mirror of https://github.com/NekoX-Dev/NekoX.git
2288 lines
87 KiB
C++
2288 lines
87 KiB
C++
#include "v2/InstanceV2Impl.h"
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#include "LogSinkImpl.h"
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#include "VideoCaptureInterfaceImpl.h"
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#include "VideoCapturerInterface.h"
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#include "v2/NativeNetworkingImpl.h"
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#include "v2/Signaling.h"
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#include "v2/ContentNegotiation.h"
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#include "CodecSelectHelper.h"
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#include "platform/PlatformInterface.h"
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#include "api/audio_codecs/audio_decoder_factory_template.h"
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#include "api/audio_codecs/audio_encoder_factory_template.h"
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#include "api/audio_codecs/opus/audio_decoder_opus.h"
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#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h"
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#include "api/audio_codecs/opus/audio_encoder_opus.h"
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#include "api/audio_codecs/L16/audio_decoder_L16.h"
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#include "api/audio_codecs/L16/audio_encoder_L16.h"
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#include "api/task_queue/default_task_queue_factory.h"
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#include "media/engine/webrtc_media_engine.h"
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#include "system_wrappers/include/field_trial.h"
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#include "api/video/builtin_video_bitrate_allocator_factory.h"
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#include "call/call.h"
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#include "api/call/audio_sink.h"
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#include "modules/audio_processing/audio_buffer.h"
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#include "absl/strings/match.h"
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#include "pc/channel.h"
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#include "audio/audio_state.h"
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#include "modules/audio_coding/neteq/default_neteq_factory.h"
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#include "modules/audio_coding/include/audio_coding_module.h"
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#include "api/candidate.h"
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#include "api/jsep_ice_candidate.h"
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#include "pc/used_ids.h"
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#include "media/base/sdp_video_format_utils.h"
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#include "pc/media_session.h"
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#include "rtc_base/rtc_certificate_generator.h"
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#include "AudioFrame.h"
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#include "ThreadLocalObject.h"
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#include "Manager.h"
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#include "NetworkManager.h"
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#include "VideoCaptureInterfaceImpl.h"
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#include "platform/PlatformInterface.h"
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#include "LogSinkImpl.h"
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#include "CodecSelectHelper.h"
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#include "AudioDeviceHelper.h"
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#include "SignalingEncryption.h"
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#ifdef WEBRTC_IOS
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#include "platform/darwin/iOS/tgcalls_audio_device_module_ios.h"
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#endif
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#include <random>
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#include <sstream>
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#include "FieldTrialsConfig.h"
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#include "third-party/json11.hpp"
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#include "ChannelManager.h"
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#include "SignalingConnection.h"
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#include "ExternalSignalingConnection.h"
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#include "SignalingSctpConnection.h"
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#include "utils/gzip.h"
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namespace tgcalls {
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namespace {
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enum class SignalingProtocolVersion {
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V1,
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V2,
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V3
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};
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SignalingProtocolVersion signalingProtocolVersion(std::string const &version) {
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if (version == "7.0.0") {
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return SignalingProtocolVersion::V1;
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} else if (version == "8.0.0") {
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return SignalingProtocolVersion::V2;
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} else if (version == "9.0.0") {
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return SignalingProtocolVersion::V3;
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} else {
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RTC_LOG(LS_ERROR) << "signalingProtocolVersion: unknown version " << version;
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return SignalingProtocolVersion::V2;
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}
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}
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bool signalingProtocolSupportsCompression(SignalingProtocolVersion version) {
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switch (version) {
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case SignalingProtocolVersion::V1:
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case SignalingProtocolVersion::V2:
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return false;
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case SignalingProtocolVersion::V3:
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return true;
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default:
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RTC_DCHECK_NOTREACHED();
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break;
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}
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return false;
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}
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static VideoCaptureInterfaceObject *GetVideoCaptureAssumingSameThread(VideoCaptureInterface *videoCapture) {
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return videoCapture
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? static_cast<VideoCaptureInterfaceImpl*>(videoCapture)->object()->getSyncAssumingSameThread()
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: nullptr;
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}
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class OutgoingAudioChannel : public sigslot::has_slots<> {
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public:
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OutgoingAudioChannel(
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webrtc::Call *call,
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ChannelManager *channelManager,
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rtc::UniqueRandomIdGenerator *uniqueRandomIdGenerator,
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webrtc::LocalAudioSinkAdapter *audioSource,
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webrtc::RtpTransport *rtpTransport,
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signaling::MediaContent const &mediaContent,
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std::shared_ptr<Threads> threads
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) :
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_threads(threads),
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_ssrc(mediaContent.ssrc),
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_call(call),
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_channelManager(channelManager),
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_audioSource(audioSource) {
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cricket::AudioOptions audioOptions;
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bool _disableOutgoingAudioProcessing = false;
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if (_disableOutgoingAudioProcessing) {
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audioOptions.echo_cancellation = false;
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audioOptions.noise_suppression = false;
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audioOptions.auto_gain_control = false;
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audioOptions.highpass_filter = false;
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//audioOptions.typing_detection = false;
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//audioOptions.residual_echo_detector = false;
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} else {
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audioOptions.echo_cancellation = true;
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audioOptions.noise_suppression = true;
