream encoder does not allow encoding coupled
+  *                    streams for which one channel is unused since this
+  *                    is never a good idea.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_multistream_encoder_init(
+      OpusMSEncoder *st,
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int application
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
+
+OPUS_EXPORT int opus_multistream_surround_encoder_init(
+      OpusMSEncoder *st,
+      opus_int32 Fs,
+      int channels,
+      int mapping_family,
+      int *streams,
+      int *coupled_streams,
+      unsigned char *mapping,
+      int application
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6) OPUS_ARG_NONNULL(7);
+
+/** Encodes a multistream Opus frame.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param[in] pcm <tt>const opus_int16*</tt>: The input signal as interleaved
+  *                                            samples.
+  *                                            This must contain
+  *                                            <code>frame_size*channels</code>
+  *                                            samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode(
+    OpusMSEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes a multistream Opus frame from floating point input.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param[in] pcm <tt>const float*</tt>: The input signal as interleaved
+  *                                       samples with a normal range of
+  *                                       +/-1.0.
+  *                                       Samples with a range beyond +/-1.0
+  *                                       are supported but will be clipped by
+  *                                       decoders using the integer API and
+  *                                       should only be used if it is known
+  *                                       that the far end supports extended
+  *                                       dynamic range.
+  *                                       This must contain
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode_float(
+      OpusMSEncoder *st,
+      const float *pcm,
+      int frame_size,
+      unsigned char *data,
+      opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Frees an <code>OpusMSEncoder</code> allocated by
+  * opus_multistream_encoder_create().
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to be freed.
+  */
+OPUS_EXPORT void opus_multistream_encoder_destroy(OpusMSEncoder *st);
+
+/** Perform a CTL function on a multistream Opus encoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_encoderctls, or @ref opus_multistream_ctls.
+  * @see opus_genericctls
+  * @see opus_encoderctls
+  * @see opus_multistream_ctls
+  */
+OPUS_EXPORT int opus_multistream_encoder_ctl(OpusMSEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/**@}*/
+
+/**\name Multistream decoder functions */
+/**@{*/
+
+/** Gets the size of an <code>OpusMSDecoder</code> structure.
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_decoder_get_size(
+      int streams,
+      int coupled_streams
+);
+
+/** Allocates and initializes a multistream decoder state.
+  * Call opus_multistream_decoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    coded channels to output channels, as described in
+  *                    @ref opus_multistream.
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSDecoder *opus_multistream_decoder_create(
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int *error
+) OPUS_ARG_NONNULL(5);
+
+/** Intialize a previously allocated decoder state object.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_multistream_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_multistream_decoder_create
+  * @see opus_multistream_deocder_get_size
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    coded channels to output channels, as described in
+  *                    @ref opus_multistream.
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_multistream_decoder_init(
+      OpusMSDecoder *st,
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
+
+/** Decode a multistream Opus packet.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode(
+    OpusMSDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int16 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode a multistream Opus packet with floating point output.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode_float(
+    OpusMSDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    float *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on a multistream Opus decoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_decoderctls, or @ref opus_multistream_ctls.
+  * @see opus_genericctls
+  * @see opus_decoderctls
+  * @see opus_multistream_ctls
+  */
+OPUS_EXPORT int opus_multistream_decoder_ctl(OpusMSDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/** Frees an <code>OpusMSDecoder</code> allocated by
+  * opus_multistream_decoder_create().
+  * @param st <tt>OpusMSDecoder</tt>: Multistream decoder state to be freed.
+  */
+OPUS_EXPORT void opus_multistream_decoder_destroy(OpusMSDecoder *st);
+
+/**@}*/
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_MULTISTREAM_H */

engine/voice_codecs/opus/opus/opus_projection.h

@@ -0,0 +1,568 @@
+/* Copyright (c) 2017 Google Inc.
+   Written by Andrew Allen */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus_projection.h
+ * @brief Opus projection reference API
+ */
+
+#ifndef OPUS_PROJECTION_H
+#define OPUS_PROJECTION_H
+
+#include "opus_multistream.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** @cond OPUS_INTERNAL_DOC */
+
+/** These are the actual encoder and decoder CTL ID numbers.
+  * They should not be used directly by applications.c
+  * In general, SETs should be even and GETs should be odd.*/
+/**@{*/
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN_REQUEST    6001
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE_REQUEST    6003
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_REQUEST         6005
+/**@}*/
+
+
+/** @endcond */
+
+/** @defgroup opus_projection_ctls Projection specific encoder and decoder CTLs
+  *
+  * These are convenience macros that are specific to the
+  * opus_projection_encoder_ctl() and opus_projection_decoder_ctl()
+  * interface.
+  * The CTLs from @ref opus_genericctls, @ref opus_encoderctls,
+  * @ref opus_decoderctls, and @ref opus_multistream_ctls may be applied to a
+  * projection encoder or decoder as well.
+  */
+/**@{*/
+
+/** Gets the gain (in dB. S7.8-format) of the demixing matrix from the encoder.
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the gain (in dB. S7.8-format)
+  *                                      of the demixing matrix.
+  * @hideinitializer
+  */
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN(x) OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN_REQUEST, __opus_check_int_ptr(x)
+
+
+/** Gets the size in bytes of the demixing matrix from the encoder.
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the size in bytes of the
+  *                                      demixing matrix.
+  * @hideinitializer
+  */
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE(x) OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE_REQUEST, __opus_check_int_ptr(x)
+
+
+/** Copies the demixing matrix to the supplied pointer location.
+  * @param[out] x <tt>unsigned char *</tt>: Returns the demixing matrix to the
+  *                                         supplied pointer location.
+  * @param y <tt>opus_int32</tt>: The size in bytes of the reserved memory at the
+  *                              pointer location.
+  * @hideinitializer
+  */
+#define OPUS_PROJECTION_GET_DEMIXING_MATRIX(x,y) OPUS_PROJECTION_GET_DEMIXING_MATRIX_REQUEST, x, __opus_check_int(y)
+
+
+/**@}*/
+
+/** Opus projection encoder state.
+ * This contains the complete state of a projection Opus encoder.
+ * It is position independent and can be freely copied.
+ * @see opus_projection_ambisonics_encoder_create
+ */
+typedef struct OpusProjectionEncoder OpusProjectionEncoder;
+
+
+/** Opus projection decoder state.
+  * This contains the complete state of a projection Opus decoder.
+  * It is position independent and can be freely copied.
+  * @see opus_projection_decoder_create
+  * @see opus_projection_decoder_init
+  */
+typedef struct OpusProjectionDecoder OpusProjectionDecoder;
+
+
+/**\name Projection encoder functions */
+/**@{*/
+
+/** Gets the size of an OpusProjectionEncoder structure.
+  * @param channels <tt>int</tt>: The total number of input channels to encode.
+  *                               This must be no more than 255.
+  * @param mapping_family <tt>int</tt>: The mapping family to use for selecting
+  *                                     the appropriate projection.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_projection_ambisonics_encoder_get_size(
+    int channels,
+    int mapping_family
+);
+
+
+/** Allocates and initializes a projection encoder state.
+  * Call opus_projection_encoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param mapping_family <tt>int</tt>: The mapping family to use for selecting
+  *                                     the appropriate projection.
+  * @param[out] streams <tt>int *</tt>: The total number of streams that will
+  *                                     be encoded from the input.
+  * @param[out] coupled_streams <tt>int *</tt>: Number of coupled (2 channel)
+  *                                 streams that will be encoded from the input.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusProjectionEncoder *opus_projection_ambisonics_encoder_create(
+    opus_int32 Fs,
+    int channels,
+    int mapping_family,
+    int *streams,
+    int *coupled_streams,
+    int application,
+    int *error
+) OPUS_ARG_NONNULL(4) OPUS_ARG_NONNULL(5);
+
+
+/** Initialize a previously allocated projection encoder state.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_projection_ambisonics_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_projection_ambisonics_encoder_create
+  * @see opus_projection_ambisonics_encoder_get_size
+  * @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than the number of channels.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than the number of input channels.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_projection_ambisonics_encoder_init(
+    OpusProjectionEncoder *st,
+    opus_int32 Fs,
+    int channels,
+    int mapping_family,
+    int *streams,
+    int *coupled_streams,
+    int application
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6);
+
+
+/** Encodes a projection Opus frame.
+  * @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
+  * @param[in] pcm <tt>const opus_int16*</tt>: The input signal as interleaved
+  *                                            samples.
+  *                                            This must contain
+  *                                            <code>frame_size*channels</code>
+  *                                            samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_encode(
+    OpusProjectionEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+
+/** Encodes a projection Opus frame from floating point input.
+  * @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
+  * @param[in] pcm <tt>const float*</tt>: The input signal as interleaved
+  *                                       samples with a normal range of
+  *                                       +/-1.0.
+  *                                       Samples with a range beyond +/-1.0
+  *                                       are supported but will be clipped by
+  *                                       decoders using the integer API and
+  *                                       should only be used if it is known
+  *                                       that the far end supports extended
+  *                                       dynamic range.
+  *                                       This must contain
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_encode_float(
+    OpusProjectionEncoder *st,
+    const float *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+
+/** Frees an <code>OpusProjectionEncoder</code> allocated by
+  * opus_projection_ambisonics_encoder_create().
+  * @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state to be freed.
+  */
+OPUS_EXPORT void opus_projection_encoder_destroy(OpusProjectionEncoder *st);
+
+
+/** Perform a CTL function on a projection Opus encoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_encoderctls, @ref opus_multistream_ctls, or
+  *                @ref opus_projection_ctls
+  * @see opus_genericctls
+  * @see opus_encoderctls
+  * @see opus_multistream_ctls
+  * @see opus_projection_ctls
+  */
+OPUS_EXPORT int opus_projection_encoder_ctl(OpusProjectionEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+
+/**@}*/
+
+/**\name Projection decoder functions */
+/**@{*/
+
+/** Gets the size of an <code>OpusProjectionDecoder</code> structure.
+  * @param channels <tt>int</tt>: The total number of output channels.
+  *                               This must be no more than 255.
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_projection_decoder_get_size(
+    int channels,
+    int streams,
+    int coupled_streams
+);
+
+
+/** Allocates and initializes a projection decoder state.
+  * Call opus_projection_decoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] demixing_matrix <tt>const unsigned char[demixing_matrix_size]</tt>: Demixing matrix
+  *                         that mapping from coded channels to output channels,
+  *                         as described in @ref opus_projection and
+  *                         @ref opus_projection_ctls.
+  * @param demixing_matrix_size <tt>opus_int32</tt>: The size in bytes of the
+  *                                                  demixing matrix, as
+  *                                                  described in @ref
+  *                                                  opus_projection_ctls.
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusProjectionDecoder *opus_projection_decoder_create(
+    opus_int32 Fs,
+    int channels,
+    int streams,
+    int coupled_streams,
+    unsigned char *demixing_matrix,
+    opus_int32 demixing_matrix_size,
+    int *error
+) OPUS_ARG_NONNULL(5);
+
+
+/** Intialize a previously allocated projection decoder state object.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_projection_decoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_projection_decoder_create
+  * @see opus_projection_deocder_get_size
+  * @param st <tt>OpusProjectionDecoder*</tt>: Projection encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] demixing_matrix <tt>const unsigned char[demixing_matrix_size]</tt>: Demixing matrix
+  *                         that mapping from coded channels to output channels,
+  *                         as described in @ref opus_projection and
+  *                         @ref opus_projection_ctls.
+  * @param demixing_matrix_size <tt>opus_int32</tt>: The size in bytes of the
+  *                                                  demixing matrix, as
+  *                                                  described in @ref
+  *                                                  opus_projection_ctls.
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_projection_decoder_init(
+    OpusProjectionDecoder *st,
+    opus_int32 Fs,
+    int channels,
+    int streams,
+    int coupled_streams,
+    unsigned char *demixing_matrix,
+    opus_int32 demixing_matrix_size
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
+
+
+/** Decode a projection Opus packet.
+  * @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_decode(
+    OpusProjectionDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int16 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+
+/** Decode a projection Opus packet with floating point output.
+  * @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_decode_float(
+    OpusProjectionDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    float *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+
+/** Perform a CTL function on a projection Opus decoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_decoderctls, @ref opus_multistream_ctls, or
+  *                @ref opus_projection_ctls.
+  * @see opus_genericctls
+  * @see opus_decoderctls
+  * @see opus_multistream_ctls
+  * @see opus_projection_ctls
+  */
+OPUS_EXPORT int opus_projection_decoder_ctl(OpusProjectionDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+
+/** Frees an <code>OpusProjectionDecoder</code> allocated by
+  * opus_projection_decoder_create().
+  * @param st <tt>OpusProjectionDecoder</tt>: Projection decoder state to be freed.
+  */
+OPUS_EXPORT void opus_projection_decoder_destroy(OpusProjectionDecoder *st);
+
+
+/**@}*/
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_PROJECTION_H */

engine/voice_codecs/opus/opus/opus_types.h

@@ -0,0 +1,166 @@
+/* (C) COPYRIGHT 1994-2002 Xiph.Org Foundation */
+/* Modified by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+/* opus_types.h based on ogg_types.h from libogg */
+
+/**
+   @file opus_types.h
+   @brief Opus reference implementation types
+*/
+#ifndef OPUS_TYPES_H
+#define OPUS_TYPES_H
+
+#define opus_int         int                     /* used for counters etc; at least 16 bits */
+#define opus_int64       long long
+#define opus_int8        signed char
+
+#define opus_uint        unsigned int            /* used for counters etc; at least 16 bits */
+#define opus_uint64      unsigned long long
+#define opus_uint8       unsigned char
+
+/* Use the real stdint.h if it's there (taken from Paul Hsieh's pstdint.h) */
+#if (defined(__STDC__) && __STDC__ && defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L) || (defined(__GNUC__) && (defined(_STDINT_H) || defined(_STDINT_H_)) || defined (HAVE_STDINT_H))
+#include <stdint.h>
+#  undef opus_int64
+#  undef opus_int8
+#  undef opus_uint64
+#  undef opus_uint8
+   typedef int8_t opus_int8;
+   typedef uint8_t opus_uint8;
+   typedef int16_t opus_int16;
+   typedef uint16_t opus_uint16;
+   typedef int32_t opus_int32;
+   typedef uint32_t opus_uint32;
+   typedef int64_t opus_int64;
+   typedef uint64_t opus_uint64;
+#elif defined(_WIN32)
+
+#  if defined(__CYGWIN__)
+#    include <_G_config.h>
+     typedef _G_int32_t opus_int32;
+     typedef _G_uint32_t opus_uint32;
+     typedef _G_int16 opus_int16;
+     typedef _G_uint16 opus_uint16;
+#  elif defined(__MINGW32__)
+     typedef short opus_int16;
+     typedef unsigned short opus_uint16;
+     typedef int opus_int32;
+     typedef unsigned int opus_uint32;
+#  elif defined(__MWERKS__)
+     typedef int opus_int32;
+     typedef unsigned int opus_uint32;
+     typedef short opus_int16;
+     typedef unsigned short opus_uint16;
+#  else
+     /* MSVC/Borland */
+     typedef __int32 opus_int32;
+     typedef unsigned __int32 opus_uint32;
+     typedef __int16 opus_int16;
+     typedef unsigned __int16 opus_uint16;
+#  endif
+
+#elif defined(__MACOS__)
+
+#  include <sys/types.h>
+   typedef SInt16 opus_int16;
+   typedef UInt16 opus_uint16;
+   typedef SInt32 opus_int32;
+   typedef UInt32 opus_uint32;
+
+#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */
+
+#  include <sys/types.h>
+   typedef int16_t opus_int16;
+   typedef u_int16_t opus_uint16;
+   typedef int32_t opus_int32;
+   typedef u_int32_t opus_uint32;
+
+#elif defined(__BEOS__)
+
+   /* Be */
+#  include <inttypes.h>
+   typedef int16 opus_int16;
+   typedef u_int16 opus_uint16;
+   typedef int32_t opus_int32;
+   typedef u_int32_t opus_uint32;
+
+#elif defined (__EMX__)
+
+   /* OS/2 GCC */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined (DJGPP)
+
+   /* DJGPP */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined(R5900)
+
+   /* PS2 EE */
+   typedef int opus_int32;
+   typedef unsigned opus_uint32;
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+
+#elif defined(__SYMBIAN32__)
+
+   /* Symbian GCC */
+   typedef signed short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef signed int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined(CONFIG_TI_C54X) || defined (CONFIG_TI_C55X)
+
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef long opus_int32;
+   typedef unsigned long opus_uint32;
+
+#elif defined(CONFIG_TI_C6X)
+
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#else
+
+   /* Give up, take a reasonable guess */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#endif
+
+#endif  /* OPUS_TYPES_H */

engine/voice_codecs/opus/voiceencoder_opus.cpp

@@ -0,0 +1,184 @@
+//========= Copyright Valve Corporation, All rights reserved. ============//
+//
+// Purpose: 
+//
+// $NoKeywords: $
+//
+//=============================================================================//
+#include "ivoicecodec.h"
+#include "iframeencoder.h"
+
+#include <stdio.h>
+#include <opus/opus.h>
+#include <opus/opus_custom.h>
+
+
+// NOTE: This has to be the last file included!
+#include "tier0/memdbgon.h"
+#include "tier0/dbg.h"
+
+#define CHANNELS 1
+
+struct opus_options
+{
+	int iSampleRate;
+	int iRawFrameSize;
+	int iPacketSize;
+};
+
+opus_options g_OpusOpts[] =
+{
+        {44100, 256, 120},
+        {22050, 120, 60},
+        {22050, 256, 60},
+        {22050, 512, 64},
+};
+
+class VoiceEncoder_Opus : public IFrameEncoder
+{
+public:
+	VoiceEncoder_Opus();
+	virtual ~VoiceEncoder_Opus();
+
+	// Interfaces IFrameDecoder
+
+	bool Init(int quality, int &rawFrameSize, int &encodedFrameSize);
+	void Release();
+	void DecodeFrame(const char *pCompressed, char *pDecompressedBytes);
+	void EncodeFrame(const char *pUncompressedBytes, char *pCompressed);
+	bool ResetState();
+
+private:
+
+	bool	InitStates();
+	void	TermStates();
+
+	OpusCustomEncoder *m_EncoderState;	// Celt internal encoder state
+	OpusCustomDecoder *m_DecoderState; // Celt internal decoder state
+	OpusCustomMode	*m_Mode;
+
+	int m_iVersion;
+};
+
+extern IVoiceCodec* CreateVoiceCodec_Frame(IFrameEncoder *pEncoder);
+
+void* CreateCeltVoiceCodec()
+{
+	IFrameEncoder *pEncoder = new VoiceEncoder_Opus;
+	return CreateVoiceCodec_Frame( pEncoder );
+}
+
+EXPOSE_INTERFACE_FN(CreateCeltVoiceCodec, IVoiceCodec, "vaudio_opus")
+
+//////////////////////////////////////////////////////////////////////
+// Construction/Destruction
+//////////////////////////////////////////////////////////////////////
+
+VoiceEncoder_Opus::VoiceEncoder_Opus()
+{
+	m_EncoderState = NULL;
+	m_DecoderState = NULL;
+	m_Mode = NULL;
+	m_iVersion = 0;
+}
+
+VoiceEncoder_Opus::~VoiceEncoder_Opus()
+{
+	TermStates();
+}
+
+bool VoiceEncoder_Opus::Init( int quality, int &rawFrameSize, int &encodedFrameSize)
+{
+	m_iVersion = quality;
+
+	rawFrameSize = g_OpusOpts[m_iVersion].iRawFrameSize * BYTES_PER_SAMPLE;
+
+	int iError = 0;
+
+	m_Mode = opus_custom_mode_create( g_OpusOpts[m_iVersion].iSampleRate, g_OpusOpts[m_iVersion].iRawFrameSize, &iError );
+
+	m_EncoderState = opus_custom_encoder_create( m_Mode, CHANNELS, NULL);
+	m_DecoderState = opus_custom_decoder_create( m_Mode, CHANNELS, NULL);
+
+	if ( !InitStates() )
+		return false;
+
+	encodedFrameSize = g_OpusOpts[m_iVersion].iPacketSize;
+
+	return true;
+}
+
+void VoiceEncoder_Opus::Release()
+{
+	delete this;
+}
+
+void VoiceEncoder_Opus::EncodeFrame(const char *pUncompressedBytes, char *pCompressed)
+{
+	unsigned char output[1024];
+
+	opus_custom_encode( m_EncoderState, (opus_int16*)pUncompressedBytes, g_OpusOpts[m_iVersion].iRawFrameSize, output, g_OpusOpts[m_iVersion].iPacketSize );
+
+	for ( int i = 0; i < g_OpusOpts[m_iVersion].iPacketSize; i++ )
+	{
+		*pCompressed = (char)output[i];
+		pCompressed++;
+	}
+}
+
+void VoiceEncoder_Opus::DecodeFrame(const char *pCompressed, char *pDecompressedBytes)
+{
+	unsigned char output[1024];
+	char *out = (char *)pCompressed;
+
+	if ( !pCompressed )
+	{
+		int decoded = opus_custom_decode( m_DecoderState, NULL, g_OpusOpts[m_iVersion].iPacketSize, (opus_int16 *)pDecompressedBytes, g_OpusOpts[m_iVersion].iRawFrameSize );
+		return;
+	}
+
+	for ( int i = 0; i < g_OpusOpts[m_iVersion].iPacketSize; i++ )
+	{
+		output[i] = ( unsigned char ) ( ( *out < 0 ) ? (*out + 256) : *out );
+		out++;
+	}
+
+	//celt_decoder_ctl( m_DecoderState, CELT_RESET_STATE_REQUEST, NULL );
+	int decoded = opus_custom_decode( m_DecoderState, output, g_OpusOpts[m_iVersion].iPacketSize, (opus_int16 *)pDecompressedBytes, g_OpusOpts[m_iVersion].iRawFrameSize );
+}
+
+bool VoiceEncoder_Opus::ResetState()
+{
+	opus_custom_encoder_ctl(m_EncoderState, OPUS_RESET_STATE );
+	opus_custom_decoder_ctl(m_DecoderState, OPUS_RESET_STATE );
+
+	return true;
+}
+
+bool VoiceEncoder_Opus::InitStates()
+{
+	if ( !m_EncoderState || !m_DecoderState )
+		return false;
+
+	opus_custom_encoder_ctl(m_EncoderState, OPUS_RESET_STATE );
+	opus_custom_decoder_ctl(m_DecoderState, OPUS_RESET_STATE );
+
+	return true;
+}
+
+void VoiceEncoder_Opus::TermStates()
+{
+	if( m_EncoderState )
+	{
+		opus_custom_encoder_destroy( m_EncoderState );
+		m_EncoderState = NULL;
+	}
+
+	if( m_DecoderState )
+	{
+		opus_custom_decoder_destroy( m_DecoderState );
+		m_DecoderState = NULL;
+	}
+
+	opus_custom_mode_destroy( m_Mode );
+}

engine/voice_codecs/opus/wscript

@@ -0,0 +1,53 @@
+#! /usr/bin/env python
+# encoding: utf-8
+
+from waflib import Utils
+import os
+
+top = '.'
+PROJECT_NAME = 'vaudio_opus'
+
+def options(opt):
+	return
+
+def configure(conf):
+	conf.define('OPUS_EXPORTS',1)
+
+def build(bld):
+	source = [
+		'voiceencoder_opus.cpp',
+		'../frame_encoder/voice_codec_frame.cpp',
+		'../../../tier1/interface.cpp'
+	]
+
+	includes = [
+		'.',
+		'../frame_encoder',
+		'../../../public',
+		'../../../public/tier1',
+		'../../../public/tier0',
+		'../../',
+		'../../../common',
+		'../../audio/public',
+		'celt'
+	]
+
+	defines = []
+
+	libs = ['tier0','tier1','vstdlib','OPUS']
+
+	install_path = bld.env.LIBDIR
+
+	bld.shlib(
+		source   = source,
+		target   = PROJECT_NAME,
+		name     = PROJECT_NAME,
+		features = 'c cxx',
+		includes = includes,
+		defines  = defines,
+		use      = libs,
+		install_path = install_path,
+		subsystem = bld.env.MSVC_SUBSYSTEM,
+		idx      = bld.get_taskgen_count()
+	)
+

engine/voice_codecs/speex/VoiceEncoder_Speex.h

@@ -45,7 +45,7 @@ damage. */
 
 #include "iframeencoder.h"
 
-#include <speex.h>
+#include <speex/speex.h>
 
 class VoiceEncoder_Speex : public IFrameEncoder
 {

engine/voice_codecs/speex/source/AUTHORS

@@ -1,15 +0,0 @@
-Jean-Marc Valin <jean-marc.valin@hermes.usherb.ca>
-   All the code except the following
-
-David Rowe <david@voicetronix.com.au> via VoiceTronix
-   lsp.c lsp.h 
-   Also ideas and feedback
-
-John Francis Edwards:
-   wave_out.[ch], some #ifdefs for windows port and MSVC project files
-
-Atsuhiko Yamanaka <ymnk@jcraft.com>:
-   Patch to speexenc.c to add Vorbis comment format
-
-Radim Kolar <hsn@cybermail.net>:
-   Patch to speexenc.c for supporting more input formats

engine/voice_codecs/speex/source/COPYING

@@ -1,26 +0,0 @@
-Redistribution and use in source and binary forms, with or without
-modification, are permitted provided that the following conditions
-are met:
-
-- Redistributions of source code must retain the above copyright
-notice, this list of conditions and the following disclaimer.
-
-- Redistributions in binary form must reproduce the above copyright
-notice, this list of conditions and the following disclaimer in the
-documentation and/or other materials provided with the distribution.
-
-- Neither the name of the Xiph.org Foundation nor the names of its
-contributors may be used to endorse or promote products derived from
-this software without specific prior written permission.
-
-THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.

engine/voice_codecs/speex/source/ChangeLog

@@ -1,6 +0,0 @@
-2002/03/27 Jean-Marc Valin:
-Working encoder and decoder for both narrowband and wideband.
-
-2002/02/27 Jean-Marc Valin:
-Got the basic encoder working as a demo with quantization only on some 
-parameters.

engine/voice_codecs/speex/source/INSTALL

@@ -1,8 +0,0 @@
-Installing Speex is as easy as:
-
-% ./configure [--prefix=<install-path>]
-% make
-% make install
-
-Note that if you are using the code from CVS, you will need to run "autogen.sh"
-instead of "configure".

engine/voice_codecs/speex/source/Makefile.am

@@ -1,11 +0,0 @@
-## Process this file with automake to produce Makefile.in. -*-Makefile-*-
-
-# Disable automatic dependency tracking if using other tools than gcc and gmake
-#AUTOMAKE_OPTIONS = no-dependencies
-
-EXTRA_DIST = Speex.spec Speex.spec.in
-
-SUBDIRS = libspeex @src@ doc win32
-
-rpm: dist
-	rpmbuild -ta ${PACKAGE}-${VERSION}.tar.gz

engine/voice_codecs/speex/source/Makefile.in

@@ -1,347 +0,0 @@
-# Makefile.in generated automatically by automake 1.4-p6 from Makefile.am
-
-# Copyright (C) 1994, 1995-8, 1999, 2001 Free Software Foundation, Inc.
-# This Makefile.in is free software; the Free Software Foundation
-# gives unlimited permission to copy and/or distribute it,
-# with or without modifications, as long as this notice is preserved.
-
-# This program is distributed in the hope that it will be useful,
-# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
-# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
-# PARTICULAR PURPOSE.
-
-# Disable automatic dependency tracking if using other tools than gcc and gmake
-#AUTOMAKE_OPTIONS = no-dependencies
-
-
-SHELL = @SHELL@
-
-srcdir = @srcdir@
-top_srcdir = @top_srcdir@
-VPATH = @srcdir@
-prefix = @prefix@
-exec_prefix = @exec_prefix@
-
-bindir = @bindir@
-sbindir = @sbindir@
-libexecdir = @libexecdir@
-datadir = @datadir@
-sysconfdir = @sysconfdir@
-sharedstatedir = @sharedstatedir@
-localstatedir = @localstatedir@
-libdir = @libdir@
-infodir = @infodir@
-mandir = @mandir@
-includedir = @includedir@
-oldincludedir = /usr/include
-
-DESTDIR =
-
-pkgdatadir = $(datadir)/@PACKAGE@
-pkglibdir = $(libdir)/@PACKAGE@
-pkgincludedir = $(includedir)/@PACKAGE@
-
-top_builddir = .
-
-ACLOCAL = @ACLOCAL@
-AUTOCONF = @AUTOCONF@
-AUTOMAKE = @AUTOMAKE@
-AUTOHEADER = @AUTOHEADER@
-
-INSTALL = @INSTALL@
-INSTALL_PROGRAM = @INSTALL_PROGRAM@ $(AM_INSTALL_PROGRAM_FLAGS)
-INSTALL_DATA = @INSTALL_DATA@
-INSTALL_SCRIPT = @INSTALL_SCRIPT@
-transform = @program_transform_name@
-
-NORMAL_INSTALL = :
-PRE_INSTALL = :
-POST_INSTALL = :
-NORMAL_UNINSTALL = :
-PRE_UNINSTALL = :
-POST_UNINSTALL = :
-host_alias = @host_alias@
-host_triplet = @host@
-AS = @AS@
-CC = @CC@
-DLLTOOL = @DLLTOOL@
-ECHO = @ECHO@
-EXEEXT = @EXEEXT@
-LIBTOOL = @LIBTOOL@
-LN_S = @LN_S@
-MAINT = @MAINT@
-MAKEINFO = @MAKEINFO@
-OBJDUMP = @OBJDUMP@
-OBJEXT = @OBJEXT@
-OGG_INCLUDES = @OGG_INCLUDES@
-OGG_LDFLAGS = @OGG_LDFLAGS@
-OGG_LIBS = @OGG_LIBS@
-PACKAGE = @PACKAGE@
-RANLIB = @RANLIB@
-SPEEX_LT_AGE = @SPEEX_LT_AGE@
-SPEEX_LT_CURRENT = @SPEEX_LT_CURRENT@
-SPEEX_LT_REVISION = @SPEEX_LT_REVISION@
-STRIP = @STRIP@
-VERSION = @VERSION@
-src = @src@
-
-EXTRA_DIST = Speex.spec Speex.spec.in
-
-SUBDIRS = libspeex @src@ doc win32
-ACLOCAL_M4 = $(top_srcdir)/aclocal.m4
-mkinstalldirs = $(SHELL) $(top_srcdir)/mkinstalldirs
-CONFIG_CLEAN_FILES =  Speex.spec
-DIST_COMMON =  README AUTHORS COPYING ChangeLog INSTALL Makefile.am \
-Makefile.in NEWS Speex.spec.in TODO acinclude.m4 aclocal.m4 \
-config.guess config.sub configure configure.in install-sh ltmain.sh \
-missing mkinstalldirs
-
-
-DISTFILES = $(DIST_COMMON) $(SOURCES) $(HEADERS) $(TEXINFOS) $(EXTRA_DIST)
-
-TAR = gtar
-GZIP_ENV = --best
-all: all-redirect
-.SUFFIXES:
-$(srcdir)/Makefile.in: @MAINTAINER_MODE_TRUE@ Makefile.am $(top_srcdir)/configure.in $(ACLOCAL_M4) 
-	cd $(top_srcdir) && $(AUTOMAKE) --gnu --include-deps Makefile
-
-Makefile: $(srcdir)/Makefile.in  $(top_builddir)/config.status
-	cd $(top_builddir) \
-	  && CONFIG_FILES=$@ CONFIG_HEADERS= $(SHELL) ./config.status
-
-$(ACLOCAL_M4): @MAINTAINER_MODE_TRUE@ configure.in  acinclude.m4
-	cd $(srcdir) && $(ACLOCAL)
-
-config.status: $(srcdir)/configure $(CONFIG_STATUS_DEPENDENCIES)
-	$(SHELL) ./config.status --recheck
-$(srcdir)/configure: @MAINTAINER_MODE_TRUE@$(srcdir)/configure.in $(ACLOCAL_M4) $(CONFIGURE_DEPENDENCIES)
-	cd $(srcdir) && $(AUTOCONF)
-Speex.spec: $(top_builddir)/config.status Speex.spec.in
-	cd $(top_builddir) && CONFIG_FILES=$@ CONFIG_HEADERS= $(SHELL) ./config.status
-
-# This directory's subdirectories are mostly independent; you can cd
-# into them and run `make' without going through this Makefile.
-# To change the values of `make' variables: instead of editing Makefiles,
-# (1) if the variable is set in `config.status', edit `config.status'
-#     (which will cause the Makefiles to be regenerated when you run `make');
-# (2) otherwise, pass the desired values on the `make' command line.
-
-@SET_MAKE@
-
-all-recursive install-data-recursive install-exec-recursive \
-installdirs-recursive install-recursive uninstall-recursive  \
-check-recursive installcheck-recursive info-recursive dvi-recursive:
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-	dot_seen=no; \
-	target=`echo $@ | sed s/-recursive//`; \
-	list='$(SUBDIRS)'; for subdir in $$list; do \
-	  echo "Making $$target in $$subdir"; \
-	  if test "$$subdir" = "."; then \
-	    dot_seen=yes; \
-	    local_target="$$target-am"; \
-	  else \
-	    local_target="$$target"; \
-	  fi; \
-	  (cd $$subdir && $(MAKE) $(AM_MAKEFLAGS) $$local_target) \
-	   || case "$$amf" in *=*) exit 1;; *k*) fail=yes;; *) exit 1;; esac; \
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-	if test "$$dot_seen" = "no"; then \
-	  $(MAKE) $(AM_MAKEFLAGS) "$$target-am" || exit 1; \
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engine/voice_codecs/speex/source/NEWS

