mirror of
https://github.com/godotengine/webrtc-native.git
synced 2026-01-02 09:48:12 +03:00
232 lines
7.9 KiB
C++
232 lines
7.9 KiB
C++
/*************************************************************************/
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/* WebRTCLibDataChannel.cpp */
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/*************************************************************************/
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/* This file is part of: */
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/* GODOT ENGINE */
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/* https://godotengine.org */
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/*************************************************************************/
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/* Copyright (c) 2007-2021 Juan Linietsky, Ariel Manzur. */
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/* Copyright (c) 2014-2021 Godot Engine contributors (cf. AUTHORS.md). */
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/* */
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/* Permission is hereby granted, free of charge, to any person obtaining */
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/* a copy of this software and associated documentation files (the */
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/* "Software"), to deal in the Software without restriction, including */
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/* without limitation the rights to use, copy, modify, merge, publish, */
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/* distribute, sublicense, and/or sell copies of the Software, and to */
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/* permit persons to whom the Software is furnished to do so, subject to */
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/* the following conditions: */
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/* */
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/* The above copyright notice and this permission notice shall be */
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/* included in all copies or substantial portions of the Software. */
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/* */
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/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, */
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/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF */
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/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/
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/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY */
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/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, */
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/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */
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/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */
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/*************************************************************************/
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#include "WebRTCLibDataChannel.hpp"
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#ifdef GDNATIVE_WEBRTC
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#include "GDNativeLibrary.hpp"
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#include "NativeScript.hpp"
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#define ERR_UNAVAILABLE Error::ERR_UNAVAILABLE
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#define FAILED Error::FAILED
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#define ERR_INVALID_PARAMETER Error::ERR_INVALID_PARAMETER
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#define OK Error::OK
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#endif
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#include <stdio.h>
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#include <string.h>
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#include <cstring>
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using namespace godot;
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using namespace godot_webrtc;
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// DataChannel
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WebRTCLibDataChannel *WebRTCLibDataChannel::new_data_channel(std::shared_ptr<rtc::DataChannel> p_channel, bool p_negotiated) {
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// Invalid channel result in NULL return
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ERR_FAIL_COND_V(!p_channel, nullptr);
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#ifdef GDNATIVE_WEBRTC
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// Instance a WebRTCDataChannelGDNative object
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WebRTCDataChannelGDNative *native = WebRTCDataChannelGDNative::_new();
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// Set our implementation as its script
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NativeScript *script = NativeScript::_new();
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script->set_library(detail::get_wrapper<GDNativeLibrary>((godot_object *)gdnlib));
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script->set_class_name("WebRTCLibDataChannel");
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native->set_script(script);
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WebRTCLibDataChannel *out = native->cast_to<WebRTCLibDataChannel>(native);
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#else
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WebRTCLibDataChannel *out = memnew(WebRTCLibDataChannel);
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#endif
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// Bind the library data channel to our object.
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out->bind_channel(p_channel, p_negotiated);
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return out;
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}
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void WebRTCLibDataChannel::bind_channel(std::shared_ptr<rtc::DataChannel> p_channel, bool p_negotiated) {
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ERR_FAIL_COND(!p_channel);
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channel = p_channel;
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negotiated = p_negotiated;
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// Binding this should be fine as long as we call close when going out of scope.