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}
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std::ostringstream contentId;
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contentId << _ssrc;
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std::vector<std::string> streamIds;
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streamIds.push_back(contentId.str());
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_outgoingAudioChannel = _channelManager->CreateVoiceChannel(call, cricket::MediaConfig(), contentId.str(), false, NativeNetworkingImpl::getDefaulCryptoOptions(), audioOptions);
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_threads->getNetworkThread()->BlockingCall([&]() {
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_outgoingAudioChannel->SetRtpTransport(rtpTransport);
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});
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std::vector<cricket::AudioCodec> codecs;
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for (const auto &payloadType : mediaContent.payloadTypes) {
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if (payloadType.name == "opus") {
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cricket::AudioCodec codec(payloadType.id, payloadType.name, payloadType.clockrate, 0, payloadType.channels);
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codec.SetParam(cricket::kCodecParamUseInbandFec, 1);
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codec.SetParam(cricket::kCodecParamPTime, 60);
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for (const auto &feedbackType : payloadType.feedbackTypes) {
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codec.AddFeedbackParam(cricket::FeedbackParam(feedbackType.type, feedbackType.subtype));
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}
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codecs.push_back(std::move(codec));
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break;
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}
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}
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auto outgoingAudioDescription = std::make_unique<cricket::AudioContentDescription>();
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for (const auto &rtpExtension : mediaContent.rtpExtensions) {
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outgoingAudioDescription->AddRtpHeaderExtension(webrtc::RtpExtension(rtpExtension.uri, rtpExtension.id));
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}
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outgoingAudioDescription->set_rtcp_mux(true);
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outgoingAudioDescription->set_rtcp_reduced_size(true);
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outgoingAudioDescription->set_direction(webrtc::RtpTransceiverDirection::kSendOnly);
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outgoingAudioDescription->set_codecs(codecs);
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outgoingAudioDescription->set_bandwidth(-1);
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outgoingAudioDescription->AddStream(cricket::StreamParams::CreateLegacy(_ssrc));
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auto incomingAudioDescription = std::make_unique<cricket::AudioContentDescription>();
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for (const auto &rtpExtension : mediaContent.rtpExtensions) {
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incomingAudioDescription->AddRtpHeaderExtension(webrtc::RtpExtension(rtpExtension.uri, rtpExtension.id));
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}
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incomingAudioDescription->set_rtcp_mux(true);
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incomingAudioDescription->set_rtcp_reduced_size(true);
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incomingAudioDescription->set_direction(webrtc::RtpTransceiverDirection::kRecvOnly);
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incomingAudioDescription->set_codecs(codecs);
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incomingAudioDescription->set_bandwidth(-1);
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_threads->getWorkerThread()->BlockingCall([&]() {
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_outgoingAudioChannel->SetPayloadTypeDemuxingEnabled(false);
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std::string errorDesc;
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_outgoingAudioChannel->SetLocalContent(outgoingAudioDescription.get(), webrtc::SdpType::kOffer, errorDesc);
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_outgoingAudioChannel->SetRemoteContent(incomingAudioDescription.get(), webrtc::SdpType::kAnswer, errorDesc);
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});
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setIsMuted(false);
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}
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~OutgoingAudioChannel() {
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_outgoingAudioChannel->Enable(false);
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_threads->getNetworkThread()->BlockingCall([&]() {
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_outgoingAudioChannel->SetRtpTransport(nullptr);
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});
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_channelManager->DestroyChannel(_outgoingAudioChannel);
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_outgoingAudioChannel = nullptr;
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}
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void setIsMuted(bool isMuted) {
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if (_isMuted != isMuted) {
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_isMuted = isMuted;
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_outgoingAudioChannel->Enable(!_isMuted);
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_threads->getWorkerThread()->BlockingCall([&]() {
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_outgoingAudioChannel->media_channel()->SetAudioSend(_ssrc, !_isMuted, nullptr, _audioSource);
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});
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}
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}
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uint32_t ssrc() const {
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return _ssrc;
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}
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void setMaxBitrate(int bitrate) {
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_threads->getWorkerThread()->BlockingCall([&]() {
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webrtc::RtpParameters initialParameters = _outgoingAudioChannel->media_channel()->GetRtpSendParameters(_ssrc);
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webrtc::RtpParameters updatedParameters = initialParameters;
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if (updatedParameters.encodings.empty()) {
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updatedParameters.encodings.push_back(webrtc::RtpEncodingParameters());
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}
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updatedParameters.encodings[0].max_bitrate_bps = bitrate;
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if (initialParameters != updatedParameters) {
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_outgoingAudioChannel->media_channel()->SetRtpSendParameters(_ssrc, updatedParameters);
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}
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});
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}
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private:
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void OnSentPacket_w(const rtc::SentPacket& sent_packet) {
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_call->OnSentPacket(sent_packet);
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}
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private:
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std::shared_ptr<Threads> _threads;
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uint32_t _ssrc = 0;
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webrtc::Call *_call = nullptr;
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ChannelManager *_channelManager = nullptr;
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webrtc::LocalAudioSinkAdapter *_audioSource = nullptr;
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cricket::VoiceChannel *_outgoingAudioChannel = nullptr;
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bool _isMuted = true;
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};
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class IncomingV2AudioChannel : public sigslot::has_slots<> {
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public:
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IncomingV2AudioChannel(
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ChannelManager *channelManager,
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webrtc::Call *call,
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webrtc::RtpTransport *rtpTransport,
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rtc::UniqueRandomIdGenerator *randomIdGenerator,
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signaling::MediaContent const &mediaContent,
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std::shared_ptr<Threads> threads) :
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_threads(threads),
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_ssrc(mediaContent.ssrc),
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_channelManager(channelManager),
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_call(call) {
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_creationTimestamp = rtc::TimeMillis();
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cricket::AudioOptions audioOptions;
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audioOptions.audio_jitter_buffer_fast_accelerate = true;