@@ -1 +0,0 @@
-2002/02/13: Creation of the "Speex" project

engine/voice_codecs/speex/source/README

@@ -1,9 +0,0 @@
-See INSTALL file for instruction on how to install Speex.
-
-The Speex is a patent-free, Open Source/Free Software voice codec. Unlike other codecs like MP3 and Ogg Vorbis, Speex is designed to compress voice at bitrates in the 2-45 kbps range. Possible applications include VoIP, internet audio streaming, archiving of speech data (e.g. voice mail), and audio books. In some sense, it is meant to be complementary to the Ogg Vorbis codec.
-
-To use the Speex command line tools:
-
-% speexenc [options] input_file.wav compressed_file.spx
-
-% speexdec [options] compressed_file.spx output_file.wav

engine/voice_codecs/speex/source/Speex.spec

@@ -1,68 +0,0 @@
-%define name     speex
-%define ver      1.0.1
-%define rel      1
-
-Summary: An open-source, patent-free speech codec
-Name: %name
-Version: %ver
-Release: %rel
-Copyright: BSD
-Group: Application/Devel
-Source: http://www.speex.org/download/%{name}-%{ver}.tar.gz
-URL: http://www.speex.org/
-Vendor: Speex
-Packager: Jean-Marc Valin (jean-marc.valin@hermes.usherb.ca)
-BuildRoot: /var/tmp/%{name}-build-root
-Docdir: /usr/share/doc
-
-%description
-Speex is a patent-free audio codec designed especially for voice (unlike 
-Vorbis which targets general audio) signals and providing good narrowband 
-and wideband quality. This project aims to be complementary to the Vorbis
-codec.
-
-%package devel
-Summary:	Speex development files
-Group:		Development/Libraries
-Requires:	%{name} = %{version}
-
-%description devel
-Speex development files.
-
-%changelog
-* Thu Oct 03 2002 Jean-Marc Valin 
-- Added devel package inspired from PLD spec file
-
-* Tue Jul 30 2002 Fredrik Rambris <boost@users.sourceforge.net> 0.5.2
-- Added buildroot and docdir and ldconfig. Makes it builadble by non-roots
-  and also doesn't write to actual library paths when building.
-
-%prep
-%setup
-
-%build
-export CFLAGS='-O3 -DRELEASE'
-./configure --prefix=/usr --enable-shared --enable-static
-make
-
-%install
-rm -rf $RPM_BUILD_ROOT
-make DESTDIR=$RPM_BUILD_ROOT install
-
-%post -p /sbin/ldconfig
-%postun -p /sbin/ldconfig
-
-%files
-%defattr(644,root,root,755)
-%doc COPYING AUTHORS ChangeLog NEWS README
-%doc doc/manual.pdf
-/usr/share/man/man1/speexenc.1*
-/usr/share/man/man1/speexdec.1*
-%attr(755,root,root) %{_bindir}/speex*
-%attr(755,root,root) %{_libdir}/libspeex*.so*
-
-%files devel
-%defattr(644,root,root,755)
-%attr(755,root,root) %{_libdir}/libspeex*.la
-%{_includedir}/speex*.h
-%{_libdir}/libspeex*.a

engine/voice_codecs/speex/source/Speex.spec.in

@@ -1,68 +0,0 @@
-%define name     @PACKAGE@
-%define ver      @VERSION@
-%define rel      1
-
-Summary: An open-source, patent-free speech codec
-Name: %name
-Version: %ver
-Release: %rel
-Copyright: BSD
-Group: Application/Devel
-Source: http://www.speex.org/download/%{name}-%{ver}.tar.gz
-URL: http://www.speex.org/
-Vendor: Speex
-Packager: Jean-Marc Valin (jean-marc.valin@hermes.usherb.ca)
-BuildRoot: /var/tmp/%{name}-build-root
-Docdir: /usr/share/doc
-
-%description
-Speex is a patent-free audio codec designed especially for voice (unlike 
-Vorbis which targets general audio) signals and providing good narrowband 
-and wideband quality. This project aims to be complementary to the Vorbis
-codec.
-
-%package devel
-Summary:	Speex development files
-Group:		Development/Libraries
-Requires:	%{name} = %{version}
-
-%description devel
-Speex development files.
-
-%changelog
-* Thu Oct 03 2002 Jean-Marc Valin 
-- Added devel package inspired from PLD spec file
-
-* Tue Jul 30 2002 Fredrik Rambris <boost@users.sourceforge.net> 0.5.2
-- Added buildroot and docdir and ldconfig. Makes it builadble by non-roots
-  and also doesn't write to actual library paths when building.
-
-%prep
-%setup
-
-%build
-export CFLAGS='-O3 -DRELEASE'
-./configure --prefix=/usr --enable-shared --enable-static
-make
-
-%install
-rm -rf $RPM_BUILD_ROOT
-make DESTDIR=$RPM_BUILD_ROOT install
-
-%post -p /sbin/ldconfig
-%postun -p /sbin/ldconfig
-
-%files
-%defattr(644,root,root,755)
-%doc COPYING AUTHORS ChangeLog NEWS README
-%doc doc/manual.pdf
-/usr/share/man/man1/speexenc.1*
-/usr/share/man/man1/speexdec.1*
-%attr(755,root,root) %{_bindir}/speex*
-%attr(755,root,root) %{_libdir}/libspeex*.so*
-
-%files devel
-%defattr(644,root,root,755)
-%attr(755,root,root) %{_libdir}/libspeex*.la
-%{_includedir}/speex*.h
-%{_libdir}/libspeex*.a

engine/voice_codecs/speex/source/TODO

@@ -1,18 +0,0 @@
-Features
--Add maximum/minimum bit-rate control for VBR
--Get the encoder to use the rate of packet loss (more conservative pitch gains)
-
-Long-term quality improvements
--Improve perceptual enhancement (including wideband)
-
-Standards
-*Complete Speex RTP profile
--MIME type registration
--MS ACM wrapper
-
-*required for 1.0
-
-ideas:
-Peelable stream (double codebook, higher bands, stereo)
-LPC from spectral domain
-Better psycho-acoustics? Masking curve from Vorbis

engine/voice_codecs/speex/source/acinclude.m4

@@ -1,84 +0,0 @@
-AC_DEFUN(AC_FIND_FILE,
-[
-$3=NONE
-for i in $2;
-do
-  for j in $1;
-  do
-    if test -r "$i/$j"; then
-      $3=$i
-      break 2
-    fi
-  done
-done
-])
-
-AC_DEFUN(AC_PATH_LIBOGG,
-[
-OGG_LIBS="-logg"
-
-AC_MSG_CHECKING([for libogg])
-
-ac_ogg_includes=NONE ac_ogg_libraries=NONE ac_ogg_bindir=NONE
-ogg_libraries=""
-ogg_includes=""
-AC_ARG_WITH(ogg-dir,
-    [  --with-ogg-dir=DIR       where the root of OGG is installed ],
-    [  ac_ogg_includes="$withval"/include
-       ac_ogg_libraries="$withval"/lib
-    ])
-
-AC_ARG_WITH(ogg-includes,
-    [  --with-ogg-includes=DIR  where the OGG includes are. ],
-    [  
-       ac_ogg_includes="$withval"
-    ])
-    
-ogg_libs_given=no
-
-AC_ARG_WITH(ogg-libraries,
-    [  --with-ogg-libraries=DIR where the OGG library is installed.],
-    [  ac_ogg_libraries="$withval"
-       ogg_libs_given=yes
-    ])
-
-ogg_incdirs="/usr/include /usr/lib/ogg/include /opt/include /usr/local/ogg/include /usr/include/ogg /usr/include /usr/local/include"
-if test ! "$ac_ogg_includes" = "NONE"; then
-  ogg_incdirs="$ac_ogg_includes $ac_ogg_includes/.. $ogg_incdirs"
-fi
-AC_FIND_FILE(ogg/ogg.h, $ogg_incdirs, ogg_incdir)
-echo "Ogg includes in $ogg_incdir"
-
-
-ogg_libdirs="$ac_ogg_libraries /usr/lib/ogg/lib /usr/lib /opt/lib /usr/local/ogg/lib /usr/local/lib /usr/lib/ogg /usr/local/lib"
-test -n "$OGGDIR" && ogg_libdirs="$OGGDIR/lib $OGGDIR $ogg_libdirs"
-if test ! "$ac_ogg_libraries" = "NONE"; then
-  ogg_libdirs="$ac_ogg_libraries $ogg_libdirs"
-fi
-
-test=NONE
-ogg_libdir=NONE
-for dir in $ogg_libdirs; do
-  try="ls -1 $dir/libogg*"
-  if test=`eval $try 2> /dev/null`; then ogg_libdir=$dir; break; else echo "tried $dir" >&AC_FD_CC ; fi
-done
-
-echo "Ogg libraries in $ogg_libdir"
-
-if test "$ogg_libdir" = "NONE" || test "$ogg_incdir" = "NONE"; then
-   have_libogg=no
-else
-   have_libogg=yes
-   AC_DEFINE(HAVE_LIBOGG)
-fi
-
-OGG_INCLUDES="-I$ogg_incdir"
-OGG_LDFLAGS="-L$ogg_libdir"
-
-
-AC_SUBST(OGG_LIBS)
-AC_SUBST(OGG_INCLUDES)
-AC_SUBST(OGG_LDFLAGS)
-
-])
-

engine/voice_codecs/speex/source/configure.in

@@ -1,69 +0,0 @@
-dnl Process this file with autoconf to produce a configure script. -*-m4-*-
-
-AC_INIT(libspeex/speex.h)
-
-SPEEX_MAJOR_VERSION=1
-SPEEX_MINOR_VERSION=0
-SPEEX_MICRO_VERSION=1
-SPEEX_VERSION=1.0.1
-
-SPEEX_LT_CURRENT=2
-SPEEX_LT_REVISION=0
-SPEEX_LT_AGE=1
-
-AC_SUBST(SPEEX_LT_CURRENT)
-AC_SUBST(SPEEX_LT_REVISION)
-AC_SUBST(SPEEX_LT_AGE)
-
-# For automake.
-VERSION=$SPEEX_VERSION
-PACKAGE=speex
-
-AM_INIT_AUTOMAKE($PACKAGE, $VERSION, no-define)
-AM_MAINTAINER_MODE
-
-AC_CANONICAL_HOST
-AM_PROG_LIBTOOL
-
-AC_C_BIGENDIAN
-AC_CHECK_HEADERS(sys/soundcard.h)
-
-AC_ARG_ENABLE(ogg,
-      [  --enable-ogg=[yes/no]    Turn on or off the use of ogg
-libraries [default=yes]],
-      [case "${enableval}" in
-        yes) useogg=true ;;
-        no)  useogg=false ;;
-        *) AC_MSG_ERROR(bad value ${enableval} for --enable-ogg) ;;
-      esac],[useogg=true])
-
-if test x$useogg = xtrue; then
-AC_PATH_LIBOGG
-fi
-
-if test "$have_libogg" = yes; then
-   src=src
-else
-   src=
-fi
-AC_SUBST(src)
-
-AC_CHECK_LIB(m, sin)
-AC_CHECK_LIB(gnugetopt, getopt_long)
-
-AC_DEFINE_UNQUOTED(VERSION, "${VERSION}")
-
-AC_ARG_ENABLE(sse, [  --enable-sse             enable SSE support], [if test "$enableval" = yes; then AC_DEFINE(_USE_SSE) fi])
-
-dnl Output the makefiles and version.h.
-
-AC_OUTPUT([Makefile libspeex/Makefile src/Makefile doc/Makefile Speex.spec
-           win32/Makefile win32/libspeex/Makefile win32/speexenc/Makefile
-           win32/speexdec/Makefile ])
-
-if test "x$src" = "x"; then 
-echo "You don't seem to have libogg installed. Only the Speex library (libspeex) will be built (no encoder/decoder executable)"
-echo "You can download libogg from http://www.ogg.org/ogg/index.html"
-fi
-
-echo "Type \"make; make install\" to compile and install Speex";

engine/voice_codecs/speex/source/doc/Makefile.am

@@ -1,5 +0,0 @@
-docdir=$(prefix)/share/doc/@PACKAGE@-@VERSION@
-
-doc_DATA = manual.pdf
-
-EXTRA_DIST = $(doc_DATA)

engine/voice_codecs/speex/source/doc/Makefile.in

@@ -1,213 +0,0 @@
-# Makefile.in generated automatically by automake 1.4-p6 from Makefile.am
-
-# Copyright (C) 1994, 1995-8, 1999, 2001 Free Software Foundation, Inc.
-# This Makefile.in is free software; the Free Software Foundation
-# gives unlimited permission to copy and/or distribute it,
-# with or without modifications, as long as this notice is preserved.
-
-# This program is distributed in the hope that it will be useful,
-# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
-# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
-# PARTICULAR PURPOSE.
-
-
-SHELL = @SHELL@
-
-srcdir = @srcdir@
-top_srcdir = @top_srcdir@
-VPATH = @srcdir@
-prefix = @prefix@
-exec_prefix = @exec_prefix@
-
-bindir = @bindir@
-sbindir = @sbindir@
-libexecdir = @libexecdir@
-datadir = @datadir@
-sysconfdir = @sysconfdir@
-sharedstatedir = @sharedstatedir@
-localstatedir = @localstatedir@
-libdir = @libdir@
-infodir = @infodir@
-mandir = @mandir@
-includedir = @includedir@
-oldincludedir = /usr/include
-
-DESTDIR =
-
-pkgdatadir = $(datadir)/@PACKAGE@
-pkglibdir = $(libdir)/@PACKAGE@
-pkgincludedir = $(includedir)/@PACKAGE@
-
-top_builddir = ..
-
-ACLOCAL = @ACLOCAL@
-AUTOCONF = @AUTOCONF@
-AUTOMAKE = @AUTOMAKE@
-AUTOHEADER = @AUTOHEADER@
-
-INSTALL = @INSTALL@
-INSTALL_PROGRAM = @INSTALL_PROGRAM@ $(AM_INSTALL_PROGRAM_FLAGS)
-INSTALL_DATA = @INSTALL_DATA@
-INSTALL_SCRIPT = @INSTALL_SCRIPT@
-transform = @program_transform_name@
-
-NORMAL_INSTALL = :
-PRE_INSTALL = :
-POST_INSTALL = :
-NORMAL_UNINSTALL = :
-PRE_UNINSTALL = :
-POST_UNINSTALL = :
-host_alias = @host_alias@
-host_triplet = @host@
-AS = @AS@
-CC = @CC@
-DLLTOOL = @DLLTOOL@
-ECHO = @ECHO@
-EXEEXT = @EXEEXT@
-LIBTOOL = @LIBTOOL@
-LN_S = @LN_S@
-MAINT = @MAINT@
-MAKEINFO = @MAKEINFO@
-OBJDUMP = @OBJDUMP@
-OBJEXT = @OBJEXT@
-OGG_INCLUDES = @OGG_INCLUDES@
-OGG_LDFLAGS = @OGG_LDFLAGS@
-OGG_LIBS = @OGG_LIBS@
-PACKAGE = @PACKAGE@
-RANLIB = @RANLIB@
-SPEEX_LT_AGE = @SPEEX_LT_AGE@
-SPEEX_LT_CURRENT = @SPEEX_LT_CURRENT@
-SPEEX_LT_REVISION = @SPEEX_LT_REVISION@
-STRIP = @STRIP@
-VERSION = @VERSION@
-src = @src@
-
-docdir = $(prefix)/share/doc/@PACKAGE@-@VERSION@
-
-doc_DATA = manual.pdf
-
-EXTRA_DIST = $(doc_DATA)
-mkinstalldirs = $(SHELL) $(top_srcdir)/mkinstalldirs
-CONFIG_CLEAN_FILES = 
-DATA =  $(doc_DATA)
-
-DIST_COMMON =  Makefile.am Makefile.in
-
-
-DISTFILES = $(DIST_COMMON) $(SOURCES) $(HEADERS) $(TEXINFOS) $(EXTRA_DIST)
-
-TAR = gtar
-GZIP_ENV = --best
-all: all-redirect
-.SUFFIXES:
-$(srcdir)/Makefile.in: @MAINTAINER_MODE_TRUE@ Makefile.am $(top_srcdir)/configure.in $(ACLOCAL_M4) 
-	cd $(top_srcdir) && $(AUTOMAKE) --gnu --include-deps doc/Makefile
-
-Makefile: $(srcdir)/Makefile.in  $(top_builddir)/config.status
-	cd $(top_builddir) \
-	  && CONFIG_FILES=$(subdir)/$@ CONFIG_HEADERS= $(SHELL) ./config.status
-
-
-install-docDATA: $(doc_DATA)
-	@$(NORMAL_INSTALL)
-	$(mkinstalldirs) $(DESTDIR)$(docdir)
-	@list='$(doc_DATA)'; for p in $$list; do \
-	  if test -f $(srcdir)/$$p; then \
-	    echo " $(INSTALL_DATA) $(srcdir)/$$p $(DESTDIR)$(docdir)/$$p"; \
-	    $(INSTALL_DATA) $(srcdir)/$$p $(DESTDIR)$(docdir)/$$p; \
-	  else if test -f $$p; then \
-	    echo " $(INSTALL_DATA) $$p $(DESTDIR)$(docdir)/$$p"; \
-	    $(INSTALL_DATA) $$p $(DESTDIR)$(docdir)/$$p; \
-	  fi; fi; \
-	done
-
-uninstall-docDATA:
-	@$(NORMAL_UNINSTALL)
-	list='$(doc_DATA)'; for p in $$list; do \
-	  rm -f $(DESTDIR)$(docdir)/$$p; \
-	done
-tags: TAGS
-TAGS:
-
-
-distdir = $(top_builddir)/$(PACKAGE)-$(VERSION)/$(subdir)
-
-subdir = doc
-
-distdir: $(DISTFILES)
-	@for file in $(DISTFILES); do \
-	  d=$(srcdir); \
-	  if test -d $$d/$$file; then \
-	    cp -pr $$d/$$file $(distdir)/$$file; \
-	  else \
-	    test -f $(distdir)/$$file \
-	    || ln $$d/$$file $(distdir)/$$file 2> /dev/null \
-	    || cp -p $$d/$$file $(distdir)/$$file || :; \
-	  fi; \
-	done
-info-am:
-info: info-am
-dvi-am:
-dvi: dvi-am
-check-am: all-am
-check: check-am
-installcheck-am:
-installcheck: installcheck-am
-install-exec-am:
-install-exec: install-exec-am
-
-install-data-am: install-docDATA
-install-data: install-data-am
-
-install-am: all-am
-	@$(MAKE) $(AM_MAKEFLAGS) install-exec-am install-data-am
-install: install-am
-uninstall-am: uninstall-docDATA
-uninstall: uninstall-am
-all-am: Makefile $(DATA)
-all-redirect: all-am
-install-strip:
-	$(MAKE) $(AM_MAKEFLAGS) AM_INSTALL_PROGRAM_FLAGS=-s install
-installdirs:
-	$(mkinstalldirs)  $(DESTDIR)$(docdir)
-
-
-mostlyclean-generic:
-
-clean-generic:
-
-distclean-generic:
-	-rm -f Makefile $(CONFIG_CLEAN_FILES)
-	-rm -f config.cache config.log stamp-h stamp-h[0-9]*
-
-maintainer-clean-generic:
-mostlyclean-am:  mostlyclean-generic
-
-mostlyclean: mostlyclean-am
-
-clean-am:  clean-generic mostlyclean-am
-
-clean: clean-am
-
-distclean-am:  distclean-generic clean-am
-	-rm -f libtool
-
-distclean: distclean-am
-
-maintainer-clean-am:  maintainer-clean-generic distclean-am
-	@echo "This command is intended for maintainers to use;"
-	@echo "it deletes files that may require special tools to rebuild."
-
-maintainer-clean: maintainer-clean-am
-
-.PHONY: uninstall-docDATA install-docDATA tags distdir info-am info \
-dvi-am dvi check check-am installcheck-am installcheck install-exec-am \
-install-exec install-data-am install-data install-am install \
-uninstall-am uninstall all-redirect all-am all installdirs \
-mostlyclean-generic distclean-generic clean-generic \
-maintainer-clean-generic clean mostlyclean distclean maintainer-clean
-
-
-# Tell versions [3.59,3.63) of GNU make to not export all variables.
-# Otherwise a system limit (for SysV at least) may be exceeded.
-.NOEXPORT:

engine/voice_codecs/speex/source/install-sh

@@ -1,251 +0,0 @@
-#!/bin/sh
-#
-# install - install a program, script, or datafile
-# This comes from X11R5 (mit/util/scripts/install.sh).
-#
-# Copyright 1991 by the Massachusetts Institute of Technology
-#
-# Permission to use, copy, modify, distribute, and sell this software and its
-# documentation for any purpose is hereby granted without fee, provided that
-# the above copyright notice appear in all copies and that both that
-# copyright notice and this permission notice appear in supporting
-# documentation, and that the name of M.I.T. not be used in advertising or
-# publicity pertaining to distribution of the software without specific,
-# written prior permission.  M.I.T. makes no representations about the
-# suitability of this software for any purpose.  It is provided "as is"
-# without express or implied warranty.
-#
-# Calling this script install-sh is preferred over install.sh, to prevent
-# `make' implicit rules from creating a file called install from it
-# when there is no Makefile.
-#
-# This script is compatible with the BSD install script, but was written
-# from scratch.  It can only install one file at a time, a restriction
-# shared with many OS's install programs.
-
-
-# set DOITPROG to echo to test this script
-
-# Don't use :- since 4.3BSD and earlier shells don't like it.
-doit="${DOITPROG-}"
-
-
-# put in absolute paths if you don't have them in your path; or use env. vars.
-
-mvprog="${MVPROG-mv}"
-cpprog="${CPPROG-cp}"
-chmodprog="${CHMODPROG-chmod}"
-chownprog="${CHOWNPROG-chown}"
-chgrpprog="${CHGRPPROG-chgrp}"
-stripprog="${STRIPPROG-strip}"
-rmprog="${RMPROG-rm}"
-mkdirprog="${MKDIRPROG-mkdir}"
-
-transformbasename=""
-transform_arg=""
-instcmd="$mvprog"
-chmodcmd="$chmodprog 0755"
-chowncmd=""
-chgrpcmd=""
-stripcmd=""
-rmcmd="$rmprog -f"
-mvcmd="$mvprog"
-src=""
-dst=""
-dir_arg=""
-
-while [ x"$1" != x ]; do
-    case $1 in
-	-c) instcmd="$cpprog"
-	    shift
-	    continue;;
-
-	-d) dir_arg=true
-	    shift
-	    continue;;
-
-	-m) chmodcmd="$chmodprog $2"
-	    shift
-	    shift
-	    continue;;
-
-	-o) chowncmd="$chownprog $2"
-	    shift
-	    shift
-	    continue;;
-
-	-g) chgrpcmd="$chgrpprog $2"
-	    shift
-	    shift
-	    continue;;
-
-	-s) stripcmd="$stripprog"
-	    shift
-	    continue;;
-
-	-t=*) transformarg=`echo $1 | sed 's/-t=//'`
-	    shift
-	    continue;;
-
-	-b=*) transformbasename=`echo $1 | sed 's/-b=//'`
-	    shift
-	    continue;;
-
-	*)  if [ x"$src" = x ]
-	    then
-		src=$1
-	    else
-		# this colon is to work around a 386BSD /bin/sh bug
-		:
-		dst=$1
-	    fi
-	    shift
-	    continue;;
-    esac
-done
-
-if [ x"$src" = x ]
-then
-	echo "install:	no input file specified"
-	exit 1
-else
-	true
-fi
-
-if [ x"$dir_arg" != x ]; then
-	dst=$src
-	src=""
-	
-	if [ -d $dst ]; then
-		instcmd=:
-		chmodcmd=""
-	else
-		instcmd=mkdir
-	fi
-else
-
-# Waiting for this to be detected by the "$instcmd $src $dsttmp" command
-# might cause directories to be created, which would be especially bad 
-# if $src (and thus $dsttmp) contains '*'.
-
-	if [ -f $src -o -d $src ]
-	then
-		true
-	else
-		echo "install:  $src does not exist"
-		exit 1
-	fi
-	
-	if [ x"$dst" = x ]
-	then
-		echo "install:	no destination specified"
-		exit 1
-	else
-		true
-	fi
-
-# If destination is a directory, append the input filename; if your system
-# does not like double slashes in filenames, you may need to add some logic
-
-	if [ -d $dst ]
-	then
-		dst="$dst"/`basename $src`
-	else
-		true
-	fi
-fi
-
-## this sed command emulates the dirname command
-dstdir=`echo $dst | sed -e 's,[^/]*$,,;s,/$,,;s,^$,.,'`
-
-# Make sure that the destination directory exists.
-#  this part is taken from Noah Friedman's mkinstalldirs script
-
-# Skip lots of stat calls in the usual case.
-if [ ! -d "$dstdir" ]; then
-defaultIFS='	
-'
-IFS="${IFS-${defaultIFS}}"
-
-oIFS="${IFS}"
-# Some sh's can't handle IFS=/ for some reason.
-IFS='%'
-set - `echo ${dstdir} | sed -e 's@/@%@g' -e 's@^%@/@'`
-IFS="${oIFS}"
-
-pathcomp=''
-
-while [ $# -ne 0 ] ; do
-	pathcomp="${pathcomp}${1}"
-	shift
-
-	if [ ! -d "${pathcomp}" ] ;
-        then
-		$mkdirprog "${pathcomp}"
-	else
-		true
-	fi
-
-	pathcomp="${pathcomp}/"
-done
-fi
-
-if [ x"$dir_arg" != x ]
-then
-	$doit $instcmd $dst &&
-
-	if [ x"$chowncmd" != x ]; then $doit $chowncmd $dst; else true ; fi &&
-	if [ x"$chgrpcmd" != x ]; then $doit $chgrpcmd $dst; else true ; fi &&
-	if [ x"$stripcmd" != x ]; then $doit $stripcmd $dst; else true ; fi &&
-	if [ x"$chmodcmd" != x ]; then $doit $chmodcmd $dst; else true ; fi
-else
-
-# If we're going to rename the final executable, determine the name now.
-
-	if [ x"$transformarg" = x ] 
-	then
-		dstfile=`basename $dst`
-	else
-		dstfile=`basename $dst $transformbasename | 
-			sed $transformarg`$transformbasename
-	fi
-
-# don't allow the sed command to completely eliminate the filename
-
-	if [ x"$dstfile" = x ] 
-	then
-		dstfile=`basename $dst`
-	else
-		true
-	fi
-
-# Make a temp file name in the proper directory.
-
-	dsttmp=$dstdir/#inst.$$#
-
-# Move or copy the file name to the temp name
-
-	$doit $instcmd $src $dsttmp &&
-
-	trap "rm -f ${dsttmp}" 0 &&
-
-# and set any options; do chmod last to preserve setuid bits
-
-# If any of these fail, we abort the whole thing.  If we want to
-# ignore errors from any of these, just make sure not to ignore
-# errors from the above "$doit $instcmd $src $dsttmp" command.
-
-	if [ x"$chowncmd" != x ]; then $doit $chowncmd $dsttmp; else true;fi &&
-	if [ x"$chgrpcmd" != x ]; then $doit $chgrpcmd $dsttmp; else true;fi &&
-	if [ x"$stripcmd" != x ]; then $doit $stripcmd $dsttmp; else true;fi &&
-	if [ x"$chmodcmd" != x ]; then $doit $chmodcmd $dsttmp; else true;fi &&
-
-# Now rename the file to the real destination.
-
-	$doit $rmcmd -f $dstdir/$dstfile &&
-	$doit $mvcmd $dsttmp $dstdir/$dstfile 
-
-fi &&
-
-
-exit 0

engine/voice_codecs/speex/source/libspeex/Makefile.am

@@ -1,77 +0,0 @@
-## Process this file with automake to produce Makefile.in. -*-Makefile-*-
-
-# $Id: Makefile.am,v 1.49 2003/03/17 22:40:57 jm Exp $
-
-# Disable automatic dependency tracking if using other tools than gcc and gmake
-#AUTOMAKE_OPTIONS = no-dependencies
-
-lib_LTLIBRARIES = libspeex.la
-
-# Sources for compilation in the library
-libspeex_la_SOURCES = nb_celp.c \
-	sb_celp.c \
-	lpc.c \
-	ltp.c \
-	lsp.c \
-	quant_lsp.c \
-	lsp_tables_nb.c \
-	gain_table.c \
-	gain_table_lbr.c \
-	cb_search.c \
-	filters.c \
-	bits.c \
-	modes.c \
-	vq.c \
-	high_lsp_tables.c \
-	vbr.c \
-	hexc_table.c \
-	exc_5_256_table.c \
-	exc_5_64_table.c \
-	exc_8_128_table.c \
-	exc_10_32_table.c \
-	exc_10_16_table.c \
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-	hexc_10_32_table.c \
-	misc.c \
-	speex_header.c \
-	speex_callbacks.c \
-	math_approx.c \
-	stereo.c
-
-
-include_HEADERS =  speex.h \
-	speex_bits.h \
-	speex_header.h \
-	speex_callbacks.h \
-	speex_stereo.h
-
-noinst_HEADERS = lsp.h \
-	nb_celp.h \
-	lpc.h \
-	ltp.h \
-	quant_lsp.h \
-	cb_search.h \
-	filters.h \
-	stack_alloc.h \
-	vq.h \
-	modes.h \
-	sb_celp.h \
-	vbr.h \
-	misc.h \
-	ltp_sse.h \
-	filters_sse.h \
-	math_approx.h
-	
-	
-libspeex_la_LDFLAGS = -version-info @SPEEX_LT_CURRENT@:@SPEEX_LT_REVISION@:@SPEEX_LT_AGE@
-
-noinst_PROGRAMS = testenc testenc_wb testenc_uwb
-
-testenc_SOURCES = testenc.c
-testenc_LDADD = libspeex.la
-
-testenc_wb_SOURCES = testenc_wb.c
-testenc_wb_LDADD = libspeex.la
-
-testenc_uwb_SOURCES = testenc_uwb.c
-testenc_uwb_LDADD = libspeex.la

engine/voice_codecs/speex/source/libspeex/Makefile.in

@@ -1,450 +0,0 @@
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-
-# Copyright (C) 1994, 1995-8, 1999, 2001 Free Software Foundation, Inc.
-# This Makefile.in is free software; the Free Software Foundation
-# gives unlimited permission to copy and/or distribute it,
-# with or without modifications, as long as this notice is preserved.
-
-# This program is distributed in the hope that it will be useful,
-# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
-# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
-# PARTICULAR PURPOSE.
-
-# $Id: Makefile.am,v 1.49 2003/03/17 22:40:57 jm Exp $
-
-# Disable automatic dependency tracking if using other tools than gcc and gmake
-#AUTOMAKE_OPTIONS = no-dependencies
-
-
-SHELL = @SHELL@
-
-srcdir = @srcdir@
-top_srcdir = @top_srcdir@
-VPATH = @srcdir@
-prefix = @prefix@
-exec_prefix = @exec_prefix@
-
-bindir = @bindir@
-sbindir = @sbindir@
-libexecdir = @libexecdir@
-datadir = @datadir@
-sysconfdir = @sysconfdir@
-sharedstatedir = @sharedstatedir@
-localstatedir = @localstatedir@
-libdir = @libdir@
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-mandir = @mandir@
-includedir = @includedir@
-oldincludedir = /usr/include
-
-DESTDIR =
-
-pkgdatadir = $(datadir)/@PACKAGE@
-pkglibdir = $(libdir)/@PACKAGE@
-pkgincludedir = $(includedir)/@PACKAGE@
-
-top_builddir = ..
-
-ACLOCAL = @ACLOCAL@
-AUTOCONF = @AUTOCONF@
-AUTOMAKE = @AUTOMAKE@
-AUTOHEADER = @AUTOHEADER@
-
-INSTALL = @INSTALL@
-INSTALL_PROGRAM = @INSTALL_PROGRAM@ $(AM_INSTALL_PROGRAM_FLAGS)
-INSTALL_DATA = @INSTALL_DATA@
-INSTALL_SCRIPT = @INSTALL_SCRIPT@
-transform = @program_transform_name@
-
-NORMAL_INSTALL = :
-PRE_INSTALL = :
-POST_INSTALL = :
-NORMAL_UNINSTALL = :
-PRE_UNINSTALL = :
-POST_UNINSTALL = :
-host_alias = @host_alias@
-host_triplet = @host@
-AS = @AS@
-CC = @CC@
-DLLTOOL = @DLLTOOL@
-ECHO = @ECHO@
-EXEEXT = @EXEEXT@
-LIBTOOL = @LIBTOOL@
-LN_S = @LN_S@
-MAINT = @MAINT@
-MAKEINFO = @MAKEINFO@
-OBJDUMP = @OBJDUMP@
-OBJEXT = @OBJEXT@
-OGG_INCLUDES = @OGG_INCLUDES@
-OGG_LDFLAGS = @OGG_LDFLAGS@
-OGG_LIBS = @OGG_LIBS@
-PACKAGE = @PACKAGE@
-RANLIB = @RANLIB@
-SPEEX_LT_AGE = @SPEEX_LT_AGE@
-SPEEX_LT_CURRENT = @SPEEX_LT_CURRENT@
-SPEEX_LT_REVISION = @SPEEX_LT_REVISION@
-STRIP = @STRIP@
-VERSION = @VERSION@
-src = @src@
-
-lib_LTLIBRARIES = libspeex.la
-
-# Sources for compilation in the library
-libspeex_la_SOURCES = nb_celp.c 	sb_celp.c 	lpc.c 	ltp.c 	lsp.c 	quant_lsp.c 	lsp_tables_nb.c 	gain_table.c 	gain_table_lbr.c 	cb_search.c 	filters.c 	bits.c 	modes.c 	vq.c 	high_lsp_tables.c 	vbr.c 	hexc_table.c 	exc_5_256_table.c 	exc_5_64_table.c 	exc_8_128_table.c 	exc_10_32_table.c 	exc_10_16_table.c 	exc_20_32_table.c 	hexc_10_32_table.c 	misc.c 	speex_header.c 	speex_callbacks.c 	math_approx.c 	stereo.c
-
-
-include_HEADERS = speex.h 	speex_bits.h 	speex_header.h 	speex_callbacks.h 	speex_stereo.h
-
-
-noinst_HEADERS = lsp.h 	nb_celp.h 	lpc.h 	ltp.h 	quant_lsp.h 	cb_search.h 	filters.h 	stack_alloc.h 	vq.h 	modes.h 	sb_celp.h 	vbr.h 	misc.h 	ltp_sse.h 	filters_sse.h 	math_approx.h
-
-
-libspeex_la_LDFLAGS = -version-info @SPEEX_LT_CURRENT@:@SPEEX_LT_REVISION@:@SPEEX_LT_AGE@
-
-noinst_PROGRAMS = testenc testenc_wb testenc_uwb
-
-testenc_SOURCES = testenc.c
-testenc_LDADD = libspeex.la
-
-testenc_wb_SOURCES = testenc_wb.c
-testenc_wb_LDADD = libspeex.la
-
-testenc_uwb_SOURCES = testenc_uwb.c
-testenc_uwb_LDADD = libspeex.la
-mkinstalldirs = $(SHELL) $(top_srcdir)/mkinstalldirs
-CONFIG_CLEAN_FILES = 
-LTLIBRARIES =  $(lib_LTLIBRARIES)
-
-
-DEFS = @DEFS@ -I. -I$(srcdir) 
-CPPFLAGS = @CPPFLAGS@
-LDFLAGS = @LDFLAGS@
-LIBS = @LIBS@
-libspeex_la_LIBADD = 
-libspeex_la_OBJECTS =  nb_celp.lo sb_celp.lo lpc.lo ltp.lo lsp.lo \
-quant_lsp.lo lsp_tables_nb.lo gain_table.lo gain_table_lbr.lo \
-cb_search.lo filters.lo bits.lo modes.lo vq.lo high_lsp_tables.lo \
-vbr.lo hexc_table.lo exc_5_256_table.lo exc_5_64_table.lo \
-exc_8_128_table.lo exc_10_32_table.lo exc_10_16_table.lo \
-exc_20_32_table.lo hexc_10_32_table.lo misc.lo speex_header.lo \
-speex_callbacks.lo math_approx.lo stereo.lo
-noinst_PROGRAMS =  testenc$(EXEEXT) testenc_wb$(EXEEXT) \
-testenc_uwb$(EXEEXT)
-PROGRAMS =  $(noinst_PROGRAMS)
-
-testenc_OBJECTS =  testenc.$(OBJEXT)
-testenc_DEPENDENCIES =  libspeex.la
-testenc_LDFLAGS = 
-testenc_wb_OBJECTS =  testenc_wb.$(OBJEXT)
-testenc_wb_DEPENDENCIES =  libspeex.la
-testenc_wb_LDFLAGS = 
-testenc_uwb_OBJECTS =  testenc_uwb.$(OBJEXT)
-testenc_uwb_DEPENDENCIES =  libspeex.la
-testenc_uwb_LDFLAGS = 
-CFLAGS = @CFLAGS@
-COMPILE = $(CC) $(DEFS) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) $(AM_CFLAGS) $(CFLAGS)
-LTCOMPILE = $(LIBTOOL) --mode=compile $(CC) $(DEFS) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) $(AM_CFLAGS) $(CFLAGS)
-CCLD = $(CC)
-LINK = $(LIBTOOL) --mode=link $(CCLD) $(AM_CFLAGS) $(CFLAGS) $(LDFLAGS) -o $@
-HEADERS =  $(include_HEADERS) $(noinst_HEADERS)
-
-DIST_COMMON =  Makefile.am Makefile.in
-
-
-DISTFILES = $(DIST_COMMON) $(SOURCES) $(HEADERS) $(TEXINFOS) $(EXTRA_DIST)
-
-TAR = gtar
-GZIP_ENV = --best
-SOURCES = $(libspeex_la_SOURCES) $(testenc_SOURCES) $(testenc_wb_SOURCES) $(testenc_uwb_SOURCES)
-OBJECTS = $(libspeex_la_OBJECTS) $(testenc_OBJECTS) $(testenc_wb_OBJECTS) $(testenc_uwb_OBJECTS)
-
-all: all-redirect
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-.SUFFIXES: .S .c .lo .o .obj .s
-$(srcdir)/Makefile.in: @MAINTAINER_MODE_TRUE@ Makefile.am $(top_srcdir)/configure.in $(ACLOCAL_M4) 
-	cd $(top_srcdir) && $(AUTOMAKE) --gnu --include-deps libspeex/Makefile
-
-Makefile: $(srcdir)/Makefile.in  $(top_builddir)/config.status
-	cd $(top_builddir) \
-	  && CONFIG_FILES=$(subdir)/$@ CONFIG_HEADERS= $(SHELL) ./config.status
-
-
-mostlyclean-libLTLIBRARIES:
-
-clean-libLTLIBRARIES:
-	-test -z "$(lib_LTLIBRARIES)" || rm -f $(lib_LTLIBRARIES)
-
-distclean-libLTLIBRARIES:
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-maintainer-clean-libLTLIBRARIES:
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-install-libLTLIBRARIES: $(lib_LTLIBRARIES)
-	@$(NORMAL_INSTALL)
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-	    $(LIBTOOL)  --mode=install $(INSTALL) $$p $(DESTDIR)$(libdir)/$$p; \
-	  else :; fi; \
-	done
-
-uninstall-libLTLIBRARIES:
-	@$(NORMAL_UNINSTALL)
-	list='$(lib_LTLIBRARIES)'; for p in $$list; do \
-	  $(LIBTOOL)  --mode=uninstall rm -f $(DESTDIR)$(libdir)/$$p; \
-	done
-
-.c.o:
-	$(COMPILE) -c $<
-
-# FIXME: We should only use cygpath when building on Windows,
-# and only if it is available.
-.c.obj:
-	$(COMPILE) -c `cygpath -w $<`
-
-.s.o:
-	$(COMPILE) -c $<
-
-.S.o:
-	$(COMPILE) -c $<
-
-mostlyclean-compile:
-	-rm -f *.o core *.core
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-	$(LIBTOOL) --mode=compile $(COMPILE) -c $<
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-install-includeHEADERS: $(include_HEADERS)
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-	  $(INSTALL_DATA) $$d$$p $(DESTDIR)$(includedir)/$$p; \
-	done
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-uninstall-includeHEADERS:
-	@$(NORMAL_UNINSTALL)
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-	done
-
-tags: TAGS
-
-ID: $(HEADERS) $(SOURCES) $(LISP)
-	list='$(SOURCES) $(HEADERS)'; \
-	unique=`for i in $$list; do echo $$i; done | \
-	  awk '    { files[$$0] = 1; } \
-	       END { for (i in files) print i; }'`; \
-	here=`pwd` && cd $(srcdir) \
-	  && mkid -f$$here/ID $$unique $(LISP)
-
-TAGS:  $(HEADERS) $(SOURCES)  $(TAGS_DEPENDENCIES) $(LISP)
-	tags=; \
-	here=`pwd`; \
-	list='$(SOURCES) $(HEADERS)'; \
-	unique=`for i in $$list; do echo $$i; done | \
-	  awk '    { files[$$0] = 1; } \
-	       END { for (i in files) print i; }'`; \
-	test -z "$(ETAGS_ARGS)$$unique$(LISP)$$tags" \
-	  || (cd $(srcdir) && etags $(ETAGS_ARGS) $$tags  $$unique $(LISP) -o $$here/TAGS)
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-mostlyclean-tags:
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-distclean-tags:
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-distdir = $(top_builddir)/$(PACKAGE)-$(VERSION)/$(subdir)
-
-subdir = libspeex
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-distdir: $(DISTFILES)
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-	    || ln $$d/$$file $(distdir)/$$file 2> /dev/null \
-	    || cp -p $$d/$$file $(distdir)/$$file || :; \
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-	done
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-.NOEXPORT:

engine/voice_codecs/speex/source/libspeex/bits.c

@@ -1,352 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: speex_bits.c
-
-   Handles bit packing/unpacking
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-
-*/
-
-#include "speex_bits.h"
-#include "misc.h"
-
-void speex_bits_init(SpeexBits *bits)
-{
-   int i;
-   bits->bytes = (char*)speex_alloc(MAX_BYTES_PER_FRAME);
-   bits->buf_size = MAX_BYTES_PER_FRAME;
-
-   for (i=0;i<bits->buf_size;i++)
-      bits->bytes[i]=0;
-   bits->nbBits=0;
-   bits->bytePtr=0;
-   bits->bitPtr=0;
-   bits->owner=1;
-   bits->overflow=0;
-}
-
-void speex_bits_init_buffer(SpeexBits *bits, void *buff, int buf_size)
-{
-   int i;
-   bits->bytes = (char*)buff;
-   bits->buf_size = buf_size;
-
-   for (i=0;i<buf_size;i++)
-      bits->bytes[i]=0;
-   bits->nbBits=0;
-   bits->bytePtr=0;
-   bits->bitPtr=0;
-   bits->owner=0;
-   bits->overflow=0;
-}
-
-void speex_bits_destroy(SpeexBits *bits)
-{
-   if (bits->owner)
-      speex_free(bits->bytes);
-   /* Will do something once the allocation is dynamic */
-}
-
-void speex_bits_reset(SpeexBits *bits)
-{
-   int i;
-   for (i=0;i<bits->buf_size;i++)
-      bits->bytes[i]=0;
-   bits->nbBits=0;
-   bits->bytePtr=0;
-   bits->bitPtr=0;
-   bits->overflow=0;
-}
-
-void speex_bits_rewind(SpeexBits *bits)
-{
-   bits->bytePtr=0;
-   bits->bitPtr=0;
-   bits->overflow=0;
-}
-
-void speex_bits_read_from(SpeexBits *bits, char *bytes, int len)
-{
-   int i;
-   if (len > bits->buf_size)
-   {
-      speex_warning_int("Packet if larger than allocated buffer: ", len);
-      if (bits->owner)
-      {
-         char *tmp = (char*)speex_realloc(bits->bytes, len);
-         if (tmp)
-         {
-            bits->buf_size=len;
-            bits->bytes=tmp;
-         } else {
-            len=bits->buf_size;
-            speex_warning("Could not resize input buffer: truncating input");
-         }
-      } else {
-         speex_warning("Do not own input buffer: truncating input");
-         len=bits->buf_size;
-      }
-   }
-   for (i=0;i<len;i++)
-      bits->bytes[i]=bytes[i];
-   bits->nbBits=len<<3;
-   bits->bytePtr=0;
-   bits->bitPtr=0;
-   bits->overflow=0;
-}
-
-static void speex_bits_flush(SpeexBits *bits)
-{
-   int i;
-   if (bits->bytePtr>0)
-   {
-      for (i=bits->bytePtr;i<((bits->nbBits+7)>>3);i++)
-         bits->bytes[i-bits->bytePtr]=bits->bytes[i];
-   }
-   bits->nbBits -= bits->bytePtr<<3;
-   bits->bytePtr=0;
-}
-
-void speex_bits_read_whole_bytes(SpeexBits *bits, char *bytes, int len)
-{
-   int i,pos;
-
-   if ((bits->nbBits>>3)+len+1 > bits->buf_size)
-   {
-      speex_warning_int("Packet if larger than allocated buffer: ", len);
-      if (bits->owner)
-      {
-         char *tmp = (char*)speex_realloc(bits->bytes, (bits->nbBits>>3)+len+1);
-         if (tmp)
-         {
-            bits->buf_size=(bits->nbBits>>3)+len+1;
-            bits->bytes=tmp;
-         } else {
-            len=bits->buf_size-(bits->nbBits>>3)-1;
-            speex_warning("Could not resize input buffer: truncating input");
-         }
-      } else {
-         speex_warning("Do not own input buffer: truncating input");
-         len=bits->buf_size;
-      }
-   }
-
-   speex_bits_flush(bits);
-   pos=bits->nbBits>>3;
-   for (i=0;i<len;i++)
-      bits->bytes[pos+i]=bytes[i];
-   bits->nbBits+=len<<3;
-}
-
-int speex_bits_write(SpeexBits *bits, char *bytes, int max_len)
-{
-   int i;
-   if (max_len > ((bits->nbBits+7)>>3))
-      max_len = ((bits->nbBits+7)>>3);
-   for (i=0;i<max_len;i++)
-      bytes[i]=bits->bytes[i];
-   return max_len;
-}
-
-int speex_bits_write_whole_bytes(SpeexBits *bits, char *bytes, int max_len)
-{
-   int i;
-   if (max_len > ((bits->nbBits)>>3))
-      max_len = ((bits->nbBits)>>3);
-   for (i=0;i<max_len;i++)
-      bytes[i]=bits->bytes[i];
-   
-   if (bits->bitPtr>0)
-      bits->bytes[0]=bits->bytes[max_len];
-   else
-      bits->bytes[0]=0;
-   for (i=1;i<((bits->nbBits)>>3)+1;i++)
-      bits->bytes[i]=0;
-   bits->bytePtr=0;
-   bits->nbBits &= 7;
-   return max_len;
-}
-
-
-void speex_bits_pack(SpeexBits *bits, int data, int nbBits)
-{
-   int i;
-   unsigned int d=data;
-
-   if (bits->bytePtr+((nbBits+bits->bitPtr)>>3) >= bits->buf_size)
-   {
-      speex_warning("Buffer too small to pack bits");
-      if (bits->owner)
-      {
-         char *tmp = (char*)speex_realloc(bits->bytes, ((bits->buf_size+5)*3)>>1);
-         if (tmp)
-         {
-            for (i=bits->buf_size;i<(((bits->buf_size+5)*3)>>1);i++)
-               tmp[i]=0;
-            bits->buf_size=((bits->buf_size+5)*3)>>1;
-            bits->bytes=tmp;
-         } else {
-            speex_warning("Could not resize input buffer: not packing");
-            return;
-         }
-      } else {
-         speex_warning("Do not own input buffer: not packing");
-         return;
-      }
-   }
-
-   while(nbBits)
-   {
-      int bit;
-      bit = (d>>(nbBits-1))&1;
-      bits->bytes[bits->bytePtr] |= bit<<(7-bits->bitPtr);
-      bits->bitPtr++;
-
-      if (bits->bitPtr==8)
-      {
-         bits->bitPtr=0;
-         bits->bytePtr++;
-      }
-      bits->nbBits++;
-      nbBits--;
-   }
-}
-
-int speex_bits_unpack_signed(SpeexBits *bits, int nbBits)
-{
-   unsigned int d=speex_bits_unpack_unsigned(bits,nbBits);
-   /* If number is negative */
-   if (d>>(nbBits-1))
-   {
-      d |= (-1)<<nbBits;
-   }
-   return d;
-}
-
-unsigned int speex_bits_unpack_unsigned(SpeexBits *bits, int nbBits)
-{
-   unsigned int d=0;
-   if ((bits->bytePtr<<3)+bits->bitPtr+nbBits>bits->nbBits)
-      bits->overflow=1;
-   if (bits->overflow)
-      return 0;
-   while(nbBits)
-   {
-      d<<=1;
-      d |= (bits->bytes[bits->bytePtr]>>(7-bits->bitPtr))&1;
-      bits->bitPtr++;
-      if (bits->bitPtr==8)
-      {
-         bits->bitPtr=0;
-         bits->bytePtr++;
-      }
-      nbBits--;
-   }
-   return d;
-}
-
-unsigned int speex_bits_peek_unsigned(SpeexBits *bits, int nbBits)
-{
-   unsigned int d=0;
-   int bitPtr, bytePtr;
-   char *bytes;
-
-   if ((bits->bytePtr<<3)+bits->bitPtr+nbBits>bits->nbBits)
-     bits->overflow=1;
-   if (bits->overflow)
-      return 0;
-
-   bitPtr=bits->bitPtr;
-   bytePtr=bits->bytePtr;
-   bytes = bits->bytes;
-   while(nbBits)
-   {
-      d<<=1;
-      d |= (bytes[bytePtr]>>(7-bitPtr))&1;
-      bitPtr++;
-      if (bitPtr==8)
-      {
-         bitPtr=0;
-         bytePtr++;
-      }
-      nbBits--;
-   }
-   return d;
-}
-
-int speex_bits_peek(SpeexBits *bits)
-{
-   if ((bits->bytePtr<<3)+bits->bitPtr+1>bits->nbBits)
-      bits->overflow=1;
-   if (bits->overflow)
-      return 0;
-   return (bits->bytes[bits->bytePtr]>>(7-bits->bitPtr))&1;
-}
-
-void speex_bits_advance(SpeexBits *bits, int n)
-{
-   int nbytes, nbits;
-
-   if ((bits->bytePtr<<3)+bits->bitPtr+n>bits->nbBits)
-      bits->overflow=1;
-   if (bits->overflow)
-      return;
-
-   nbytes = n >> 3;
-   nbits = n & 7;
-   
-   bits->bytePtr += nbytes;
-   bits->bitPtr += nbits;
-   
-   if (bits->bitPtr>7)
-   {
-      bits->bitPtr-=8;
-      bits->bytePtr++;
-   }
-}
-
-int speex_bits_remaining(SpeexBits *bits)
-{
-   if (bits->overflow)
-      return -1;
-   else
-      return bits->nbBits-((bits->bytePtr<<3)+bits->bitPtr);
-}
-
-int speex_bits_nbytes(SpeexBits *bits)
-{
-   return ((bits->nbBits+7)>>3);
-}
-
-void speex_bits_insert_terminator(SpeexBits *bits)
-{
-   if (bits->bitPtr<7)
-      speex_bits_pack(bits, 0, 1);
-   while (bits->bitPtr<7)
-      speex_bits_pack(bits, 1, 1);
-}

engine/voice_codecs/speex/source/libspeex/cb_search.c

@@ -1,387 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: cb_search.c
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-
-#include "cb_search.h"
-#include "filters.h"
-#include "stack_alloc.h"
-#include "vq.h"
-#include "misc.h"
-
-void split_cb_search_shape_sign(
-float target[],			/* target vector */
-float ak[],			/* LPCs for this subframe */
-float awk1[],			/* Weighted LPCs for this subframe */
-float awk2[],			/* Weighted LPCs for this subframe */
-void *par,                      /* Codebook/search parameters*/
-int   p,                        /* number of LPC coeffs */
-int   nsf,                      /* number of samples in subframe */
-float *exc,
-float *r,
-SpeexBits *bits,
-char *stack,
-int   complexity
-)
-{
-   int i,j,k,m,n,q;
-   float *resp;
-   float *t, *e, *E, *r2;
-   float *tmp;
-   float *ndist, *odist;
-   int *itmp;
-   float **ot, **nt;
-   int **nind, **oind;
-   int *ind;
-   signed char *shape_cb;
-   int shape_cb_size, subvect_size, nb_subvect;
-   split_cb_params *params;
-   int N=2;
-   int *best_index;
-   float *best_dist;
-   int have_sign;
-
-   N=complexity;
-   if (N>10)
-      N=10;
-
-   ot=PUSH(stack, N, float*);
-   nt=PUSH(stack, N, float*);
-   oind=PUSH(stack, N, int*);
-   nind=PUSH(stack, N, int*);
-
-   params = (split_cb_params *) par;
-   subvect_size = params->subvect_size;
-   nb_subvect = params->nb_subvect;
-   shape_cb_size = 1<<params->shape_bits;
-   shape_cb = params->shape_cb;
-   have_sign = params->have_sign;
-   resp = PUSH(stack, shape_cb_size*subvect_size, float);
-   t = PUSH(stack, nsf, float);
-   e = PUSH(stack, nsf, float);
-   r2 = PUSH(stack, nsf, float);
-   E = PUSH(stack, shape_cb_size, float);
-   ind = PUSH(stack, nb_subvect, int);
-
-   tmp = PUSH(stack, 2*N*nsf, float);
-   for (i=0;i<N;i++)
-   {
-      ot[i]=tmp;
-      tmp += nsf;
-      nt[i]=tmp;
-      tmp += nsf;
-   }
-
-   best_index = PUSH(stack, N, int);
-   best_dist = PUSH(stack, N, float);
-   ndist = PUSH(stack, N, float);
-   odist = PUSH(stack, N, float);
-   
-   itmp = PUSH(stack, 2*N*nb_subvect, int);
-   for (i=0;i<N;i++)
-   {
-      nind[i]=itmp;
-      itmp+=nb_subvect;
-      oind[i]=itmp;
-      itmp+=nb_subvect;
-      for (j=0;j<nb_subvect;j++)
-         nind[i][j]=oind[i][j]=-1;
-   }
-
-   for (j=0;j<N;j++)
-      for (i=0;i<nsf;i++)
-         ot[j][i]=target[i];
-
-   for (i=0;i<nsf;i++)
-      t[i]=target[i];
-
-   /* Pre-compute codewords response and energy */
-   for (i=0;i<shape_cb_size;i++)
-   {
-      float *res;
-      signed char *shape;
-
-      res = resp+i*subvect_size;
-      shape = shape_cb+i*subvect_size;
-
-      /* Compute codeword response using convolution with impulse response */
-      for(j=0;j<subvect_size;j++)
-      {
-         res[j]=0;
-         for (k=0;k<=j;k++)
-            res[j] += 0.03125*shape[k]*r[j-k];
-      }
-      
-      /* Compute codeword energy */
-      E[i]=0;
-      for(j=0;j<subvect_size;j++)
-         E[i]+=res[j]*res[j];
-   }
-
-   for (j=0;j<N;j++)
-      odist[j]=0;
-   /*For all subvectors*/
-   for (i=0;i<nb_subvect;i++)
-   {
-      /*"erase" nbest list*/
-      for (j=0;j<N;j++)
-         ndist[j]=-1;
-
-      /*For all n-bests of previous subvector*/
-      for (j=0;j<N;j++)
-      {
-         float *x=ot[j]+subvect_size*i;
-         /*Find new n-best based on previous n-best j*/
-         if (have_sign)
-            vq_nbest_sign(x, resp, subvect_size, shape_cb_size, E, N, best_index, best_dist);
-         else
-            vq_nbest(x, resp, subvect_size, shape_cb_size, E, N, best_index, best_dist);
-
-         /*For all new n-bests*/
-         for (k=0;k<N;k++)
-         {
-            float *ct;
-            float err=0;
-            ct = ot[j];
-            /*update target*/
-
-            /*previous target*/
-            for (m=i*subvect_size;m<(i+1)*subvect_size;m++)
-               t[m]=ct[m];
-
-            /* New code: update only enough of the target to calculate error*/
-            {
-               int rind;
-               float *res;
-               float sign=1;
-               rind = best_index[k];
-               if (rind>=shape_cb_size)
-               {
-                  sign=-1;
-                  rind-=shape_cb_size;
-               }
-               res = resp+rind*subvect_size;
-               if (sign>0)
-                  for (m=0;m<subvect_size;m++)
-                     t[subvect_size*i+m] -= res[m];
-               else
-                  for (m=0;m<subvect_size;m++)
-                     t[subvect_size*i+m] += res[m];
-            }
-            
-            /*compute error (distance)*/
-            err=odist[j];
-            for (m=i*subvect_size;m<(i+1)*subvect_size;m++)
-               err += t[m]*t[m];
-            /*update n-best list*/
-            if (err<ndist[N-1] || ndist[N-1]<-.5)
-            {
-
-               /*previous target (we don't care what happened before*/
-               for (m=(i+1)*subvect_size;m<nsf;m++)
-                  t[m]=ct[m];
-               /* New code: update the rest of the target only if it's worth it */
-               for (m=0;m<subvect_size;m++)
-               {
-                  float g;
-                  int rind;
-                  float sign=1;
-                  rind = best_index[k];
-                  if (rind>=shape_cb_size)
-                  {
-                     sign=-1;
-                     rind-=shape_cb_size;
-                  }
-
-                  g=sign*0.03125*shape_cb[rind*subvect_size+m];
-                  q=subvect_size-m;
-                  for (n=subvect_size*(i+1);n<nsf;n++,q++)
-                     t[n] -= g*r[q];
-               }
-
-
-               for (m=0;m<N;m++)
-               {
-                  if (err < ndist[m] || ndist[m]<-.5)
-                  {
-                     for (n=N-1;n>m;n--)
-                     {
-                        for (q=(i+1)*subvect_size;q<nsf;q++)
-                           nt[n][q]=nt[n-1][q];
-                        for (q=0;q<nb_subvect;q++)
-                           nind[n][q]=nind[n-1][q];
-                        ndist[n]=ndist[n-1];
-                     }
-                     for (q=(i+1)*subvect_size;q<nsf;q++)
-                        nt[m][q]=t[q];
-                     for (q=0;q<nb_subvect;q++)
-                        nind[m][q]=oind[j][q];
-                     nind[m][i]=best_index[k];
-                     ndist[m]=err;
-                     break;
-                  }
-               }
-            }
-         }
-         if (i==0)
-           break;
-      }
-
-      /*update old-new data*/
-      /* just swap pointers instead of a long copy */
-      {
-         float **tmp2;
-         tmp2=ot;
-         ot=nt;
-         nt=tmp2;
-      }
-      for (j=0;j<N;j++)
-         for (m=0;m<nb_subvect;m++)
-            oind[j][m]=nind[j][m];
-      for (j=0;j<N;j++)
-         odist[j]=ndist[j];
-   }
-
-   /*save indices*/
-   for (i=0;i<nb_subvect;i++)
-   {
-      ind[i]=nind[0][i];
-      speex_bits_pack(bits,ind[i],params->shape_bits+have_sign);
-   }
-   
-   /* Put everything back together */
-   for (i=0;i<nb_subvect;i++)
-   {
-      int rind;
-      float sign=1;
-      rind = ind[i];
-      if (rind>=shape_cb_size)
-      {
-         sign=-1;
-         rind-=shape_cb_size;
-      }
-
-      for (j=0;j<subvect_size;j++)
-         e[subvect_size*i+j]=sign*0.03125*shape_cb[rind*subvect_size+j];
-   }   
-   /* Update excitation */
-   for (j=0;j<nsf;j++)
-      exc[j]+=e[j];
-   
-   /* Update target */
-   syn_percep_zero(e, ak, awk1, awk2, r2, nsf,p, stack);
-   for (j=0;j<nsf;j++)
-      target[j]-=r2[j];
-
-}
-
-
-void split_cb_shape_sign_unquant(
-float *exc,
-void *par,                      /* non-overlapping codebook */
-int   nsf,                      /* number of samples in subframe */
-SpeexBits *bits,
-char *stack
-)
-{
-   int i,j;
-   int *ind, *signs;
-   signed char *shape_cb;
-   int shape_cb_size, subvect_size, nb_subvect;
-   split_cb_params *params;
-   int have_sign;
-
-   params = (split_cb_params *) par;
-   subvect_size = params->subvect_size;
-   nb_subvect = params->nb_subvect;
-   shape_cb_size = 1<<params->shape_bits;
-   shape_cb = params->shape_cb;
-   have_sign = params->have_sign;
-
-   ind = PUSH(stack, nb_subvect, int);
-   signs = PUSH(stack, nb_subvect, int);
-
-   /* Decode codewords and gains */
-   for (i=0;i<nb_subvect;i++)
-   {
-      if (have_sign)
-         signs[i] = speex_bits_unpack_unsigned(bits, 1);
-      else
-         signs[i] = 0;
-      ind[i] = speex_bits_unpack_unsigned(bits, params->shape_bits);
-   }
-   /* Compute decoded excitation */
-   for (i=0;i<nb_subvect;i++)
-   {
-      float s=1;
-      if (signs[i])
-         s=-1;
-      for (j=0;j<subvect_size;j++)
-         exc[subvect_size*i+j]+=s*0.03125*shape_cb[ind[i]*subvect_size+j];
-   }
-}
-
-void noise_codebook_quant(
-float target[],			/* target vector */
-float ak[],			/* LPCs for this subframe */
-float awk1[],			/* Weighted LPCs for this subframe */
-float awk2[],			/* Weighted LPCs for this subframe */
-void *par,                      /* Codebook/search parameters*/
-int   p,                        /* number of LPC coeffs */
-int   nsf,                      /* number of samples in subframe */
-float *exc,
-float *r,
-SpeexBits *bits,
-char *stack,
-int   complexity
-)
-{
-   int i;
-   float *tmp=PUSH(stack, nsf, float);
-   residue_percep_zero(target, ak, awk1, awk2, tmp, nsf, p, stack);
-
-   for (i=0;i<nsf;i++)
-      exc[i]+=tmp[i];
-   for (i=0;i<nsf;i++)
-      target[i]=0;
-
-}
-
-
-void noise_codebook_unquant(
-float *exc,
-void *par,                      /* non-overlapping codebook */
-int   nsf,                      /* number of samples in subframe */
-SpeexBits *bits,
-char *stack
-)
-{
-   speex_rand_vec(1, exc, nsf);
-}

engine/voice_codecs/speex/source/libspeex/cb_search.h

@@ -1,95 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin & David Rowe
-   File: cb_search.c
-   Overlapped codebook search
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#ifndef CB_SEARCH_H
-#define CB_SEARCH_H
-
-#include "speex_bits.h"
-
-typedef struct split_cb_params {
-   int     subvect_size;
-   int     nb_subvect;
-   signed char  *shape_cb;
-   int     shape_bits;
-   int     have_sign;
-} split_cb_params;
-
-
-void split_cb_search_shape_sign(
-float target[],			/* target vector */
-float ak[],			/* LPCs for this subframe */
-float awk1[],			/* Weighted LPCs for this subframe */
-float awk2[],			/* Weighted LPCs for this subframe */
-void *par,                      /* Codebook/search parameters*/
-int   p,                        /* number of LPC coeffs */
-int   nsf,                      /* number of samples in subframe */
-float *exc,
-float *r,
-SpeexBits *bits,
-char *stack,
-int   complexity
-);
-
-void split_cb_shape_sign_unquant(
-float *exc,
-void *par,                      /* non-overlapping codebook */
-int   nsf,                      /* number of samples in subframe */
-SpeexBits *bits,
-char *stack
-);
-
-
-void noise_codebook_quant(
-float target[],			/* target vector */
-float ak[],			/* LPCs for this subframe */
-float awk1[],			/* Weighted LPCs for this subframe */
-float awk2[],			/* Weighted LPCs for this subframe */
-void *par,                      /* Codebook/search parameters*/
-int   p,                        /* number of LPC coeffs */
-int   nsf,                      /* number of samples in subframe */
-float *exc,
-float *r,
-SpeexBits *bits,
-char *stack,
-int   complexity
-);
-
-
-void noise_codebook_unquant(
-float *exc,
-void *par,                      /* non-overlapping codebook */
-int   nsf,                      /* number of samples in subframe */
-SpeexBits *bits,
-char *stack
-);
-
-#endif

engine/voice_codecs/speex/source/libspeex/exc_10_16_table.c

@@ -1,50 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: exc_10_16_table.c
-   Codebook for excitation in narrowband CELP mode (3200 bps)
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-
-signed char exc_10_16_table[160] = {
-22,39,14,44,11,35,-2,23,-4,6,
-46,-28,13,-27,-23,12,4,20,-5,9,
-37,-18,-23,23,0,9,-6,-20,4,-1,
--17,-5,-4,17,0,1,9,-2,1,2,
-2,-12,8,-25,39,15,9,16,-55,-11,
-9,11,5,10,-2,-60,8,13,-6,11,
--16,27,-47,-12,11,1,16,-7,9,-3,
--29,9,-14,25,-19,34,36,12,40,-10,
--3,-24,-14,-37,-21,-35,-2,-36,3,-6,
-67,28,6,-17,-3,-12,-16,-15,-17,-7,
--59,-36,-13,1,7,1,2,10,2,11,
-13,10,8,-2,7,3,5,4,2,2,
--3,-8,4,-5,6,7,-42,15,35,-2,
--46,38,28,-20,-9,1,7,-3,0,-2,
-0,0,0,0,0,0,0,0,0,0,
--15,-28,52,32,5,-5,-17,-20,-10,-1};

engine/voice_codecs/speex/source/libspeex/exc_10_32_table.c

@@ -1,66 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: exc_10_32_table.c
-   Codebook for excitation in narrowband CELP mode (4000 bps)
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-
-signed char exc_10_32_table[320] = {
-7,17,17,27,25,22,12,4,-3,0,
-28,-36,39,-24,-15,3,-9,15,-5,10,
-31,-28,11,31,-21,9,-11,-11,-2,-7,
--25,14,-22,31,4,-14,19,-12,14,-5,
-4,-7,4,-5,9,0,-2,42,-47,-16,
-1,8,0,9,23,-57,0,28,-11,6,
--31,55,-45,3,-5,4,2,-2,4,-7,
--3,6,-2,7,-3,12,5,8,54,-10,
-8,-7,-8,-24,-25,-27,-14,-5,8,5,
-44,23,5,-9,-11,-11,-13,-9,-12,-8,
--29,-8,-22,6,-15,3,-12,-1,-5,-3,
-34,-1,29,-16,17,-4,12,2,1,4,
--2,-4,2,-1,11,-3,-52,28,30,-9,
--32,25,44,-20,-24,4,6,-1,0,0,
-0,0,0,0,0,0,0,0,0,0,
--25,-10,22,29,13,-13,-22,-13,-4,0,
--4,-16,10,15,-36,-24,28,25,-1,-3,
-66,-33,-11,-15,6,0,3,4,-2,5,
-24,-20,-47,29,19,-2,-4,-1,0,-1,
--2,3,1,8,-11,5,5,-57,28,28,
-0,-16,4,-4,12,-6,-1,2,-20,61,
--9,24,-22,-42,29,6,17,8,4,2,
--65,15,8,10,5,6,5,3,2,-2,
--3,5,-9,4,-5,23,13,23,-3,-63,
-3,-5,-4,-6,0,-3,23,-36,-46,9,
-5,5,8,4,9,-5,1,-3,10,1,
--6,10,-11,24,-47,31,22,-12,14,-10,
-6,11,-7,-7,7,-31,51,-12,-6,7,
-6,-17,9,-11,-20,52,-19,3,-6,-6,
--8,-5,23,-41,37,1,-21,10,-14,8,
-7,5,-15,-15,23,39,-26,-33,7,2,
--32,-30,-21,-8,4,12,17,15,14,11};

engine/voice_codecs/speex/source/libspeex/exc_20_32_table.c

@@ -1,66 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: exc_20_32_table.c
-   Codebook for excitation in narrowband CELP mode (2000 bps)
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-
-signed char exc_20_32_table[640] = {
-12,32,25,46,36,33,9,14,-3,6,1,-8,0,-10,-5,-7,-7,-7,-5,-5,
-31,-27,24,-32,-4,10,-11,21,-3,19,23,-9,22,24,-10,-1,-10,-13,-7,-11,
-42,-33,31,19,-8,0,-10,-16,1,-21,-17,10,-8,14,8,4,11,-2,5,-2,
--33,11,-16,33,11,-4,9,-4,11,2,6,-5,8,-5,11,-4,-6,26,-36,-16,
-0,4,-2,-8,12,6,-1,34,-46,-22,9,9,21,9,5,-66,-5,26,2,10,
-13,2,19,9,12,-81,3,13,13,0,-14,22,-35,6,-7,-4,6,-6,10,-6,
--31,38,-33,0,-10,-11,5,-12,12,-17,5,0,-6,13,-9,10,8,25,33,2,
--12,8,-6,10,-2,21,7,17,43,5,11,-7,-9,-20,-36,-20,-23,-4,-4,-3,
-27,-9,-9,-49,-39,-38,-11,-9,6,5,23,25,5,3,3,4,1,2,-3,-1,
-87,39,17,-21,-9,-19,-9,-15,-13,-14,-17,-11,-10,-11,-8,-6,-1,-3,-3,-1,
--54,-34,-27,-8,-11,-4,-5,0,0,4,8,6,9,7,9,7,6,5,5,5,
-48,10,19,-10,12,-1,9,-3,2,5,-3,2,-2,-2,0,-2,-26,6,9,-7,
--16,-9,2,7,7,-5,-43,11,22,-11,-9,34,37,-15,-13,-6,1,-1,1,1,
--64,56,52,-11,-27,5,4,3,1,2,1,3,-1,-4,-4,-10,-7,-4,-4,2,
--1,-7,-7,-12,-10,-15,-9,-5,-5,-11,-16,-13,6,16,4,-13,-16,-10,-4,2,
--47,-13,25,47,19,-14,-20,-8,-17,0,-3,-13,1,6,-17,-14,15,1,10,6,
--24,0,-10,19,-69,-8,14,49,17,-5,33,-29,3,-4,0,2,-8,5,-6,2,
-120,-56,-12,-47,23,-9,6,-5,1,2,-5,1,-10,4,-1,-1,4,-1,0,-3,
-30,-52,-67,30,22,11,-1,-4,3,0,7,2,0,1,-10,-4,-8,-13,5,1,
-1,-1,5,13,-9,-3,-10,-62,22,48,-4,-6,2,3,5,1,1,4,1,13,
-3,-20,10,-9,13,-2,-4,9,-20,44,-1,20,-32,-67,19,0,28,11,8,2,
--11,15,-19,-53,31,2,34,10,6,-4,-58,8,10,13,14,1,12,2,0,0,
--128,37,-8,44,-9,26,-3,18,2,6,11,-1,9,1,5,3,0,1,1,2,
-12,3,-2,-3,7,25,9,18,-6,-37,3,-8,-16,3,-10,-7,17,-34,-44,11,
-17,-15,-3,-16,-1,-13,11,-46,-65,-2,8,13,2,4,4,5,15,5,9,6,
-0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
--9,19,-12,12,-28,38,29,-1,12,2,5,23,-10,3,4,-15,21,-4,3,3,
-6,17,-9,-4,-8,-20,26,5,-10,6,1,-19,18,-15,-12,47,-6,-2,-7,-9,
--1,-17,-2,-2,-14,30,-14,2,-7,-4,-1,-12,11,-25,16,-3,-12,11,-7,7,
--17,1,19,-28,31,-7,-10,7,-10,3,12,5,-16,6,24,41,-29,-54,0,1,
-7,-1,5,-6,13,10,-4,-8,8,-9,-27,-53,-38,-1,10,19,17,16,12,12,
-0,3,-7,-4,13,12,-31,-14,6,-5,3,5,17,43,50,25,10,1,-6,-2};

engine/voice_codecs/speex/source/libspeex/exc_5_256_table.c

@@ -1,290 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: exc_5_256_table.c
-   Codebook for excitation in narrowband CELP mode (12800 bps)
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-
-signed char exc_5_256_table[1280] = {
--8,-37,5,-43,5,
-73,61,39,12,-3,
--61,-32,2,42,30,
--3,17,-27,9,34,
-20,-1,-5,2,23,
--7,-46,26,53,-47,
-20,-2,-33,-89,-51,
--64,27,11,15,-34,
--5,-56,25,-9,-1,
--29,1,40,67,-23,
--16,16,33,19,7,
-14,85,22,-10,-10,
--12,-7,-1,52,89,
-29,11,-20,-37,-46,
--15,17,-24,-28,24,
-2,1,0,23,-101,
-23,14,-1,-23,-18,
-9,5,-13,38,1,
--28,-28,4,27,51,
--26,34,-40,35,47,
-54,38,-54,-26,-6,
-42,-25,13,-30,-36,
-18,41,-4,-33,23,
--32,-7,-4,51,-3,
-17,-52,56,-47,36,
--2,-21,36,10,8,
--33,31,19,9,-5,
--40,10,-9,-21,19,
-18,-78,-18,-5,0,
--26,-36,-47,-51,-44,
-18,40,27,-2,29,
-49,-26,2,32,-54,
-30,-73,54,3,-5,
-36,22,53,10,-1,
--84,-53,-29,-5,3,
--44,53,-51,4,22,
-71,-35,-1,33,-5,
--27,-7,36,17,-23,
--39,16,-9,-55,-15,
--20,39,-35,6,-39,
--14,18,48,-64,-17,
--15,9,39,81,37,
--68,37,47,-21,-6,
--104,13,6,9,-2,
-35,8,-23,18,42,
-45,21,33,-5,-49,
-9,-6,-43,-56,39,
-2,-16,-25,87,1,
--3,-9,17,-25,-11,
--9,-1,10,2,-14,
--14,4,-1,-10,28,
--23,40,-32,26,-9,
-26,4,-27,-23,3,
-42,-60,1,49,-3,
-27,10,-52,-40,-2,
-18,45,-23,17,-44,
-3,-3,17,-46,52,
--40,-47,25,75,31,
--49,53,30,-30,-32,
--36,38,-6,-15,-16,
-54,-27,-48,3,38,
--29,-32,-22,-14,-4,
--23,-13,32,-39,9,
-8,-45,-13,34,-16,
-49,40,32,31,28,
-23,23,32,47,59,
--68,8,62,44,25,
--14,-24,-65,-16,36,
-67,-25,-38,-21,4,
--33,-2,42,5,-63,
-40,11,26,-42,-23,
--61,79,-31,23,-20,
-10,-32,53,-25,-36,
-10,-26,-5,3,0,
--71,5,-10,-37,1,
--24,21,-54,-17,1,
--29,-25,-15,-27,32,
-68,45,-16,-37,-18,
--5,1,0,-77,71,
--6,3,-20,71,-67,
-29,-35,10,-30,19,
-4,16,17,5,0,
--14,19,2,28,26,
-59,3,2,24,39,
-55,-50,-45,-18,-17,
-33,-35,14,-1,1,
-8,87,-35,-29,0,
--27,13,-7,23,-13,
-37,-40,50,-35,14,
-19,-7,-14,49,54,
--5,22,-2,-29,-8,
--27,38,13,27,48,
-12,-41,-21,-15,28,
-7,-16,-24,-19,-20,
-11,-20,9,2,13,
-23,-20,11,27,-27,
-71,-69,8,2,-6,
-22,12,16,16,9,
--16,-8,-17,1,25,
-1,40,-37,-33,66,
-94,53,4,-22,-25,
--41,-42,25,35,-16,
--15,57,31,-29,-32,
-21,16,-60,45,15,
--1,7,57,-26,-47,
--29,11,8,15,19,
--105,-8,54,27,10,
--17,6,-12,-1,-10,
-4,0,23,-10,31,
-13,11,10,12,-64,
-23,-3,-8,-19,16,
-52,24,-40,16,10,
-40,5,9,0,-13,
--7,-21,-8,-6,-7,
--21,59,16,-53,18,
--60,11,-47,14,-18,
-25,-13,-24,4,-39,
-16,-28,54,26,-67,
-30,27,-20,-52,20,
--12,55,12,18,-16,
-39,-14,-6,-26,56,
--88,-55,12,25,26,
--37,6,75,0,-34,
--81,54,-30,1,-7,
-49,-23,-14,21,10,
--62,-58,-57,-47,-34,
-15,-4,34,-78,31,
-25,-11,7,50,-10,
-42,-63,14,-36,-4,
-57,55,57,53,42,
--42,-1,15,40,37,
-15,25,-11,6,1,
-31,-2,-6,-1,-7,
--64,34,28,30,-1,
-3,21,0,-88,-12,
--56,25,-28,40,8,
--28,-14,9,12,2,
--6,-17,22,49,-6,
--26,14,28,-20,4,
--12,50,35,40,13,
--38,-58,-29,17,30,
-22,60,26,-54,-39,
--12,58,-28,-63,10,
--21,-8,-12,26,-62,
-6,-10,-11,-22,-6,
--7,4,1,18,2,
--70,11,14,4,13,
-19,-24,-34,24,67,
-17,51,-21,13,23,
-54,-30,48,1,-13,
-80,26,-16,-2,13,
--4,6,-30,29,-24,
-73,-58,30,-27,20,
--2,-21,41,45,30,
--27,-3,-5,-18,-20,
--49,-3,-35,10,42,
--19,-67,-53,-11,9,
-13,-15,-33,-51,-30,
-15,7,25,-30,4,
-28,-22,-34,54,-29,
-39,-46,20,16,34,
--4,47,75,1,-44,
--55,-24,7,-1,9,
--42,50,-8,-36,41,
-68,0,-4,-10,-23,
--15,-50,64,36,-9,
--27,12,25,-38,-47,
--37,32,-49,51,-36,
-2,-4,69,-26,19,
-7,45,67,46,13,
--63,46,15,-47,4,
--41,13,-6,5,-21,
-37,26,-55,-7,33,
--1,-28,10,-17,-64,
--14,0,-36,-17,93,
--3,-9,-66,44,-21,
-3,-12,38,-6,-13,
--12,19,13,43,-43,
--10,-12,6,-5,9,
--49,32,-5,2,4,
-5,15,-16,10,-21,
-8,-62,-8,64,8,
-79,-1,-66,-49,-18,
-5,40,-5,-30,-45,
-1,-6,21,-32,93,
--18,-30,-21,32,21,
--18,22,8,5,-41,
--54,80,22,-10,-7,
--8,-23,-64,66,56,
--14,-30,-41,-46,-14,
--29,-37,27,-14,42,
--2,-9,-29,34,14,
-33,-14,22,4,10,
-26,26,28,32,23,
--72,-32,3,0,-14,
-35,-42,-78,-32,6,
-29,-18,-45,-5,7,
--33,-45,-3,-22,-34,
-8,-8,4,-51,-25,
--9,59,-78,21,-5,
--25,-48,66,-15,-17,
--24,-49,-13,25,-23,
--64,-6,40,-24,-19,
--11,57,-33,-8,1,
-10,-52,-54,28,39,
-49,34,-11,-61,-41,
--43,10,15,-15,51,
-30,15,-51,32,-34,
--2,-34,14,18,16,
-1,1,-3,-3,1,
-1,-18,6,16,48,
-12,-5,-42,7,36,
-48,7,-20,-10,7,
-12,2,54,39,-38,
-37,54,4,-11,-8,
--46,-10,5,-10,-34,
-46,-12,29,-37,39,
-36,-11,24,56,17,
-14,20,25,0,-25,
--28,55,-7,-5,27,
-3,9,-26,-8,6,
--24,-10,-30,-31,-34,
-18,4,22,21,40,
--1,-29,-37,-8,-21,
-92,-29,11,-3,11,
-73,23,22,7,4,
--44,-9,-11,21,-13,
-11,9,-78,-1,47,
-114,-12,-37,-19,-5,
--11,-22,19,12,-30,
-7,38,45,-21,-8,
--9,55,-45,56,-21,
-7,17,46,-57,-87,
--6,27,31,31,7,
--56,-12,46,21,-5,
--12,36,3,3,-21,
-43,19,12,-7,9,
--14,0,-9,-33,-91,
-7,26,3,-11,64,
-83,-31,-46,25,2,
-9,5,2,2,-1,
-20,-17,10,-5,-27,
--8,20,8,-19,16,
--21,-13,-31,5,5,
-42,24,9,34,-20,
-28,-61,22,11,-39,
-64,-20,-1,-30,-9,
--20,24,-25,-24,-29,
-22,-60,6,-5,41,
--9,-87,14,34,15,
--57,52,69,15,-3,
--102,58,16,3,6,
-60,-75,-32,26,7,
--57,-27,-32,-24,-21,
--29,-16,62,-46,31,
-30,-27,-15,7,15};