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p_channel->onMessage([this](auto message) {
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if (std::holds_alternative<rtc::string>(message)) {
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rtc::string str = std::get<rtc::string>(message);
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queue_packet(reinterpret_cast<const uint8_t *>(str.c_str()), str.size(), true);
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} else if (std::holds_alternative<rtc::binary>(message)) {
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rtc::binary bin = std::get<rtc::binary>(message);
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queue_packet(reinterpret_cast<const uint8_t *>(&bin[0]), bin.size(), false);
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} else {
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ERR_PRINT("Message parsing bug. Unknown message type.");
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}
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});
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p_channel->onOpen([this]() {
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channel_state = STATE_OPEN;
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});
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p_channel->onClosed([this]() {
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channel_state = STATE_CLOSED;
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});
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p_channel->onError([](auto error) {
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ERR_PRINT("Channel Error: " + String(std::string(error).c_str()));
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});
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}
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void WebRTCLibDataChannel::queue_packet(const uint8_t *data, uint32_t size, bool p_is_string) {
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mutex->lock();
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std::vector<uint8_t> packet;
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packet.resize(size);
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memcpy(&packet[0], data, size);
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packet_queue.push(QueuedPacket(packet, p_is_string));
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mutex->unlock();
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}
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void WebRTCLibDataChannel::_set_write_mode(WriteMode p_mode) {
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ERR_FAIL_COND(p_mode != WRITE_MODE_TEXT && p_mode != WRITE_MODE_BINARY);
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write_mode = p_mode;
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}
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WebRTCDataChannel::WriteMode WebRTCLibDataChannel::_get_write_mode() const {
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return write_mode;
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}
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bool WebRTCLibDataChannel::_was_string_packet() const {
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return current_packet.second;
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}
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WebRTCDataChannel::ChannelState WebRTCLibDataChannel::_get_ready_state() const {
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ERR_FAIL_COND_V(!channel, STATE_CLOSED);
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return channel_state;
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}
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String WebRTCLibDataChannel::_get_label() const {
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ERR_FAIL_COND_V(!channel, "");
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return channel->label().c_str();
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}
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bool WebRTCLibDataChannel::_is_ordered() const {
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ERR_FAIL_COND_V(!channel, false);
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return channel->reliability().unordered == false;
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}
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int64_t WebRTCLibDataChannel::_get_id() const {
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ERR_FAIL_COND_V(!channel, -1);
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return channel->id().value_or(-1);
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}
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int64_t WebRTCLibDataChannel::_get_max_packet_life_time() const {
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ERR_FAIL_COND_V(!channel, 0);
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return channel->reliability().type == rtc::Reliability::Type::Timed ? std::get<std::chrono::milliseconds>(channel->reliability().rexmit).count() : -1;
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}
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int64_t WebRTCLibDataChannel::_get_max_retransmits() const {
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ERR_FAIL_COND_V(!channel, 0);
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return channel->reliability().type == rtc::Reliability::Type::Rexmit ? std::get<int>(channel->reliability().rexmit) : -1;
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}
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String WebRTCLibDataChannel::_get_protocol() const {
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ERR_FAIL_COND_V(!channel, "");
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return channel->protocol().c_str();
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}
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bool WebRTCLibDataChannel::_is_negotiated() const {
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ERR_FAIL_COND_V(!channel, false);
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return negotiated;
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}
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int64_t WebRTCLibDataChannel::_get_buffered_amount() const {
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ERR_FAIL_COND_V(!channel, 0);
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return channel->bufferedAmount();
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}
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Error WebRTCLibDataChannel::_poll() {
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return OK;
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}
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void WebRTCLibDataChannel::_close() try {
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if (channel) {
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channel->close();
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}
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} catch (...) {
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}
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Error WebRTCLibDataChannel::_get_packet(const uint8_t **r_buffer, int32_t *r_len) {
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ERR_FAIL_COND_V(packet_queue.empty(), ERR_UNAVAILABLE);
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mutex->lock();
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// Update current packet and pop queue
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current_packet = packet_queue.front();
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packet_queue.pop();
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// Set out buffer and size (buffer will be gone at next get_packet or close)
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*r_buffer = ¤t_packet.first[0];
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*r_len = current_packet.first.size();
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mutex->unlock();
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return OK;
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}
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Error WebRTCLibDataChannel::_put_packet(const uint8_t *p_buffer, int64_t p_len) try {
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ERR_FAIL_COND_V(!channel, FAILED);
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ERR_FAIL_COND_V(channel->isClosed(), FAILED);
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if (write_mode == WRITE_MODE_TEXT) {
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std::string str(p_len, '\x00');
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std::strncpy(str.data(), (const char *)p_buffer, p_len);
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channel->send(str);
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} else if (write_mode == WRITE_MODE_BINARY) {
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channel->send(reinterpret_cast<const std::byte *>(p_buffer), p_len);
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} else {
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ERR_FAIL_V(ERR_INVALID_PARAMETER);
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}
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return OK;
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} catch (const std::exception &e) {
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ERR_PRINT(e.what());
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ERR_FAIL_V(FAILED);
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}
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int64_t WebRTCLibDataChannel::_get_available_packet_count() const {
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return packet_queue.size();
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}
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int64_t WebRTCLibDataChannel::_get_max_packet_size() const {
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return 16384; // See RFC-8831 section 6.6: https://datatracker.ietf.org/doc/rfc8831/
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}
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WebRTCLibDataChannel::WebRTCLibDataChannel() {
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mutex = new std::mutex;
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}
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WebRTCLibDataChannel::~WebRTCLibDataChannel() {
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_close();
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channel = nullptr;
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delete mutex;
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}
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