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audioOptions.audio_jitter_buffer_min_delay_ms = 50;
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std::ostringstream contentId;
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contentId << _ssrc;
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std::string streamId = contentId.str();
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_audioChannel = _channelManager->CreateVoiceChannel(call, cricket::MediaConfig(), contentId.str(), false, NativeNetworkingImpl::getDefaulCryptoOptions(), audioOptions);
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_threads->getNetworkThread()->BlockingCall([&]() {
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_audioChannel->SetRtpTransport(rtpTransport);
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});
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std::vector<cricket::AudioCodec> codecs;
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for (const auto &payloadType : mediaContent.payloadTypes) {
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cricket::AudioCodec codec(payloadType.id, payloadType.name, payloadType.clockrate, 0, payloadType.channels);
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for (const auto ¶meter : payloadType.parameters) {
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codec.SetParam(parameter.first, parameter.second);
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}
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for (const auto &feedbackType : payloadType.feedbackTypes) {
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codec.AddFeedbackParam(cricket::FeedbackParam(feedbackType.type, feedbackType.subtype));
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}
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codecs.push_back(std::move(codec));
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}
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auto outgoingAudioDescription = std::make_unique<cricket::AudioContentDescription>();
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for (const auto &rtpExtension : mediaContent.rtpExtensions) {
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outgoingAudioDescription->AddRtpHeaderExtension(webrtc::RtpExtension(rtpExtension.uri, rtpExtension.id));
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}
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outgoingAudioDescription->set_rtcp_mux(true);
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outgoingAudioDescription->set_rtcp_reduced_size(true);
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outgoingAudioDescription->set_direction(webrtc::RtpTransceiverDirection::kRecvOnly);
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outgoingAudioDescription->set_codecs(codecs);
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outgoingAudioDescription->set_bandwidth(-1);
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auto incomingAudioDescription = std::make_unique<cricket::AudioContentDescription>();
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for (const auto &rtpExtension : mediaContent.rtpExtensions) {
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incomingAudioDescription->AddRtpHeaderExtension(webrtc::RtpExtension(rtpExtension.uri, rtpExtension.id));
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}
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incomingAudioDescription->set_rtcp_mux(true);
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incomingAudioDescription->set_rtcp_reduced_size(true);
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incomingAudioDescription->set_direction(webrtc::RtpTransceiverDirection::kSendOnly);
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incomingAudioDescription->set_codecs(codecs);
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incomingAudioDescription->set_bandwidth(-1);
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cricket::StreamParams streamParams = cricket::StreamParams::CreateLegacy(mediaContent.ssrc);
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streamParams.set_stream_ids({ streamId });
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incomingAudioDescription->AddStream(streamParams);
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threads->getWorkerThread()->BlockingCall([&]() {
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_audioChannel->SetPayloadTypeDemuxingEnabled(false);
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std::string errorDesc;
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_audioChannel->SetLocalContent(outgoingAudioDescription.get(), webrtc::SdpType::kOffer, errorDesc);
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_audioChannel->SetRemoteContent(incomingAudioDescription.get(), webrtc::SdpType::kAnswer, errorDesc);
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});
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outgoingAudioDescription.reset();
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incomingAudioDescription.reset();
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//std::unique_ptr<AudioSinkImpl> audioLevelSink(new AudioSinkImpl(onAudioLevelUpdated, _ssrc, std::move(onAudioFrame)));
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//_audioChannel->media_channel()->SetRawAudioSink(ssrc.networkSsrc, std::move(audioLevelSink));
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_audioChannel->Enable(true);
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}
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~IncomingV2AudioChannel() {
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_audioChannel->Enable(false);
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_threads->getNetworkThread()->BlockingCall([&]() {
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_audioChannel->SetRtpTransport(nullptr);
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});
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_channelManager->DestroyChannel(_audioChannel);
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_audioChannel = nullptr;
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}
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void setVolume(double value) {
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_audioChannel->media_channel()->SetOutputVolume(_ssrc, value);
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}
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void updateActivity() {
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_activityTimestamp = rtc::TimeMillis();
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}
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int64_t getActivity() {
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return _activityTimestamp;
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}
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uint32_t ssrc() const {
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return _ssrc;
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}
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private:
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void OnSentPacket_w(const rtc::SentPacket& sent_packet) {
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_call->OnSentPacket(sent_packet);
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}
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private:
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std::shared_ptr<Threads> _threads;
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uint32_t _ssrc = 0;
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// Memory is managed by _channelManager
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cricket::VoiceChannel *_audioChannel = nullptr;
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// Memory is managed externally
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ChannelManager *_channelManager = nullptr;
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webrtc::Call *_call = nullptr;
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int64_t _creationTimestamp = 0;
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int64_t _activityTimestamp = 0;
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};
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class OutgoingVideoChannel : public sigslot::has_slots<>, public std::enable_shared_from_this<OutgoingVideoChannel> {
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public:
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OutgoingVideoChannel(
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std::shared_ptr<Threads> threads,
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ChannelManager *channelManager,
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webrtc::Call *call,
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webrtc::RtpTransport *rtpTransport,
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rtc::UniqueRandomIdGenerator *randomIdGenerator,
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webrtc::VideoBitrateAllocatorFactory *videoBitrateAllocatorFactory,
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std::function<void()> rotationUpdated,
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signaling::MediaContent const &mediaContent,
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bool isScreencast
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) :
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_threads(threads),
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_mainSsrc(mediaContent.ssrc),
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_call(call),
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_channelManager(channelManager),
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_rotationUpdated(rotationUpdated) {
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cricket::VideoOptions videoOptions;
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videoOptions.is_screencast = isScreencast;
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std::ostringstream contentId;
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contentId << mediaContent.ssrc;
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_outgoingVideoChannel = _channelManager->CreateVideoChannel(call, cricket::MediaConfig(), contentId.str(), false, NativeNetworkingImpl::getDefaulCryptoOptions(), videoOptions, videoBitrateAllocatorFactory);