engine/voice_codecs/speex/source/libspeex/exc_5_64_table.c

@@ -1,98 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: exc_5_64_table.c
-   Codebook for excitation in narrowband CELP mode (9600 bps)
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-
-signed char exc_5_64_table[320]={
-1,5,-15,49,-66,
--48,-4,50,-44,7,
-37,16,-18,25,-26,
--26,-15,19,19,-27,
--47,28,57,5,-17,
--32,-41,68,21,-2,
-64,56,8,-16,-13,
--26,-9,-16,11,6,
--39,25,-19,22,-31,
-20,-45,55,-43,10,
--16,47,-40,40,-20,
--51,3,-17,-14,-15,
--24,53,-20,-46,46,
-27,-68,32,3,-18,
--5,9,-31,16,-9,
--10,-1,-23,48,95,
-47,25,-41,-32,-3,
-15,-25,-55,36,41,
--27,20,5,13,14,
--22,5,2,-23,18,
-46,-15,17,-18,-34,
--5,-8,27,-55,73,
-16,2,-1,-17,40,
--78,33,0,2,19,
-4,53,-16,-15,-16,
--28,-3,-13,49,8,
--7,-29,27,-13,32,
-20,32,-61,16,14,
-41,44,40,24,20,
-7,4,48,-60,-77,
-17,-6,-48,65,-15,
-32,-30,-71,-10,-3,
--6,10,-2,-7,-29,
--56,67,-30,7,-5,
-86,-6,-10,0,5,
--31,60,34,-38,-3,
-24,10,-2,30,23,
-24,-41,12,70,-43,
-15,-17,6,13,16,
--13,8,30,-15,-8,
-5,23,-34,-98,-4,
--13,13,-48,-31,70,
-12,31,25,24,-24,
-26,-7,33,-16,8,
-5,-11,-14,-8,-65,
-13,10,-2,-9,0,
--3,-68,5,35,7,
-0,-31,-1,-17,-9,
--9,16,-37,-18,-1,
-69,-48,-28,22,-21,
--11,5,49,55,23,
--86,-36,16,2,13,
-63,-51,30,-11,13,
-24,-18,-6,14,-19,
-1,41,9,-5,27,
--36,-44,-34,-37,-21,
--26,31,-39,15,43,
-5,-8,29,20,-8,
--20,-52,-28,-1,13,
-26,-34,-10,-9,27,
--8,8,27,-66,4,
-12,-22,49,10,-77,
-32,-18,3,-38,12,
--3,-1,2,2,0};

engine/voice_codecs/speex/source/libspeex/exc_8_128_table.c

@@ -1,162 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: exc_8_128_table.c
-   Codebook for excitation in narrowband CELP mode (7000 bps)
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-
-signed char exc_8_128_table[1024] = {
--14,9,13,-32,2,-10,31,-10,
--8,-8,6,-4,-1,10,-64,23,
-6,20,13,6,8,-22,16,34,
-7,42,-49,-28,5,26,4,-15,
-41,34,41,32,33,24,23,14,
-8,40,34,4,-24,-41,-19,-15,
-13,-13,33,-54,24,27,-44,33,
-27,-15,-15,24,-19,14,-36,14,
--9,24,-12,-4,37,-5,16,-34,
-5,10,33,-15,-54,-16,12,25,
-12,1,2,0,3,-1,-4,-4,
-11,2,-56,54,27,-20,13,-6,
--46,-41,-33,-11,-5,7,12,14,
--14,-5,8,20,6,3,4,-8,
--5,-42,11,8,-14,25,-2,2,
-13,11,-22,39,-9,9,5,-45,
--9,7,-9,12,-7,34,-17,-102,
-7,2,-42,18,35,-9,-34,11,
--5,-2,3,22,46,-52,-25,-9,
--94,8,11,-5,-5,-5,4,-7,
--35,-7,54,5,-32,3,24,-9,
--22,8,65,37,-1,-12,-23,-6,
--9,-28,55,-33,14,-3,2,18,
--60,41,-17,8,-16,17,-11,0,
--11,29,-28,37,9,-53,33,-14,
--9,7,-25,-7,-11,26,-32,-8,
-24,-21,22,-19,19,-10,29,-14,
-0,0,0,0,0,0,0,0,
--5,-52,10,41,6,-30,-4,16,
-32,22,-27,-22,32,-3,-28,-3,
-3,-35,6,17,23,21,8,2,
-4,-45,-17,14,23,-4,-31,-11,
--3,14,1,19,-11,2,61,-8,
-9,-12,7,-10,12,-3,-24,99,
--48,23,50,-37,-5,-23,0,8,
--14,35,-64,-5,46,-25,13,-1,
--49,-19,-15,9,34,50,25,11,
--6,-9,-16,-20,-32,-33,-32,-27,
-10,-8,12,-15,56,-14,-32,33,
-3,-9,1,65,-9,-9,-10,-2,
--6,-23,9,17,3,-28,13,-32,
-4,-2,-10,4,-16,76,12,-52,
-6,13,33,-6,4,-14,-9,-3,
-1,-15,-16,28,1,-15,11,16,
-9,4,-21,-37,-40,-6,22,12,
--15,-23,-14,-17,-16,-9,-10,-9,
-13,-39,41,5,-9,16,-38,25,
-46,-47,4,49,-14,17,-2,6,
-18,5,-6,-33,-22,44,50,-2,
-1,3,-6,7,7,-3,-21,38,
--18,34,-14,-41,60,-13,6,16,
--24,35,19,-13,-36,24,3,-17,
--14,-10,36,44,-44,-29,-3,3,
--54,-8,12,55,26,4,-2,-5,
-2,-11,22,-23,2,22,1,-25,
--39,66,-49,21,-8,-2,10,-14,
--60,25,6,10,27,-25,16,5,
--2,-9,26,-13,-20,58,-2,7,
-52,-9,2,5,-4,-15,23,-1,
--38,23,8,27,-6,0,-27,-7,
-39,-10,-14,26,11,-45,-12,9,
--5,34,4,-35,10,43,-22,-11,
-56,-7,20,1,10,1,-26,9,
-94,11,-27,-14,-13,1,-11,0,
-14,-5,-6,-10,-4,-15,-8,-41,
-21,-5,1,-28,-8,22,-9,33,
--23,-4,-4,-12,39,4,-7,3,
--60,80,8,-17,2,-6,12,-5,
-1,9,15,27,31,30,27,23,
-61,47,26,10,-5,-8,-12,-13,
-5,-18,25,-15,-4,-15,-11,12,
--2,-2,-16,-2,-6,24,12,11,
--4,9,1,-9,14,-45,57,12,
-20,-35,26,11,-64,32,-10,-10,
-42,-4,-9,-16,32,24,7,10,
-52,-11,-57,29,0,8,0,-6,
-17,-17,-56,-40,7,20,18,12,
--6,16,5,7,-1,9,1,10,
-29,12,16,13,-2,23,7,9,
--3,-4,-5,18,-64,13,55,-25,
-9,-9,24,14,-25,15,-11,-40,
--30,37,1,-19,22,-5,-31,13,
--2,0,7,-4,16,-67,12,66,
--36,24,-8,18,-15,-23,19,0,
--45,-7,4,3,-13,13,35,5,
-13,33,10,27,23,0,-7,-11,
-43,-74,36,-12,2,5,-8,6,
--33,11,-16,-14,-5,-7,-3,17,
--34,27,-16,11,-9,15,33,-31,
-8,-16,7,-6,-7,63,-55,-17,
-11,-1,20,-46,34,-30,6,9,
-19,28,-9,5,-24,-8,-23,-2,
-31,-19,-16,-5,-15,-18,0,26,
-18,37,-5,-15,-2,17,5,-27,
-21,-33,44,12,-27,-9,17,11,
-25,-21,-31,-7,13,33,-8,-25,
--7,7,-10,4,-6,-9,48,-82,
--23,-8,6,11,-23,3,-3,49,
--29,25,31,4,14,16,9,-4,
--18,10,-26,3,5,-44,-9,9,
--47,-55,15,9,28,1,4,-3,
-46,6,-6,-38,-29,-31,-15,-6,
-3,0,14,-6,8,-54,-50,33,
--5,1,-14,33,-48,26,-4,-5,
--3,-5,-3,-5,-28,-22,77,55,
--1,2,10,10,-9,-14,-66,-49,
-11,-36,-6,-20,10,-10,16,12,
-4,-1,-16,45,-44,-50,31,-2,
-25,42,23,-32,-22,0,11,20,
--40,-35,-40,-36,-32,-26,-21,-13,
-52,-22,6,-24,-20,17,-5,-8,
-36,-25,-11,21,-26,6,34,-8,
-7,20,-3,5,-25,-8,18,-5,
--9,-4,1,-9,20,20,39,48,
--24,9,5,-65,22,29,4,3,
--43,-11,32,-6,9,19,-27,-10,
--47,-14,24,10,-7,-36,-7,-1,
--4,-5,-5,16,53,25,-26,-29,
--4,-12,45,-58,-34,33,-5,2,
--1,27,-48,31,-15,22,-5,4,
-7,7,-25,-3,11,-22,16,-12,
-8,-3,7,-11,45,14,-73,-19,
-56,-46,24,-20,28,-12,-2,-1,
--36,-3,-33,19,-6,7,2,-15,
-5,-31,-45,8,35,13,20,0,
--9,48,-13,-43,-3,-13,2,-5,
-72,-68,-27,2,1,-2,-7,5,
-36,33,-40,-12,-4,-5,23,19};

engine/voice_codecs/speex/source/libspeex/filters.c

@@ -1,292 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: filters.c
-   Various analysis/synthesis filters
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#include "filters.h"
-#include "stack_alloc.h"
-#include <math.h>
-
-
-void bw_lpc(float gamma, float *lpc_in, float *lpc_out, int order)
-{
-   int i;
-   float tmp=1;
-   for (i=0;i<order+1;i++)
-   {
-      lpc_out[i] = tmp * lpc_in[i];
-      tmp *= gamma;
-   }
-}
-
-#ifdef _USE_SSE
-#include "filters_sse.h"
-#else
-void filter_mem2(float *x, float *num, float *den, float *y, int N, int ord, float *mem)
-{
-   int i,j;
-   float xi,yi;
-   for (i=0;i<N;i++)
-   {
-      xi=x[i];
-      y[i] = num[0]*xi + mem[0];
-      yi=y[i];
-      for (j=0;j<ord-1;j++)
-      {
-         mem[j] = mem[j+1] + num[j+1]*xi - den[j+1]*yi;
-      }
-      mem[ord-1] = num[ord]*xi - den[ord]*yi;
-   }
-}
-
-
-void iir_mem2(float *x, float *den, float *y, int N, int ord, float *mem)
-{
-   int i,j;
-   for (i=0;i<N;i++)
-   {
-      y[i] = x[i] + mem[0];
-      for (j=0;j<ord-1;j++)
-      {
-         mem[j] = mem[j+1] - den[j+1]*y[i];
-      }
-      mem[ord-1] = - den[ord]*y[i];
-   }
-}
-#endif
-
-void fir_mem2(float *x, float *num, float *y, int N, int ord, float *mem)
-{
-   int i,j;
-   float xi;
-   for (i=0;i<N;i++)
-   {
-      xi=x[i];
-      y[i] = num[0]*xi + mem[0];
-      for (j=0;j<ord-1;j++)
-      {
-         mem[j] = mem[j+1] + num[j+1]*xi;
-      }
-      mem[ord-1] = num[ord]*xi;
-   }
-}
-
-void syn_percep_zero(float *xx, float *ak, float *awk1, float *awk2, float *y, int N, int ord, char *stack)
-{
-   int i;
-   float *mem = PUSH(stack,ord, float);
-   for (i=0;i<ord;i++)
-      mem[i]=0;
-   filter_mem2(xx, awk1, ak, y, N, ord, mem);
-   for (i=0;i<ord;i++)
-     mem[i]=0;
-   iir_mem2(y, awk2, y, N, ord, mem);
-}
-
-void residue_percep_zero(float *xx, float *ak, float *awk1, float *awk2, float *y, int N, int ord, char *stack)
-{
-   int i;
-   float *mem = PUSH(stack,ord, float);
-   for (i=0;i<ord;i++)
-      mem[i]=0;
-   filter_mem2(xx, ak, awk1, y, N, ord, mem);
-   for (i=0;i<ord;i++)
-     mem[i]=0;
-   fir_mem2(y, awk2, y, N, ord, mem);
-}
-
-
-void qmf_decomp(float *xx, float *aa, float *y1, float *y2, int N, int M, float *mem, char *stack)
-{
-   int i,j,k,M2;
-   float *a;
-   float *x;
-   float *x2;
-   
-   a = PUSH(stack, M, float);
-   x = PUSH(stack, N+M-1, float);
-   x2=x+M-1;
-   M2=M>>1;
-   for (i=0;i<M;i++)
-      a[M-i-1]=aa[i];
-   for (i=0;i<M-1;i++)
-      x[i]=mem[M-i-2];
-   for (i=0;i<N;i++)
-      x[i+M-1]=xx[i];
-   for (i=0,k=0;i<N;i+=2,k++)
-   {
-      y1[k]=0;
-      y2[k]=0;
-      for (j=0;j<M2;j++)
-      {
-         y1[k]+=a[j]*(x[i+j]+x2[i-j]);
-         y2[k]-=a[j]*(x[i+j]-x2[i-j]);
-         j++;
-         y1[k]+=a[j]*(x[i+j]+x2[i-j]);
-         y2[k]+=a[j]*(x[i+j]-x2[i-j]);
-      }
-   }
-   for (i=0;i<M-1;i++)
-     mem[i]=xx[N-i-1];
-}
-
-/* By segher */
-void fir_mem_up(float *x, float *a, float *y, int N, int M, float *mem, char *stack)
-   /* assumptions:
-      all odd x[i] are zero -- well, actually they are left out of the array now
-      N and M are multiples of 4 */
-{
-   int i, j;
-   float *xx=PUSH(stack, M+N-1, float);
-
-   for (i = 0; i < N/2; i++)
-      xx[2*i] = x[N/2-1-i];
-   for (i = 0; i < M - 1; i += 2)
-      xx[N+i] = mem[i+1];
-
-   for (i = 0; i < N; i += 4) {
-      float y0, y1, y2, y3;
-      float x0;
-
-      y0 = y1 = y2 = y3 = 0.f;
-      x0 = xx[N-4-i];
-
-      for (j = 0; j < M; j += 4) {
-         float x1;
-         float a0, a1;
-
-         a0 = a[j];
-         a1 = a[j+1];
-         x1 = xx[N-2+j-i];
-
-         y0 += a0 * x1;
-         y1 += a1 * x1;
-         y2 += a0 * x0;
-         y3 += a1 * x0;
-
-         a0 = a[j+2];
-         a1 = a[j+3];
-         x0 = xx[N+j-i];
-
-         y0 += a0 * x0;
-         y1 += a1 * x0;
-         y2 += a0 * x1;
-         y3 += a1 * x1;
-      }
-      y[i] = y0;
-      y[i+1] = y1;
-      y[i+2] = y2;
-      y[i+3] = y3;
-   }
-
-   for (i = 0; i < M - 1; i += 2)
-      mem[i+1] = xx[i];
-}
-
-
-void comp_filter_mem_init (CombFilterMem *mem)
-{
-   mem->last_pitch=0;
-   mem->last_pitch_gain[0]=mem->last_pitch_gain[1]=mem->last_pitch_gain[2]=0;
-   mem->smooth_gain=1;
-}
-
-void comb_filter(
-float *exc,          /*decoded excitation*/
-float *new_exc,      /*enhanced excitation*/
-float *ak,           /*LPC filter coefs*/
-int p,               /*LPC order*/
-int nsf,             /*sub-frame size*/
-int pitch,           /*pitch period*/
-float *pitch_gain,   /*pitch gain (3-tap)*/
-float  comb_gain,    /*gain of comb filter*/
-CombFilterMem *mem
-)
-{
-   int i;
-   float exc_energy=0, new_exc_energy=0;
-   float gain;
-   float step;
-   float fact;
-   /*Compute excitation energy prior to enhancement*/
-   for (i=0;i<nsf;i++)
-      exc_energy+=exc[i]*exc[i];
-
-   /*Some gain adjustment is pitch is too high or if unvoiced*/
-   {
-      float g=0;
-      g = .5*fabs(pitch_gain[0]+pitch_gain[1]+pitch_gain[2] +
-      mem->last_pitch_gain[0] + mem->last_pitch_gain[1] + mem->last_pitch_gain[2]);
-      if (g>1.3)
-         comb_gain*=1.3/g;
-      if (g<.5)
-         comb_gain*=2*g;
-   }
-   step = 1.0/nsf;
-   fact=0;
-   /*Apply pitch comb-filter (filter out noise between pitch harmonics)*/
-   for (i=0;i<nsf;i++)
-   {
-      fact += step;
-
-      new_exc[i] = exc[i] + comb_gain * fact * (
-                                         pitch_gain[0]*exc[i-pitch+1] +
-                                         pitch_gain[1]*exc[i-pitch] +
-                                         pitch_gain[2]*exc[i-pitch-1]
-                                         )
-      + comb_gain * (1-fact) * (
-                                         mem->last_pitch_gain[0]*exc[i-mem->last_pitch+1] +
-                                         mem->last_pitch_gain[1]*exc[i-mem->last_pitch] +
-                                         mem->last_pitch_gain[2]*exc[i-mem->last_pitch-1]
-                                         );
-   }
-
-   mem->last_pitch_gain[0] = pitch_gain[0];
-   mem->last_pitch_gain[1] = pitch_gain[1];
-   mem->last_pitch_gain[2] = pitch_gain[2];
-   mem->last_pitch = pitch;
-
-   /*Gain after enhancement*/
-   for (i=0;i<nsf;i++)
-      new_exc_energy+=new_exc[i]*new_exc[i];
-
-   /*Compute scaling factor and normalize energy*/
-   gain = sqrt(exc_energy)/sqrt(.1+new_exc_energy);
-   if (gain < .5)
-      gain=.5;
-   if (gain>1)
-      gain=1;
-
-   for (i=0;i<nsf;i++)
-   {
-      mem->smooth_gain = .96*mem->smooth_gain + .04*gain;
-      new_exc[i] *= mem->smooth_gain;
-   }
-}

engine/voice_codecs/speex/source/libspeex/filters.h

@@ -1,79 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: filters.h
-   Various analysis/synthesis filters
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#ifndef FILTERS_H
-#define FILTERS_H
-
-
-typedef struct CombFilterMem {
-   int   last_pitch;
-   float last_pitch_gain[3];
-   float smooth_gain;
-} CombFilterMem;
-
-
-void qmf_decomp(float *xx, float *aa, float *y1, float *y2, int N, int M, float *mem, char *stack);
-void fir_mem_up(float *x, float *a, float *y, int N, int M, float *mem, char *stack);
-
-
-void filter_mem2(float *x, float *num, float *den, float *y, int N, int ord, float *mem);
-void fir_mem2(float *x, float *num, float *y, int N, int ord, float *mem);
-void iir_mem2(float *x, float *den, float *y, int N, int ord, float *mem);
-
-/* Apply bandwidth expansion on LPC coef */
-void bw_lpc(float gamma, float *lpc_in, float *lpc_out, int order);
-
-
-
-/* FIR filter */
-void fir_decim_mem(float *x, float *a, float *y, int N, int M, float *mem);
-
-void syn_percep_zero(float *x, float *ak, float *awk1, float *awk2, float *y, int N, int ord, char *stack);
-
-void residue_percep_zero(float *xx, float *ak, float *awk1, float *awk2, float *y, int N, int ord, char *stack);
-
-void comp_filter_mem_init (CombFilterMem *mem);
-
-void comb_filter(
-float *exc,          /*decoded excitation*/
-float *new_exc,      /*enhanced excitation*/
-float *ak,           /*LPC filter coefs*/
-int p,               /*LPC order*/
-int nsf,             /*sub-frame size*/
-int pitch,           /*pitch period*/
-float *pitch_gain,   /*pitch gain (3-tap)*/
-float  comb_gain,    /*gain of comb filter*/
-CombFilterMem *mem
-);
-
-
-#endif

engine/voice_codecs/speex/source/libspeex/filters_sse.h

@@ -1,289 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: filters.c
-   Various analysis/synthesis filters
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-void filter_mem2(float *x, float *_num, float *_den, float *y, int N, int ord, float *_mem)
-{
-   float __num[20], __den[20], __mem[20];
-   float *num, *den, *mem;
-   int i;
-
-   num = (float*)(((int)(__num+4))&0xfffffff0)-1;
-   den = (float*)(((int)(__den+4))&0xfffffff0)-1;
-   mem = (float*)(((int)(__mem+4))&0xfffffff0)-1;
-   for (i=0;i<=10;i++)
-      num[i]=den[i]=0;
-   for (i=0;i<10;i++)
-      mem[i]=0;
-
-   for (i=0;i<ord+1;i++)
-   {
-      num[i]=_num[i];
-      den[i]=_den[i];
-   }
-   for (i=0;i<ord;i++)
-      mem[i]=_mem[i];
-   for (i=0;i<N;i+=4)
-   {
-
-      __asm__ __volatile__ 
-      (
-       "\tmovss (%1), %%xmm0\n"
-       "\tmovss (%0), %%xmm1\n"
-       "\taddss %%xmm0, %%xmm1\n"
-       "\tmovss %%xmm1, (%2)\n"
-       "\tshufps $0x00, %%xmm0, %%xmm0\n"
-       "\tshufps $0x00, %%xmm1, %%xmm1\n"
-
-       "\tmovaps 4(%3),  %%xmm2\n"
-       "\tmovaps 4(%4),  %%xmm3\n"
-       "\tmulps  %%xmm0, %%xmm2\n"
-       "\tmulps  %%xmm1, %%xmm3\n"
-       "\tmovaps 20(%3), %%xmm4\n"
-       "\tmulps  %%xmm0, %%xmm4\n"
-       "\taddps  4(%0),  %%xmm2\n"
-       "\tmovaps 20(%4), %%xmm5\n"
-       "\tmulps  %%xmm1, %%xmm5\n"
-       "\taddps  20(%0), %%xmm4\n"
-       "\tsubps  %%xmm3, %%xmm2\n"
-       "\tmovups %%xmm2, (%0)\n"
-       "\tsubps  %%xmm5, %%xmm4\n"
-       "\tmovups %%xmm4, 16(%0)\n"
-
-       "\tmovss  36(%3), %%xmm2\n"
-       "\tmulss  %%xmm0, %%xmm2\n"
-       "\tmovss  36(%4), %%xmm3\n"
-       "\tmulss  %%xmm1, %%xmm3\n"
-       "\taddss  36(%0), %%xmm2\n"
-       "\tmovss  40(%3), %%xmm4\n"
-       "\tmulss  %%xmm0, %%xmm4\n"
-       "\tmovss  40(%4), %%xmm5\n"
-       "\tmulss  %%xmm1, %%xmm5\n"
-       "\tsubss  %%xmm3, %%xmm2\n"
-       "\tmovss  %%xmm2, 32(%0)       \n"
-       "\tsubss  %%xmm5, %%xmm4\n"
-       "\tmovss  %%xmm4, 36(%0)\n"
-
-
-
-       "\tmovss 4(%1), %%xmm0\n"
-       "\tmovss (%0), %%xmm1\n"
-       "\taddss %%xmm0, %%xmm1\n"
-       "\tmovss %%xmm1, 4(%2)\n"
-       "\tshufps $0x00, %%xmm0, %%xmm0\n"
-       "\tshufps $0x00, %%xmm1, %%xmm1\n"
-
-       "\tmovaps 4(%3),  %%xmm2\n"
-       "\tmovaps 4(%4),  %%xmm3\n"
-       "\tmulps  %%xmm0, %%xmm2\n"
-       "\tmulps  %%xmm1, %%xmm3\n"
-       "\tmovaps 20(%3), %%xmm4\n"
-       "\tmulps  %%xmm0, %%xmm4\n"
-       "\taddps  4(%0),  %%xmm2\n"
-       "\tmovaps 20(%4), %%xmm5\n"
-       "\tmulps  %%xmm1, %%xmm5\n"
-       "\taddps  20(%0), %%xmm4\n"
-       "\tsubps  %%xmm3, %%xmm2\n"
-       "\tmovups %%xmm2, (%0)\n"
-       "\tsubps  %%xmm5, %%xmm4\n"
-       "\tmovups %%xmm4, 16(%0)\n"
-
-       "\tmovss  36(%3), %%xmm2\n"
-       "\tmulss  %%xmm0, %%xmm2\n"
-       "\tmovss  36(%4), %%xmm3\n"
-       "\tmulss  %%xmm1, %%xmm3\n"
-       "\taddss  36(%0), %%xmm2\n"
-       "\tmovss  40(%3), %%xmm4\n"
-       "\tmulss  %%xmm0, %%xmm4\n"
-       "\tmovss  40(%4), %%xmm5\n"
-       "\tmulss  %%xmm1, %%xmm5\n"
-       "\tsubss  %%xmm3, %%xmm2\n"
-       "\tmovss  %%xmm2, 32(%0)       \n"
-       "\tsubss  %%xmm5, %%xmm4\n"
-       "\tmovss  %%xmm4, 36(%0)\n"
-
-
-
-       "\tmovss 8(%1), %%xmm0\n"
-       "\tmovss (%0), %%xmm1\n"
-       "\taddss %%xmm0, %%xmm1\n"
-       "\tmovss %%xmm1, 8(%2)\n"
-       "\tshufps $0x00, %%xmm0, %%xmm0\n"
-       "\tshufps $0x00, %%xmm1, %%xmm1\n"
-
-       "\tmovaps 4(%3),  %%xmm2\n"
-       "\tmovaps 4(%4),  %%xmm3\n"
-       "\tmulps  %%xmm0, %%xmm2\n"
-       "\tmulps  %%xmm1, %%xmm3\n"
-       "\tmovaps 20(%3), %%xmm4\n"
-       "\tmulps  %%xmm0, %%xmm4\n"
-       "\taddps  4(%0),  %%xmm2\n"
-       "\tmovaps 20(%4), %%xmm5\n"
-       "\tmulps  %%xmm1, %%xmm5\n"
-       "\taddps  20(%0), %%xmm4\n"
-       "\tsubps  %%xmm3, %%xmm2\n"
-       "\tmovups %%xmm2, (%0)\n"
-       "\tsubps  %%xmm5, %%xmm4\n"
-       "\tmovups %%xmm4, 16(%0)\n"
-
-       "\tmovss  36(%3), %%xmm2\n"
-       "\tmulss  %%xmm0, %%xmm2\n"
-       "\tmovss  36(%4), %%xmm3\n"
-       "\tmulss  %%xmm1, %%xmm3\n"
-       "\taddss  36(%0), %%xmm2\n"
-       "\tmovss  40(%3), %%xmm4\n"
-       "\tmulss  %%xmm0, %%xmm4\n"
-       "\tmovss  40(%4), %%xmm5\n"
-       "\tmulss  %%xmm1, %%xmm5\n"
-       "\tsubss  %%xmm3, %%xmm2\n"
-       "\tmovss  %%xmm2, 32(%0)       \n"
-       "\tsubss  %%xmm5, %%xmm4\n"
-       "\tmovss  %%xmm4, 36(%0)\n"
-
-
-
-       "\tmovss 12(%1), %%xmm0\n"
-       "\tmovss (%0), %%xmm1\n"
-       "\taddss %%xmm0, %%xmm1\n"
-       "\tmovss %%xmm1, 12(%2)\n"
-       "\tshufps $0x00, %%xmm0, %%xmm0\n"
-       "\tshufps $0x00, %%xmm1, %%xmm1\n"
-
-       "\tmovaps 4(%3),  %%xmm2\n"
-       "\tmovaps 4(%4),  %%xmm3\n"
-       "\tmulps  %%xmm0, %%xmm2\n"
-       "\tmulps  %%xmm1, %%xmm3\n"
-       "\tmovaps 20(%3), %%xmm4\n"
-       "\tmulps  %%xmm0, %%xmm4\n"
-       "\taddps  4(%0),  %%xmm2\n"
-       "\tmovaps 20(%4), %%xmm5\n"
-       "\tmulps  %%xmm1, %%xmm5\n"
-       "\taddps  20(%0), %%xmm4\n"
-       "\tsubps  %%xmm3, %%xmm2\n"
-       "\tmovups %%xmm2, (%0)\n"
-       "\tsubps  %%xmm5, %%xmm4\n"
-       "\tmovups %%xmm4, 16(%0)\n"
-
-       "\tmovss  36(%3), %%xmm2\n"
-       "\tmulss  %%xmm0, %%xmm2\n"
-       "\tmovss  36(%4), %%xmm3\n"
-       "\tmulss  %%xmm1, %%xmm3\n"
-       "\taddss  36(%0), %%xmm2\n"
-       "\tmovss  40(%3), %%xmm4\n"
-       "\tmulss  %%xmm0, %%xmm4\n"
-       "\tmovss  40(%4), %%xmm5\n"
-       "\tmulss  %%xmm1, %%xmm5\n"
-       "\tsubss  %%xmm3, %%xmm2\n"
-       "\tmovss  %%xmm2, 32(%0)       \n"
-       "\tsubss  %%xmm5, %%xmm4\n"
-       "\tmovss  %%xmm4, 36(%0)\n"
-
-       : : "r" (mem), "r" (x+i), "r" (y+i), "r" (num), "r" (den)
-       : "memory" );
-
-   }
-   for (i=0;i<ord;i++)
-      _mem[i]=mem[i];
-
-}
-
-
-void iir_mem2(float *x, float *_den, float *y, int N, int ord, float *_mem)
-{
-   float  __den[20], __mem[20];
-   float *den, *mem;
-   int i;
-
-   den = (float*)(((int)(__den+4))&0xfffffff0)-1;
-   mem = (float*)(((int)(__mem+4))&0xfffffff0)-1;
-   for (i=0;i<=10;i++)
-      den[i]=0;
-   for (i=0;i<10;i++)
-      mem[i]=0;
-   for (i=0;i<ord+1;i++)
-   {
-      den[i]=_den[i];
-   }
-   for (i=0;i<ord;i++)
-      mem[i]=_mem[i];
-
-   for (i=0;i<N;i++)
-   {
-#if 0
-      y[i] = x[i] + mem[0];
-      for (j=0;j<ord-1;j++)
-      {
-         mem[j] = mem[j+1] - den[j+1]*y[i];
-      }
-      mem[ord-1] = - den[ord]*y[i];
-#else
-      __asm__ __volatile__ 
-      (
-       "\tmovss (%1), %%xmm0\n"
-       "\tmovss (%0), %%xmm1\n"
-       "\taddss %%xmm0, %%xmm1\n"
-       "\tmovss %%xmm1, (%2)\n"
-       "\tshufps $0x00, %%xmm0, %%xmm0\n"
-       "\tshufps $0x00, %%xmm1, %%xmm1\n"
-
-       
-       "\tmovaps 4(%3),  %%xmm2\n"
-       "\tmovaps 20(%3), %%xmm3\n"
-       "\tmulps  %%xmm1, %%xmm2\n"
-       "\tmulps  %%xmm1, %%xmm3\n"
-       "\tmovss  36(%3), %%xmm4\n"
-       "\tmovss  40(%3), %%xmm5\n"
-       "\tmulss  %%xmm1, %%xmm4\n"
-       "\tmulss  %%xmm1, %%xmm5\n"
-       "\tmovaps 4(%0),  %%xmm6\n"
-       "\tsubps  %%xmm2, %%xmm6\n"
-       "\tmovups %%xmm6, (%0)\n"
-       "\tmovaps 20(%0), %%xmm7\n"
-       "\tsubps  %%xmm3, %%xmm7\n"
-       "\tmovups %%xmm7, 16(%0)\n"
-
-
-       "\tmovss  36(%0), %%xmm7\n"
-       "\tsubss  %%xmm4, %%xmm7\n"
-       "\tmovss  %%xmm7, 32(%0)       \n"
-       "\txorps  %%xmm2, %%xmm2\n"
-       "\tsubss  %%xmm5, %%xmm2\n"
-       "\tmovss  %%xmm2, 36(%0)\n"
-
-       : : "r" (mem), "r" (x+i), "r" (y+i), "r" (den)
-       : "memory" );
-#endif
-   }
-   for (i=0;i<ord;i++)
-      _mem[i]=mem[i];
-
-}
-

engine/voice_codecs/speex/source/libspeex/gain_table.c

@@ -1,160 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: gain_table.c
-   Codebook for 3-tap pitch prediction gain (128 entries)
-  
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions are
-   met:
-
-   1. Redistributions of source code must retain the above copyright notice,
-   this list of conditions and the following disclaimer.  
-
-   2. Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-
-   3. The name of the author may not be used to endorse or promote products
-   derived from this software without specific prior written permission.
-
-   THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
-   IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
-   OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
-   DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
-   INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
-   (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
-   SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
-   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
-   STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
-   ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
-   POSSIBILITY OF SUCH DAMAGE.
-*/
-
-signed char gain_cdbk_nb[384] = {
--32,-32,-32,
--28,-67,-5,
--42,-6,-32,
--57,-10,-54,
--16,27,-41,
-19,-19,-40,
--45,24,-21,
--8,-14,-18,
-1,14,-58,
--18,-88,-39,
--38,21,-18,
--19,20,-43,
-10,17,-48,
--52,-58,-13,
--44,-1,-11,
--12,-11,-34,
-14,0,-46,
--37,-35,-34,
--25,44,-30,
-6,-4,-63,
--31,43,-41,
--23,30,-43,
--43,26,-14,
--33,1,-13,
--13,18,-37,
--46,-73,-45,
--36,24,-25,
--36,-11,-20,
--25,12,-18,
--36,-69,-59,
--45,6,8,
--22,-14,-24,
--1,13,-44,
--39,-48,-26,
--32,31,-37,
--33,15,-46,
--24,30,-36,
--41,31,-23,
--50,22,-4,
--22,2,-21,
--17,30,-34,
--7,-60,-28,
--38,42,-28,
--44,-11,21,
--16,8,-44,
--39,-55,-43,
--11,-35,26,
--9,0,-34,
--8,121,-81,
-7,-16,-22,
--37,33,-31,
--27,-7,-36,
--34,70,-57,
--37,-11,-48,
--40,17,-1,
--33,6,-6,
--9,0,-20,
--21,69,-33,
--29,33,-31,
--55,12,-1,
--33,27,-22,
--50,-33,-47,
--50,54,51,
--1,-5,-44,
--4,22,-40,
--39,-66,-25,
--33,1,-26,
--24,-23,-25,
--11,21,-45,
--25,-45,-19,
--43,105,-16,
-5,-21,1,
--16,11,-33,
--13,-99,-4,
--37,33,-15,
--25,37,-63,
--36,24,-31,
--53,-56,-38,
--41,-4,4,
--33,13,-30,
-49,52,-94,
--5,-30,-15,
-1,38,-40,
--23,12,-36,
--17,40,-47,
--37,-41,-39,
--49,34,0,
--18,-7,-4,
--16,17,-27,
-30,5,-62,
-4,48,-68,
--43,11,-11,
--18,19,-15,
--23,-62,-39,
--42,10,-2,
--21,-13,-13,
--9,13,-47,
--23,-62,-24,
--44,60,-21,
--18,-3,-52,
--22,22,-36,
--75,57,16,
--19,3,10,
--29,23,-38,
--5,-62,-51,
--51,40,-18,
--42,13,-24,
--34,14,-20,
--56,-75,-26,
--26,32,15,
--26,17,-29,
--7,28,-52,
--12,-30,5,
--5,-48,-5,
-2,2,-43,
-21,16,16,
--25,-45,-32,
--43,18,-10,
-9,0,-1,
--1,7,-30,
-19,-48,-4,
--28,25,-29,
--22,0,-31,
--32,17,-10,
--64,-41,-62,
--52,15,16,
--30,-22,-32,
--7,9,-38};