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_threads->getNetworkThread()->BlockingCall([&]() {
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_outgoingVideoChannel->SetRtpTransport(rtpTransport);
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});
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std::vector<cricket::VideoCodec> unsortedCodecs;
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for (const auto &payloadType : mediaContent.payloadTypes) {
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cricket::VideoCodec codec(payloadType.id, payloadType.name);
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for (const auto ¶meter : payloadType.parameters) {
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codec.SetParam(parameter.first, parameter.second);
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}
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for (const auto &feedbackType : payloadType.feedbackTypes) {
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codec.AddFeedbackParam(cricket::FeedbackParam(feedbackType.type, feedbackType.subtype));
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}
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unsortedCodecs.push_back(std::move(codec));
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}
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std::vector<std::string> codecPreferences = {
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#ifndef WEBRTC_DISABLE_H265
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cricket::kH265CodecName,
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#endif
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cricket::kH264CodecName
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};
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std::vector<cricket::VideoCodec> codecs;
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for (const auto &name : codecPreferences) {
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for (const auto &codec : unsortedCodecs) {
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if (codec.name == name) {
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codecs.push_back(codec);
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}
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}
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}
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for (const auto &codec : unsortedCodecs) {
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if (std::find(codecs.begin(), codecs.end(), codec) == codecs.end()) {
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codecs.push_back(codec);
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}
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}
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auto outgoingVideoDescription = std::make_unique<cricket::VideoContentDescription>();
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for (const auto &rtpExtension : mediaContent.rtpExtensions) {
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outgoingVideoDescription->AddRtpHeaderExtension(rtpExtension);
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}
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outgoingVideoDescription->set_rtcp_mux(true);
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outgoingVideoDescription->set_rtcp_reduced_size(true);
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outgoingVideoDescription->set_direction(webrtc::RtpTransceiverDirection::kSendOnly);
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outgoingVideoDescription->set_codecs(codecs);
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outgoingVideoDescription->set_bandwidth(-1);
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cricket::StreamParams videoSendStreamParams;
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for (const auto &ssrcGroup : mediaContent.ssrcGroups) {
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for (auto ssrc : ssrcGroup.ssrcs) {
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if (!videoSendStreamParams.has_ssrc(ssrc)) {
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videoSendStreamParams.ssrcs.push_back(ssrc);
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}
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}
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cricket::SsrcGroup mappedGroup(ssrcGroup.semantics, ssrcGroup.ssrcs);
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videoSendStreamParams.ssrc_groups.push_back(std::move(mappedGroup));
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}
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videoSendStreamParams.cname = "cname";
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outgoingVideoDescription->AddStream(videoSendStreamParams);
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auto incomingVideoDescription = std::make_unique<cricket::VideoContentDescription>();
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for (const auto &rtpExtension : mediaContent.rtpExtensions) {
|
|
incomingVideoDescription->AddRtpHeaderExtension(webrtc::RtpExtension(rtpExtension.uri, rtpExtension.id));
|
|
}
|
|
incomingVideoDescription->set_rtcp_mux(true);
|
|
incomingVideoDescription->set_rtcp_reduced_size(true);
|
|
incomingVideoDescription->set_direction(webrtc::RtpTransceiverDirection::kRecvOnly);
|
|
incomingVideoDescription->set_codecs(codecs);
|
|
incomingVideoDescription->set_bandwidth(-1);
|
|
|
|
threads->getWorkerThread()->BlockingCall([&]() {
|
|
_outgoingVideoChannel->SetPayloadTypeDemuxingEnabled(false);
|
|
std::string errorDesc;
|
|
_outgoingVideoChannel->SetLocalContent(outgoingVideoDescription.get(), webrtc::SdpType::kOffer, errorDesc);
|
|
_outgoingVideoChannel->SetRemoteContent(incomingVideoDescription.get(), webrtc::SdpType::kAnswer, errorDesc);
|
|
|
|
webrtc::RtpParameters rtpParameters = _outgoingVideoChannel->media_channel()->GetRtpSendParameters(mediaContent.ssrc);
|
|
|
|
if (isScreencast) {
|
|
rtpParameters.degradation_preference = webrtc::DegradationPreference::MAINTAIN_RESOLUTION;
|
|
}
|
|
|
|
_outgoingVideoChannel->media_channel()->SetRtpSendParameters(mediaContent.ssrc, rtpParameters);
|
|
});
|
|
|
|
_outgoingVideoChannel->Enable(false);
|
|
|
|
threads->getWorkerThread()->BlockingCall([&]() {
|
|
_outgoingVideoChannel->media_channel()->SetVideoSend(mediaContent.ssrc, NULL, nullptr);
|
|
});
|
|
}
|
|
|
|
~OutgoingVideoChannel() {
|
|
_outgoingVideoChannel->Enable(false);
|
|
_threads->getNetworkThread()->BlockingCall([&]() {
|
|
_outgoingVideoChannel->SetRtpTransport(nullptr);
|
|
});
|
|
_channelManager->DestroyChannel(_outgoingVideoChannel);
|
|
_outgoingVideoChannel = nullptr;
|
|
}
|
|
|
|
void setVideoCapture(std::shared_ptr<VideoCaptureInterface> videoCapture) {
|
|
_videoCapture = videoCapture;
|
|
|
|
if (_videoCapture) {
|
|
_outgoingVideoChannel->Enable(true);
|
|
auto videoCaptureImpl = GetVideoCaptureAssumingSameThread(_videoCapture.get());
|
|
|
|
_threads->getWorkerThread()->BlockingCall([&]() {
|
|
_outgoingVideoChannel->media_channel()->SetVideoSend(_mainSsrc, NULL, videoCaptureImpl->source().get());
|
|
});
|
|
|
|
const auto weak = std::weak_ptr<OutgoingVideoChannel>(shared_from_this());
|
|
videoCaptureImpl->setRotationUpdated([threads = _threads, weak](int angle) {
|
|
threads->getMediaThread()->PostTask([=] {
|
|
const auto strong = weak.lock();
|
|
if (!strong) {
|
|
return;
|
|
}
|
|
signaling::MediaStateMessage::VideoRotation videoRotation = signaling::MediaStateMessage::VideoRotation::Rotation0;
|
|
switch (angle) {
|
|
case 0: {
|
|
videoRotation = signaling::MediaStateMessage::VideoRotation::Rotation0;
|
|
break;
|
|
}
|
|
case 90: {
|
|
videoRotation = signaling::MediaStateMessage::VideoRotation::Rotation90;
|
|
break;
|
|
}
|
|
case 180: {
|
|
videoRotation = signaling::MediaStateMessage::VideoRotation::Rotation180;
|
|
break;
|
|
}
|
|
case 270: {
|
|
videoRotation = signaling::MediaStateMessage::VideoRotation::Rotation270;
|
|
break;
|
|
}
|
|
default: {
|
|
videoRotation = signaling::MediaStateMessage::VideoRotation::Rotation0;
|
|
break;
|
|
}
|
|
}
|
|
if (strong->_videoRotation != videoRotation) {
|
|
strong->_videoRotation = videoRotation;
|
|
strong->_rotationUpdated();
|
|
}
|
|
});
|
|
});
|
|
|
|
switch (videoCaptureImpl->getRotation()) {
|
|
case 0: {
|
|
_videoRotation = signaling::MediaStateMessage::VideoRotation::Rotation0;
|
|
break;
|
|
}
|
|
case 90: {
|
|
_videoRotation = signaling::MediaStateMessage::VideoRotation::Rotation90;
|
|
break;
|
|
}
|
|
case 180: {
|
|
_videoRotation = signaling::MediaStateMessage::VideoRotation::Rotation180;
|
|
break;
|
|
}
|
|
case 270: {
|
|
_videoRotation = signaling::MediaStateMessage::VideoRotation::Rotation270;
|
|
break;
|
|
}
|
|
default: {
|
|
_videoRotation = signaling::MediaStateMessage::VideoRotation::Rotation0;
|
|
break;
|
|
}
|
|
}
|
|
} else {
|
|
_videoRotation = signaling::MediaStateMessage::VideoRotation::Rotation0;
|
|
_outgoingVideoChannel->Enable(false);
|
|
|
|
_threads->getWorkerThread()->BlockingCall([&]() {
|
|
_outgoingVideoChannel->media_channel()->SetVideoSend(_mainSsrc, NULL, nullptr);
|
|
});
|
|
}
|
|
}
|
|
|
|
uint32_t ssrc() const {
|
|
return _mainSsrc;
|
|
}
|
|
|
|
void setMaxBitrate(int bitrate) {
|
|
_threads->getWorkerThread()->BlockingCall([&]() {
|
|
webrtc::RtpParameters initialParameters = _outgoingVideoChannel->media_channel()->GetRtpSendParameters(_mainSsrc);
|
|
webrtc::RtpParameters updatedParameters = initialParameters;
|
|
|
|
if (updatedParameters.encodings.empty()) {
|
|
updatedParameters.encodings.push_back(webrtc::RtpEncodingParameters());
|
|
}
|
|
|
|
updatedParameters.encodings[0].max_bitrate_bps = bitrate;
|
|
|
|
if (initialParameters != updatedParameters) {
|
|
_outgoingVideoChannel->media_channel()->SetRtpSendParameters(_mainSsrc, updatedParameters);
|
|
}
|
|
});
|
|
}
|
|
|
|
public:
|
|
std::shared_ptr<VideoCaptureInterface> videoCapture() {
|
|
return _videoCapture;
|
|
}
|
|
|
|
signaling::MediaStateMessage::VideoRotation getRotation() {
|
|
return _videoRotation;
|
|
}
|
|
|
|
private:
|
|
void OnSentPacket_w(const rtc::SentPacket& sent_packet) {
|
|
_call->OnSentPacket(sent_packet);
|
|
}
|
|
|
|
private:
|
|
std::shared_ptr<Threads> _threads;
|
|
|
|
uint32_t _mainSsrc = 0;
|