engine/voice_codecs/speex/source/libspeex/gain_table_lbr.c

@@ -1,64 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: gain_table_lbr.c
-   Codebook for 3-tap pitch prediction gain (32 entries)
-  
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions are
-   met:
-
-   1. Redistributions of source code must retain the above copyright notice,
-   this list of conditions and the following disclaimer.  
-
-   2. Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-
-   3. The name of the author may not be used to endorse or promote products
-   derived from this software without specific prior written permission.
-
-   THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
-   IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
-   OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
-   DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
-   INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
-   (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
-   SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
-   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
-   STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
-   ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
-   POSSIBILITY OF SUCH DAMAGE.
-*/
-
-signed char gain_cdbk_lbr[96] = {
--32,-32,-32,
--31,-58,-16,
--41,-24,-43,
--56,-22,-55,
--13,33,-41,
--4,-39,-9,
--41,15,-12,
--8,-15,-12,
-1,2,-44,
--22,-66,-42,
--38,28,-23,
--21,14,-37,
-0,21,-50,
--53,-71,-27,
--37,-1,-19,
--19,-5,-28,
-6,65,-44,
--33,-48,-33,
--40,57,-14,
--17,4,-45,
--31,38,-33,
--23,28,-40,
--43,29,-12,
--34,13,-23,
--16,15,-27,
--14,-82,-15,
--31,25,-32,
--21,5,-5,
--47,-63,-51,
--46,12,3,
--28,-17,-29,
--10,14,-40};

engine/voice_codecs/speex/source/libspeex/hexc_10_32_table.c

@@ -1,66 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: hexc_10_32_table.c
-   Codebook for high-band excitation in SB-CELP mode (4000 bps)
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-signed char hexc_10_32_table[320] = {
--3, -2, -1, 0, -4, 5, 35, -40, -9, 13, 
--44, 5, -27, -1, -7, 6, -11, 7, -8, 7, 
-19, -14, 15, -4, 9, -10, 10, -8, 10, -9, 
--1, 1, 0, 0, 2, 5, -18, 22, -53, 50, 
-1, -23, 50, -36, 15, 3, -13, 14, -10, 6, 
-1, 5, -3, 4, -2, 5, -32, 25, 5, -2, 
--1, -4, 1, 11, -29, 26, -6, -15, 30, -18, 
-0, 15, -17, 40, -41, 3, 9, -2, -2, 3, 
--3, -1, -5, 2, 21, -6, -16, -21, 23, 2, 
-60, 15, 16, -16, -9, 14, 9, -1, 7, -9, 
-0, 1, 1, 0, -1, -6, 17, -28, 54, -45, 
--1, 1, -1, -6, -6, 2, 11, 26, -29, -2, 
-46, -21, 34, 12, -23, 32, -23, 16, -10, 3, 
-66, 19, -20, 24, 7, 11, -3, 0, -3, -1, 
--50, -46, 2, -18, -3, 4, -1, -2, 3, -3, 
--19, 41, -36, 9, 11, -24, 21, -16, 9, -3, 
--25, -3, 10, 18, -9, -2, -5, -1, -5, 6, 
--4, -3, 2, -26, 21, -19, 35, -15, 7, -13, 
-17, -19, 39, -43, 48, -31, 16, -9, 7, -2, 
--5, 3, -4, 9, -19, 27, -55, 63, -35, 10, 
-26, -44, -2, 9, 4, 1, -6, 8, -9, 5, 
--8, -1, -3, -16, 45, -42, 5, 15, -16, 10, 
-0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 
--16, 24, -55, 47, -38, 27, -19, 7, -3, 1, 
-16, 27, 20, -19, 18, 5, -7, 1, -5, 2, 
--6, 8, -22, 0, -3, -3, 8, -1, 7, -8, 
-1, -3, 5, 0, 17, -48, 58, -52, 29, -7, 
--2, 3, -10, 6, -26, 58, -31, 1, -6, 3, 
-93, -29, 39, 3, 17, 5, 6, -1, -1, -1, 
-27, 13, 10, 19, -7, -34, 12, 10, -4, 9, 
--76, 9, 8, -28, -2, -11, 2, -1, 3, 1, 
--83, 38, -39, 4, -16, -6, -2, -5, 5, -2, 
-};

engine/voice_codecs/speex/source/libspeex/hexc_table.c

@@ -1,162 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: hexc_table.c
-   Codebook for high-band excitation in SB-CELP mode (8000 bps with sign)
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-signed char hexc_table[1024] = {
--24, 21, -20, 5, -5, -7, 14, -10, 
-2, -27, 16, -20, 0, -32, 26, 19, 
-8, -11, -41, 31, 28, -27, -32, 34, 
-42, 34, -17, 22, -10, 13, -29, 18, 
--12, -26, -24, 11, 22, 5, -5, -5, 
-54, -68, -43, 57, -25, 24, 4, 4, 
-26, -8, -12, -17, 54, 30, -45, 1, 
-10, -15, 18, -41, 11, 68, -67, 37, 
--16, -24, -16, 38, -22, 6, -29, 30, 
-66, -27, 5, 7, -16, 13, 2, -12, 
--7, -3, -20, 36, 4, -28, 9, 3, 
-32, 48, 26, 39, 3, 0, 7, -21, 
--13, 5, -82, -7, 73, -20, 34, -9, 
--5, 1, -1, 10, -5, -10, -1, 9, 
-1, -9, 10, 0, -14, 11, -1, -2, 
--1, 11, 20, 96, -81, -22, -12, -9, 
--58, 9, 24, -30, 26, -35, 27, -12, 
-13, -18, 56, -59, 15, -7, 23, -15, 
--1, 6, -25, 14, -22, -20, 47, -11, 
-16, 2, 38, -23, -19, -30, -9, 40, 
--11, 5, 4, -6, 8, 26, -21, -11, 
-131, 4, 1, 6, -9, 2, -7, -2, 
--3, 7, -5, 10, -19, 7, -106, 91, 
--3, 9, -4, 21, -8, 26, -80, 8, 
-1, -2, -10, -17, -17, -27, 32, 71, 
-6, -29, 11, -23, 54, -38, 29, -22, 
-39, 87, -31, -12, -20, 3, -2, -2, 
-2, 20, 0, -1, -35, 27, 9, -6, 
--12, 3, -12, -6, 13, 1, 14, -22, 
--59, -15, -17, -25, 13, -7, 7, 3, 
-0, 1, -7, 6, -3, 61, -37, -23, 
--23, -29, 38, -31, 27, 1, -8, 2, 
--27, 23, -26, 36, -34, 5, 24, -24, 
--6, 7, 3, -59, 78, -62, 44, -16, 
-1, 6, 0, 17, 8, 45, 0, -110, 
-6, 14, -2, 32, -77, -56, 62, -3, 
-3, -13, 4, -16, 102, -15, -36, -1, 
-9, -113, 6, 23, 0, 9, 9, 5, 
--8, -1, -14, 5, -12, 121, -53, -27, 
--8, -9, 22, -13, 3, 2, -3, 1, 
--2, -71, 95, 38, -19, 15, -16, -5, 
-71, 10, 2, -32, -13, -5, 15, -1, 
--2, -14, -85, 30, 29, 6, 3, 2, 
-0, 0, 0, 0, 0, 0, 0, 0, 
-2, -65, -56, -9, 18, 18, 23, -14, 
--2, 0, 12, -29, 26, -12, 1, 2, 
--12, -64, 90, -6, 4, 1, 5, -5, 
--110, -3, -31, 22, -29, 9, 0, 8, 
--40, -5, 21, -5, -5, 13, 10, -18, 
-40, 1, 35, -20, 30, -28, 11, -6, 
-19, 7, 14, 18, -64, 9, -6, 16, 
-51, 68, 8, 16, 12, -8, 0, -9, 
-20, -22, 25, 7, -4, -13, 41, -35, 
-93, -18, -54, 11, -1, 1, -9, 4, 
--66, 66, -31, 20, -22, 25, -23, 11, 
-10, 9, 19, 15, 11, -5, -31, -10, 
--23, -28, -6, -6, -3, -4, 5, 3, 
--28, 22, -11, -42, 25, -25, -16, 41, 
-34, 47, -6, 2, 42, -19, -22, 5, 
--39, 32, 6, -35, 22, 17, -30, 8, 
--26, -11, -11, 3, -12, 33, 33, -37, 
-21, -1, 6, -4, 3, 0, -5, 5, 
-12, -12, 57, 27, -61, -3, 20, -17, 
-2, 0, 4, 0, -2, -33, -58, 81, 
--23, 39, -10, -5, 2, 6, -7, 5, 
-4, -3, -2, -13, -23, -72, 107, 15, 
--5, 0, -7, -3, -6, 5, -4, 15, 
-47, 12, -31, 25, -16, 8, 22, -25, 
--62, -56, -18, 14, 28, 12, 2, -11, 
-74, -66, 41, -20, -7, 16, -20, 16, 
--8, 0, -16, 4, -19, 92, 12, -59, 
--14, -39, 49, -25, -16, 23, -27, 19, 
--3, -33, 19, 85, -29, 6, -7, -10, 
-16, -7, -12, 1, -6, 2, 4, -2, 
-64, 10, -25, 41, -2, -31, 15, 0, 
-110, 50, 69, 35, 28, 19, -10, 2, 
--43, -49, -56, -15, -16, 10, 3, 12, 
--1, -8, 1, 26, -12, -1, 7, -11, 
--27, 41, 25, 1, -11, -18, 22, -7, 
--1, -47, -8, 23, -3, -17, -7, 18, 
--125, 59, -5, 3, 18, 1, 2, 3, 
-27, -35, 65, -53, 50, -46, 37, -21, 
--28, 7, 14, -37, -5, -5, 12, 5, 
--8, 78, -19, 21, -6, -16, 8, -7, 
-5, 2, 7, 2, 10, -6, 12, -60, 
-44, 11, -36, -32, 31, 0, 2, -2, 
-2, 1, -3, 7, -10, 17, -21, 10, 
-6, -2, 19, -2, 59, -38, -86, 38, 
-8, -41, -30, -45, -33, 7, 15, 28, 
-29, -7, 24, -40, 7, 7, 5, -2, 
-9, 24, -23, -18, 6, -29, 30, 2, 
-28, 49, -11, -46, 10, 43, -13, -9, 
--1, -3, -7, -7, -17, -6, 97, -33, 
--21, 3, 5, 1, 12, -43, -8, 28, 
-7, -43, -7, 17, -20, 19, -1, 2, 
--13, 9, 54, 34, 9, -28, -11, -9, 
--17, 110, -59, 44, -26, 0, 3, -12, 
--47, 73, -34, -43, 38, -33, 16, -5, 
--46, -4, -6, -2, -25, 19, -29, 28, 
--13, 5, 14, 27, -40, -43, 4, 32, 
--13, -2, -35, -4, 112, -42, 9, -12, 
-37, -28, 17, 14, -19, 35, -39, 23, 
-3, -14, -1, -57, -5, 94, -9, 3, 
--39, 5, 30, -10, -32, 42, -13, -14, 
--97, -63, 30, -9, 1, -7, 12, 5, 
-20, 17, -9, -36, -30, 25, 47, -9, 
--15, 12, -22, 98, -8, -50, 15, -27, 
-21, -16, -11, 2, 12, -10, 10, -3, 
-33, 36, -96, 0, -17, 31, -9, 9, 
-3, -20, 13, -11, 8, -4, 10, -10, 
-9, 1, 112, -70, -27, 5, -21, 2, 
--57, -3, -29, 10, 19, -21, 21, -10, 
--66, -3, 91, -35, 30, -12, 0, -7, 
-59, -28, 26, 2, 14, -18, 1, 1, 
-11, 17, 20, -54, -59, 27, 4, 29, 
-32, 5, 19, 12, -4, 1, 7, -10, 
-5, -2, 10, 0, 23, -5, 28, -104, 
-46, 11, 16, 3, 29, 1, -8, -14, 
-1, 7, -50, 88, -62, 26, 8, -17, 
--14, 50, 0, 32, -12, -3, -27, 18, 
--8, -5, 8, 3, -20, -11, 37, -12, 
-9, 33, 46, -101, -1, -4, 1, 6, 
--1, 28, -42, -15, 16, 5, -1, -2, 
--55, 85, 38, -9, -4, 11, -2, -9, 
--6, 3, -20, -10, -77, 89, 24, -3, 
--104, -57, -26, -31, -20, -6, -9, 14, 
-20, -23, 46, -15, -31, 28, 1, -15, 
--2, 6, -2, 31, 45, -76, 23, -25, 
-};

engine/voice_codecs/speex/source/libspeex/high_lsp_tables.c

@@ -1,163 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: high_lsp_tables.c
-   Codebooks for high-band LSPs in SB-CELP mode
-  
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions are
-   met:
-
-   1. Redistributions of source code must retain the above copyright notice,
-   this list of conditions and the following disclaimer.  
-
-   2. Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-
-   3. The name of the author may not be used to endorse or promote products
-   derived from this software without specific prior written permission.
-
-   THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
-   IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
-   OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
-   DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
-   INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
-   (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
-   SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
-   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
-   STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
-   ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
-   POSSIBILITY OF SUCH DAMAGE.
-*/
- 
-signed char high_lsp_cdbk[512]={
-39,12,-14,-20,-29,-61,-67,-76,
--32,-71,-67,68,77,46,34,5,
--13,-48,-46,-72,-81,-84,-60,-58,
--40,-28,82,93,68,45,29,3,
--19,-47,-28,-43,-35,-30,-8,-13,
--39,-91,-91,-123,-96,10,10,-6,
--18,-55,-60,-91,-56,-36,-27,-16,
--48,-75,40,28,-10,-28,35,9,
-37,19,1,-20,-31,-41,-18,-25,
--35,-68,-80,45,27,-1,47,13,
-0,-29,-35,-57,-50,-79,-73,-38,
--19,5,35,14,-10,-23,16,-8,
-5,-24,-40,-62,-23,-27,-22,-16,
--18,-46,-72,-77,43,21,33,1,
--80,-70,-70,-64,-56,-52,-39,-33,
--31,-38,-19,-19,-15,32,33,-2,
-7,-15,-15,-24,-23,-33,-41,-56,
--24,-57,5,89,64,41,27,5,
--9,-47,-60,-97,-97,-124,-20,-9,
--44,-73,31,29,-4,64,48,7,
--35,-57,0,-3,-26,-47,-3,-6,
--40,-76,-79,-48,12,81,55,10,
-9,-24,-43,-73,-57,-69,16,5,
--28,-53,18,29,20,0,-4,-11,
-6,-13,23,7,-17,-35,-37,-37,
--30,-68,-63,6,24,-9,-14,3,
-21,-13,-27,-57,-49,-80,-24,-41,
--5,-16,-5,1,45,25,12,-7,
-3,-15,-6,-16,-15,-8,6,-13,
--42,-81,-80,-87,14,1,-10,-3,
--43,-69,-46,-24,-28,-29,36,6,
--43,-56,-12,12,54,79,43,9,
-54,22,2,8,-12,-43,-46,-52,
--38,-69,-89,-5,75,38,33,5,
--13,-53,-62,-87,-89,-113,-99,-55,
--34,-37,62,55,33,16,21,-2,
--17,-46,-29,-38,-38,-48,-39,-42,
--36,-75,-72,-88,-48,-30,21,2,
--15,-57,-64,-98,-84,-76,25,1,
--46,-80,-12,18,-7,3,34,6,
-38,31,23,4,-1,20,14,-15,
--43,-78,-91,-24,14,-3,54,16,
-0,-27,-28,-44,-56,-83,-92,-89,
--3,34,56,41,36,22,20,-8,
--7,-35,-42,-62,-49,3,12,-10,
--50,-87,-96,-66,92,70,38,9,
--70,-71,-62,-42,-39,-43,-11,-7,
--50,-79,-58,-50,-31,32,31,-6,
--4,-25,7,-17,-38,-70,-58,-27,
--43,-83,-28,59,36,20,31,2,
--27,-71,-80,-109,-98,-75,-33,-32,
--31,-2,33,15,-6,43,33,-5,
-0,-22,-10,-27,-34,-49,-11,-20,
--41,-91,-100,-121,-39,57,41,10,
--19,-50,-38,-59,-60,-70,-18,-20,
--8,-31,-8,-15,1,-14,-26,-25,
-33,21,32,17,1,-19,-19,-26,
--58,-81,-35,-22,45,30,11,-11,
-3,-26,-48,-87,-67,-83,-58,3,
--1,-26,-20,44,10,25,39,5,
--9,-35,-27,-38,7,10,4,-9,
--42,-85,-102,-127,52,44,28,10,
--47,-61,-40,-39,-17,-1,-10,-33,
--42,-74,-48,21,-4,70,52,10};
-
-
-signed char high_lsp_cdbk2[512]={
--36,-62,6,-9,-10,-14,-56,23,
-1,-26,23,-48,-17,12,8,-7,
-23,29,-36,-28,-6,-29,-17,-5,
-40,23,10,10,-46,-13,36,6,
-4,-30,-29,62,32,-32,-1,22,
--14,1,-4,-22,-45,2,54,4,
--30,-57,-59,-12,27,-3,-31,8,
--9,5,10,-14,32,66,19,9,
-2,-25,-37,23,-15,18,-38,-31,
-5,-9,-21,15,0,22,62,30,
-15,-12,-14,-46,77,21,33,3,
-34,29,-19,50,2,11,9,-38,
--12,-37,62,1,-15,54,32,6,
-2,-24,20,35,-21,2,19,24,
--13,55,4,9,39,-19,30,-1,
--21,73,54,33,8,18,3,15,
-6,-19,-47,6,-3,-48,-50,1,
-26,20,8,-23,-50,65,-14,-55,
--17,-31,-37,-28,53,-1,-17,-53,
-1,57,11,-8,-25,-30,-37,64,
-5,-52,-45,15,23,31,15,14,
--25,24,33,-2,-44,-56,-18,6,
--21,-43,4,-12,17,-37,20,-10,
-34,15,2,15,55,21,-11,-31,
--6,46,25,16,-9,-25,-8,-62,
-28,17,20,-32,-29,26,30,25,
--19,2,-16,-17,26,-51,2,50,
-42,19,-66,23,29,-2,3,19,
--19,-37,32,15,6,30,-34,13,
-11,-5,40,31,10,-42,4,-9,
-26,-9,-70,17,-2,-23,20,-22,
--55,51,-24,-31,22,-22,15,-13,
-3,-10,-28,-16,56,4,-63,11,
--18,-15,-18,-38,-35,16,-7,34,
--1,-21,-49,-47,9,-37,7,8,
-69,55,20,6,-33,-45,-10,-9,
-6,-9,12,71,15,-3,-42,-7,
--24,32,-35,-2,-42,-17,-5,0,
--2,-33,-54,13,-12,-34,47,23,
-19,55,7,-8,74,31,14,16,
--23,-26,19,12,-18,-49,-28,-31,
--20,2,-14,-20,-47,78,40,13,
--23,-11,21,-6,18,1,47,5,
-38,35,32,46,22,8,13,16,
--14,18,51,19,40,39,11,-26,
--1,-17,47,2,-53,-15,31,-22,
-38,21,-15,-16,5,-33,53,15,
--38,86,11,-3,-24,49,13,-4,
--11,-18,28,20,-12,-27,-26,35,
--25,-35,-3,-20,-61,30,10,-55,
--12,-22,-52,-54,-14,19,-32,-12,
-45,15,-8,-48,-9,11,-32,8,
--16,-34,-13,51,18,38,-2,-32,
--17,22,-2,-18,-28,-70,59,27,
--28,-19,-10,-20,-9,-9,-8,-21,
-21,-8,35,-2,45,-3,-9,12,
-0,30,7,-39,43,27,-38,-91,
-30,26,19,-55,-4,63,14,-17,
-13,9,13,2,7,4,6,61,
-72,-1,-17,29,-1,-22,-17,8,
--28,-37,63,44,41,3,2,14,
-9,-6,75,-8,-7,-12,-15,-12,
-13,9,-4,30,-22,-65,15,0,
--45,4,-4,1,5,22,11,23};

engine/voice_codecs/speex/source/libspeex/lpc.c

@@ -1,119 +0,0 @@
-/*
-  Copyright 1992, 1993, 1994 by Jutta Degener and Carsten Bormann,
-  Technische Universitaet Berlin
-
-  Any use of this software is permitted provided that this notice is not
-  removed and that neither the authors nor the Technische Universitaet Berlin
-  are deemed to have made any representations as to the suitability of this
-  software for any purpose nor are held responsible for any defects of
-  this software.  THERE IS ABSOLUTELY NO WARRANTY FOR THIS SOFTWARE.
-
-  As a matter of courtesy, the authors request to be informed about uses
-  this software has found, about bugs in this software, and about any
-  improvements that may be of general interest.
-
-  Berlin, 28.11.1994
-  Jutta Degener
-  Carsten Bormann
-
-
-   Code slightly modified by Jean-Marc Valin
-
-   Speex License:
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-
-
-/* LPC- and Reflection Coefficients
- *
- * The next two functions calculate linear prediction coefficients
- * and/or the related reflection coefficients from the first P_MAX+1
- * values of the autocorrelation function.
- */
-
-/* Invented by N. Levinson in 1947, modified by J. Durbin in 1959.
- */
-
-#include "lpc.h"
-
-float                      /* returns minimum mean square error    */
-wld(
-    float       * lpc, /*      [0...p-1] LPC coefficients      */
-    const float * ac,  /*  in: [0...p] autocorrelation values  */
-    float       * ref, /* out: [0...p-1] reflection coef's     */
-    int p
-    )
-{
-   int i, j;  float r, error = ac[0];
-
-   if (ac[0] == 0) {
-      for (i = 0; i < p; i++) ref[i] = 0; return 0; }
-
-   for (i = 0; i < p; i++) {
-
-      /* Sum up this iteration's reflection coefficient.
-       */
-      r = -ac[i + 1];
-      for (j = 0; j < i; j++) r -= lpc[j] * ac[i - j];
-      ref[i] = r /= error;
-
-      /*  Update LPC coefficients and total error.
-       */
-      lpc[i] = r;
-      for (j = 0; j < i/2; j++) {
-         float tmp  = lpc[j];
-         lpc[j]     += r * lpc[i-1-j];
-         lpc[i-1-j] += r * tmp;
-      }
-      if (i % 2) lpc[j] += lpc[j] * r;
-
-      error *= 1.0 - r * r;
-   }
-   return error;
-}
-
-
-/* Compute the autocorrelation
- *                      ,--,
- *              ac(i) = >  x(n) * x(n-i)  for all n
- *                      `--'
- * for lags between 0 and lag-1, and x == 0 outside 0...n-1
- */
-void _spx_autocorr(
-              const float * x,   /*  in: [0...n-1] samples x   */
-              float *ac,   /* out: [0...lag-1] ac values */
-              int lag, int   n)
-{
-   float d; int i;
-   while (lag--) {
-      for (i = lag, d = 0; i < n; i++) d += x[i] * x[i-lag];
-      ac[lag] = d;
-   }
-}

engine/voice_codecs/speex/source/libspeex/lpc.h

@@ -1,50 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: lpc.h
-   Functions for LPC (Linear Prediction Coefficients) analysis
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#ifndef LPC_H
-#define LPC_H
-
-void _spx_autocorr(
-              const float * x,   /*  in: [0...n-1] samples x   */
-              float *ac,   /* out: [0...lag-1] ac values */
-              int lag, int   n);
-
-float                      /* returns minimum mean square error    */
-wld(
-    float       * lpc, /*      [0...p-1] LPC coefficients      */
-    const float * ac,  /*  in: [0...p] autocorrelation values  */
-    float       * ref, /* out: [0...p-1] reflection coef's     */
-    int p
-    );
-
-
-#endif

engine/voice_codecs/speex/source/libspeex/lsp.c

@@ -1,328 +0,0 @@
-/*---------------------------------------------------------------------------*\
-Original copyright
-	FILE........: AKSLSPD.C
-	TYPE........: Turbo C
-	COMPANY.....: Voicetronix
-	AUTHOR......: David Rowe
-	DATE CREATED: 24/2/93
-
-Modified by Jean-Marc Valin
-
-   This file contains functions for converting Linear Prediction
-   Coefficients (LPC) to Line Spectral Pair (LSP) and back. Note that the
-   LSP coefficients are not in radians format but in the x domain of the
-   unit circle.
-
-   Speex License:
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#include <math.h>
-#include "lsp.h"
-#include "stack_alloc.h"
-
-
-#ifndef M_PI
-#define M_PI           3.14159265358979323846  /* pi */
-#endif
-
-#ifndef NULL
-#define NULL 0
-#endif
-
-/*---------------------------------------------------------------------------*\
-
-	FUNCTION....: cheb_poly_eva()
-
-	AUTHOR......: David Rowe
-	DATE CREATED: 24/2/93
-
-    This function evaluates a series of Chebyshev polynomials
-
-\*---------------------------------------------------------------------------*/
-
-
-
-static float cheb_poly_eva(float *coef,float x,int m,char *stack)
-/*  float coef[]  	coefficients of the polynomial to be evaluated 	*/
-/*  float x   		the point where polynomial is to be evaluated 	*/
-/*  int m 		order of the polynomial 			*/
-{
-    int i;
-    float *T,sum;
-    int m2=m>>1;
-
-    /* Allocate memory for Chebyshev series formulation */
-    T=PUSH(stack, m2+1, float);
-
-    /* Initialise values */
-    T[0]=1;
-    T[1]=x;
-
-    /* Evaluate Chebyshev series formulation using iterative approach  */
-    /* Evaluate polynomial and return value also free memory space */
-    sum = coef[m2] + coef[m2-1]*x;
-    x *= 2;
-    for(i=2;i<=m2;i++)
-    {
-       T[i] = x*T[i-1] - T[i-2];
-       sum += coef[m2-i] * T[i];
-    }
-    
-    return sum;
-}
-
-
-/*---------------------------------------------------------------------------*\
-
-	FUNCTION....: lpc_to_lsp()
-
-	AUTHOR......: David Rowe
-	DATE CREATED: 24/2/93
-
-    This function converts LPC coefficients to LSP
-    coefficients.
-
-\*---------------------------------------------------------------------------*/
-
-
-int lpc_to_lsp (float *a,int lpcrdr,float *freq,int nb,float delta, char *stack)
-/*  float *a 		     	lpc coefficients			*/
-/*  int lpcrdr			order of LPC coefficients (10) 		*/
-/*  float *freq 	      	LSP frequencies in the x domain       	*/
-/*  int nb			number of sub-intervals (4) 		*/
-/*  float delta			grid spacing interval (0.02) 		*/
-
-
-{
-
-    float psuml,psumr,psumm,temp_xr,xl,xr,xm=0;
-    float temp_psumr/*,temp_qsumr*/;
-    int i,j,m,flag,k;
-    float *Q;                 	/* ptrs for memory allocation 		*/
-    float *P;
-    float *px;                	/* ptrs of respective P'(z) & Q'(z)	*/
-    float *qx;
-    float *p;
-    float *q;
-    float *pt;                	/* ptr used for cheb_poly_eval()
-				whether P' or Q' 			*/
-    int roots=0;              	/* DR 8/2/94: number of roots found 	*/
-    flag = 1;                	/*  program is searching for a root when,
-				1 else has found one 			*/
-    m = lpcrdr/2;            	/* order of P'(z) & Q'(z) polynomials 	*/
-
-
-    /* Allocate memory space for polynomials */
-    Q = PUSH(stack, (m+1), float);
-    P = PUSH(stack, (m+1), float);
-
-    /* determine P'(z)'s and Q'(z)'s coefficients where
-      P'(z) = P(z)/(1 + z^(-1)) and Q'(z) = Q(z)/(1-z^(-1)) */
-
-    px = P;                      /* initialise ptrs 			*/
-    qx = Q;
-    p = px;
-    q = qx;
-    *px++ = 1.0;
-    *qx++ = 1.0;
-    for(i=1;i<=m;i++){
-	*px++ = a[i]+a[lpcrdr+1-i]-*p++;
-	*qx++ = a[i]-a[lpcrdr+1-i]+*q++;
-    }
-    px = P;
-    qx = Q;
-    for(i=0;i<m;i++){
-	*px = 2**px;
-	*qx = 2**qx;
-	 px++;
-	 qx++;
-    }
-    px = P;             	/* re-initialise ptrs 			*/
-    qx = Q;
-
-    /* Search for a zero in P'(z) polynomial first and then alternate to Q'(z).
-    Keep alternating between the two polynomials as each zero is found 	*/
-
-    xr = 0;             	/* initialise xr to zero 		*/
-    xl = 1.0;               	/* start at point xl = 1 		*/
-
-
-    for(j=0;j<lpcrdr;j++){
-	if(j%2)            	/* determines whether P' or Q' is eval. */
-	    pt = qx;
-	else
-	    pt = px;
-
-	psuml = cheb_poly_eva(pt,xl,lpcrdr,stack);	/* evals poly. at xl 	*/
-	flag = 1;
-	while(flag && (xr >= -1.0)){
-           float dd;
-           /* Modified by JMV to provide smaller steps around x=+-1 */
-           dd=(delta*(1-.9*xl*xl));
-           if (fabs(psuml)<.2)
-              dd *= .5;
-
-           xr = xl - dd;                        	/* interval spacing 	*/
-	    psumr = cheb_poly_eva(pt,xr,lpcrdr,stack);/* poly(xl-delta_x) 	*/
-	    temp_psumr = psumr;
-	    temp_xr = xr;
-
-    /* if no sign change increment xr and re-evaluate poly(xr). Repeat til
-    sign change.
-    if a sign change has occurred the interval is bisected and then
-    checked again for a sign change which determines in which
-    interval the zero lies in.
-    If there is no sign change between poly(xm) and poly(xl) set interval
-    between xm and xr else set interval between xl and xr and repeat till
-    root is located within the specified limits 			*/
-
-	    if((psumr*psuml)<0.0){
-		roots++;
-
-		psumm=psuml;
-		for(k=0;k<=nb;k++){
-		    xm = (xl+xr)/2;        	/* bisect the interval 	*/
-		    psumm=cheb_poly_eva(pt,xm,lpcrdr,stack);
-		    if(psumm*psuml>0.){
-			psuml=psumm;
-			xl=xm;
-		    }
-		    else{
-			psumr=psumm;
-			xr=xm;
-		    }
-		}
-
-	       /* once zero is found, reset initial interval to xr 	*/
-	       freq[j] = (xm);
-	       xl = xm;
-	       flag = 0;       		/* reset flag for next search 	*/
-	    }
-	    else{
-		psuml=temp_psumr;
-		xl=temp_xr;
-	    }
-	}
-    }
-    return(roots);
-}
-
-
-/*---------------------------------------------------------------------------*\
-
-	FUNCTION....: lsp_to_lpc()
-
-	AUTHOR......: David Rowe
-	DATE CREATED: 24/2/93
-
-    lsp_to_lpc: This function converts LSP coefficients to LPC
-    coefficients.
-
-\*---------------------------------------------------------------------------*/
-
-
-void lsp_to_lpc(float *freq,float *ak,int lpcrdr, char *stack)
-/*  float *freq 	array of LSP frequencies in the x domain	*/
-/*  float *ak 		array of LPC coefficients 			*/
-/*  int lpcrdr  	order of LPC coefficients 			*/
-
-
-{
-    int i,j;
-    float xout1,xout2,xin1,xin2;
-    float *Wp;
-    float *pw,*n1,*n2,*n3,*n4=NULL;
-    int m = lpcrdr/2;
-
-    Wp = PUSH(stack, 4*m+2, float);
-    pw = Wp;
-
-    /* initialise contents of array */
-
-    for(i=0;i<=4*m+1;i++){       	/* set contents of buffer to 0 */
-	*pw++ = 0.0;
-    }
-
-    /* Set pointers up */
-
-    pw = Wp;
-    xin1 = 1.0;
-    xin2 = 1.0;
-
-    /* reconstruct P(z) and Q(z) by  cascading second order
-      polynomials in form 1 - 2xz(-1) +z(-2), where x is the
-      LSP coefficient */
-
-    for(j=0;j<=lpcrdr;j++){
-       int i2=0;
-	for(i=0;i<m;i++,i2+=2){
-	    n1 = pw+(i*4);
-	    n2 = n1 + 1;
-	    n3 = n2 + 1;
-	    n4 = n3 + 1;
-	    xout1 = xin1 - 2*(freq[i2]) * *n1 + *n2;
-	    xout2 = xin2 - 2*(freq[i2+1]) * *n3 + *n4;
-	    *n2 = *n1;
-	    *n4 = *n3;
-	    *n1 = xin1;
-	    *n3 = xin2;
-	    xin1 = xout1;
-	    xin2 = xout2;
-	}
-	xout1 = xin1 + *(n4+1);
-	xout2 = xin2 - *(n4+2);
-	ak[j] = (xout1 + xout2)*0.5;
-	*(n4+1) = xin1;
-	*(n4+2) = xin2;
-
-	xin1 = 0.0;
-	xin2 = 0.0;
-    }
-
-}
-
-/*Added by JMV
-  Makes sure the LSPs are stable*/
-void lsp_enforce_margin(float *lsp, int len, float margin)
-{
-   int i;
-   if (lsp[0]<margin)
-      lsp[0]=margin;
-   if (lsp[len-1]>M_PI-margin)
-      lsp[len-1]=M_PI-margin;
-   for (i=1;i<len-1;i++)
-   {
-      if (lsp[i]<lsp[i-1]+margin)
-         lsp[i]=lsp[i-1]+margin;
-
-      if (lsp[i]>lsp[i+1]-margin)
-         lsp[i]= .5* (lsp[i] + lsp[i+1]-margin);
-   }
-}

engine/voice_codecs/speex/source/libspeex/lsp.h

@@ -1,57 +0,0 @@
-/*---------------------------------------------------------------------------*\
-Original Copyright
-	FILE........: AK2LSPD.H
-	TYPE........: Turbo C header file
-	COMPANY.....: Voicetronix
-	AUTHOR......: James Whitehall
-	DATE CREATED: 21/11/95
-
-Modified by Jean-Marc Valin
-
-    This file contains functions for converting Linear Prediction
-    Coefficients (LPC) to Line Spectral Pair (LSP) and back. Note that the
-    LSP coefficients are not in radians format but in the x domain of the
-    unit circle.
-
-\*---------------------------------------------------------------------------*/
-/* Speex License:
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#ifndef __AK2LSPD__
-#define __AK2LSPD__
-
-int lpc_to_lsp (float *a, int lpcrdr, float *freq, int nb, float delta, char *stack);
-void lsp_to_lpc(float *freq, float *ak, int lpcrdr, char *stack);
-
-/*Added by JMV*/
-void lsp_enforce_margin(float *lsp, int len, float margin);
-
-
-#endif	/* __AK2LSPD__ */