|
webrtc::Call *_call = nullptr;
|
|
ChannelManager *_channelManager = nullptr;
|
|
cricket::VideoChannel *_outgoingVideoChannel = nullptr;
|
|
|
|
std::function<void()> _rotationUpdated;
|
|
|
|
std::shared_ptr<VideoCaptureInterface> _videoCapture;
|
|
signaling::MediaStateMessage::VideoRotation _videoRotation = signaling::MediaStateMessage::VideoRotation::Rotation0;
|
|
};
|
|
|
|
class VideoSinkImpl : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
|
|
public:
|
|
VideoSinkImpl() {
|
|
}
|
|
|
|
virtual ~VideoSinkImpl() {
|
|
}
|
|
|
|
virtual void OnFrame(const webrtc::VideoFrame& frame) override {
|
|
//_lastFrame = frame;
|
|
for (int i = (int)(_sinks.size()) - 1; i >= 0; i--) {
|
|
auto strong = _sinks[i].lock();
|
|
if (!strong) {
|
|
_sinks.erase(_sinks.begin() + i);
|
|
} else {
|
|
strong->OnFrame(frame);
|
|
}
|
|
}
|
|
}
|
|
|
|
virtual void OnDiscardedFrame() override {
|
|
for (int i = (int)(_sinks.size()) - 1; i >= 0; i--) {
|
|
auto strong = _sinks[i].lock();
|
|
if (!strong) {
|
|
_sinks.erase(_sinks.begin() + i);
|
|
} else {
|
|
strong->OnDiscardedFrame();
|
|
}
|
|
}
|
|
}
|
|
|
|
void addSink(std::weak_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> impl) {
|
|
_sinks.push_back(impl);
|
|
if (_lastFrame) {
|
|
auto strong = impl.lock();
|
|
if (strong) {
|
|
strong->OnFrame(_lastFrame.value());
|
|
}
|
|
}
|
|
}
|
|
|
|
private:
|
|
std::vector<std::weak_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>>> _sinks;
|
|
absl::optional<webrtc::VideoFrame> _lastFrame;
|
|
};
|
|
|
|
class IncomingV2VideoChannel : public sigslot::has_slots<> {
|
|
public:
|
|
IncomingV2VideoChannel(
|
|
ChannelManager *channelManager,
|
|
webrtc::Call *call,
|
|
webrtc::RtpTransport *rtpTransport,
|
|
rtc::UniqueRandomIdGenerator *randomIdGenerator,
|
|
signaling::MediaContent const &mediaContent,
|
|
std::shared_ptr<Threads> threads) :
|
|
_threads(threads),
|
|
_channelManager(channelManager),
|
|
_call(call) {
|
|
_videoSink.reset(new VideoSinkImpl());
|
|
|
|
_videoBitrateAllocatorFactory = webrtc::CreateBuiltinVideoBitrateAllocatorFactory();
|
|
|
|
std::ostringstream contentId;
|
|
contentId << mediaContent.ssrc;
|
|
|
|
_videoChannel = _channelManager->CreateVideoChannel(call, cricket::MediaConfig(), contentId.str(), false, NativeNetworkingImpl::getDefaulCryptoOptions(), cricket::VideoOptions(), _videoBitrateAllocatorFactory.get());
|
|
_threads->getNetworkThread()->BlockingCall([&]() {
|
|
_videoChannel->SetRtpTransport(rtpTransport);
|
|
});
|
|
|
|
std::vector<cricket::VideoCodec> codecs;
|
|
for (const auto &payloadType : mediaContent.payloadTypes) {
|
|
cricket::VideoCodec codec(payloadType.id, payloadType.name);
|
|
for (const auto ¶meter : payloadType.parameters) {
|
|
codec.SetParam(parameter.first, parameter.second);
|
|
}
|
|
for (const auto &feedbackType : payloadType.feedbackTypes) {
|
|
codec.AddFeedbackParam(cricket::FeedbackParam(feedbackType.type, feedbackType.subtype));
|
|
}
|
|
codecs.push_back(std::move(codec));
|
|
}
|
|
|
|
auto outgoingVideoDescription = std::make_unique<cricket::VideoContentDescription>();
|
|
for (const auto &rtpExtension : mediaContent.rtpExtensions) {
|
|
outgoingVideoDescription->AddRtpHeaderExtension(webrtc::RtpExtension(rtpExtension.uri, rtpExtension.id));
|
|
}
|
|
outgoingVideoDescription->set_rtcp_mux(true);
|
|
outgoingVideoDescription->set_rtcp_reduced_size(true);
|
|
outgoingVideoDescription->set_direction(webrtc::RtpTransceiverDirection::kRecvOnly);
|
|
outgoingVideoDescription->set_codecs(codecs);
|
|
outgoingVideoDescription->set_bandwidth(-1);
|
|
|
|
cricket::StreamParams videoRecvStreamParams;
|
|
|
|
_mainVideoSsrc = mediaContent.ssrc;
|
|
|
|
std::vector<uint32_t> allSsrcs;
|
|
for (const auto &group : mediaContent.ssrcGroups) {
|
|
for (auto ssrc : group.ssrcs) {
|
|
if (std::find(allSsrcs.begin(), allSsrcs.end(), ssrc) == allSsrcs.end()) {
|
|
allSsrcs.push_back(ssrc);
|
|
}
|
|
}
|
|
|
|
cricket::SsrcGroup parsedGroup(group.semantics, group.ssrcs);
|
|
videoRecvStreamParams.ssrc_groups.push_back(parsedGroup);
|
|
}
|
|
videoRecvStreamParams.ssrcs = allSsrcs;
|
|
|
|
videoRecvStreamParams.cname = "cname";
|
|
videoRecvStreamParams.set_stream_ids({ contentId.str() });
|
|
|
|
auto incomingVideoDescription = std::make_unique<cricket::VideoContentDescription>();
|
|
for (const auto &rtpExtension : mediaContent.rtpExtensions) {
|
|
incomingVideoDescription->AddRtpHeaderExtension(webrtc::RtpExtension(rtpExtension.uri, rtpExtension.id));
|
|
}
|
|
incomingVideoDescription->set_rtcp_mux(true);
|
|
incomingVideoDescription->set_rtcp_reduced_size(true);
|
|
incomingVideoDescription->set_direction(webrtc::RtpTransceiverDirection::kSendOnly);
|
|
incomingVideoDescription->set_codecs(codecs);
|
|
incomingVideoDescription->set_bandwidth(-1);
|
|
|
|
incomingVideoDescription->AddStream(videoRecvStreamParams);
|
|
|
|
threads->getWorkerThread()->BlockingCall([&]() {
|
|
_videoChannel->SetPayloadTypeDemuxingEnabled(false);
|
|
std::string errorDesc;
|
|
_videoChannel->SetLocalContent(outgoingVideoDescription.get(), webrtc::SdpType::kOffer, errorDesc);
|
|
_videoChannel->SetRemoteContent(incomingVideoDescription.get(), webrtc::SdpType::kAnswer, errorDesc);
|
|
|
|
_videoChannel->media_channel()->SetSink(_mainVideoSsrc, _videoSink.get());
|
|
});
|
|
|
|
_videoChannel->Enable(true);
|
|
}
|
|
|
|
~IncomingV2VideoChannel() {
|
|
_videoChannel->Enable(false);
|
|
_threads->getNetworkThread()->BlockingCall([&]() {
|
|
_videoChannel->SetRtpTransport(nullptr);
|
|
});
|
|
_channelManager->DestroyChannel(_videoChannel);
|
|
_videoChannel = nullptr;
|
|
}
|
|
|
|
void addSink(std::weak_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> impl) {
|
|
_videoSink->addSink(impl);
|
|
}
|
|
|
|
uint32_t ssrc() const {
|
|
return _mainVideoSsrc;
|
|
}
|
|
|
|
private:
|
|
void OnSentPacket_w(const rtc::SentPacket& sent_packet) {
|
|
_call->OnSentPacket(sent_packet);
|
|
}
|
|
|
|
private:
|
|
std::shared_ptr<Threads> _threads;
|
|
uint32_t _mainVideoSsrc = 0;
|
|
std::unique_ptr<VideoSinkImpl> _videoSink;
|
|
std::unique_ptr<webrtc::VideoBitrateAllocatorFactory> _videoBitrateAllocatorFactory;
|
|
// Memory is managed by _channelManager
|
|
cricket::VideoChannel *_videoChannel;
|
|
// Memory is managed externally
|
|
ChannelManager *_channelManager = nullptr;
|
|
webrtc::Call *_call = nullptr;
|
|
};
|
|
|
|
template<typename T>
|
|
struct StateLogRecord {
|
|
int64_t timestamp = 0;
|
|
T record;
|
|
|
|
explicit StateLogRecord(int32_t timestamp_, T &&record_) :
|
|
timestamp(timestamp_),
|
|
record(std::move(record_)) {
|
|
}
|
|
};
|
|
|
|
struct NetworkStateLogRecord {
|
|
bool isConnected = false;
|
|
bool isFailed = false;
|
|
absl::optional<NativeNetworkingImpl::RouteDescription> route;
|
|
absl::optional<NativeNetworkingImpl::ConnectionDescription> connection;
|
|
|
|
bool operator==(NetworkStateLogRecord const &rhs) const {
|
|
if (isConnected != rhs.isConnected) {
|
|
return false;
|
|
}
|
|
if (isFailed != rhs.isFailed) {
|
|
return false;
|
|
}
|
|
if (route != rhs.route) {
|
|
return false;
|
|
}
|
|
if (connection != rhs.connection) {
|
|
return false;
|
|
}
|
|
|
|
return true;
|
|
}
|
|
};
|
|
|
|
struct NetworkBitrateLogRecord {
|
|
int32_t bitrate = 0;
|
|
};
|
|
|
|
} // namespace
|
|
|
|
class InstanceV2ImplInternal : public std::enable_shared_from_this<InstanceV2ImplInternal> {
|
|
public:
|
|
InstanceV2ImplInternal(Descriptor &&descriptor, std::shared_ptr<Threads> threads) :
|
|
_signalingProtocolVersion(signalingProtocolVersion(descriptor.version)),
|
|
_threads(threads),
|
|
_rtcServers(descriptor.rtcServers),
|
|
_proxy(std::move(descriptor.proxy)),
|
|
_enableP2P(descriptor.config.enableP2P),
|
|
_encryptionKey(std::move(descriptor.encryptionKey)),
|
|
_stateUpdated(descriptor.stateUpdated),
|
|
_signalBarsUpdated(descriptor.signalBarsUpdated),
|
|
_audioLevelsUpdated(descriptor.audioLevelsUpdated),
|
|
_remoteBatteryLevelIsLowUpdated(descriptor.remoteBatteryLevelIsLowUpdated),
|
|
_remoteMediaStateUpdated(descriptor.remoteMediaStateUpdated),
|
|
_remotePrefferedAspectRatioUpdated(descriptor.remotePrefferedAspectRatioUpdated),
|
|
_signalingDataEmitted(descriptor.signalingDataEmitted),
|
|
_createAudioDeviceModule(descriptor.createAudioDeviceModule),
|
|
_statsLogPath(descriptor.config.statsLogPath),
|
|
_eventLog(std::make_unique<webrtc::RtcEventLogNull>()),
|
|
_taskQueueFactory(webrtc::CreateDefaultTaskQueueFactory()),
|
|
_initialInputDeviceId(std::move(descriptor.initialInputDeviceId)),
|
|
_initialOutputDeviceId(std::move(descriptor.initialOutputDeviceId)),
|
|
_videoCapture(descriptor.videoCapture),
|
|
_platformContext(descriptor.platformContext) {
|
|
webrtc::field_trial::InitFieldTrialsFromString(
|
|
"WebRTC-DataChannel-Dcsctp/Enabled/"
|
|
"WebRTC-Audio-MinimizeResamplingOnMobile/Enabled/"
|
|
"WebRTC-Audio-iOS-Holding/Enabled/"
|
|
);
|
|
}
|
|
|
|
~InstanceV2ImplInternal() {
|
|
_incomingAudioChannel.reset();
|
|
_incomingVideoChannel.reset();
|
|
_incomingScreencastChannel.reset();
|
|
_outgoingAudioChannel.reset();
|
|
_outgoingVideoChannel.reset();
|
|
_outgoingScreencastChannel.reset();
|
|
_currentSink.reset();
|
|
|
|
_channelManager.reset();
|
|
|
|
_threads->getWorkerThread()->BlockingCall([&]() {
|
|
_call.reset();
|
|
_audioDeviceModule = nullptr;
|
|
});
|
|
|
|
_contentNegotiationContext.reset();
|
|
|
|
_networking->perform([](NativeNetworkingImpl *networking) {
|
|
networking->stop();
|
|
});
|
|
|
|
_threads->getNetworkThread()->BlockingCall([]() {
|
|
});
|
|
}
|
|
|
|
void start() {
|
|
_startTimestamp = rtc::TimeMillis();
|
|
|
|
const auto weak = std::weak_ptr<InstanceV2ImplInternal>(shared_from_this());
|
|
|
|
if (_signalingProtocolVersion == SignalingProtocolVersion::V3) {
|
|
_signalingConnection = std::make_unique<SignalingSctpConnection>(
|
|
_threads,
|
|
[threads = _threads, weak](const std::vector<uint8_t> &data) {
|
|
threads->getMediaThread()->PostTask([weak, data] {
|
|
const auto strong = weak.lock();
|
|
if (!strong) {
|
|
return;
|
|
}
|
|
|
|
strong->onSignalingData(data);
|
|
});
|
|
},
|
|
[signalingDataEmitted = _signalingDataEmitted](const std::vector<uint8_t> &data) {
|
|
signalingDataEmitted(data);
|
|
}
|
|
);
|
|
}
|
|
if (!_signalingConnection) {
|
|
_signalingConnection = std::make_unique<ExternalSignalingConnection>(
|
|
[threads = _threads, weak](const std::vector<uint8_t> &data) {
|
|
threads->getMediaThread()->PostTask([weak, data] {
|
|
const auto strong = weak.lock();
|
|
if (!strong) {
|
|
return;
|
|
}
|
|
|
|
strong->onSignalingData(data);
|
|
});
|
|
},
|
|
[signalingDataEmitted = _signalingDataEmitted](const std::vector<uint8_t> &data) {
|
|
signalingDataEmitted(data);
|
|
}
|
|
);
|
|
}
|
|
|
|
_signalingConnection->start();
|
|
|
|
absl::optional<Proxy> proxy;
|
|
if (_proxy) {
|
|
proxy = *(_proxy.get());
|
|
}
|
|
|
|
_networking.reset(new ThreadLocalObject<NativeNetworkingImpl>(_threads->getNetworkThread(), [weak, threads = _threads, isOutgoing = _encryptionKey.isOutgoing, rtcServers = _rtcServers, proxy, enableP2P = _enableP2P]() {
|
|
return new NativeNetworkingImpl(NativeNetworkingImpl::Configuration{
|
|
.isOutgoing = isOutgoing,
|
|
.enableStunMarking = false,
|
|
.enableTCP = false,
|
|
.enableP2P = enableP2P,
|
|
.rtcServers = rtcServers,
|
|
.proxy = proxy,
|
|
.stateUpdated = [threads, weak](const NativeNetworkingImpl::State &state) {
|
|
threads->getMediaThread()->PostTask([=] {
|
|
const auto strong = weak.lock();
|
|
if (!strong) {
|
|
return;
|
|
}
|
|
strong->onNetworkStateUpdated(state);
|
|
});
|
|
},
|
|
.candidateGathered = [threads, weak](const cricket::Candidate &candidate) {
|
|
threads->getMediaThread()->PostTask([=] {
|
|
const auto strong = weak.lock();
|
|
if (!strong) {
|
|
return;
|
|
}
|
|
|
|
strong->sendCandidate(candidate);
|
|
});
|
|
},
|
|
.transportMessageReceived = [threads, weak](rtc::CopyOnWriteBuffer const &packet, bool isMissing) {
|
|
threads->getMediaThread()->PostTask([=] {
|
|
const auto strong = weak.lock();
|
|
if (!strong) {
|
|
return;
|
|
}
|
|
});
|
|
},
|
|
.rtcpPacketReceived = [threads, weak](rtc::CopyOnWriteBuffer const &packet, int64_t timestamp) {
|
|
const auto strong = weak.lock();
|
|
if (!strong) {
|
|
return;
|
|
}
|
|
strong->_call->Receiver()->DeliverPacket(webrtc::MediaType::ANY, packet, timestamp);
|
|
},
|
|
.dataChannelStateUpdated = [threads, weak](bool isDataChannelOpen) {
|
|
threads->getMediaThread()->PostTask([=] {
|
|
const auto strong = weak.lock();
|
|
if (!strong) {
|
|
return;
|
|
}
|
|
strong->onDataChannelStateUpdated(isDataChannelOpen);
|
|
});
|
|
},
|
|
.dataChannelMessageReceived = [threads, weak](std::string const &message) {
|
|
threads->getMediaThread()->PostTask([=] {
|
|
const auto strong = weak.lock();
|
|
if (!strong) {
|
|
return;
|
|
}
|
|
strong->onDataChannelMessage(message);
|
|
});
|
|
},
|
|
.threads = threads
|
|
});
|
|
}));
|
|
|
|
PlatformInterface::SharedInstance()->configurePlatformAudio();
|
|
|
|
|
|
_threads->getWorkerThread()->BlockingCall([&]() {
|
|
_audioDeviceModule = createAudioDeviceModule();
|
|
});
|
|
|
|
cricket::MediaEngineDependencies mediaDeps;
|
|
mediaDeps.task_queue_factory = _taskQueueFactory.get();
|
|
mediaDeps.audio_encoder_factory = webrtc::CreateAudioEncoderFactory<webrtc::AudioEncoderOpus, webrtc::AudioEncoderL16>();
|
|
mediaDeps.audio_decoder_factory = webrtc::CreateAudioDecoderFactory<webrtc::AudioDecoderOpus, webrtc::AudioDecoderL16>();
|
|
|
|
mediaDeps.video_encoder_factory = PlatformInterface::SharedInstance()->makeVideoEncoderFactory(_platformContext, true);
|
|
mediaDeps.video_decoder_factory = PlatformInterface::SharedInstance()->makeVideoDecoderFactory(_platformContext);
|
|
|
|
mediaDeps.adm = _audioDeviceModule;
|
|
|
|
|
|
|
|
_availableVideoFormats = mediaDeps.video_encoder_factory->GetSupportedFormats();
|
|
|
|
std::unique_ptr<cricket::MediaEngineInterface> mediaEngine = cricket::CreateMediaEngine(std::move(mediaDeps));
|
|
|
|
_channelManager = ChannelManager::Create(
|
|
std::move(mediaEngine),
|
|
_threads->getWorkerThread(),
|
|
_threads->getNetworkThread()
|
|
);
|
|
|
|
webrtc::Call::Config callConfig(_eventLog.get(), _threads->getNetworkThread());
|
|
callConfig.task_queue_factory = _taskQueueFactory.get();
|
|
callConfig.trials = &fieldTrialsBasedConfig;
|
|
|
|
_threads->getNetworkThread()->BlockingCall([&]() {
|
|
_rtpTransport = _networking->getSyncAssumingSameThread()->getRtpTransport();
|
|
});
|
|
|
|
_threads->getWorkerThread()->BlockingCall([&]() {
|
|
callConfig.audio_state = _channelManager->media_engine()->voice().GetAudioState();
|
|
_call.reset(webrtc::Call::Create(callConfig));
|
|
});
|
|
|
|
_uniqueRandomIdGenerator.reset(new rtc::UniqueRandomIdGenerator());
|
|
|
|
_contentNegotiationContext = std::make_unique<ContentNegotiationContext>(fieldTrialsBasedConfig, _encryptionKey.isOutgoing, _uniqueRandomIdGenerator.get());
|
|
_contentNegotiationContext->copyCodecsFromChannelManager(_channelManager->media_engine(), false);
|
|
|
|
_outgoingAudioChannelId = _contentNegotiationContext->addOutgoingChannel(signaling::MediaContent::Type::Audio);
|
|
|
|
_videoBitrateAllocatorFactory = webrtc::CreateBuiltinVideoBitrateAllocatorFactory();
|
|
|
|
_networking->perform([](NativeNetworkingImpl *networking) {
|
|
networking->start();
|
|
});
|
|
|
|
if (_videoCapture) {
|
|
setVideoCapture(_videoCapture);
|
|
}
|
|
|
|
beginSignaling();
|
|
|
|
adjustBitratePreferences(true);
|
|
|
|
beginQualityTimer(0);
|
|
beginLogTimer(0);
|
|
|
|
|
|
}
|
|
|
|
void beginQualityTimer(int delayMs) {
|
|
const auto weak = std::weak_ptr<InstanceV2ImplInternal>(shared_from_this());
|
|
_threads->getMediaThread()->PostDelayedTask([weak]() {
|
|
auto strong = weak.lock();
|
|
if (!strong) {
|
|
return;
|
|
}
|
|
|
|
|
|
|
|
strong->beginQualityTimer(500);
|
|
}, webrtc::TimeDelta::Millis(delayMs));
|
|
}
|
|
|
|
void beginLogTimer(int delayMs) {
|
|
const auto weak = std::weak_ptr<InstanceV2ImplInternal>(shared_from_this());
|
|
_threads->getMediaThread()->PostDelayedTask([weak]() {
|
|
auto strong = weak.lock();
|
|
if (!strong) {
|
|
return;
|
|
}
|
|
|
|
strong->writeStateLogRecords();
|
|
|
|
strong->beginLogTimer(1000);
|
|
}, webrtc::TimeDelta::Millis(delayMs));
|
|
}
|
|
|
|
void writeStateLogRecords() {
|
|
const auto weak = std::weak_ptr<InstanceV2ImplInternal>(shared_from_this());
|
|
_threads->getWorkerThread()->PostTask([weak]() {
|
|
auto strong = weak.lock();
|
|
if (!strong) {
|
|
return;
|
|
}
|
|
|
|
auto stats = strong->_call->GetStats();
|
|
float sendBitrateKbps = ((float)stats.send_bandwidth_bps / 1000.0f);
|
|
|
|
strong->_threads->getMediaThread()->PostTask([weak, sendBitrateKbps]() {
|
|
auto strong = weak.lock();
|
|
if (!strong) {
|
|
return;
|
|
}
|
|
|
|
float bitrateNorm = 16.0f;
|
|
if (strong->_outgoingVideoChannel) {
|
|
bitrateNorm = 600.0f;
|
|
}
|
|
|
|
float signalBarsNorm = 4.0f;
|
|
float adjustedQuality = sendBitrateKbps / bitrateNorm;
|
|
adjustedQuality = fmaxf(0.0f, adjustedQuality);
|
|
adjustedQuality = fminf(1.0f, adjustedQuality);
|
|
if (strong->_signalBarsUpdated) {
|
|
strong->_signalBarsUpdated((int)(adjustedQuality * signalBarsNorm));
|
|
}
|
|
|
|
NetworkBitrateLogRecord networkBitrateLogRecord;
|
|
networkBitrateLogRecord.bitrate = (int32_t)sendBitrateKbps;
|
|
|
|
strong->_networkBitrateLogRecords.emplace_back(rtc::TimeMillis(), std::move(networkBitrateLogRecord));
|
|
});
|
|
});
|
|
}
|
|
|
|
void sendSignalingMessage(signaling::Message const &message) {
|
|
auto data = message.serialize();
|
|
sendRawSignalingMessage(data);
|
|
}
|
|
|
|
void sendRawSignalingMessage(std::vector<uint8_t> const &data) {
|
|
RTC_LOG(LS_INFO) << "sendSignalingMessage: " << std::string(data.begin(), data.end());
|
|
|
|
if (_signalingConnection && _signalingEncryptedConnection) {
|
|
switch (_signalingProtocolVersion) {
|
|
case SignalingProtocolVersion::V1:
|
|
case SignalingProtocolVersion::V3: {
|
|
std::vector<uint8_t> packetData;
|
|
if (signalingProtocolSupportsCompression(_signalingProtocolVersion)) {
|
|
if (const auto compressedData = gzipData(data)) {
|
|
packetData = std::move(compressedData.value());
|
|
} else {
|
|
RTC_LOG(LS_ERROR) << "Could not gzip signaling message";
|
|
}
|
|
} else {
|
|
packetData = data;
|
|
}
|
|
|
|
if (const auto message = _signalingEncryptedConnection->encryptRawPacket(rtc::CopyOnWriteBuffer(packetData.data(), packetData.size()))) {
|
|
_signalingConnection->send(std::vector<uint8_t>(message.value().data(), message.value().data() + message.value().size()));
|
|
} else {
|
|
RTC_LOG(LS_ERROR) << "Could not encrypt signaling message";
|
|
}
|
|
break;
|
|
}
|
|
case SignalingProtocolVersion::V2: {
|
|
rtc::CopyOnWriteBuffer message;
|
|
message.AppendData(data.data(), data.size());
|
|
|
|
commitSendSignalingMessage(_signalingEncryptedConnection->prepareForSendingRawMessage(message, true));
|
|
|
|
break;
|
|
}
|
|
default: {
|
|
RTC_DCHECK_NOTREACHED();
|
|
|
|
break;
|
|
}
|
|
}
|
|
} else {
|
|
RTC_LOG(LS_ERROR) << "sendSignalingMessage encryption not available";
|
|
}
|
|
}
|
|
|
|
void commitSendSignalingMessage(absl::optional<EncryptedConnection::EncryptedPacket> packet) {
|
|
if (!packet) {
|
|
return;
|
|
}
|
|
|
|
if (_signalingConnection) {
|
|
_signalingConnection->send(packet.value().bytes);
|
|
}
|
|
}
|
|
|
|
void beginSignaling() {
|
|
const auto weak = std::weak_ptr<InstanceV2ImplInternal>(shared_from_this());
|
|
|
|
_signalingEncryptedConnection = std::make_unique<EncryptedConnection>(
|
|
EncryptedConnection::Type::Signaling,
|
|
_encryptionKey,
|
|
[weak, threads = _threads](int delayMs, int cause) {
|
|
if (delayMs == 0) {
|
|
threads->getMediaThread()->PostTask([weak, cause]() {
|
|
const auto strong = weak.lock();
|
|
if (!strong) {
|
|
return;
|
|
}
|
|
|
|
strong->sendPendingSignalingServiceData(cause);
|
|
});
|
|
} else {
|
|
threads->getMediaThread()->PostDelayedTask([weak, cause]() {
|
|
const auto strong = weak.lock();
|
|
if (!strong) {
|
|
return;
|
|
}
|
|
|
|
strong->sendPendingSignalingServiceData(cause);
|
|
}, webrtc::TimeDelta::Millis(delayMs));
|
|
}
|
|
}
|
|
);
|
|
|
|
if (_encryptionKey.isOutgoing) {
|
|
sendInitialSetup();
|
|
}
|
|
}
|
|
|
|
void sendPendingSignalingServiceData(int cause) {
|
|
commitSendSignalingMessage(_signalingEncryptedConnection->prepareForSendingService(cause));
|
|
}
|
|
|
|
void createNegotiatedChannels() {
|
|
const auto coordinatedState = _contentNegotiationContext->coordinatedState();
|
|
if (!coordinatedState) {
|
|
return;
|
|
}
|
|
|
|
if (_outgoingAudioChannelId) {
|
|
const auto audioSsrc = _contentNegotiationContext->outgoingChannelSsrc(_outgoingAudioChannelId.value());
|
|
if (audioSsrc) {
|
|
if (_outgoingAudioChannel && _outgoingAudioChannel->ssrc() != audioSsrc.value()) {
|
|
_outgoingAudioChannel.reset();
|
|
}
|
|
|
|
absl::optional<signaling::MediaContent> outgoingAudioContent;