engine/voice_codecs/speex/source/libspeex/lsp_tables_nb.c

@@ -1,360 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: lsp_tables_nb.c
-   Codebooks for LSPs in narrowband CELP mode
-  
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions are
-   met:
-
-   1. Redistributions of source code must retain the above copyright notice,
-   this list of conditions and the following disclaimer.  
-
-   2. Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-
-   3. The name of the author may not be used to endorse or promote products
-   derived from this software without specific prior written permission.
-
-   THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
-   IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
-   OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
-   DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
-   INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
-   (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
-   SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
-   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
-   STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
-   ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
-   POSSIBILITY OF SUCH DAMAGE.
-*/
-
-signed char cdbk_nb[640]={
-30,19,38,34,40,32,46,43,58,43,
-5,-18,-25,-40,-33,-55,-52,20,34,28,
--20,-63,-97,-92,61,53,47,49,53,75,
--14,-53,-77,-79,0,-3,-5,19,22,26,
--9,-53,-55,66,90,72,85,68,74,52,
--4,-41,-58,-31,-18,-31,27,32,30,18,
-24,3,8,5,-12,-3,26,28,74,63,
--2,-39,-67,-77,-106,-74,59,59,73,65,
-44,40,71,72,82,83,98,88,89,60,
--6,-31,-47,-48,-13,-39,-9,7,2,79,
--1,-39,-60,-17,87,81,65,50,45,19,
--21,-67,-91,-87,-41,-50,7,18,39,74,
-10,-31,-28,39,24,13,23,5,56,45,
-29,10,-5,-13,-11,-35,-18,-8,-10,-8,
--25,-71,-77,-21,2,16,50,63,87,87,
-5,-32,-40,-51,-68,0,12,6,54,34,
-5,-12,32,52,68,64,69,59,65,45,
-14,-16,-31,-40,-65,-67,41,49,47,37,
--11,-52,-75,-84,-4,57,48,42,42,33,
--11,-51,-68,-6,13,0,8,-8,26,32,
--23,-53,0,36,56,76,97,105,111,97,
--1,-28,-39,-40,-43,-54,-44,-40,-18,35,
-16,-20,-19,-28,-42,29,47,38,74,45,
-3,-29,-48,-62,-80,-104,-33,56,59,59,
-10,17,46,72,84,101,117,123,123,106,
--7,-33,-49,-51,-70,-67,-27,-31,70,67,
--16,-62,-85,-20,82,71,86,80,85,74,
--19,-58,-75,-45,-29,-33,-18,-25,45,57,
--12,-42,-5,12,28,36,52,64,81,82,
-13,-9,-27,-28,22,3,2,22,26,6,
--6,-44,-51,2,15,10,48,43,49,34,
--19,-62,-84,-89,-102,-24,8,17,61,68,
-39,24,23,19,16,-5,12,15,27,15,
--8,-44,-49,-60,-18,-32,-28,52,54,62,
--8,-48,-77,-70,66,101,83,63,61,37,
--12,-50,-75,-64,33,17,13,25,15,77,
-1,-42,-29,72,64,46,49,31,61,44,
--8,-47,-54,-46,-30,19,20,-1,-16,0,
-16,-12,-18,-9,-26,-27,-10,-22,53,45,
--10,-47,-75,-82,-105,-109,8,25,49,77,
-50,65,114,117,124,118,115,96,90,61,
--9,-45,-63,-60,-75,-57,8,11,20,29,
-0,-35,-49,-43,40,47,35,40,55,38,
--24,-76,-103,-112,-27,3,23,34,52,75,
-8,-29,-43,12,63,38,35,29,24,8,
-25,11,1,-15,-18,-43,-7,37,40,21,
--20,-56,-19,-19,-4,-2,11,29,51,63,
--2,-44,-62,-75,-89,30,57,51,74,51,
-50,46,68,64,65,52,63,55,65,43,
-18,-9,-26,-35,-55,-69,3,6,8,17,
--15,-61,-86,-97,1,86,93,74,78,67,
--1,-38,-66,-48,48,39,29,25,17,-1,
-13,13,29,39,50,51,69,82,97,98,
--2,-36,-46,-27,-16,-30,-13,-4,-7,-4,
-25,-5,-11,-6,-25,-21,33,12,31,29,
--8,-38,-52,-63,-68,-89,-33,-1,10,74,
--2,-15,59,91,105,105,101,87,84,62,
--7,-33,-50,-35,-54,-47,25,17,82,81,
--13,-56,-83,21,58,31,42,25,72,65,
--24,-66,-91,-56,9,-2,21,10,69,75,
-2,-24,11,22,25,28,38,34,48,33,
-7,-29,-26,17,15,-1,14,0,-2,0,
--6,-41,-67,6,-2,-9,19,2,85,74,
--22,-67,-84,-71,-50,3,11,-9,2,62};
-
-signed char cdbk_nb_low1[320]={
--34,-52,-15,45,2,
-23,21,52,24,-33,
--9,-1,9,-44,-41,
--13,-17,44,22,-17,
--6,-4,-1,22,38,
-26,16,2,50,27,
--35,-34,-9,-41,6,
-0,-16,-34,51,8,
--14,-31,-49,15,-33,
-45,49,33,-11,-37,
--62,-54,45,11,-5,
--72,11,-1,-12,-11,
-24,27,-11,-43,46,
-43,33,-12,-9,-1,
-1,-4,-23,-57,-71,
-11,8,16,17,-8,
--20,-31,-41,53,48,
--16,3,65,-24,-8,
--23,-32,-37,-32,-49,
--10,-17,6,38,5,
--9,-17,-46,8,52,
-3,6,45,40,39,
--7,-6,-34,-74,31,
-8,1,-16,43,68,
--11,-19,-31,4,6,
-0,-6,-17,-16,-38,
--16,-30,2,9,-39,
--16,-1,43,-10,48,
-3,3,-16,-31,-3,
-62,68,43,13,3,
--10,8,20,-56,12,
-12,-2,-18,22,-15,
--40,-36,1,7,41,
-0,1,46,-6,-62,
--4,-12,-2,-11,-83,
--13,-2,91,33,-10,
-0,4,-11,-16,79,
-32,37,14,9,51,
--21,-28,-56,-34,0,
-21,9,-26,11,28,
--42,-54,-23,-2,-15,
-31,30,8,-39,-66,
--39,-36,31,-28,-40,
--46,35,40,22,24,
-33,48,23,-34,14,
-40,32,17,27,-3,
-25,26,-13,-61,-17,
-11,4,31,60,-6,
--26,-41,-64,13,16,
--26,54,31,-11,-23,
--9,-11,-34,-71,-21,
--34,-35,55,50,29,
--22,-27,-50,-38,57,
-33,42,57,48,26,
-11,0,-49,-31,26,
--4,-14,5,78,37,
-17,0,-49,-12,-23,
-26,14,2,2,-43,
--17,-12,10,-8,-4,
-8,18,12,-6,20,
--12,-6,-13,-25,34,
-15,40,49,7,8,
-13,20,20,-19,-22,
--2,-8,2,51,-51};
-
-signed char cdbk_nb_low2[320]={
--6,53,-21,-24,4,
-26,17,-4,-37,25,
-17,-36,-13,31,3,
--6,27,15,-10,31,
-28,26,-10,-10,-40,
-16,-7,15,13,41,
--9,0,-4,50,-6,
--7,14,38,22,0,
--48,2,1,-13,-19,
-32,-3,-60,11,-17,
--1,-24,-34,-1,35,
--5,-27,28,44,13,
-25,15,42,-11,15,
-51,35,-36,20,8,
--4,-12,-29,19,-47,
-49,-15,-4,16,-29,
--39,14,-30,4,25,
--9,-5,-51,-14,-3,
--40,-32,38,5,-9,
--8,-4,-1,-22,71,
--3,14,26,-18,-22,
-24,-41,-25,-24,6,
-23,19,-10,39,-26,
--27,65,45,2,-7,
--26,-8,22,-12,16,
-15,16,-35,-5,33,
--21,-8,0,23,33,
-34,6,21,36,6,
--7,-22,8,-37,-14,
-31,38,11,-4,-3,
--39,-32,-8,32,-23,
--6,-12,16,20,-28,
--4,23,13,-52,-1,
-22,6,-33,-40,-6,
-4,-62,13,5,-26,
-35,39,11,2,57,
--11,9,-20,-28,-33,
-52,-5,-6,-2,22,
--14,-16,-48,35,1,
--58,20,13,33,-1,
--74,56,-18,-22,-31,
-12,6,-14,4,-2,
--9,-47,10,-3,29,
--17,-5,61,14,47,
--12,2,72,-39,-17,
-92,64,-53,-51,-15,
--30,-38,-41,-29,-28,
-27,9,36,9,-35,
--42,81,-21,20,25,
--16,-5,-17,-35,21,
-15,-28,48,2,-2,
-9,-19,29,-40,30,
--18,-18,18,-16,-57,
-15,-20,-12,-15,-37,
--15,33,-39,21,-22,
--13,35,11,13,-38,
--63,29,23,-27,32,
-18,3,-26,42,33,
--64,-66,-17,16,56,
-2,36,3,31,21,
--41,-39,8,-57,14,
-37,-2,19,-36,-19,
--23,-29,-16,1,-3,
--8,-10,31,64,-65};
-
-signed char cdbk_nb_high1[320]={
--26,-8,29,21,4,
-19,-39,33,-7,-36,
-56,54,48,40,29,
--4,-24,-42,-66,-43,
--60,19,-2,37,41,
--10,-37,-60,-64,18,
--22,77,73,40,25,
-4,19,-19,-66,-2,
-11,5,21,14,26,
--25,-86,-4,18,1,
-26,-37,10,37,-1,
-24,-12,-59,-11,20,
--6,34,-16,-16,42,
-19,-28,-51,53,32,
-4,10,62,21,-12,
--34,27,4,-48,-48,
--50,-49,31,-7,-21,
--42,-25,-4,-43,-22,
-59,2,27,12,-9,
--6,-16,-8,-32,-58,
--16,-29,-5,41,23,
--30,-33,-46,-13,-10,
--38,52,52,1,-17,
--9,10,26,-25,-6,
-33,-20,53,55,25,
--32,-5,-42,23,21,
-66,5,-28,20,9,
-75,29,-7,-42,-39,
-15,3,-23,21,6,
-11,1,-29,14,63,
-10,54,26,-24,-51,
--49,7,-23,-51,15,
--66,1,60,25,10,
-0,-30,-4,-15,17,
-19,59,40,4,-5,
-33,6,-22,-58,-70,
--5,23,-6,60,44,
--29,-16,-47,-29,52,
--19,50,28,16,35,
-31,36,0,-21,6,
-21,27,22,42,7,
--66,-40,-8,7,19,
-46,0,-4,60,36,
-45,-7,-29,-6,-32,
--39,2,6,-9,33,
-20,-51,-34,18,-6,
-19,6,11,5,-19,
--29,-2,42,-11,-45,
--21,-55,57,37,2,
--14,-67,-16,-27,-38,
-69,48,19,2,-17,
-20,-20,-16,-34,-17,
--25,-61,10,73,45,
-16,-40,-64,-17,-29,
--22,56,17,-39,8,
--11,8,-25,-18,-13,
--19,8,54,57,36,
--17,-26,-4,6,-21,
-40,42,-4,20,31,
-53,10,-34,-53,31,
--17,35,0,15,-6,
--20,-63,-73,22,25,
-29,17,8,-29,-39,
--69,18,15,-15,-5};
-
-signed char cdbk_nb_high2[320]={
-11,47,16,-9,-46,
--32,26,-64,34,-5,
-38,-7,47,20,2,
--73,-99,-3,-45,20,
-70,-52,15,-6,-7,
--82,31,21,47,51,
-39,-3,9,0,-41,
--7,-15,-54,2,0,
-27,-31,9,-45,-22,
--38,-24,-24,8,-33,
-23,5,50,-36,-17,
--18,-51,-2,13,19,
-43,12,-15,-12,61,
-38,38,7,13,0,
-6,-1,3,62,9,
-27,22,-33,38,-35,
--9,30,-43,-9,-32,
--1,4,-4,1,-5,
--11,-8,38,31,11,
--10,-42,-21,-37,1,
-43,15,-13,-35,-19,
--18,15,23,-26,59,
-1,-21,53,8,-41,
--50,-14,-28,4,21,
-25,-28,-40,5,-40,
--41,4,51,-33,-8,
--8,1,17,-60,12,
-25,-41,17,34,43,
-19,45,7,-37,24,
--15,56,-2,35,-10,
-48,4,-47,-2,5,
--5,-54,5,-3,-33,
--10,30,-2,-44,-24,
--38,9,-9,42,4,
-6,-56,44,-16,9,
--40,-26,18,-20,10,
-28,-41,-21,-4,13,
--18,32,-30,-3,37,
-15,22,28,50,-40,
-3,-29,-64,7,51,
--19,-11,17,-27,-40,
--64,24,-12,-7,-27,
-3,37,48,-1,2,
--9,-38,-34,46,1,
-27,-6,19,-13,26,
-10,34,20,25,40,
-50,-6,-7,30,9,
--24,0,-23,71,-61,
-22,58,-34,-4,2,
--49,-33,25,30,-8,
--6,-16,77,2,38,
--8,-35,-6,-30,56,
-78,31,33,-20,13,
--39,20,22,4,21,
--8,4,-6,10,-83,
--41,9,-25,-43,15,
--7,-12,-34,-39,-37,
--33,19,30,16,-33,
-42,-25,25,-68,44,
--15,-11,-4,23,50,
-14,4,-39,-43,20,
--30,60,9,-20,7,
-16,19,-33,37,29,
-16,-35,7,38,-27};

engine/voice_codecs/speex/source/libspeex/ltp.c

@@ -1,548 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: ltp.c
-   Long-Term Prediction functions
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#include <math.h>
-#include "ltp.h"
-#include "stack_alloc.h"
-#include "filters.h"
-#include "speex_bits.h"
-
-#ifdef _USE_SSE
-#include "ltp_sse.h"
-#else
-static float inner_prod(float *x, float *y, int len)
-{
-   int i;
-   float sum1=0,sum2=0,sum3=0,sum4=0;
-   for (i=0;i<len;)
-   {
-      sum1 += x[i]*y[i];
-      sum2 += x[i+1]*y[i+1];
-      sum3 += x[i+2]*y[i+2];
-      sum4 += x[i+3]*y[i+3];
-      i+=4;
-   }
-   return sum1+sum2+sum3+sum4;
-}
-#endif
-
-/*Original, non-optimized version*/
-/*static float inner_prod(float *x, float *y, int len)
-{
-   int i;
-   float sum=0;
-   for (i=0;i<len;i++)
-      sum += x[i]*y[i];
-   return sum;
-}
-*/
-
-
-void open_loop_nbest_pitch(float *sw, int start, int end, int len, int *pitch, float *gain, int N, char *stack)
-{
-   int i,j,k;
-   /*float corr=0;*/
-   /*float energy;*/
-   float *best_score;
-   float e0;
-   float *corr, *energy, *score;
-
-   best_score = PUSH(stack,N, float);
-   corr = PUSH(stack,end-start+1, float);
-   energy = PUSH(stack,end-start+2, float);
-   score = PUSH(stack,end-start+1, float);
-   for (i=0;i<N;i++)
-   {
-        best_score[i]=-1;
-        gain[i]=0;
-   }
-   energy[0]=inner_prod(sw-start, sw-start, len);
-   e0=inner_prod(sw, sw, len);
-   for (i=start;i<=end;i++)
-   {
-      /* Update energy for next pitch*/
-      energy[i-start+1] = energy[i-start] + sw[-i-1]*sw[-i-1] - sw[-i+len-1]*sw[-i+len-1];
-   }
-   for (i=start;i<=end;i++)
-   {
-      corr[i-start]=0;
-      score[i-start]=0;
-   }
-
-   for (i=start;i<=end;i++)
-   {
-      /* Compute correlation*/
-      corr[i-start]=inner_prod(sw, sw-i, len);
-      score[i-start]=corr[i-start]*corr[i-start]/(energy[i-start]+1);
-   }
-   for (i=start;i<=end;i++)
-   {
-      if (score[i-start]>best_score[N-1])
-      {
-         float g1, g;
-         g1 = corr[i-start]/(energy[i-start]+10);
-         g = sqrt(g1*corr[i-start]/(e0+10));
-         if (g>g1)
-            g=g1;
-         if (g<0)
-            g=0;
-         for (j=0;j<N;j++)
-         {
-            if (score[i-start] > best_score[j])
-            {
-               for (k=N-1;k>j;k--)
-               {
-                  best_score[k]=best_score[k-1];
-                  pitch[k]=pitch[k-1];
-                  gain[k] = gain[k-1];
-               }
-               best_score[j]=score[i-start];
-               pitch[j]=i;
-               gain[j]=g;
-               break;
-            }
-         }
-      }
-   }
-
-}
-
-
-
-
-/** Finds the best quantized 3-tap pitch predictor by analysis by synthesis */
-float pitch_gain_search_3tap(
-float target[],                 /* Target vector */
-float ak[],                     /* LPCs for this subframe */
-float awk1[],                   /* Weighted LPCs #1 for this subframe */
-float awk2[],                   /* Weighted LPCs #2 for this subframe */
-float exc[],                    /* Excitation */
-void *par,
-int   pitch,                    /* Pitch value */
-int   p,                        /* Number of LPC coeffs */
-int   nsf,                      /* Number of samples in subframe */
-SpeexBits *bits,
-char *stack,
-float *exc2,
-float *r,
-int  *cdbk_index
-)
-{
-   int i,j;
-   float *tmp, *tmp2;
-   float *x[3];
-   float *e[3];
-   float corr[3];
-   float A[3][3];
-   float gain[3];
-   int   gain_cdbk_size;
-   signed char *gain_cdbk;
-   float err1,err2;
-
-   ltp_params *params;
-   params = (ltp_params*) par;
-   gain_cdbk=params->gain_cdbk;
-   gain_cdbk_size=1<<params->gain_bits;
-   tmp = PUSH(stack, 3*nsf, float);
-   tmp2 = PUSH(stack, 3*nsf, float);
-
-   x[0]=tmp;
-   x[1]=tmp+nsf;
-   x[2]=tmp+2*nsf;
-
-   e[0]=tmp2;
-   e[1]=tmp2+nsf;
-   e[2]=tmp2+2*nsf;
-   
-   for (i=2;i>=0;i--)
-   {
-      int pp=pitch+1-i;
-      for (j=0;j<nsf;j++)
-      {
-         if (j-pp<0)
-            e[i][j]=exc2[j-pp];
-         else if (j-pp-pitch<0)
-            e[i][j]=exc2[j-pp-pitch];
-         else
-            e[i][j]=0;
-      }
-
-      if (i==2)
-         syn_percep_zero(e[i], ak, awk1, awk2, x[i], nsf, p, stack);
-      else {
-         for (j=0;j<nsf-1;j++)
-            x[i][j+1]=x[i+1][j];
-         x[i][0]=0;
-         for (j=0;j<nsf;j++)
-            x[i][j]+=e[i][0]*r[j];
-      }
-   }
-
-   for (i=0;i<3;i++)
-      corr[i]=inner_prod(x[i],target,nsf);
-   
-   for (i=0;i<3;i++)
-      for (j=0;j<=i;j++)
-         A[i][j]=A[j][i]=inner_prod(x[i],x[j],nsf);
-   
-   {
-      float C[9];
-      signed char *ptr=gain_cdbk;
-      int best_cdbk=0;
-      float best_sum=0;
-      C[0]=corr[2];
-      C[1]=corr[1];
-      C[2]=corr[0];
-      C[3]=A[1][2];
-      C[4]=A[0][1];
-      C[5]=A[0][2];
-      C[6]=A[2][2];
-      C[7]=A[1][1];
-      C[8]=A[0][0];
-      
-      for (i=0;i<gain_cdbk_size;i++)
-      {
-         float sum=0;
-         float g0,g1,g2;
-         ptr = gain_cdbk+3*i;
-         g0=0.015625*ptr[0]+.5;
-         g1=0.015625*ptr[1]+.5;
-         g2=0.015625*ptr[2]+.5;
-
-         sum += C[0]*g0;
-         sum += C[1]*g1;
-         sum += C[2]*g2;
-         sum -= C[3]*g0*g1;
-         sum -= C[4]*g2*g1;
-         sum -= C[5]*g2*g0;
-         sum -= .5*C[6]*g0*g0;
-         sum -= .5*C[7]*g1*g1;
-         sum -= .5*C[8]*g2*g2;
-
-         /* If 1, force "safe" pitch values to handle packet loss better */
-         if (0) {
-            float tot = fabs(ptr[1]);
-            if (ptr[0]>0)
-               tot+=ptr[0];
-            if (ptr[2]>0)
-               tot+=ptr[2];
-            if (tot>1)
-               continue;
-         }
-
-         if (sum>best_sum || i==0)
-         {
-            best_sum=sum;
-            best_cdbk=i;
-         }
-      }
-      gain[0] = 0.015625*gain_cdbk[best_cdbk*3]  + .5;
-      gain[1] = 0.015625*gain_cdbk[best_cdbk*3+1]+ .5;
-      gain[2] = 0.015625*gain_cdbk[best_cdbk*3+2]+ .5;
-
-      *cdbk_index=best_cdbk;
-   }
-   
-   for (i=0;i<nsf;i++)
-      exc[i]=gain[0]*e[2][i]+gain[1]*e[1][i]+gain[2]*e[0][i];
-   
-   err1=0;
-   err2=0;
-   for (i=0;i<nsf;i++)
-      err1+=target[i]*target[i];
-   for (i=0;i<nsf;i++)
-      err2+=(target[i]-gain[2]*x[0][i]-gain[1]*x[1][i]-gain[0]*x[2][i])
-      * (target[i]-gain[2]*x[0][i]-gain[1]*x[1][i]-gain[0]*x[2][i]);
-
-   return err2;
-}
-
-
-/** Finds the best quantized 3-tap pitch predictor by analysis by synthesis */
-int pitch_search_3tap(
-float target[],                 /* Target vector */
-float *sw,
-float ak[],                     /* LPCs for this subframe */
-float awk1[],                   /* Weighted LPCs #1 for this subframe */
-float awk2[],                   /* Weighted LPCs #2 for this subframe */
-float exc[],                    /* Excitation */
-void *par,
-int   start,                    /* Smallest pitch value allowed */
-int   end,                      /* Largest pitch value allowed */
-float pitch_coef,               /* Voicing (pitch) coefficient */
-int   p,                        /* Number of LPC coeffs */
-int   nsf,                      /* Number of samples in subframe */
-SpeexBits *bits,
-char *stack,
-float *exc2,
-float *r,
-int complexity
-)
-{
-   int i,j;
-   int cdbk_index, pitch=0, best_gain_index=0;
-   float *best_exc;
-   int best_pitch=0;
-   float err, best_err=-1;
-   int N;
-   ltp_params *params;
-   int *nbest;
-   float *gains;
-
-   N=complexity;
-   if (N>10)
-      N=10;
-
-   nbest=PUSH(stack, N, int);
-   gains = PUSH(stack, N, float);
-   params = (ltp_params*) par;
-
-   if (N==0 || end<start)
-   {
-      speex_bits_pack(bits, 0, params->pitch_bits);
-      speex_bits_pack(bits, 0, params->gain_bits);
-      for (i=0;i<nsf;i++)
-         exc[i]=0;
-      return start;
-   }
-   
-   best_exc=PUSH(stack,nsf, float);
-   
-   if (N>end-start+1)
-      N=end-start+1;
-   open_loop_nbest_pitch(sw, start, end, nsf, nbest, gains, N, stack);
-   for (i=0;i<N;i++)
-   {
-      pitch=nbest[i];
-      for (j=0;j<nsf;j++)
-         exc[j]=0;
-      err=pitch_gain_search_3tap(target, ak, awk1, awk2, exc, par, pitch, p, nsf,
-                                 bits, stack, exc2, r, &cdbk_index);
-      if (err<best_err || best_err<0)
-      {
-         for (j=0;j<nsf;j++)
-            best_exc[j]=exc[j];
-         best_err=err;
-         best_pitch=pitch;
-         best_gain_index=cdbk_index;
-      }
-   }
-   
-   /*printf ("pitch: %d %d\n", best_pitch, best_gain_index);*/
-   speex_bits_pack(bits, best_pitch-start, params->pitch_bits);
-   speex_bits_pack(bits, best_gain_index, params->gain_bits);
-   /*printf ("encode pitch: %d %d\n", best_pitch, best_gain_index);*/
-   for (i=0;i<nsf;i++)
-      exc[i]=best_exc[i];
-
-   return pitch;
-}
-
-void pitch_unquant_3tap(
-float exc[],                    /* Excitation */
-int   start,                    /* Smallest pitch value allowed */
-int   end,                      /* Largest pitch value allowed */
-float pitch_coef,               /* Voicing (pitch) coefficient */
-void *par,
-int   nsf,                      /* Number of samples in subframe */
-int *pitch_val,
-float *gain_val,
-SpeexBits *bits,
-char *stack,
-int count_lost,
-int subframe_offset,
-float last_pitch_gain)
-{
-   int i;
-   int pitch;
-   int gain_index;
-   float gain[3];
-   signed char *gain_cdbk;
-   ltp_params *params;
-   params = (ltp_params*) par;
-   gain_cdbk=params->gain_cdbk;
-
-   pitch = speex_bits_unpack_unsigned(bits, params->pitch_bits);
-   pitch += start;
-   gain_index = speex_bits_unpack_unsigned(bits, params->gain_bits);
-   /*printf ("decode pitch: %d %d\n", pitch, gain_index);*/
-   gain[0] = 0.015625*gain_cdbk[gain_index*3]+.5;
-   gain[1] = 0.015625*gain_cdbk[gain_index*3+1]+.5;
-   gain[2] = 0.015625*gain_cdbk[gain_index*3+2]+.5;
-
-   if (count_lost && pitch > subframe_offset)
-   {
-      float gain_sum;
-
-      if (1) {
-	 float tmp = count_lost < 4 ? last_pitch_gain : 0.4 * last_pitch_gain;
-         if (tmp>.95)
-            tmp=.95;
-         gain_sum = fabs(gain[1]);
-         if (gain[0]>0)
-            gain_sum += gain[0];
-         else
-            gain_sum -= .5*gain[0];
-         if (gain[2]>0)
-            gain_sum += gain[2];
-         else
-            gain_sum -= .5*gain[2];
-	 if (gain_sum > tmp) {
-	    float fact = tmp/gain_sum;
-	    for (i=0;i<3;i++)
-	       gain[i]*=fact;
-
-	 }
-
-      }
-
-      if (0) {
-      gain_sum = fabs(gain[0])+fabs(gain[1])+fabs(gain[2]);
-	 if (gain_sum>.95) {
-         float fact = .95/gain_sum;
-         for (i=0;i<3;i++)
-            gain[i]*=fact;
-      }
-   }
-   }
-
-   *pitch_val = pitch;
-   /**gain_val = gain[0]+gain[1]+gain[2];*/
-   gain_val[0]=gain[0];
-   gain_val[1]=gain[1];
-   gain_val[2]=gain[2];
-
-   {
-      float *e[3];
-      float *tmp2;
-      tmp2=PUSH(stack, 3*nsf, float);
-      e[0]=tmp2;
-      e[1]=tmp2+nsf;
-      e[2]=tmp2+2*nsf;
-      
-      for (i=0;i<3;i++)
-      {
-         int j;
-         int pp=pitch+1-i;
-#if 0
-         for (j=0;j<nsf;j++)
-         {
-            if (j-pp<0)
-               e[i][j]=exc[j-pp];
-            else if (j-pp-pitch<0)
-               e[i][j]=exc[j-pp-pitch];
-            else
-               e[i][j]=0;
-         }
-#else
-         {
-            int tmp1, tmp3;
-            tmp1=nsf;
-            if (tmp1>pp)
-               tmp1=pp;
-            for (j=0;j<tmp1;j++)
-               e[i][j]=exc[j-pp];
-            tmp3=nsf;
-            if (tmp3>pp+pitch)
-               tmp3=pp+pitch;
-            for (j=tmp1;j<tmp3;j++)
-               e[i][j]=exc[j-pp-pitch];
-            for (j=tmp3;j<nsf;j++)
-               e[i][j]=0;
-         }
-#endif
-      }
-      for (i=0;i<nsf;i++)
-           exc[i]=gain[0]*e[2][i]+gain[1]*e[1][i]+gain[2]*e[0][i];
-   }
-}
-
-
-/** Forced pitch delay and gain */
-int forced_pitch_quant(
-float target[],                 /* Target vector */
-float *sw,
-float ak[],                     /* LPCs for this subframe */
-float awk1[],                   /* Weighted LPCs #1 for this subframe */
-float awk2[],                   /* Weighted LPCs #2 for this subframe */
-float exc[],                    /* Excitation */
-void *par,
-int   start,                    /* Smallest pitch value allowed */
-int   end,                      /* Largest pitch value allowed */
-float pitch_coef,               /* Voicing (pitch) coefficient */
-int   p,                        /* Number of LPC coeffs */
-int   nsf,                      /* Number of samples in subframe */
-SpeexBits *bits,
-char *stack,
-float *exc2,
-float *r,
-int complexity
-)
-{
-   int i;
-   if (pitch_coef>.99)
-      pitch_coef=.99;
-   for (i=0;i<nsf;i++)
-   {
-      exc[i]=exc[i-start]*pitch_coef;
-   }
-   return start;
-}
-
-/** Unquantize forced pitch delay and gain */
-void forced_pitch_unquant(
-float exc[],                    /* Excitation */
-int   start,                    /* Smallest pitch value allowed */
-int   end,                      /* Largest pitch value allowed */
-float pitch_coef,               /* Voicing (pitch) coefficient */
-void *par,
-int   nsf,                      /* Number of samples in subframe */
-int *pitch_val,
-float *gain_val,
-SpeexBits *bits,
-char *stack,
-int count_lost,
-int subframe_offset,
-float last_pitch_gain)
-{
-   int i;
-   /*pitch_coef=.9;*/
-   if (pitch_coef>.99)
-      pitch_coef=.99;
-   for (i=0;i<nsf;i++)
-   {
-      exc[i]=exc[i-start]*pitch_coef;
-   }
-   *pitch_val = start;
-   gain_val[0]=gain_val[2]=0;
-   gain_val[1] = pitch_coef;
-}

engine/voice_codecs/speex/source/libspeex/ltp.h

@@ -1,138 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: ltp.h
-   Long-Term Prediction functions
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#include "speex_bits.h"
-
-
-typedef struct ltp_params {
-   signed char *gain_cdbk;
-   int     gain_bits;
-   int     pitch_bits;
-} ltp_params;
-
-
-void open_loop_nbest_pitch(float *sw, int start, int end, int len, int *pitch, float *gain, int N, char *stack);
-
-
-/** Finds the best quantized 3-tap pitch predictor by analysis by synthesis */
-int pitch_search_3tap(
-float target[],                 /* Target vector */
-float *sw,
-float ak[],                     /* LPCs for this subframe */
-float awk1[],                   /* Weighted LPCs #1 for this subframe */
-float awk2[],                   /* Weighted LPCs #2 for this subframe */
-float exc[],                    /* Overlapping codebook */
-void *par,
-int   start,                    /* Smallest pitch value allowed */
-int   end,                      /* Largest pitch value allowed */
-float pitch_coef,               /* Voicing (pitch) coefficient */
-int   p,                        /* Number of LPC coeffs */
-int   nsf,                      /* Number of samples in subframe */
-SpeexBits *bits,
-char *stack,
-float *exc2,
-float *r,
-int   complexity
-);
-
-/*Unquantize adaptive codebook and update pitch contribution*/
-void pitch_unquant_3tap(
-float exc[],                    /* Excitation */
-int   start,                    /* Smallest pitch value allowed */
-int   end,                      /* Largest pitch value allowed */
-float pitch_coef,               /* Voicing (pitch) coefficient */
-void *par,
-int   nsf,                      /* Number of samples in subframe */
-int *pitch_val,
-float *gain_val,
-SpeexBits *bits,
-char *stack,
-int lost,
-int subframe_offset,
-float last_pitch_gain
-);
-
-float pitch_gain_search_3tap(
-float target[],                 /* Target vector */
-float ak[],                     /* LPCs for this subframe */
-float awk1[],                   /* Weighted LPCs #1 for this subframe */
-float awk2[],                   /* Weighted LPCs #2 for this subframe */
-float exc[],                    /* Excitation */
-void *par,
-int   pitch,                    /* Pitch value */
-int   p,                        /* Number of LPC coeffs */
-int   nsf,                      /* Number of samples in subframe */
-SpeexBits *bits,
-char *stack,
-float *exc2,
-float *r,
-int  *cdbk_index
-);
-
-
-/** Forced pitch delay and gain */
-int forced_pitch_quant(
-float target[],                 /* Target vector */
-float *sw,
-float ak[],                     /* LPCs for this subframe */
-float awk1[],                   /* Weighted LPCs #1 for this subframe */
-float awk2[],                   /* Weighted LPCs #2 for this subframe */
-float exc[],                    /* Excitation */
-void *par,
-int   start,                    /* Smallest pitch value allowed */
-int   end,                      /* Largest pitch value allowed */
-float pitch_coef,               /* Voicing (pitch) coefficient */
-int   p,                        /* Number of LPC coeffs */
-int   nsf,                      /* Number of samples in subframe */
-SpeexBits *bits,
-char *stack,
-float *exc2,
-float *r,
-int complexity
-);
-
-/** Unquantize forced pitch delay and gain */
-void forced_pitch_unquant(
-float exc[],                    /* Excitation */
-int   start,                    /* Smallest pitch value allowed */
-int   end,                      /* Largest pitch value allowed */
-float pitch_coef,               /* Voicing (pitch) coefficient */
-void *par,
-int   nsf,                      /* Number of samples in subframe */
-int *pitch_val,
-float *gain_val,
-SpeexBits *bits,
-char *stack,
-int lost,
-int subframe_offset,
-float last_pitch_gain
-);

engine/voice_codecs/speex/source/libspeex/ltp_sse.h

@@ -1,95 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: ltp.c
-   Lont-Term Prediction functions
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-
-static float inner_prod(float *a, float *b, int len)
-{
-  float sum;
-  __asm__ __volatile__ (
-  "\tpush %%eax\n"
-  "\tpush %%edi\n"
-  "\tpush %%ecx\n"
-  "\txorps %%xmm3, %%xmm3\n"
-  "\txorps %%xmm4, %%xmm4\n"
-
-  "\tsub $20, %%ecx\n"
-
-".mul20_loop%=:\n"
-
-  "\tmovups (%%eax), %%xmm0\n"
-  "\tmovups (%%edi), %%xmm1\n"
-  "\tmulps %%xmm0, %%xmm1\n"
-
-  "\tmovups 16(%%eax), %%xmm5\n"
-  "\tmovups 16(%%edi), %%xmm6\n"
-  "\tmulps %%xmm5, %%xmm6\n"
-  "\taddps %%xmm1, %%xmm3\n"
-
-  "\tmovups 32(%%eax), %%xmm0\n"
-  "\tmovups 32(%%edi), %%xmm1\n"
-  "\tmulps %%xmm0, %%xmm1\n"
-  "\taddps %%xmm6, %%xmm4\n"
-
-  "\tmovups 48(%%eax), %%xmm5\n"
-  "\tmovups 48(%%edi), %%xmm6\n"
-  "\tmulps %%xmm5, %%xmm6\n"
-  "\taddps %%xmm1, %%xmm3\n"
-
-  "\tmovups 64(%%eax), %%xmm0\n"
-  "\tmovups 64(%%edi), %%xmm1\n"
-  "\tmulps %%xmm0, %%xmm1\n"
-  "\taddps %%xmm6, %%xmm4\n"
-  "\taddps %%xmm1, %%xmm3\n"
-
-
-  "\tadd $80, %%eax\n"
-  "\tadd $80, %%edi\n"
-
-  "\tsub $20,  %%ecx\n"
-
-  "\tjae .mul20_loop%=\n"
-
-  "\taddps %%xmm4, %%xmm3\n"
-
-  "\tmovhlps %%xmm3, %%xmm4\n"
-  "\taddps %%xmm4, %%xmm3\n"
-  "\tmovaps %%xmm3, %%xmm4\n"
-  "\tshufps $0x55, %%xmm4, %%xmm4\n"
-  "\taddss %%xmm4, %%xmm3\n"
-  "\tmovss %%xmm3, (%%edx)\n"
-  
-  "\tpop %%ecx\n"
-  "\tpop %%edi\n"
-  "\tpop %%eax\n"
-  : : "a" (a), "D" (b), "c" (len), "d" (&sum) : "memory");
-  return sum;
-}

engine/voice_codecs/speex/source/libspeex/math_approx.c

@@ -1,105 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: math_approx.c
-   Various math approximation functions for Speex
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#include <math.h>
-#include "math_approx.h"
-
-#ifdef SLOW_TRIG
-
-float cos_sin[102] = {
-   1.00000000, 0.00000000,
-   0.99804751, 0.06245932,
-   0.99219767, 0.12467473,
-   0.98247331, 0.18640330,
-   0.96891242, 0.24740396,
-   0.95156795, 0.30743851,
-   0.93050762, 0.36627253,
-   0.90581368, 0.42367626,
-   0.87758256, 0.47942554,
-   0.84592450, 0.53330267,
-   0.81096312, 0.58509727,
-   0.77283495, 0.63460708,
-   0.73168887, 0.68163876,
-   0.68768556, 0.72600866,
-   0.64099686, 0.76754350,
-   0.59180508, 0.80608111,
-   0.54030231, 0.84147098,
-   0.48668967, 0.87357494,
-   0.43117652, 0.90226759,
-   0.37397963, 0.92743692,
-   0.31532236, 0.94898462,
-   0.25543377, 0.96682656,
-   0.19454771, 0.98089306,
-   0.13290194, 0.99112919,
-   0.07073720, 0.99749499,
-   0.00829623, 0.99996559,
-   -0.05417714, 0.99853134,
-   -0.11643894, 0.99319785,
-   -0.17824606, 0.98398595,
-   -0.23935712, 0.97093160,
-   -0.29953351, 0.95408578,
-   -0.35854022, 0.93351428,
-   -0.41614684, 0.90929743,
-   -0.47212841, 0.88152979,
-   -0.52626633, 0.85031979,
-   -0.57834920, 0.81578931,
-   -0.62817362, 0.77807320,
-   -0.67554504, 0.73731872,
-   -0.72027847, 0.69368503,
-   -0.76219923, 0.64734252,
-   -0.80114362, 0.59847214,
-   -0.83695955, 0.54726475,
-   -0.86950718, 0.49392030,
-   -0.89865940, 0.43864710,
-   -0.92430238, 0.38166099,
-   -0.94633597, 0.32318451,
-   -0.96467415, 0.26344599,
-   -0.97924529, 0.20267873,
-   -0.98999250, 0.14112001,
-   -0.99687381, 0.07901022,
-   -0.99986235, 0.01659189
-};
-
-float speex_cos(float x)
-{
-   int ind;
-   float delta;
-   ind = (int)floor(x*16+.5);
-   delta = x-0.062500*ind;
-   ind <<= 1;
-   return cos_sin[ind] - delta*(cos_sin[ind+1] + 
-                                  .5*delta*(cos_sin[ind] - 
-                                            .3333333*delta*cos_sin[ind+1]));
-}
-
-#endif
-

engine/voice_codecs/speex/source/libspeex/math_approx.h

@@ -1,39 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: math_approx.c
-   Various math approximation functions for Speex
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#ifndef MATH_APPROX_H
-#define MATH_APPROX_H
-
-float speex_cos(float x);
-
-
-#endif

engine/voice_codecs/speex/source/libspeex/misc.c

@@ -1,145 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: mics.c
-   Various utility routines for Speex
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#include "misc.h"
-#include <stdlib.h>
-#include <string.h>
-#include <stdio.h>
-
-#ifndef RELEASE
-void print_vec(float *vec, int len, char *name)
-{
-   int i;
-   printf ("%s ", name);
-   for (i=0;i<len;i++)
-      printf (" %f", vec[i]);
-   printf ("\n");
-}
-#endif
-
-unsigned int be_int(unsigned int i)
-{
-   unsigned int ret=i;
-#ifndef WORDS_BIGENDIAN
-   ret =  i>>24;
-   ret += (i>>8)&0x0000ff00;
-   ret += (i<<8)&0x00ff0000;
-   ret += (i<<24);
-#endif
-   return ret;
-}
-
-unsigned int le_int(unsigned int i)
-{
-   unsigned int ret=i;
-#ifdef WORDS_BIGENDIAN
-   ret =  i>>24;
-   ret += (i>>8)&0x0000ff00;
-   ret += (i<<8)&0x00ff0000;
-   ret += (i<<24);
-#endif
-   return ret;
-}
-
-unsigned short be_short(unsigned short s)
-{
-   unsigned short ret=s;
-#ifndef WORDS_BIGENDIAN
-   ret =  s>>8;
-   ret += s<<8;
-#endif
-   return ret;
-}
-
-unsigned short le_short(unsigned short s)
-{
-   unsigned short ret=s;
-#ifdef WORDS_BIGENDIAN
-   ret =  s>>8;
-   ret += s<<8;
-#endif
-   return ret;
-}
-
-void *speex_alloc (int size)
-{
-   return calloc(size,1);
-}
-
-void *speex_realloc (void *ptr, int size)
-{
-   return realloc(ptr, size);
-}
-
-void speex_free (void *ptr)
-{
-   free(ptr);
-}
-
-void *speex_move (void *dest, void *src, int n)
-{
-   return memmove(dest,src,n);
-}
-
-void speex_error(char *str)
-{
-   fprintf (stderr, "Fatal error: %s\n", str);
-   exit(1);
-}
-
-void speex_warning(char *str)
-{
-   fprintf (stderr, "warning: %s\n", str);
-}
-
-void speex_warning_int(char *str, int val)
-{
-   fprintf (stderr, "warning: %s %d\n", str, val);
-}
-
-void speex_rand_vec(float std, float *data, int len)
-{
-   int i;
-   for (i=0;i<len;i++)
-      data[i]+=3*std*((((float)rand())/RAND_MAX)-.5);
-}
-
-float speex_rand(float std)
-{
-   return 3*std*((((float)rand())/RAND_MAX)-.5);
-}
-
-void _speex_putc(int ch, void *file)
-{
-   FILE *f = (FILE *)file;
-   fputc(ch, f);
-}