|
|
for (const auto &content : coordinatedState->outgoingContents) {
|
|
if (content.type == signaling::MediaContent::Type::Audio && content.ssrc == audioSsrc.value()) {
|
|
outgoingAudioContent = content;
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (outgoingAudioContent) {
|
|
if (!_outgoingAudioChannel) {
|
|
_outgoingAudioChannel.reset(new OutgoingAudioChannel(
|
|
_call.get(),
|
|
_channelManager.get(),
|
|
_uniqueRandomIdGenerator.get(),
|
|
&_audioSource,
|
|
_rtpTransport,
|
|
outgoingAudioContent.value(),
|
|
_threads
|
|
));
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
if (_outgoingVideoChannelId) {
|
|
const auto videoSsrc = _contentNegotiationContext->outgoingChannelSsrc(_outgoingVideoChannelId.value());
|
|
if (videoSsrc) {
|
|
if (_outgoingVideoChannel && _outgoingVideoChannel->ssrc() != videoSsrc.value()) {
|
|
_outgoingVideoChannel.reset();
|
|
}
|
|
|
|
absl::optional<signaling::MediaContent> outgoingVideoContent;
|
|
for (const auto &content : coordinatedState->outgoingContents) {
|
|
if (content.type == signaling::MediaContent::Type::Video && content.ssrc == videoSsrc.value()) {
|
|
outgoingVideoContent = content;
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (outgoingVideoContent) {
|
|
if (!_outgoingVideoChannel) {
|
|
const auto weak = std::weak_ptr<InstanceV2ImplInternal>(shared_from_this());
|
|
|
|
_outgoingVideoChannel.reset(new OutgoingVideoChannel(
|
|
_threads,
|
|
_channelManager.get(),
|
|
_call.get(),
|
|
_rtpTransport,
|
|
_uniqueRandomIdGenerator.get(),
|
|
_videoBitrateAllocatorFactory.get(),
|
|
[threads = _threads, weak]() {
|
|
threads->getMediaThread()->PostTask([=] {
|
|
const auto strong = weak.lock();
|
|
if (!strong) {
|
|
return;
|
|
}
|
|
strong->sendMediaState();
|
|
});
|
|
},
|
|
outgoingVideoContent.value(),
|
|
false
|
|
));
|
|
|
|
if (_videoCapture) {
|
|
_outgoingVideoChannel->setVideoCapture(_videoCapture);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
if (_outgoingScreencastChannelId) {
|
|
const auto screencastSsrc = _contentNegotiationContext->outgoingChannelSsrc(_outgoingScreencastChannelId.value());
|
|
if (screencastSsrc) {
|
|
if (_outgoingScreencastChannel && _outgoingScreencastChannel->ssrc() != screencastSsrc.value()) {
|
|
_outgoingScreencastChannel.reset();
|
|
}
|
|
|
|
absl::optional<signaling::MediaContent> outgoingScreencastContent;
|
|
for (const auto &content : coordinatedState->outgoingContents) {
|
|
if (content.type == signaling::MediaContent::Type::Video && content.ssrc == screencastSsrc.value()) {
|
|
outgoingScreencastContent = content;
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (outgoingScreencastContent) {
|
|
if (!_outgoingScreencastChannel) {
|
|
const auto weak = std::weak_ptr<InstanceV2ImplInternal>(shared_from_this());
|
|
|
|
_outgoingScreencastChannel.reset(new OutgoingVideoChannel(
|
|
_threads,
|
|
_channelManager.get(),
|
|
_call.get(),
|
|
_rtpTransport,
|
|
_uniqueRandomIdGenerator.get(),
|
|
_videoBitrateAllocatorFactory.get(),
|
|
[threads = _threads, weak]() {
|
|
threads->getMediaThread()->PostTask([=] {
|
|
const auto strong = weak.lock();
|
|
if (!strong) {
|
|
return;
|
|
}
|
|
strong->sendMediaState();
|
|
});
|
|
},
|
|
outgoingScreencastContent.value(),
|
|
true
|
|
));
|
|
|
|
if (_screencastCapture) {
|
|
_outgoingScreencastChannel->setVideoCapture(_screencastCapture);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
for (const auto &content : coordinatedState->incomingContents) {
|
|
switch (content.type) {
|
|
case signaling::MediaContent::Type::Audio: {
|
|
if (_incomingAudioChannel && _incomingAudioChannel->ssrc() != content.ssrc) {
|
|
_incomingAudioChannel.reset();
|
|
}
|
|
|
|
if (!_incomingAudioChannel) {
|
|
_incomingAudioChannel.reset(new IncomingV2AudioChannel(
|
|
_channelManager.get(),
|
|
_call.get(),
|
|
_rtpTransport,
|
|
_uniqueRandomIdGenerator.get(),
|
|
content,
|
|
_threads
|
|
));
|
|
}
|
|
|
|
break;
|
|
}
|
|
case signaling::MediaContent::Type::Video: {
|
|
if (_incomingVideoChannel && _incomingVideoChannel->ssrc() != content.ssrc) {
|
|
_incomingVideoChannel.reset();
|
|
}
|
|
|
|
if (!_incomingVideoChannel) {
|
|
_incomingVideoChannel.reset(new IncomingV2VideoChannel(
|
|
_channelManager.get(),
|
|
_call.get(),
|
|
_rtpTransport,
|
|
_uniqueRandomIdGenerator.get(),
|
|
content,
|
|
_threads
|
|
));
|
|
_incomingVideoChannel->addSink(_currentSink);
|
|
}
|
|
|
|
break;
|
|
}
|
|
default: {
|
|
RTC_FATAL() << "Unknown media type";
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
/*
|
|
|
|
if (_negotiatedOutgoingScreencastContent) {
|
|
const auto weak = std::weak_ptr<InstanceV2ImplInternal>(shared_from_this());
|
|
|
|
_outgoingScreencastChannel.reset(new OutgoingVideoChannel(
|
|
_threads,
|
|
_channelManager.get(),
|
|
_call.get(),
|
|
_rtpTransport,
|
|
_uniqueRandomIdGenerator.get(),
|
|
_videoBitrateAllocatorFactory.get(),
|
|
[threads = _threads, weak]() {
|
|
threads->getMediaThread()->PostTask([=] {
|
|
const auto strong = weak.lock();
|
|
if (!strong) {
|
|
return;
|
|
}
|
|
strong->sendMediaState();
|
|
});
|
|
},
|
|
_negotiatedOutgoingScreencastContent.value(),
|
|
true
|
|
));
|
|
|
|
if (_screencastCapture) {
|
|
_outgoingScreencastChannel->setVideoCapture(_screencastCapture);
|
|
}
|
|
}
|
|
|
|
if (_negotiatedOutgoingAudioContent) {
|
|
_outgoingAudioChannel.reset(new OutgoingAudioChannel(
|
|
_call.get(),
|
|
_channelManager.get(),
|
|
_uniqueRandomIdGenerator.get(),
|
|
&_audioSource,
|
|
_rtpTransport,
|
|
_negotiatedOutgoingAudioContent.value(),
|
|
_threads
|
|
));
|
|
}*/
|
|
|
|
adjustBitratePreferences(true);
|
|
sendMediaState();
|
|
}
|
|
|
|
void sendInitialSetup() {
|
|
const auto weak = std::weak_ptr<InstanceV2ImplInternal>(shared_from_this());
|
|
|
|
_networking->perform([weak, threads = _threads, isOutgoing = _encryptionKey.isOutgoing](NativeNetworkingImpl *networking) {
|
|
auto localFingerprint = networking->getLocalFingerprint();
|
|
std::string hash = localFingerprint->algorithm;
|
|
std::string fingerprint = localFingerprint->GetRfc4572Fingerprint();
|
|
std::string setup;
|
|
if (isOutgoing) {
|
|
setup = "actpass";
|
|
} else {
|
|
setup = "passive";
|
|
}
|
|
|
|
auto localIceParams = networking->getLocalIceParameters();
|
|
std::string ufrag = localIceParams.ufrag;
|
|
std::string pwd = localIceParams.pwd;
|
|
bool supportsRenomination = localIceParams.supportsRenomination;
|
|
|
|
threads->getMediaThread()->PostTask([weak, ufrag, pwd, supportsRenomination, hash, fingerprint, setup, localIceParams]() {
|
|
const auto strong = weak.lock();
|
|
if (!strong) {
|
|
return;
|
|
}
|
|
|
|
signaling::InitialSetupMessage data;
|
|
|
|
data.ufrag = ufrag;
|
|
data.pwd = pwd;
|
|
data.supportsRenomination = supportsRenomination;
|
|
|
|
signaling::DtlsFingerprint dtlsFingerprint;
|
|
dtlsFingerprint.hash = hash;
|
|
dtlsFingerprint.fingerprint = fingerprint;
|
|
dtlsFingerprint.setup = setup;
|
|
data.fingerprints.push_back(std::move(dtlsFingerprint));
|
|
|
|
signaling::Message message;
|
|
message.data = std::move(data);
|
|
strong->sendSignalingMessage(message);
|
|
});
|
|
});
|
|
}
|
|
|
|
void sendOfferIfNeeded() {
|
|
if (const auto offer = _contentNegotiationContext->getPendingOffer()) {
|
|
signaling::NegotiateChannelsMessage data;
|
|
|
|
data.exchangeId = offer->exchangeId;
|
|
|
|
data.contents = offer->contents;
|
|
|
|
signaling::Message message;
|
|
message.data = std::move(data);
|
|
sendSignalingMessage(message);
|
|
}
|
|
}
|
|
|
|
void receiveSignalingData(const std::vector<uint8_t> &data) {
|
|
if (_signalingConnection) {
|
|
_signalingConnection->receiveExternal(data);
|
|
} else {
|
|
RTC_LOG(LS_ERROR) << "receiveSignalingData: signalingConnection is not available";
|
|
}
|
|
}
|
|
|
|
void onSignalingData(const std::vector<uint8_t> &data) {
|
|
if (_signalingEncryptedConnection) {
|
|
switch (_signalingProtocolVersion) {
|
|
case SignalingProtocolVersion::V1:
|
|
case SignalingProtocolVersion::V3: {
|
|
if (const auto message = _signalingEncryptedConnection->decryptRawPacket(rtc::CopyOnWriteBuffer(data.data(), data.size()))) {
|
|
processSignalingMessage(message.value());
|
|
} else {
|
|
RTC_LOG(LS_ERROR) << "receiveSignalingData could not decrypt signaling data";
|
|
}
|
|
|
|
break;
|
|
}
|
|
case SignalingProtocolVersion::V2: {
|
|
if (const auto packet = _signalingEncryptedConnection->handleIncomingRawPacket((const char *)data.data(), data.size())) {
|
|
processSignalingMessage(packet.value().main.message);
|
|
|
|
for (const auto &additional : packet.value().additional) {
|
|
processSignalingMessage(additional.message);
|
|
}
|
|
}
|
|
|
|
break;
|
|
}
|
|
default: {
|
|
RTC_DCHECK_NOTREACHED();
|
|
|
|
break;
|
|
}
|
|
}
|
|
} else {
|
|
RTC_LOG(LS_ERROR) << "receiveSignalingData encryption not available";
|
|
}
|
|
}
|
|
|
|
void processSignalingMessage(rtc::CopyOnWriteBuffer const &data) {
|
|
std::vector<uint8_t> decryptedData = std::vector<uint8_t>(data.data(), data.data() + data.size());
|
|
|
|
if (isGzip(decryptedData)) {
|
|
if (const auto decompressedData = gunzipData(decryptedData, 2 * 1024 * 1024)) {
|
|
processSignalingData(decompressedData.value());
|
|
} else {
|
|
RTC_LOG(LS_ERROR) << "receiveSignalingData could not decompress gzipped data";
|
|
}
|
|
} else {
|
|
processSignalingData(decryptedData);
|
|
}
|
|
}
|
|
|
|
void processSignalingData(const std::vector<uint8_t> &data) {
|
|
RTC_LOG(LS_INFO) << "processSignalingData: " << std::string(data.begin(), data.end());
|
|
|
|
const auto message = signaling::Message::parse(data);
|
|
if (!message) {
|
|
return;
|
|
}
|
|
const auto messageData = &message->data;
|
|
if (const auto initialSetup = absl::get_if<signaling::InitialSetupMessage>(messageData)) {
|
|
PeerIceParameters remoteIceParameters;
|
|
remoteIceParameters.ufrag = initialSetup->ufrag;
|
|
remoteIceParameters.pwd = initialSetup->pwd;
|
|
remoteIceParameters.supportsRenomination = initialSetup->supportsRenomination;
|
|
|
|
std::unique_ptr<rtc::SSLFingerprint> fingerprint;
|
|
std::string sslSetup;
|
|
if (initialSetup->fingerprints.size() != 0) {
|
|
fingerprint = rtc::SSLFingerprint::CreateUniqueFromRfc4572(initialSetup->fingerprints[0].hash, initialSetup->fingerprints[0].fingerprint);
|
|
sslSetup = initialSetup->fingerprints[0].setup;
|
|
}
|
|
|
|
_networking->perform([threads = _threads, remoteIceParameters = std::move(remoteIceParameters), fingerprint = std::move(fingerprint), sslSetup = std::move(sslSetup)](NativeNetworkingImpl *networking) {