engine/voice_codecs/speex/source/libspeex/misc.h

@@ -1,83 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin */
-/**
-   @file misc.h
-   @brief Various compatibility routines for Speex
-*/
-/*
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#ifndef MISC_H
-#define MISC_H
-
-#ifndef VERSION
-#define VERSION "speex-1.0"
-#endif
-
-/*Disable some warnings on VC++*/
-#ifdef _MSC_VER
-#pragma warning(disable : 4244)
-#pragma warning(disable : 4305)
-#endif
-
-#ifndef RELEASE
-void print_vec(float *vec, int len, char *name);
-#endif
-
-unsigned int be_int(unsigned int i);
-unsigned int le_int(unsigned int i);
-
-
-unsigned short be_short(unsigned short s);
-unsigned short le_short(unsigned short s);
-
-/** Speex wrapper for calloc. To do your own dynamic allocation, all you need to do is replace this function, speex_realloc and speex_free */
-void *speex_alloc (int size);
-
-/** Speex wrapper for realloc. To do your own dynamic allocation, all you need to do is replace this function, speex_alloc and speex_free */
-void *speex_realloc (void *ptr, int size);
-
-/** Speex wrapper for calloc. To do your own dynamic allocation, all you need to do is replace this function, speex_realloc and speex_alloc */
-void speex_free (void *ptr);
-
-/** Speex wrapper for mem_move */
-void *speex_move (void *dest, void *src, int n);
-
-void speex_error(char *str);
-
-void speex_warning(char *str);
-
-void speex_warning_int(char *str, int val);
-
-void speex_rand_vec(float std, float *data, int len);
-
-float speex_rand(float std);
-
-void _speex_putc(int ch, void *file);
-
-#endif

engine/voice_codecs/speex/source/libspeex/modes.c

@@ -1,650 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: modes.c
-
-   Describes the different modes of the codec
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-
-*/
-
-#include "modes.h"
-#include "ltp.h"
-#include "quant_lsp.h"
-#include "cb_search.h"
-#include "sb_celp.h"
-#include "nb_celp.h"
-#include "vbr.h"
-#include "misc.h"
-
-#ifndef NULL
-#define NULL 0
-#endif
-
-SpeexMode *speex_mode_list[SPEEX_NB_MODES] = {&speex_nb_mode, &speex_wb_mode, &speex_uwb_mode};
-
-/* Extern declarations for all codebooks we use here */
-extern signed char gain_cdbk_nb[];
-extern signed char gain_cdbk_lbr[];
-extern signed char hexc_table[];
-extern signed char exc_5_256_table[];
-extern signed char exc_5_64_table[];
-extern signed char exc_8_128_table[];
-extern signed char exc_10_32_table[];
-extern signed char exc_10_16_table[];
-extern signed char exc_20_32_table[];
-extern signed char hexc_10_32_table[];
-
-static int nb_mode_query(void *mode, int request, void *ptr);
-static int wb_mode_query(void *mode, int request, void *ptr);
-
-/* Parameters for Long-Term Prediction (LTP)*/
-static ltp_params ltp_params_nb = {
-   gain_cdbk_nb,
-   7,
-   7
-};
-
-/* Parameters for Long-Term Prediction (LTP)*/
-static ltp_params ltp_params_vlbr = {
-   gain_cdbk_lbr,
-   5,
-   0
-};
-
-/* Parameters for Long-Term Prediction (LTP)*/
-static ltp_params ltp_params_lbr = {
-   gain_cdbk_lbr,
-   5,
-   7
-};
-
-/* Parameters for Long-Term Prediction (LTP)*/
-static ltp_params ltp_params_med = {
-   gain_cdbk_lbr,
-   5,
-   7
-};
-
-/* Split-VQ innovation parameters for very low bit-rate narrowband */
-static split_cb_params split_cb_nb_vlbr = {
-   10,               /*subvect_size*/
-   4,               /*nb_subvect*/
-   exc_10_16_table, /*shape_cb*/
-   4,               /*shape_bits*/
-   0,
-};
-
-/* Split-VQ innovation parameters for very low bit-rate narrowband */
-static split_cb_params split_cb_nb_ulbr = {
-   20,               /*subvect_size*/
-   2,               /*nb_subvect*/
-   exc_20_32_table, /*shape_cb*/
-   5,               /*shape_bits*/
-   0,
-};
-
-/* Split-VQ innovation parameters for low bit-rate narrowband */
-static split_cb_params split_cb_nb_lbr = {
-   10,              /*subvect_size*/
-   4,               /*nb_subvect*/
-   exc_10_32_table, /*shape_cb*/
-   5,               /*shape_bits*/
-   0,
-};
-
-
-/* Split-VQ innovation parameters narrowband */
-static split_cb_params split_cb_nb = {
-   5,               /*subvect_size*/
-   8,               /*nb_subvect*/
-   exc_5_64_table, /*shape_cb*/
-   6,               /*shape_bits*/
-   0,
-};
-
-/* Split-VQ innovation parameters narrowband */
-static split_cb_params split_cb_nb_med = {
-   8,               /*subvect_size*/
-   5,               /*nb_subvect*/
-   exc_8_128_table, /*shape_cb*/
-   7,               /*shape_bits*/
-   0,
-};
-
-/* Split-VQ innovation for low-band wideband */
-static split_cb_params split_cb_sb = {
-   5,               /*subvect_size*/
-   8,              /*nb_subvect*/
-   exc_5_256_table,    /*shape_cb*/
-   8,               /*shape_bits*/
-   0,
-};
-
-/* Split-VQ innovation for high-band wideband */
-static split_cb_params split_cb_high = {
-   8,               /*subvect_size*/
-   5,               /*nb_subvect*/
-   hexc_table,       /*shape_cb*/
-   7,               /*shape_bits*/
-   1,
-};
-
-
-/* Split-VQ innovation for high-band wideband */
-static split_cb_params split_cb_high_lbr = {
-   10,               /*subvect_size*/
-   4,               /*nb_subvect*/
-   hexc_10_32_table,       /*shape_cb*/
-   5,               /*shape_bits*/
-   0,
-};
-
-/* 2150 bps "vocoder-like" mode for comfort noise */
-static SpeexSubmode nb_submode1 = {
-   0,
-   1,
-   0,
-   0,
-   /* LSP quantization */
-   lsp_quant_lbr,
-   lsp_unquant_lbr,
-   /* No pitch quantization */
-   forced_pitch_quant,
-   forced_pitch_unquant,
-   NULL,
-   /* No innovation quantization (noise only) */
-   noise_codebook_quant,
-   noise_codebook_unquant,
-   NULL,
-   .7, .7, -1,
-   43
-};
-
-/* 3.95 kbps very low bit-rate mode */
-static SpeexSubmode nb_submode8 = {
-   0,
-   1,
-   0,
-   0,
-   /*LSP quantization*/
-   lsp_quant_lbr,
-   lsp_unquant_lbr,
-   /*No pitch quantization*/
-   forced_pitch_quant,
-   forced_pitch_unquant,
-   NULL,
-   /*Innovation quantization*/
-   split_cb_search_shape_sign,
-   split_cb_shape_sign_unquant,
-   &split_cb_nb_ulbr,
-
-   0.7, 0.5, .65,
-   79
-};
-
-/* 5.95 kbps very low bit-rate mode */
-static SpeexSubmode nb_submode2 = {
-   0,
-   0,
-   0,
-   0,
-   /*LSP quantization*/
-   lsp_quant_lbr,
-   lsp_unquant_lbr,
-   /*No pitch quantization*/
-   pitch_search_3tap,
-   pitch_unquant_3tap,
-   &ltp_params_vlbr,
-   /*Innovation quantization*/
-   split_cb_search_shape_sign,
-   split_cb_shape_sign_unquant,
-   &split_cb_nb_vlbr,
-
-   0.7, 0.5, .55,
-   119
-};
-
-/* 8 kbps low bit-rate mode */
-static SpeexSubmode nb_submode3 = {
-   -1,
-   0,
-   1,
-   0,
-   /*LSP quantization*/
-   lsp_quant_lbr,
-   lsp_unquant_lbr,
-   /*Pitch quantization*/
-   pitch_search_3tap,
-   pitch_unquant_3tap,
-   &ltp_params_lbr,
-   /*Innovation quantization*/
-   split_cb_search_shape_sign,
-   split_cb_shape_sign_unquant,
-   &split_cb_nb_lbr,
-
-   0.7, 0.55, .45,
-   160
-};
-
-/* 11 kbps medium bit-rate mode */
-static SpeexSubmode nb_submode4 = {
-   -1,
-   0,
-   1,
-   0,
-   /*LSP quantization*/
-   lsp_quant_lbr,
-   lsp_unquant_lbr,
-   /*Pitch quantization*/
-   pitch_search_3tap,
-   pitch_unquant_3tap,
-   &ltp_params_med,
-   /*Innovation quantization*/
-   split_cb_search_shape_sign,
-   split_cb_shape_sign_unquant,
-   &split_cb_nb_med,
-
-   0.7, 0.63, .35,
-   220
-};
-
-/* 15 kbps high bit-rate mode */
-static SpeexSubmode nb_submode5 = {
-   -1,
-   0,
-   3,
-   0,
-   /*LSP quantization*/
-   lsp_quant_nb,
-   lsp_unquant_nb,
-   /*Pitch quantization*/
-   pitch_search_3tap,
-   pitch_unquant_3tap,
-   &ltp_params_nb,
-   /*Innovation quantization*/
-   split_cb_search_shape_sign,
-   split_cb_shape_sign_unquant,
-   &split_cb_nb,
-
-   0.7, 0.65, .25,
-   300
-};
-
-/* 18.2 high bit-rate mode */
-static SpeexSubmode nb_submode6 = {
-   -1,
-   0,
-   3,
-   0,
-   /*LSP quantization*/
-   lsp_quant_nb,
-   lsp_unquant_nb,
-   /*Pitch quantization*/
-   pitch_search_3tap,
-   pitch_unquant_3tap,
-   &ltp_params_nb,
-   /*Innovation quantization*/
-   split_cb_search_shape_sign,
-   split_cb_shape_sign_unquant,
-   &split_cb_sb,
-
-   0.68, 0.65, .1,
-   364
-};
-
-/* 24.6 kbps high bit-rate mode */
-static SpeexSubmode nb_submode7 = {
-   -1,
-   0,
-   3,
-   1,
-   /*LSP quantization*/
-   lsp_quant_nb,
-   lsp_unquant_nb,
-   /*Pitch quantization*/
-   pitch_search_3tap,
-   pitch_unquant_3tap,
-   &ltp_params_nb,
-   /*Innovation quantization*/
-   split_cb_search_shape_sign,
-   split_cb_shape_sign_unquant,
-   &split_cb_nb,
-
-   0.65, 0.65, -1,
-   492
-};
-
-
-/* Default mode for narrowband */
-static SpeexNBMode nb_mode = {
-   160,    /*frameSize*/
-   40,     /*subframeSize*/
-   10,     /*lpcSize*/
-   640,    /*bufSize*/
-   17,     /*pitchStart*/
-   144,    /*pitchEnd*/
-   0.9,    /*gamma1*/
-   0.6,    /*gamma2*/
-   .01,   /*lag_factor*/
-   1.0001, /*lpc_floor*/
-   0.0,    /*preemph*/
-   {NULL, &nb_submode1, &nb_submode2, &nb_submode3, &nb_submode4, &nb_submode5, &nb_submode6, &nb_submode7,
-   &nb_submode8, NULL, NULL, NULL, NULL, NULL, NULL, NULL},
-   5,
-   {1, 8, 2, 3, 3, 4, 4, 5, 5, 6, 7}
-};
-
-
-/* Default mode for narrowband */
-SpeexMode speex_nb_mode = {
-   &nb_mode,
-   nb_mode_query,
-   "narrowband",
-   0,
-   4,
-   &nb_encoder_init,
-   &nb_encoder_destroy,
-   &nb_encode,
-   &nb_decoder_init,
-   &nb_decoder_destroy,
-   &nb_decode,
-   &nb_encoder_ctl,
-   &nb_decoder_ctl,
-};
-
-
-/* Wideband part */
-
-static SpeexSubmode wb_submode1 = {
-   0,
-   0,
-   1,
-   0,
-   /*LSP quantization*/
-   lsp_quant_high,
-   lsp_unquant_high,
-   /*Pitch quantization*/
-   NULL,
-   NULL,
-   NULL,
-   /*No innovation quantization*/
-   NULL,
-   NULL,
-   NULL,
-
-   .75, .75, -1,
-   36
-};
-
-
-static SpeexSubmode wb_submode2 = {
-   0,
-   0,
-   1,
-   0,
-   /*LSP quantization*/
-   lsp_quant_high,
-   lsp_unquant_high,
-   /*Pitch quantization*/
-   NULL,
-   NULL,
-   NULL,
-   /*Innovation quantization*/
-   split_cb_search_shape_sign,
-   split_cb_shape_sign_unquant,
-   &split_cb_high_lbr,
-
-   .85, .6, -1,
-   112
-};
-
-
-static SpeexSubmode wb_submode3 = {
-   0,
-   0,
-   1,
-   0,
-   /*LSP quantization*/
-   lsp_quant_high,
-   lsp_unquant_high,
-   /*Pitch quantization*/
-   NULL,
-   NULL,
-   NULL,
-   /*Innovation quantization*/
-   split_cb_search_shape_sign,
-   split_cb_shape_sign_unquant,
-   &split_cb_high,
-
-   .75, .7, -1,
-   192
-};
-
-static SpeexSubmode wb_submode4 = {
-   0,
-   0,
-   1,
-   1,
-   /*LSP quantization*/
-   lsp_quant_high,
-   lsp_unquant_high,
-   /*Pitch quantization*/
-   NULL,
-   NULL,
-   NULL,
-   /*Innovation quantization*/
-   split_cb_search_shape_sign,
-   split_cb_shape_sign_unquant,
-   &split_cb_high,
-
-   .75, .75, -1,
-   352
-};
-
-
-/* Split-band wideband CELP mode*/
-static SpeexSBMode sb_wb_mode = {
-   &speex_nb_mode,
-   160,    /*frameSize*/
-   40,     /*subframeSize*/
-   8,     /*lpcSize*/
-   640,    /*bufSize*/
-   .9,    /*gamma1*/
-   0.6,    /*gamma2*/
-   .002,   /*lag_factor*/
-   1.0001, /*lpc_floor*/
-   0.0,    /*preemph*/
-   0.9,
-   {NULL, &wb_submode1, &wb_submode2, &wb_submode3, &wb_submode4, NULL, NULL, NULL},
-   3,
-   {1, 8, 2, 3, 4, 5, 5, 6, 6, 7, 7},
-   {1, 1, 1, 1, 1, 1, 2, 2, 3, 3, 4},
-   vbr_hb_thresh,
-   5
-};
-
-
-SpeexMode speex_wb_mode = {
-   &sb_wb_mode,
-   wb_mode_query,
-   "wideband (sub-band CELP)",
-   1,
-   4,
-   &sb_encoder_init,
-   &sb_encoder_destroy,
-   &sb_encode,
-   &sb_decoder_init,
-   &sb_decoder_destroy,
-   &sb_decode,
-   &sb_encoder_ctl,
-   &sb_decoder_ctl,
-};
-
-
-
-/* "Ultra-wideband" mode stuff */
-
-
-
-/* Split-band "ultra-wideband" (32 kbps) CELP mode*/
-static SpeexSBMode sb_uwb_mode = {
-   &speex_wb_mode,
-   320,    /*frameSize*/
-   80,     /*subframeSize*/
-   8,     /*lpcSize*/
-   1280,    /*bufSize*/
-   .9,    /*gamma1*/
-   0.6,    /*gamma2*/
-   .002,   /*lag_factor*/
-   1.0001, /*lpc_floor*/
-   0.0,    /*preemph*/
-   0.7,
-   {NULL, &wb_submode1, NULL, NULL, NULL, NULL, NULL, NULL},
-   1,
-   {0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10},
-   {0, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1},
-   vbr_uhb_thresh,
-   2
-};
-
-
-SpeexMode speex_uwb_mode = {
-   &sb_uwb_mode,
-   wb_mode_query,
-   "ultra-wideband (sub-band CELP)",
-   2,
-   4,
-   &sb_encoder_init,
-   &sb_encoder_destroy,
-   &sb_encode,
-   &sb_decoder_init,
-   &sb_decoder_destroy,
-   &sb_decode,
-   &sb_encoder_ctl,
-   &sb_decoder_ctl,
-};
-
-
-
-
-void *speex_encoder_init(SpeexMode *mode)
-{
-   return mode->enc_init(mode);
-}
-
-void *speex_decoder_init(SpeexMode *mode)
-{
-   return mode->dec_init(mode);
-}
-
-void speex_encoder_destroy(void *state)
-{
-   (*((SpeexMode**)state))->enc_destroy(state);
-}
-
-int speex_encode(void *state, float *in, SpeexBits *bits)
-{
-   return (*((SpeexMode**)state))->enc(state, in, bits);
-}
-
-void speex_decoder_destroy(void *state)
-{
-   (*((SpeexMode**)state))->dec_destroy(state);
-}
-
-int speex_decode(void *state, SpeexBits *bits, float *out)
-{
-   return (*((SpeexMode**)state))->dec(state, bits, out);
-}
-
-
-int speex_encoder_ctl(void *state, int request, void *ptr)
-{
-   return (*((SpeexMode**)state))->enc_ctl(state, request, ptr);
-}
-
-int speex_decoder_ctl(void *state, int request, void *ptr)
-{
-   return (*((SpeexMode**)state))->dec_ctl(state, request, ptr);
-}
-
-
-
-static int nb_mode_query(void *mode, int request, void *ptr)
-{
-   SpeexNBMode *m = (SpeexNBMode*)mode;
-   
-   switch (request)
-   {
-   case SPEEX_MODE_FRAME_SIZE:
-      *((int*)ptr)=m->frameSize;
-      break;
-   case SPEEX_SUBMODE_BITS_PER_FRAME:
-      if (*((int*)ptr)==0)
-         *((int*)ptr) = NB_SUBMODE_BITS+1;
-      else if (m->submodes[*((int*)ptr)]==NULL)
-         *((int*)ptr) = -1;
-      else
-         *((int*)ptr) = m->submodes[*((int*)ptr)]->bits_per_frame;
-      break;
-   default:
-      speex_warning_int("Unknown nb_mode_query request: ", request);
-      return -1;
-   }
-   return 0;
-}
-
-static int wb_mode_query(void *mode, int request, void *ptr)
-{
-   SpeexSBMode *m = (SpeexSBMode*)mode;
-
-   switch (request)
-   {
-   case SPEEX_MODE_FRAME_SIZE:
-      *((int*)ptr)=2*m->frameSize;
-      break;
-   case SPEEX_SUBMODE_BITS_PER_FRAME:
-      if (*((int*)ptr)==0)
-         *((int*)ptr) = SB_SUBMODE_BITS+1;
-      else if (m->submodes[*((int*)ptr)]==NULL)
-         *((int*)ptr) = -1;
-      else
-         *((int*)ptr) = m->submodes[*((int*)ptr)]->bits_per_frame;
-      break;
-   default:
-      speex_warning_int("Unknown wb_mode_query request: ", request);
-      return -1;
-   }
-   return 0;
-}
-
-
-int speex_mode_query(SpeexMode *mode, int request, void *ptr)
-{
-   return mode->query(mode->mode, request, ptr);
-}

engine/voice_codecs/speex/source/libspeex/modes.h

@@ -1,146 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin */
-/**
-   @file modes.h
-   @brief Describes the different modes of the codec
-*/
-/*
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-
-*/
-
-#ifndef MODES_H
-#define MODES_H
-
-#include "speex.h"
-#include "speex_bits.h"
-
-
-#define NB_SUBMODES 16
-#define NB_SUBMODE_BITS 4
-
-#define SB_SUBMODES 8
-#define SB_SUBMODE_BITS 3
-
-
-/** Quantizes LSPs */
-typedef void (*lsp_quant_func)(float *, float *, int, SpeexBits *);
-
-/** Decodes quantized LSPs */
-typedef void (*lsp_unquant_func)(float *, int, SpeexBits *);
-
-
-/** Long-term predictor quantization */
-typedef int (*ltp_quant_func)(float *, float *, float *, float *, 
-                              float *, float *, void *, int, int, float, 
-                              int, int, SpeexBits*, char *, float *, float *, int);
-
-/** Long-term un-quantize */
-typedef void (*ltp_unquant_func)(float *, int, int, float, void *, int, int *,
-                                 float *, SpeexBits*, char*, int, int, float);
-
-
-/** Innovation quantization function */
-typedef void (*innovation_quant_func)(float *, float *, float *, float *, void *, int, int, 
-                                      float *, float *, SpeexBits *, char *, int);
-
-/** Innovation unquantization function */
-typedef void (*innovation_unquant_func)(float *, void *, int, SpeexBits*, char *);
-
-/** Description of a Speex sub-mode (wither narrowband or wideband */
-typedef struct SpeexSubmode {
-   int     lbr_pitch;          /**< Set to -1 for "normal" modes, otherwise encode pitch using a global pitch and allowing a +- lbr_pitch variation (for low not-rates)*/
-   int     forced_pitch_gain;  /**< Use the same (forced) pitch gain for all sub-frames */
-   int     have_subframe_gain; /**< Number of bits to use as sub-frame innovation gain */
-   int     double_codebook;    /**< Apply innovation quantization twice for higher quality (and higher bit-rate)*/
-   /*LSP functions*/
-   lsp_quant_func    lsp_quant; /**< LSP quantization function */
-   lsp_unquant_func  lsp_unquant; /**< LSP unquantization function */
-
-   /*Lont-term predictor functions*/
-   ltp_quant_func    ltp_quant; /**< Long-term predictor (pitch) quantizer */
-   ltp_unquant_func  ltp_unquant; /**< Long-term predictor (pitch) un-quantizer */
-   void             *ltp_params; /**< Pitch parameters (options) */
-
-   /*Quantization of innovation*/
-   innovation_quant_func innovation_quant; /**< Innovation quantization */
-   innovation_unquant_func innovation_unquant; /**< Innovation un-quantization */
-   void             *innovation_params; /**< Innovation quantization parameters*/
-
-   /*Synthesis filter enhancement*/
-   float             lpc_enh_k1; /**< Enhancer constant */
-   float             lpc_enh_k2; /**< Enhancer constant */
-   float             comb_gain;  /**< Gain of enhancer comb filter */
-
-   int               bits_per_frame; /**< Number of bits per frame after encoding*/
-} SpeexSubmode;
-
-/** Struct defining the encoding/decoding mode*/
-typedef struct SpeexNBMode {
-   int     frameSize;      /**< Size of frames used for encoding */
-   int     subframeSize;   /**< Size of sub-frames used for encoding */
-   int     lpcSize;        /**< Order of LPC filter */
-   int     bufSize;        /**< Size of signal buffer to use in encoder */
-   int     pitchStart;     /**< Smallest pitch value allowed */
-   int     pitchEnd;       /**< Largest pitch value allowed */
-
-   float   gamma1;         /**< Perceptual filter parameter #1 */
-   float   gamma2;         /**< Perceptual filter parameter #2 */
-   float   lag_factor;     /**< Lag-windowing parameter */
-   float   lpc_floor;      /**< Noise floor for LPC analysis */
-   float   preemph;        /**< Pre-emphasis */
-
-   SpeexSubmode *submodes[NB_SUBMODES]; /**< Sub-mode data for the mode */
-   int     defaultSubmode; /**< Default sub-mode to use when encoding */
-   int     quality_map[11]; /**< Mode corresponding to each quality setting */
-} SpeexNBMode;
-
-
-/** Struct defining the encoding/decoding mode for SB-CELP (wideband) */
-typedef struct SpeexSBMode {
-   SpeexMode *nb_mode;    /**< Embedded narrowband mode */
-   int     frameSize;     /**< Size of frames used for encoding */
-   int     subframeSize;  /**< Size of sub-frames used for encoding */
-   int     lpcSize;       /**< Order of LPC filter */
-   int     bufSize;       /**< Signal buffer size in encoder */
-   float   gamma1;        /**< Perceptual filter parameter #1 */
-   float   gamma2;        /**< Perceptual filter parameter #1 */
-   float   lag_factor;    /**< Lag-windowing parameter */
-   float   lpc_floor;     /**< Noise floor for LPC analysis */
-   float   preemph;       /**< Pre-emphasis */
-   float   folding_gain;
-
-   SpeexSubmode *submodes[SB_SUBMODES]; /**< Sub-mode data for the mode */
-   int     defaultSubmode; /**< Default sub-mode to use when encoding */
-   int     low_quality_map[11]; /**< Mode corresponding to each quality setting */
-   int     quality_map[11]; /**< Mode corresponding to each quality setting */
-   float   (*vbr_thresh)[11];
-   int     nb_modes;
-} SpeexSBMode;
-
-
-#endif

engine/voice_codecs/speex/source/libspeex/nb_celp.h

@@ -1,202 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin */
-/**
-    @file nb_celp.h
-    @brief Narrowband CELP encoder/decoder
-*/
-/*
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-
-*/
-
-#ifndef NB_CELP_H
-#define NB_CELP_H
-
-#include "modes.h"
-#include "speex_bits.h"
-#include "speex_callbacks.h"
-#include "vbr.h"
-#include "filters.h"
-
-/**Structure representing the full state of the narrowband encoder*/
-typedef struct EncState {
-   SpeexMode *mode;       /**< Mode corresponding to the state */
-   int    first;          /**< Is this the first frame? */
-   int    frameSize;      /**< Size of frames */
-   int    subframeSize;   /**< Size of sub-frames */
-   int    nbSubframes;    /**< Number of sub-frames */
-   int    windowSize;     /**< Analysis (LPC) window length */
-   int    lpcSize;        /**< LPC order */
-   int    bufSize;        /**< Buffer size */
-   int    min_pitch;      /**< Minimum pitch value allowed */
-   int    max_pitch;      /**< Maximum pitch value allowed */
-
-   int    safe_pitch;     /**< Don't use too large values for pitch (in case we lose a packet) */
-   int    bounded_pitch;  /**< Next frame should not rely on previous frames for pitch */
-   int    ol_pitch;       /**< Open-loop pitch */
-   int    ol_voiced;      /**< Open-loop voiced/non-voiced decision */
-   int   *pitch;
-   float  gamma1;         /**< Perceptual filter: A(z/gamma1) */
-   float  gamma2;         /**< Perceptual filter: A(z/gamma2) */
-   float  lag_factor;     /**< Lag windowing Gaussian width */
-   float  lpc_floor;      /**< Noise floor multiplier for A[0] in LPC analysis*/
-   float  preemph;        /**< Pre-emphasis: P(z) = 1 - a*z^-1*/
-   float  pre_mem;        /**< 1-element memory for pre-emphasis */
-   float  pre_mem2;       /**< 1-element memory for pre-emphasis */
-   char  *stack;          /**< Pseudo-stack allocation for temporary memory */
-   float *inBuf;          /**< Input buffer (original signal) */
-   float *frame;          /**< Start of original frame */
-   float *excBuf;         /**< Excitation buffer */
-   float *exc;            /**< Start of excitation frame */
-   float *exc2Buf;        /**< "Pitch enhanced" excitation */
-   float *exc2;           /**< "Pitch enhanced" excitation */
-   float *swBuf;          /**< Weighted signal buffer */
-   float *sw;             /**< Start of weighted signal frame */
-   float *innov;          /**< Innovation for the frame */
-   float *window;         /**< Temporary (Hanning) window */
-   float *buf2;           /**< 2nd temporary buffer */
-   float *autocorr;       /**< auto-correlation */
-   float *lagWindow;      /**< Window applied to auto-correlation */
-   float *lpc;            /**< LPCs for current frame */
-   float *lsp;            /**< LSPs for current frame */
-   float *qlsp;           /**< Quantized LSPs for current frame */
-   float *old_lsp;        /**< LSPs for previous frame */
-   float *old_qlsp;       /**< Quantized LSPs for previous frame */
-   float *interp_lsp;     /**< Interpolated LSPs */
-   float *interp_qlsp;    /**< Interpolated quantized LSPs */
-   float *interp_lpc;     /**< Interpolated LPCs */
-   float *interp_qlpc;    /**< Interpolated quantized LPCs */
-   float *bw_lpc1;        /**< LPCs after bandwidth expansion by gamma1 for perceptual weighting*/
-   float *bw_lpc2;        /**< LPCs after bandwidth expansion by gamma2 for perceptual weighting*/
-   float *rc;             /**< Reflection coefficients */
-   float *mem_sp;         /**< Filter memory for signal synthesis */
-   float *mem_sw;         /**< Filter memory for perceptually-weighted signal */
-   float *mem_sw_whole;   /**< Filter memory for perceptually-weighted signal (whole frame)*/
-   float *mem_exc;        /**< Filter memory for excitation (whole frame) */
-   float *pi_gain;        /**< Gain of LPC filter at theta=pi (fe/2) */
-
-   VBRState *vbr;         /**< State of the VBR data */
-   float  vbr_quality;    /**< Quality setting for VBR encoding */
-   float  relative_quality; /**< Relative quality that will be needed by VBR */
-   int    vbr_enabled;    /**< 1 for enabling VBR, 0 otherwise */
-   int    vad_enabled;    /**< 1 for enabling VAD, 0 otherwise */
-   int    dtx_enabled;    /**< 1 for enabling DTX, 0 otherwise */
-   int    dtx_count;      /**< Number of consecutive DTX frames */
-   int    abr_enabled;    /**< ABR setting (in bps), 0 if off */
-   float  abr_drift;
-   float  abr_drift2;
-   float  abr_count;
-   int    complexity;     /**< Complexity setting (0-10 from least complex to most complex) */
-   int    sampling_rate;
-
-   SpeexSubmode **submodes; /**< Sub-mode data */
-   int    submodeID;      /**< Activated sub-mode */
-   int    submodeSelect;  /**< Mode chosen by the user (may differ from submodeID if VAD is on) */
-} EncState;
-
-/**Structure representing the full state of the narrowband decoder*/
-typedef struct DecState {
-   SpeexMode *mode;       /**< Mode corresponding to the state */
-   int    first;          /**< Is this the first frame? */
-   int    count_lost;     /**< Was the last frame lost? */
-   int    frameSize;      /**< Size of frames */
-   int    subframeSize;   /**< Size of sub-frames */
-   int    nbSubframes;    /**< Number of sub-frames */
-   int    windowSize;     /**< Analysis (LPC) window length */
-   int    lpcSize;        /**< LPC order */
-   int    bufSize;        /**< Buffer size */
-   int    min_pitch;      /**< Minimum pitch value allowed */
-   int    max_pitch;      /**< Maximum pitch value allowed */
-   int    sampling_rate;
-   float  last_ol_gain;   /**< Open-loop gain for previous frame */
-
-
-   float  gamma1;         /**< Perceptual filter: A(z/gamma1) */
-   float  gamma2;         /**< Perceptual filter: A(z/gamma2) */
-   float  preemph;        /**< Pre-emphasis: P(z) = 1 - a*z^-1*/
-   float  pre_mem;        /**< 1-element memory for pre-emphasis */
-   char  *stack;          /**< Pseudo-stack allocation for temporary memory */
-   float *inBuf;          /**< Input buffer (original signal) */
-   float *frame;          /**< Start of original frame */
-   float *excBuf;         /**< Excitation buffer */
-   float *exc;            /**< Start of excitation frame */
-   float *innov;          /**< Innovation for the frame */
-   float *qlsp;           /**< Quantized LSPs for current frame */
-   float *old_qlsp;       /**< Quantized LSPs for previous frame */
-   float *interp_qlsp;    /**< Interpolated quantized LSPs */
-   float *interp_qlpc;    /**< Interpolated quantized LPCs */
-   float *mem_sp;         /**< Filter memory for synthesis signal */
-   float *pi_gain;        /**< Gain of LPC filter at theta=pi (fe/2) */
-   int    last_pitch;     /**< Pitch of last correctly decoded frame */
-   float  last_pitch_gain; /**< Pitch gain of last correctly decoded frame */
-   float  pitch_gain_buf[3];  /**< Pitch gain of last decoded frames */
-   int    pitch_gain_buf_idx; /**< Tail of the buffer */
-
-   SpeexSubmode **submodes; /**< Sub-mode data */
-   int    submodeID;      /**< Activated sub-mode */
-   int    lpc_enh_enabled; /**< 1 when LPC enhancer is on, 0 otherwise */
-   CombFilterMem *comb_mem;
-   SpeexCallback speex_callbacks[SPEEX_MAX_CALLBACKS];
-
-   SpeexCallback user_callback;
-
-   /*Vocoder data*/
-   float  voc_m1;
-   float  voc_m2;
-   float  voc_mean;
-   int    voc_offset;
-
-   int    dtx_enabled;
-} DecState;
-
-/** Initializes encoder state*/
-void *nb_encoder_init(SpeexMode *m);
-
-/** De-allocates encoder state resources*/
-void nb_encoder_destroy(void *state);
-
-/** Encodes one frame*/
-int nb_encode(void *state, float *in, SpeexBits *bits);
-
-
-/** Initializes decoder state*/
-void *nb_decoder_init(SpeexMode *m);
-
-/** De-allocates decoder state resources*/
-void nb_decoder_destroy(void *state);
-
-/** Decodes one frame*/
-int nb_decode(void *state, SpeexBits *bits, float *out);
-
-/** ioctl-like function for controlling a narrowband encoder */
-int nb_encoder_ctl(void *state, int request, void *ptr);
-
-/** ioctl-like function for controlling a narrowband decoder */
-int nb_decoder_ctl(void *state, int request, void *ptr);
-
-
-#endif

engine/voice_codecs/speex/source/libspeex/quant_lsp.c

@@ -1,311 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: quant_lsp.c
-   LSP vector quantization
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#include "quant_lsp.h"
-#include <math.h>
-
-extern int lsp_nb_vqid[64];
-static float quant_weight[MAX_LSP_SIZE];
-
-/* Note: x is modified*/
-static int lsp_quant(float *x, signed char *cdbk, int nbVec, int nbDim)
-{
-   int i,j;
-   float dist, tmp;
-   float best_dist=0;
-   int best_id=0;
-   signed char *ptr=cdbk;
-   for (i=0;i<nbVec;i++)
-   {
-      dist=0;
-      for (j=0;j<nbDim;j++)
-      {
-         tmp=(x[j]-*ptr++);
-         dist+=tmp*tmp;
-      }
-      if (dist<best_dist || i==0)
-      {
-         best_dist=dist;
-         best_id=i;
-      }
-   }
-
-   for (j=0;j<nbDim;j++)
-      x[j] -= cdbk[best_id*nbDim+j];
-    
-   return best_id;
-}
-
-/* Note: x is modified*/
-static int lsp_weight_quant(float *x, float *weight, signed char *cdbk, int nbVec, int nbDim)
-{
-   int i,j;
-   float dist, tmp;
-   float best_dist=0;
-   int best_id=0;
-   signed char *ptr=cdbk;
-   for (i=0;i<nbVec;i++)
-   {
-      dist=0;
-      for (j=0;j<nbDim;j++)
-      {
-         tmp=(x[j]-*ptr++);
-         dist+=weight[j]*tmp*tmp;
-      }
-      if (dist<best_dist || i==0)
-      {
-         best_dist=dist;
-         best_id=i;
-      }
-   }
-   
-   for (j=0;j<nbDim;j++)
-      x[j] -= cdbk[best_id*nbDim+j];
-   return best_id;
-}
-
-
-void lsp_quant_nb(float *lsp, float *qlsp, int order, SpeexBits *bits)
-{
-   int i;
-   float tmp1, tmp2;
-   int id;
-
-   for (i=0;i<order;i++)
-      qlsp[i]=lsp[i];
-
-   quant_weight[0] = 1/(qlsp[1]-qlsp[0]);
-   quant_weight[order-1] = 1/(qlsp[order-1]-qlsp[order-2]);
-   for (i=1;i<order-1;i++)
-   {
-#if 1
-      tmp1 = 1/((.15+qlsp[i]-qlsp[i-1])*(.15+qlsp[i]-qlsp[i-1]));
-      tmp2 = 1/((.15+qlsp[i+1]-qlsp[i])*(.15+qlsp[i+1]-qlsp[i]));
-#else
-      tmp1 = 1/(qlsp[i]-qlsp[i-1]);
-      tmp2 = 1/(qlsp[i+1]-qlsp[i]);
-#endif
-      quant_weight[i] = tmp1 > tmp2 ? tmp1 : tmp2;
-   }
-   for (i=0;i<order;i++)
-      qlsp[i]-=(.25*i+.25);
-   for (i=0;i<order;i++)
-      qlsp[i]*=256;
-
-   id = lsp_quant(qlsp, cdbk_nb, NB_CDBK_SIZE, order);
-   speex_bits_pack(bits, id, 6);
-
-   for (i=0;i<order;i++)
-      qlsp[i]*=2;
- 
-   id = lsp_weight_quant(qlsp, quant_weight, cdbk_nb_low1, NB_CDBK_SIZE_LOW1, 5);
-   speex_bits_pack(bits, id, 6);
-
-   for (i=0;i<5;i++)
-      qlsp[i]*=2;
-
-   id = lsp_weight_quant(qlsp, quant_weight, cdbk_nb_low2, NB_CDBK_SIZE_LOW2, 5);
-   speex_bits_pack(bits, id, 6);
-
-   id = lsp_weight_quant(qlsp+5, quant_weight+5, cdbk_nb_high1, NB_CDBK_SIZE_HIGH1, 5);
-   speex_bits_pack(bits, id, 6);
-
-   for (i=5;i<10;i++)
-      qlsp[i]*=2;
-
-   id = lsp_weight_quant(qlsp+5, quant_weight+5, cdbk_nb_high2, NB_CDBK_SIZE_HIGH2, 5);
-   speex_bits_pack(bits, id, 6);
-
-   for (i=0;i<order;i++)
-      qlsp[i]*=.00097656;
-
-   for (i=0;i<order;i++)
-      qlsp[i]=lsp[i]-qlsp[i];
-}
-
-void lsp_unquant_nb(float *lsp, int order, SpeexBits *bits)
-{
-   int i, id;
-   for (i=0;i<order;i++)
-      lsp[i]=.25*i+.25;
-
-
-   id=speex_bits_unpack_unsigned(bits, 6);
-   for (i=0;i<10;i++)
-      lsp[i] += 0.0039062*cdbk_nb[id*10+i];
-
-   id=speex_bits_unpack_unsigned(bits, 6);
-   for (i=0;i<5;i++)
-      lsp[i] += 0.0019531 * cdbk_nb_low1[id*5+i];
-
-   id=speex_bits_unpack_unsigned(bits, 6);
-   for (i=0;i<5;i++)
-      lsp[i] +=  0.00097656 * cdbk_nb_low2[id*5+i];
-
-   id=speex_bits_unpack_unsigned(bits, 6);
-   for (i=0;i<5;i++)
-      lsp[i+5] += 0.0019531 * cdbk_nb_high1[id*5+i];
-   
-   id=speex_bits_unpack_unsigned(bits, 6);
-   for (i=0;i<5;i++)
-      lsp[i+5] += 0.00097656 * cdbk_nb_high2[id*5+i];
-}
-
-
-void lsp_quant_lbr(float *lsp, float *qlsp, int order, SpeexBits *bits)
-{
-   int i;
-   float tmp1, tmp2;
-   int id;
-
-   for (i=0;i<order;i++)
-      qlsp[i]=lsp[i];
-
-   quant_weight[0] = 1/(qlsp[1]-qlsp[0]);
-   quant_weight[order-1] = 1/(qlsp[order-1]-qlsp[order-2]);
-   for (i=1;i<order-1;i++)
-   {
-#if 1
-      tmp1 = 1/((.15+qlsp[i]-qlsp[i-1])*(.15+qlsp[i]-qlsp[i-1]));
-      tmp2 = 1/((.15+qlsp[i+1]-qlsp[i])*(.15+qlsp[i+1]-qlsp[i]));
-#else
-      tmp1 = 1/(qlsp[i]-qlsp[i-1]);
-      tmp2 = 1/(qlsp[i+1]-qlsp[i]);
-#endif
-      quant_weight[i] = tmp1 > tmp2 ? tmp1 : tmp2;
-   }
-
-   for (i=0;i<order;i++)
-      qlsp[i]-=(.25*i+.25);
-   for (i=0;i<order;i++)
-      qlsp[i]*=256;
-   
-   id = lsp_quant(qlsp, cdbk_nb, NB_CDBK_SIZE, order);
-   speex_bits_pack(bits, id, 6);
-
-   for (i=0;i<order;i++)
-      qlsp[i]*=2;
-   
-   id = lsp_weight_quant(qlsp, quant_weight, cdbk_nb_low1, NB_CDBK_SIZE_LOW1, 5);
-   speex_bits_pack(bits, id, 6);
-
-   id = lsp_weight_quant(qlsp+5, quant_weight+5, cdbk_nb_high1, NB_CDBK_SIZE_HIGH1, 5);
-   speex_bits_pack(bits, id, 6);
-
-   for (i=0;i<order;i++)
-      qlsp[i]*=0.0019531;
-
-   for (i=0;i<order;i++)
-      qlsp[i]=lsp[i]-qlsp[i];
-}
-
-void lsp_unquant_lbr(float *lsp, int order, SpeexBits *bits)
-{
-   int i, id;
-   for (i=0;i<order;i++)
-      lsp[i]=.25*i+.25;
-
-
-   id=speex_bits_unpack_unsigned(bits, 6);
-   for (i=0;i<10;i++)
-      lsp[i] += 0.0039062*cdbk_nb[id*10+i];
-
-   id=speex_bits_unpack_unsigned(bits, 6);
-   for (i=0;i<5;i++)
-      lsp[i] += 0.0019531*cdbk_nb_low1[id*5+i];
-
-   id=speex_bits_unpack_unsigned(bits, 6);
-   for (i=0;i<5;i++)
-      lsp[i+5] += 0.0019531*cdbk_nb_high1[id*5+i];
-   
-}
-
-
-extern signed char high_lsp_cdbk[];
-extern signed char high_lsp_cdbk2[];
-
-
-void lsp_quant_high(float *lsp, float *qlsp, int order, SpeexBits *bits)
-{
-   int i;
-   float tmp1, tmp2;
-   int id;
-   for (i=0;i<order;i++)
-      qlsp[i]=lsp[i];
-
-   quant_weight[0] = 1/(qlsp[1]-qlsp[0]);
-   quant_weight[order-1] = 1/(qlsp[order-1]-qlsp[order-2]);
-   for (i=1;i<order-1;i++)
-   {
-      tmp1 = 1/(qlsp[i]-qlsp[i-1]);
-      tmp2 = 1/(qlsp[i+1]-qlsp[i]);
-      quant_weight[i] = tmp1 > tmp2 ? tmp1 : tmp2;
-   }
-
-   for (i=0;i<order;i++)
-      qlsp[i]-=(.3125*i+.75);
-   for (i=0;i<order;i++)
-      qlsp[i]*=256;
-
-   id = lsp_quant(qlsp, high_lsp_cdbk, 64, order);
-   speex_bits_pack(bits, id, 6);
-
-   for (i=0;i<order;i++)
-      qlsp[i]*=2;
-
-   id = lsp_weight_quant(qlsp, quant_weight, high_lsp_cdbk2, 64, order);
-   speex_bits_pack(bits, id, 6);
-
-   for (i=0;i<order;i++)
-      qlsp[i]*=0.0019531;
-
-   for (i=0;i<order;i++)
-      qlsp[i]=lsp[i]-qlsp[i];
-}
-
-void lsp_unquant_high(float *lsp, int order, SpeexBits *bits)
-{
-
-   int i, id;
-   for (i=0;i<order;i++)
-      lsp[i]=.3125*i+.75;
-
-
-   id=speex_bits_unpack_unsigned(bits, 6);
-   for (i=0;i<order;i++)
-      lsp[i] += 0.0039062*high_lsp_cdbk[id*order+i];
-
-
-   id=speex_bits_unpack_unsigned(bits, 6);
-   for (i=0;i<order;i++)
-      lsp[i] += 0.0019531*high_lsp_cdbk2[id*order+i];
-}

engine/voice_codecs/speex/source/libspeex/quant_lsp.h

@@ -1,71 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: quant_lsp.h
-   LSP vector quantization
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#ifndef QUANT_LSP_H
-#define QUANT_LSP_H
-
-#include "speex_bits.h"
-
-#define MAX_LSP_SIZE 20
-
-#define NB_CDBK_SIZE 64
-#define NB_CDBK_SIZE_LOW1 64
-#define NB_CDBK_SIZE_LOW2 64
-#define NB_CDBK_SIZE_HIGH1 64
-#define NB_CDBK_SIZE_HIGH2 64
-
-/*Narrowband codebooks*/
-extern signed char cdbk_nb[];
-extern signed char cdbk_nb_low1[];
-extern signed char cdbk_nb_low2[];
-extern signed char cdbk_nb_high1[];
-extern signed char cdbk_nb_high2[];
-
-/* Quantizes narrowband LSPs with 30 bits */
-void lsp_quant_nb(float *lsp, float *qlsp, int order, SpeexBits *bits);
-
-/* Decodes quantized narrowband LSPs */
-void lsp_unquant_nb(float *lsp, int order, SpeexBits *bits);
-
-/* Quantizes low bit-rate narrowband LSPs with 18 bits */
-void lsp_quant_lbr(float *lsp, float *qlsp, int order, SpeexBits *bits);
-
-/* Decodes quantized low bit-rate narrowband LSPs */
-void lsp_unquant_lbr(float *lsp, int order, SpeexBits *bits);
-
-/* Quantizes high-band LSPs with 12 bits */
-void lsp_quant_high(float *lsp, float *qlsp, int order, SpeexBits *bits);
-
-/* Decodes high-band LSPs */
-void lsp_unquant_high(float *lsp, int order, SpeexBits *bits);
-
-#endif

engine/voice_codecs/speex/source/libspeex/sb_celp.h

@@ -1,167 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin */
-/**
-   @file sb_celp.h
-   @brief Sub-band CELP mode used for wideband encoding
-*/
-/*
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-
-*/
-
-#ifndef SB_CELP_H
-#define SB_CELP_H
-
-#include "modes.h"
-#include "speex_bits.h"
-#include "nb_celp.h"
-
-/**Structure representing the full state of the sub-band encoder*/
-typedef struct SBEncState {
-   SpeexMode *mode;            /**< Pointer to the mode (containing for vtable info) */
-   void *st_low;               /**< State of the low-band (narrowband) encoder */
-   int    full_frame_size;     /**< Length of full-band frames*/
-   int    frame_size;          /**< Length of high-band frames*/
-   int    subframeSize;        /**< Length of high-band sub-frames*/
-   int    nbSubframes;         /**< Number of high-band sub-frames*/
-   int    windowSize;          /**< Length of high-band LPC window*/
-   int    lpcSize;             /**< Order of high-band LPC analysis */
-   int    bufSize;             /**< Buffer size */
-   int    first;               /**< First frame? */
-   float  lag_factor;          /**< Lag-windowing control parameter */
-   float  lpc_floor;           /**< Controls LPC analysis noise floor */
-   float  gamma1;              /**< Perceptual weighting coef 1 */
-   float  gamma2;              /**< Perceptual weighting coef 2 */
-
-   char  *stack;               /**< Temporary allocation stack */
-   float *x0d, *x1d; /**< QMF filter signals*/
-   float *high;                /**< High-band signal (buffer) */
-   float *y0, *y1;             /**< QMF synthesis signals */
-   float *h0_mem, *h1_mem, *g0_mem, *g1_mem; /**< QMF memories */
-
-   float *excBuf;              /**< High-band excitation */
-   float *exc;                 /**< High-band excitation (for QMF only)*/
-   float *buf;                 /**< Temporary buffer */
-   float *res;                 /**< Zero-input response (ringing) */
-   float *sw;                  /**< Perceptually weighted signal */
-   float *target;              /**< Weighted target signal (analysis by synthesis) */
-   float *window;              /**< LPC analysis window */
-   float *lagWindow;           /**< Auto-correlation window */
-   float *autocorr;            /**< Auto-correlation (for LPC analysis) */
-   float *rc;                  /**< Reflection coefficients (unused) */
-   float *lpc;                 /**< LPC coefficients */
-   float *lsp;                 /**< LSP coefficients */
-   float *qlsp;                /**< Quantized LSPs */
-   float *old_lsp;             /**< LSPs of previous frame */
-   float *old_qlsp;            /**< Quantized LSPs of previous frame */
-   float *interp_lsp;          /**< Interpolated LSPs for current sub-frame */
-   float *interp_qlsp;         /**< Interpolated quantized LSPs for current sub-frame */
-   float *interp_lpc;          /**< Interpolated LPCs for current sub-frame */
-   float *interp_qlpc;         /**< Interpolated quantized LPCs for current sub-frame */
-   float *bw_lpc1;             /**< Bandwidth-expanded version of LPCs (#1) */
-   float *bw_lpc2;             /**< Bandwidth-expanded version of LPCs (#2) */
-
-   float *mem_sp;              /**< Synthesis signal memory */
-   float *mem_sp2;
-   float *mem_sw;              /**< Perceptual signal memory */
-   float *pi_gain;
-
-   float  vbr_quality;         /**< Quality setting for VBR encoding */
-   int    vbr_enabled;         /**< 1 for enabling VBR, 0 otherwise */
-   int    abr_enabled;         /**< ABR setting (in bps), 0 if off */
-   float  abr_drift;
-   float  abr_drift2;
-   float  abr_count;
-   int    vad_enabled;         /**< 1 for enabling VAD, 0 otherwise */
-   float  relative_quality;
-
-   SpeexSubmode **submodes;
-   int    submodeID;
-   int    submodeSelect;
-   int    complexity;
-   int    sampling_rate;
-
-} SBEncState;
-
-
-/**Structure representing the full state of the sub-band decoder*/
-typedef struct SBDecState {
-   SpeexMode *mode;            /**< Pointer to the mode (containing for vtable info) */
-   void *st_low;               /**< State of the low-band (narrowband) encoder */
-   int    full_frame_size;
-   int    frame_size;
-   int    subframeSize;
-   int    nbSubframes;
-   int    lpcSize;
-   int    first;
-   int    sampling_rate;
-   int    lpc_enh_enabled;
-
-   char  *stack;
-   float *x0d, *x1d;
-   float *high;
-   float *y0, *y1;
-   float *h0_mem, *h1_mem, *g0_mem, *g1_mem;
-
-   float *exc;
-   float *qlsp;
-   float *old_qlsp;
-   float *interp_qlsp;
-   float *interp_qlpc;
-
-   float *mem_sp;
-   float *pi_gain;
-
-   SpeexSubmode **submodes;
-   int    submodeID;
-} SBDecState;
-
-
-/**Initializes encoder state*/
-void *sb_encoder_init(SpeexMode *m);
-
-/**De-allocates encoder state resources*/
-void sb_encoder_destroy(void *state);
-
-/**Encodes one frame*/
-int sb_encode(void *state, float *in, SpeexBits *bits);
-
-
-/**Initializes decoder state*/
-void *sb_decoder_init(SpeexMode *m);
-
-/**De-allocates decoder state resources*/
-void sb_decoder_destroy(void *state);
-
-/**Decodes one frame*/
-int sb_decode(void *state, SpeexBits *bits, float *out);
-
-int sb_encoder_ctl(void *state, int request, void *ptr);
-
-int sb_decoder_ctl(void *state, int request, void *ptr);
-
-#endif

engine/voice_codecs/speex/source/libspeex/speex_callbacks.c

@@ -1,135 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin
-   File speex_callbacks.c
-   Callback handling and in-band signalling
-
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-
-*/
-
-#include "speex_callbacks.h"
-#include "misc.h"
-
-int speex_inband_handler(SpeexBits *bits, SpeexCallback *callback_list, void *state)
-{
-   int id;
-   SpeexCallback *callback;
-   /*speex_bits_advance(bits, 5);*/
-   id=speex_bits_unpack_unsigned(bits, 4);
-   callback = callback_list+id;
-
-   if (callback->func)
-   {
-      return callback->func(bits, state, callback->data);
-   } else
-      /*If callback is not registered, skip the right number of bits*/
-   {
-      int adv;
-      if (id<2)
-         adv = 1;
-      else if (id<8)
-         adv = 4;
-      else if (id<10)
-         adv = 8;
-      else if (id<12)
-         adv = 16;
-      else if (id<14)
-         adv = 32;
-      else 
-         adv = 64;
-      speex_bits_advance(bits, adv);
-   }
-   return 0;
-}
-
-int speex_std_mode_request_handler(SpeexBits *bits, void *state, void *data)
-{
-   int m;
-   m = speex_bits_unpack_unsigned(bits, 4);
-   speex_encoder_ctl(data, SPEEX_SET_MODE, &m);
-   return 0;
-}
-
-int speex_std_low_mode_request_handler(SpeexBits *bits, void *state, void *data)
-{
-   int m;
-   m = speex_bits_unpack_unsigned(bits, 4);
-   speex_encoder_ctl(data, SPEEX_SET_LOW_MODE, &m);
-   return 0;
-}
-
-int speex_std_high_mode_request_handler(SpeexBits *bits, void *state, void *data)
-{
-   int m;
-   m = speex_bits_unpack_unsigned(bits, 4);
-   speex_encoder_ctl(data, SPEEX_SET_HIGH_MODE, &m);
-   return 0;
-}
-
-int speex_std_vbr_request_handler(SpeexBits *bits, void *state, void *data)
-{
-   int vbr;
-   vbr = speex_bits_unpack_unsigned(bits, 1);
-   speex_encoder_ctl(data, SPEEX_SET_VBR, &vbr);
-   return 0;
-}
-
-int speex_std_enh_request_handler(SpeexBits *bits, void *state, void *data)
-{
-   int enh;
-   enh = speex_bits_unpack_unsigned(bits, 1);
-   speex_decoder_ctl(data, SPEEX_SET_ENH, &enh);
-   return 0;
-}
-
-int speex_std_vbr_quality_request_handler(SpeexBits *bits, void *state, void *data)
-{
-   int qual;
-   qual = speex_bits_unpack_unsigned(bits, 4);
-   speex_encoder_ctl(data, SPEEX_SET_VBR_QUALITY, &qual);
-   return 0;
-}
-
-
-int speex_std_char_handler(SpeexBits *bits, void *state, void *data)
-{
-   unsigned char ch;
-   ch = speex_bits_unpack_unsigned(bits, 8);
-   _speex_putc(ch, data);
-   return 0;
-}
-
-
-
-/* Default handler for user callbacks: skip it */
-int speex_default_user_handler(SpeexBits *bits, void *state, void *data)
-{
-   int req_size = speex_bits_unpack_unsigned(bits, 4);
-   speex_bits_advance(bits, 5+8*req_size);
-   return 0;
-}

engine/voice_codecs/speex/source/libspeex/speex_header.c

@@ -1,162 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: speex_header.c
-   Describes the Speex header
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-
-*/
-
-#include "speex_header.h"
-#include "misc.h"
-#include "speex.h"
-
-#ifndef NULL
-#define NULL 0
-#endif
-
-#define ENDIAN_SWITCH(x) {x=le_int(x);}
-
-
-/*
-typedef struct SpeexHeader {
-   char speex_string[8];
-   char speex_version[SPEEX_HEADER_VERSION_LENGTH];
-   int speex_version_id;
-   int header_size;
-   int rate;
-   int mode;
-   int mode_bitstream_version;
-   int nb_channels;
-   int bitrate;
-   int frame_size;
-   int vbr;
-   int frames_per_packet;
-   int extra_headers;
-   int reserved1;
-   int reserved2;
-} SpeexHeader;
-*/
-
-void speex_init_header(SpeexHeader *header, int rate, int nb_channels, SpeexMode *m)
-{
-   int i;
-   char *h="Speex   ";
-   /*
-   strncpy(header->speex_string, "Speex   ", 8);
-   strncpy(header->speex_version, VERSION, SPEEX_HEADER_VERSION_LENGTH-1);
-   header->speex_version[SPEEX_HEADER_VERSION_LENGTH-1]=0;
-   */
-   for (i=0;i<8;i++)
-      header->speex_string[i]=h[i];
-   for (i=0;i<SPEEX_HEADER_VERSION_LENGTH-1 && VERSION[i];i++)
-      header->speex_version[i]=VERSION[i];
-   for (;i<SPEEX_HEADER_VERSION_LENGTH;i++)
-      header->speex_version[i]=0;
-   
-   header->speex_version_id = 1;
-   header->header_size = sizeof(SpeexHeader);
-   
-   header->rate = rate;
-   header->mode = m->modeID;
-   header->mode_bitstream_version = m->bitstream_version;
-   if (m->modeID<0)
-      speex_warning("This mode is meant to be used alone");
-   header->nb_channels = nb_channels;
-   header->bitrate = -1;
-   speex_mode_query(m, SPEEX_MODE_FRAME_SIZE, &header->frame_size);
-   header->vbr = 0;
-   
-   header->frames_per_packet = 0;
-   header->extra_headers = 0;
-   header->reserved1 = 0;
-   header->reserved2 = 0;
-}
-
-char *speex_header_to_packet(SpeexHeader *header, int *size)
-{
-   SpeexHeader *le_header;
-   le_header = (SpeexHeader*)speex_alloc(sizeof(SpeexHeader));
-   
-   speex_move(le_header, header, sizeof(SpeexHeader));
-   
-   /*Make sure everything is now little-endian*/
-   ENDIAN_SWITCH(le_header->speex_version_id);
-   ENDIAN_SWITCH(le_header->header_size);
-   ENDIAN_SWITCH(le_header->rate);
-   ENDIAN_SWITCH(le_header->mode);
-   ENDIAN_SWITCH(le_header->mode_bitstream_version);
-   ENDIAN_SWITCH(le_header->nb_channels);
-   ENDIAN_SWITCH(le_header->bitrate);
-   ENDIAN_SWITCH(le_header->frame_size);
-   ENDIAN_SWITCH(le_header->vbr);
-   ENDIAN_SWITCH(le_header->frames_per_packet);
-   ENDIAN_SWITCH(le_header->extra_headers);
-
-   *size = sizeof(SpeexHeader);
-   return (char *)le_header;
-}
-
-SpeexHeader *speex_packet_to_header(char *packet, int size)
-{
-   int i;
-   SpeexHeader *le_header;
-   char *h = "Speex   ";
-   for (i=0;i<8;i++)
-      if (packet[i]!=h[i])
-      {
-         speex_warning ("This doesn't look like a Speex file");
-         return NULL;
-      }
-   
-   /*FIXME: Do we allow larger headers?*/
-   if (size < sizeof(SpeexHeader))
-   {
-      speex_warning("Speex header too small");
-      return NULL;
-   }
-   
-   le_header = (SpeexHeader*)speex_alloc(sizeof(SpeexHeader));
-   
-   speex_move(le_header, packet, sizeof(SpeexHeader));
-   
-   /*Make sure everything is converted correctly from little-endian*/
-   ENDIAN_SWITCH(le_header->speex_version_id);
-   ENDIAN_SWITCH(le_header->header_size);
-   ENDIAN_SWITCH(le_header->rate);
-   ENDIAN_SWITCH(le_header->mode);
-   ENDIAN_SWITCH(le_header->mode_bitstream_version);
-   ENDIAN_SWITCH(le_header->nb_channels);
-   ENDIAN_SWITCH(le_header->bitrate);
-   ENDIAN_SWITCH(le_header->frame_size);
-   ENDIAN_SWITCH(le_header->vbr);
-   ENDIAN_SWITCH(le_header->frames_per_packet);
-   ENDIAN_SWITCH(le_header->extra_headers);
-
-   return le_header;
-
-}

engine/voice_codecs/speex/source/libspeex/stack_alloc.h

@@ -1,62 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: stack_alloc.h
-   
-   Temporary memory allocation on stack
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#ifndef STACK_ALLOC_H
-#define STACK_ALLOC_H
-
-#if 0
-
-/*Aligns the stack to a 'size' boundary */
-#define ALIGN(stack, size) (stack=(void*)((((int)stack)+((size)-1)) & (-(size))))
-/*Aligns the stack to a 'size' boundary minus k */
-#define ALIGN_1(stack, size, k) (stack=(void*)(((((int)stack)+((size)-1+(k))) & (-(size)))-(k)))
-
-/* Allocates 'size' elements of type 'type' on the stack */
-#define PUSH(stack, size, type) (ALIGN(stack,sizeof(type)),stack=(void*)(((int)stack)+((size)*sizeof(type))),(type*)(((int)stack)-((size)*sizeof(type))))
-
-
-#else
-
-/*Aligns the stack to a 'size' boundary */
-#define ALIGN(stack, size) ((stack) += ((size) - (int)(stack)) & ((size) - 1))
-
-/* Allocates 'size' elements of type 'type' on the stack */
-#define PUSH(stack, size, type) (ALIGN((stack),sizeof(type)),(stack)+=((size)*sizeof(type)),(type*)((stack)-((size)*sizeof(type))))
-
-/* Allocates a struct stack */
-#define PUSHS(stack, type) (ALIGN((stack),sizeof(long)),(stack)+=(sizeof(type)),(type*)((stack)-(sizeof(type))))
-
-
-#endif
-
-#endif

engine/voice_codecs/speex/source/libspeex/stereo.c

@@ -1,122 +0,0 @@
-/* Copyright (C) 2002 Jean-Marc Valin 
-   File: stereo.c
-
-   Redistribution and use in source and binary forms, with or without
-   modification, are permitted provided that the following conditions
-   are met:
-   
-   - Redistributions of source code must retain the above copyright
-   notice, this list of conditions and the following disclaimer.
-   
-   - Redistributions in binary form must reproduce the above copyright
-   notice, this list of conditions and the following disclaimer in the
-   documentation and/or other materials provided with the distribution.
-   
-   - Neither the name of the Xiph.org Foundation nor the names of its
-   contributors may be used to endorse or promote products derived from
-   this software without specific prior written permission.
-   
-   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
-   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
-   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
-   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
-   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
-   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
-   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
-   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#include "speex_stereo.h"
-#include "speex_callbacks.h"
-#include "vq.h"
-#include <math.h>
-
-/*float e_ratio_quant[4] = {1, 1.26, 1.587, 2};*/
-static float e_ratio_quant[4] = {.25, .315, .397, .5};
-
-void speex_encode_stereo(float *data, int frame_size, SpeexBits *bits)
-{
-   int i, tmp;
-   float e_left=0, e_right=0, e_tot=0;
-   float balance, e_ratio;
-   for (i=0;i<frame_size;i++)
-   {
-      e_left  += data[2*i]*data[2*i];
-      e_right += data[2*i+1]*data[2*i+1];
-      data[i] =  .5*(data[2*i]+data[2*i+1]);
-      e_tot   += data[i]*data[i];
-   }
-   balance=(e_left+1)/(e_right+1);
-   e_ratio = e_tot/(1+e_left+e_right);
-
-   /*Quantization*/
-   speex_bits_pack(bits, 14, 5);
-   speex_bits_pack(bits, SPEEX_INBAND_STEREO, 4);
-   
-   balance=4*log(balance);
-
-   /*Pack sign*/
-   if (balance>0)
-      speex_bits_pack(bits, 0, 1);
-   else
-      speex_bits_pack(bits, 1, 1);
-   balance=floor(.5+fabs(balance));
-   if (balance>30)
-      balance=31;
-   
-   speex_bits_pack(bits, (int)balance, 5);
-   
-   /*Quantize energy ratio*/
-   tmp=vq_index(&e_ratio, e_ratio_quant, 1, 4);
-   speex_bits_pack(bits, tmp, 2);
-}
-
-void speex_decode_stereo(float *data, int frame_size, SpeexStereoState *stereo)
-{
-   float balance, e_ratio;
-   int i;
-   float e_tot=0, e_left, e_right, e_sum;
-
-   balance=stereo->balance;
-   e_ratio=stereo->e_ratio;
-   for (i=frame_size-1;i>=0;i--)
-   {
-      e_tot += data[i]*data[i];
-   }
-   e_sum=e_tot/e_ratio;
-   e_left  = e_sum*balance / (1+balance);
-   e_right = e_sum-e_left;
-
-   e_left  = sqrt(e_left/(e_tot+.01));
-   e_right = sqrt(e_right/(e_tot+.01));
-
-   for (i=frame_size-1;i>=0;i--)
-   {
-      float ftmp=data[i];
-      stereo->smooth_left  = .98*stereo->smooth_left  + .02*e_left;
-      stereo->smooth_right = .98*stereo->smooth_right + .02*e_right;
-      data[2*i] = stereo->smooth_left*ftmp;
-      data[2*i+1] = stereo->smooth_right*ftmp;
-   }
-}
-
-int speex_std_stereo_request_handler(SpeexBits *bits, void *state, void *data)
-{
-   SpeexStereoState *stereo;
-   float sign=1;
-   int tmp;
-
-   stereo = (SpeexStereoState*)data;
-   if (speex_bits_unpack_unsigned(bits, 1))
-      sign=-1;
-   tmp = speex_bits_unpack_unsigned(bits, 5);
-   stereo->balance = exp(sign*.25*tmp);
-
-   tmp = speex_bits_unpack_unsigned(bits, 2);
-   stereo->e_ratio = e_ratio_quant[tmp];
-
-   return 0;
-}

engine/voice_codecs/speex/source/libspeex/testenc.c

@@ -1,127 +0,0 @@
-#include "speex.h"
-#include <stdio.h>
-#include <stdlib.h>
-#include "speex_callbacks.h"
-
-#define FRAME_SIZE 160
-#include <math.h>
-int main(int argc, char **argv)
-{
-   char *inFile, *outFile, *bitsFile;
-   FILE *fin, *fout, *fbits=NULL;
-   short in[FRAME_SIZE];
-   float input[FRAME_SIZE], bak[FRAME_SIZE], bak2[FRAME_SIZE];
-   char cbits[200];
-   int nbBits;
-   int i;
-   void *st;
-   void *dec;
-   SpeexBits bits;
-   int tmp;
-   int bitCount=0;
-   SpeexCallback callback;
-
-   for (i=0;i<FRAME_SIZE;i++)
-      bak2[i]=0;
-   st = speex_encoder_init(&speex_nb_mode);
-   dec = speex_decoder_init(&speex_nb_mode);
-
-   callback.callback_id = SPEEX_INBAND_CHAR;
-   callback.func = speex_std_char_handler;
-   callback.data = stderr;
-   speex_decoder_ctl(dec, SPEEX_SET_HANDLER, &callback);
-
-   callback.callback_id = SPEEX_INBAND_MODE_REQUEST;
-   callback.func = speex_std_mode_request_handler;
-   callback.data = st;
-   speex_decoder_ctl(dec, SPEEX_SET_HANDLER, &callback);
-
-   tmp=0;
-   speex_decoder_ctl(dec, SPEEX_SET_ENH, &tmp);
-   tmp=0;
-   speex_encoder_ctl(st, SPEEX_SET_VBR, &tmp);
-   tmp=8;
-   speex_encoder_ctl(st, SPEEX_SET_QUALITY, &tmp);
-   tmp=1;
-   speex_encoder_ctl(st, SPEEX_SET_COMPLEXITY, &tmp);
-
-   speex_mode_query(&speex_nb_mode, SPEEX_MODE_FRAME_SIZE, &tmp);
-   fprintf (stderr, "frame size: %d\n", tmp);
-
-   if (argc != 4 && argc != 3)
-   {
-      fprintf (stderr, "Usage: encode [in file] [out file] [bits file]\nargc = %d", argc);
-      exit(1);
-   }
-   inFile = argv[1];
-   fin = fopen(inFile, "r");
-   outFile = argv[2];
-   fout = fopen(outFile, "w");
-   if (argc==4)
-   {
-      bitsFile = argv[3];
-      fbits = fopen(bitsFile, "w");
-   }
-   speex_bits_init(&bits);
-   while (!feof(fin))
-   {
-      fread(in, sizeof(short), FRAME_SIZE, fin);
-      if (feof(fin))
-         break;
-      for (i=0;i<FRAME_SIZE;i++)
-         bak[i]=input[i]=in[i];
-      speex_bits_reset(&bits);
-      /*
-      speex_bits_pack(&bits, 14, 5);
-      speex_bits_pack(&bits, SPEEX_INBAND_CHAR, 4);
-      speex_bits_pack(&bits, 'A', 8);
-      
-      speex_bits_pack(&bits, 14, 5);
-      speex_bits_pack(&bits, SPEEX_INBAND_MODE_REQUEST, 4);
-      speex_bits_pack(&bits, 7, 4);
-
-      speex_bits_pack(&bits, 15, 5);
-      speex_bits_pack(&bits, 2, 4);
-      speex_bits_pack(&bits, 0, 16);
-      */
-      speex_encode(st, input, &bits);
-      nbBits = speex_bits_write(&bits, cbits, 200);
-      bitCount+=bits.nbBits;
-      printf ("Encoding frame in %d bits\n", nbBits*8);
-      if (argc==4)
-         fwrite(cbits, 1, nbBits, fbits);
-      {
-         float enoise=0, esig=0, snr;
-         for (i=0;i<FRAME_SIZE;i++)
-         {
-            enoise+=(bak2[i]-input[i])*(bak2[i]-input[i]);
-            esig += bak2[i]*bak2[i];
-         }
-         snr = 10*log10((esig+1)/(enoise+1));
-         printf ("real SNR = %f\n", snr);
-      }
-      speex_bits_rewind(&bits);
-      
-      speex_decode(dec, &bits, input);
-      
-      /* Save the bits here */
-      for (i=0;i<FRAME_SIZE;i++)
-      {
-         if (input[i]>32000)
-            input[i]=32000;
-         else if (input[i]<-32000)
-            input[i]=-32000;
-      }
-      speex_bits_reset(&bits);
-      for (i=0;i<FRAME_SIZE;i++)
-         in[i]=(short)input[i];
-      for (i=0;i<FRAME_SIZE;i++)
-         bak2[i]=bak[i];
-      fwrite(in, sizeof(short), FRAME_SIZE, fout);
-   }
-   fprintf (stderr, "Total encoded size: %d bits\n", bitCount);
-   
-   speex_encoder_destroy(st);
-   speex_decoder_destroy(dec);
-   return 1;
-}

engine/voice_codecs/speex/source/libspeex/testenc_uwb.c

@@ -1,106 +0,0 @@
-#include "speex.h"
-#include <stdio.h>
-#include <stdlib.h>
-
-#define FRAME_SIZE 640
-#include <math.h>
-int main(int argc, char **argv)
-{
-   char *inFile, *outFile, *bitsFile;
-   FILE *fin, *fout, *fbits=NULL;
-   short in[FRAME_SIZE];
-   float input[FRAME_SIZE], bak[FRAME_SIZE], bak2[FRAME_SIZE];
-   char cbits[200];
-   int nbBits;
-   int i;
-   void *st;
-   void *dec;
-   SpeexBits bits;
-   int tmp;
-   int bitCount=0;
-
-   for (i=0;i<FRAME_SIZE;i++)
-      bak2[i]=0;
-   st = speex_encoder_init(&speex_uwb_mode);
-   dec = speex_decoder_init(&speex_uwb_mode);
-   
-   tmp=0;
-   /*speex_decoder_ctl(dec, SPEEX_SET_ENH, &tmp);
-   tmp=0;
-   speex_encoder_ctl(st, SPEEX_SET_VBR, &tmp);
-   tmp=10;
-   speex_encoder_ctl(st, SPEEX_SET_QUALITY, &tmp);
-   tmp=3;
-   speex_encoder_ctl(st, SPEEX_SET_COMPLEXITY, &tmp);
-   tmp=3;
-   speex_encoder_ctl(st, SPEEX_SET_HIGH_MODE, &tmp);
-   tmp=6;
-   speex_encoder_ctl(st, SPEEX_SET_LOW_MODE, &tmp);
-   */
-   if (argc != 4 && argc != 3)
-   { 
-      fprintf (stderr, "Usage: encode [in file] [out file] [bits file]\nargc = %d", argc);
-      exit(1);
-   }
-   inFile = argv[1];
-   fin = fopen(inFile, "r");
-   outFile = argv[2];
-   fout = fopen(outFile, "w");
-   if (argc==4)
-   {
-      bitsFile = argv[3];
-      fbits = fopen(bitsFile, "w");
-   }
-   speex_bits_init(&bits);
-   while (!feof(fin))
-   {
-      fread(in, sizeof(short), FRAME_SIZE, fin);
-      if (feof(fin))
-         break;
-      for (i=0;i<FRAME_SIZE;i++)
-         bak[i]=input[i]=in[i];
-      speex_bits_reset(&bits);
-      speex_encode(st, input, &bits);
-      nbBits = speex_bits_write(&bits, cbits, 200);
-      bitCount+=bits.nbBits;
-      printf ("Encoding frame in %d bits\n", nbBits*8);
-      if (argc==4)
-         fwrite(cbits, 1, nbBits, fbits);
-      {
-         float enoise=0, esig=0, snr;
-         for (i=0;i<FRAME_SIZE;i++)
-         {
-            enoise+=(bak2[i]-input[i])*(bak2[i]-input[i]);
-            esig += bak2[i]*bak2[i];
-         }
-         snr = 10*log10((esig+1)/(enoise+1));
-         printf ("real SNR = %f\n", snr);
-      }
-      speex_bits_rewind(&bits);
-      
-      speex_decode(dec, &bits, input);
-
-      /* Save the bits here */
-      for (i=0;i<FRAME_SIZE;i++)
-      {
-         if (input[i]>32000)
-            input[i]=32000;
-         else if (input[i]<-32000)
-            input[i]=-32000;
-      }
-      speex_bits_reset(&bits);
-      for (i=0;i<FRAME_SIZE;i++)
-         in[i]=(short)input[i];
-      for (i=0;i<FRAME_SIZE;i++)
-         bak2[i]=bak[i];
-      fwrite(in, sizeof(short), FRAME_SIZE, fout);
-   }
-   
-   fprintf (stderr, "Total encoded size: %d bits\n", bitCount);
-   speex_encoder_destroy(st);
-   speex_decoder_destroy(dec);
-   speex_bits_destroy(&bits);
-   fclose(fin);
-   fclose(fout);
-   return 1;
-}