|
|
networking->setRemoteParams(remoteIceParameters, fingerprint.get(), sslSetup);
|
|
});
|
|
|
|
_handshakeCompleted = true;
|
|
|
|
if (_encryptionKey.isOutgoing) {
|
|
sendOfferIfNeeded();
|
|
} else {
|
|
sendInitialSetup();
|
|
}
|
|
|
|
commitPendingIceCandidates();
|
|
} else if (const auto offerAnwer = absl::get_if<signaling::NegotiateChannelsMessage>(messageData)) {
|
|
auto negotiationContents = std::make_unique<ContentNegotiationContext::NegotiationContents>();
|
|
negotiationContents->exchangeId = offerAnwer->exchangeId;
|
|
negotiationContents->contents = offerAnwer->contents;
|
|
|
|
if (const auto response = _contentNegotiationContext->setRemoteNegotiationContent(std::move(negotiationContents))) {
|
|
signaling::NegotiateChannelsMessage data;
|
|
|
|
data.exchangeId = response->exchangeId;
|
|
data.contents = response->contents;
|
|
|
|
signaling::Message message;
|
|
message.data = std::move(data);
|
|
sendSignalingMessage(message);
|
|
}
|
|
|
|
sendOfferIfNeeded();
|
|
|
|
createNegotiatedChannels();
|
|
} else if (const auto candidatesList = absl::get_if<signaling::CandidatesMessage>(messageData)) {
|
|
for (const auto &candidate : candidatesList->iceCandidates) {
|
|
webrtc::JsepIceCandidate parseCandidate{ std::string(), 0 };
|
|
if (!parseCandidate.Initialize(candidate.sdpString, nullptr)) {
|
|
RTC_LOG(LS_ERROR) << "Could not parse candidate: " << candidate.sdpString;
|
|
continue;
|
|
}
|
|
_pendingIceCandidates.push_back(parseCandidate.candidate());
|
|
}
|
|
|
|
if (_handshakeCompleted) {
|
|
commitPendingIceCandidates();
|
|
}
|
|
} else if (const auto mediaState = absl::get_if<signaling::MediaStateMessage>(messageData)) {
|
|
AudioState mappedAudioState;
|
|
if (mediaState->isMuted) {
|
|
mappedAudioState = AudioState::Muted;
|
|
} else {
|
|
mappedAudioState = AudioState::Active;
|
|
}
|
|
|
|
VideoState mappedVideoState;
|
|
switch (mediaState->videoState) {
|
|
case signaling::MediaStateMessage::VideoState::Inactive: {
|
|
mappedVideoState = VideoState::Inactive;
|
|
break;
|
|
}
|
|
case signaling::MediaStateMessage::VideoState::Suspended: {
|
|
mappedVideoState = VideoState::Paused;
|
|
break;
|
|
}
|
|
case signaling::MediaStateMessage::VideoState::Active: {
|
|
mappedVideoState = VideoState::Active;
|
|
break;
|
|
}
|
|
default: {
|
|
RTC_FATAL() << "Unknown videoState";
|
|
break;
|
|
}
|
|
}
|
|
|
|
VideoState mappedScreencastState;
|
|
switch (mediaState->screencastState) {
|
|
case signaling::MediaStateMessage::VideoState::Inactive: {
|
|
mappedScreencastState = VideoState::Inactive;
|
|
break;
|
|
}
|
|
case signaling::MediaStateMessage::VideoState::Suspended: {
|
|
mappedScreencastState = VideoState::Paused;
|
|
break;
|
|
}
|
|
case signaling::MediaStateMessage::VideoState::Active: {
|
|
mappedScreencastState = VideoState::Active;
|
|
break;
|
|
}
|
|
default: {
|
|
RTC_FATAL() << "Unknown videoState";
|
|
break;
|
|
}
|
|
}
|
|
|
|
VideoState effectiveVideoState = mappedVideoState;
|
|
if (mappedScreencastState == VideoState::Active || mappedScreencastState == VideoState::Paused) {
|
|
effectiveVideoState = mappedScreencastState;
|
|
}
|
|
|
|
if (_remoteMediaStateUpdated) {
|
|
_remoteMediaStateUpdated(mappedAudioState, effectiveVideoState);
|
|
}
|
|
|
|
if (_remoteBatteryLevelIsLowUpdated) {
|
|
_remoteBatteryLevelIsLowUpdated(mediaState->isBatteryLow);
|
|
}
|
|
}
|
|
}
|
|
|
|
void commitPendingIceCandidates() {
|
|
if (_pendingIceCandidates.size() == 0) {
|
|
return;
|
|
}
|
|
_networking->perform([threads = _threads, parsedCandidates = _pendingIceCandidates](NativeNetworkingImpl *networking) {
|
|
networking->addCandidates(parsedCandidates);
|
|
});
|
|
_pendingIceCandidates.clear();
|
|
}
|
|
|
|
void onNetworkStateUpdated(NativeNetworkingImpl::State const &state) {
|
|
State mappedState;
|
|
if (state.isFailed) {
|
|
mappedState = State::Failed;
|
|
} else if (state.isReadyToSendData) {
|
|
mappedState = State::Established;
|
|
} else {
|
|
mappedState = State::Reconnecting;
|
|
}
|
|
|
|
NetworkStateLogRecord record;
|
|
record.isConnected = state.isReadyToSendData;
|
|
record.route = state.route;
|
|
record.connection = state.connection;
|
|
record.isFailed = state.isFailed;
|
|
|
|
if (!_currentNetworkStateLogRecord || !(_currentNetworkStateLogRecord.value() == record)) {
|
|
_currentNetworkStateLogRecord = record;
|
|
_networkStateLogRecords.emplace_back(rtc::TimeMillis(), std::move(record));
|
|
}
|
|
|
|
_networkState = state;
|
|
_stateUpdated(mappedState);
|
|
}
|
|
|
|
void onDataChannelStateUpdated(bool isDataChannelOpen) {
|
|
if (_isDataChannelOpen != isDataChannelOpen) {
|
|
_isDataChannelOpen = isDataChannelOpen;
|
|
|
|
if (_isDataChannelOpen) {
|
|
sendMediaState();
|
|
}
|
|
}
|
|
}
|
|
|
|
void sendDataChannelMessage(signaling::Message const &message) {
|
|
if (!_isDataChannelOpen) {
|
|
RTC_LOG(LS_ERROR) << "sendDataChannelMessage called, but data channel is not open";
|
|
return;
|
|
}
|
|
auto data = message.serialize();
|
|
std::string stringData(data.begin(), data.end());
|
|
RTC_LOG(LS_INFO) << "sendDataChannelMessage: " << stringData;
|
|
_networking->perform([stringData = std::move(stringData)](NativeNetworkingImpl *networking) {
|
|
networking->sendDataChannelMessage(stringData);
|
|
});
|
|
}
|
|
|
|
void onDataChannelMessage(std::string const &message) {
|
|
RTC_LOG(LS_INFO) << "dataChannelMessage received: " << message;
|
|
std::vector<uint8_t> data(message.begin(), message.end());
|
|
processSignalingData(data);
|
|
}
|
|
|
|
void sendMediaState() {
|
|
if (!_isDataChannelOpen) {
|
|
return;
|
|
}
|
|
signaling::Message message;
|
|
signaling::MediaStateMessage data;
|
|
data.isMuted = _isMicrophoneMuted;
|
|
data.isBatteryLow = _isBatteryLow;
|
|
if (_outgoingVideoChannel) {
|
|
if (_outgoingVideoChannel->videoCapture()) {
|
|
data.videoState = signaling::MediaStateMessage::VideoState::Active;
|
|
} else{
|
|
data.videoState = signaling::MediaStateMessage::VideoState::Inactive;
|
|
}
|
|
data.videoRotation = _outgoingVideoChannel->getRotation();
|
|
} else {
|
|
data.videoState = signaling::MediaStateMessage::VideoState::Inactive;
|
|
data.videoRotation = signaling::MediaStateMessage::VideoRotation::Rotation0;
|
|
}
|
|
if (_outgoingScreencastChannel) {
|
|
if (_outgoingScreencastChannel->videoCapture()) {
|
|
data.screencastState = signaling::MediaStateMessage::VideoState::Active;
|
|
} else{
|
|
data.screencastState = signaling::MediaStateMessage::VideoState::Inactive;
|
|
}
|
|
} else {
|
|
data.screencastState = signaling::MediaStateMessage::VideoState::Inactive;
|
|
}
|
|
message.data = std::move(data);
|
|
sendDataChannelMessage(message);
|
|
}
|
|
|
|
void sendCandidate(const cricket::Candidate &candidate) {
|
|
cricket::Candidate patchedCandidate = candidate;
|
|
patchedCandidate.set_component(1);
|
|
|
|
signaling::CandidatesMessage data;
|
|
|
|
signaling::IceCandidate serializedCandidate;
|
|
|
|
webrtc::JsepIceCandidate iceCandidate{ std::string(), 0 };
|
|
iceCandidate.SetCandidate(patchedCandidate);
|
|
std::string serialized;
|
|
const auto success = iceCandidate.ToString(&serialized);
|
|
assert(success);
|
|
(void)success;
|
|
|
|
serializedCandidate.sdpString = serialized;
|
|
|
|
data.iceCandidates.push_back(std::move(serializedCandidate));
|
|
|
|
signaling::Message message;
|
|
message.data = std::move(data);
|
|
sendSignalingMessage(message);
|
|
}
|
|
|
|
void setVideoCapture(std::shared_ptr<VideoCaptureInterface> videoCapture) {
|
|
auto videoCaptureImpl = GetVideoCaptureAssumingSameThread(videoCapture.get());
|
|
if (videoCaptureImpl) {
|
|
if (videoCaptureImpl->isScreenCapture()) {
|
|
_videoCapture = nullptr;
|
|
_screencastCapture = videoCapture;
|
|
|
|
if (_outgoingVideoChannel) {
|
|
_outgoingVideoChannel->setVideoCapture(nullptr);
|
|
}
|
|
if (_outgoingVideoChannelId) {
|
|
_contentNegotiationContext->removeOutgoingChannel(_outgoingVideoChannelId.value());
|
|
_outgoingVideoChannelId.reset();
|
|
}
|
|
|
|
if (_outgoingScreencastChannel) {
|
|
_outgoingScreencastChannel->setVideoCapture(videoCapture);
|
|
}
|
|
if (!_outgoingScreencastChannelId) {
|
|
_outgoingScreencastChannelId = _contentNegotiationContext->addOutgoingChannel(signaling::MediaContent::Type::Video);
|
|
}
|
|
} else {
|
|
_videoCapture = videoCapture;
|
|
_screencastCapture = nullptr;
|
|
|
|
if (_outgoingVideoChannel) {
|
|
_outgoingVideoChannel->setVideoCapture(videoCapture);
|
|
}
|
|
if (!_outgoingVideoChannelId) {
|
|
_outgoingVideoChannelId = _contentNegotiationContext->addOutgoingChannel(signaling::MediaContent::Type::Video);
|
|
}
|
|
|
|
if (_outgoingScreencastChannel) {
|
|
_outgoingScreencastChannel->setVideoCapture(nullptr);
|
|
}
|
|
if (_outgoingScreencastChannelId) {
|
|
_contentNegotiationContext->removeOutgoingChannel(_outgoingScreencastChannelId.value());
|
|
_outgoingScreencastChannelId.reset();
|
|
}
|
|
}
|
|
} else {
|
|
_videoCapture = nullptr;
|
|
_screencastCapture = nullptr;
|
|
|
|
if (_outgoingVideoChannel) {
|
|
_outgoingVideoChannel->setVideoCapture(nullptr);
|
|
}
|
|
|
|
if (_outgoingScreencastChannel) {
|
|
_outgoingScreencastChannel->setVideoCapture(nullptr);
|
|
}
|
|
|
|
if (_outgoingVideoChannelId) {
|
|
_contentNegotiationContext->removeOutgoingChannel(_outgoingVideoChannelId.value());
|
|
_outgoingVideoChannelId.reset();
|
|
}
|
|
|
|
if (_outgoingScreencastChannelId) {
|
|
_contentNegotiationContext->removeOutgoingChannel(_outgoingScreencastChannelId.value());
|
|
_outgoingScreencastChannelId.reset();
|
|
}
|
|
}
|
|
|
|
if (_handshakeCompleted) {
|
|
sendOfferIfNeeded();
|
|
sendMediaState();
|
|
adjustBitratePreferences(true);
|
|
createNegotiatedChannels();
|
|
}
|
|
}
|
|
|
|
void setRequestedVideoAspect(float aspect) {
|
|
}
|
|
|
|
void setNetworkType(NetworkType networkType) {
|
|
|
|
}
|
|
|
|
void setMuteMicrophone(bool muteMicrophone) {
|
|
if (_isMicrophoneMuted != muteMicrophone) {
|
|
_isMicrophoneMuted = muteMicrophone;
|
|
|
|
if (_outgoingAudioChannel) {
|
|
_outgoingAudioChannel->setIsMuted(muteMicrophone);
|
|
}
|
|
|
|
sendMediaState();
|
|
}
|
|
}
|
|
|
|
void setIncomingVideoOutput(std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) {
|
|
_currentSink = sink;
|
|
if (_incomingVideoChannel) {
|
|
_incomingVideoChannel->addSink(sink);
|
|
}
|
|
if (_incomingScreencastChannel) {
|
|
_incomingScreencastChannel->addSink(sink);
|
|
}
|
|
}
|
|
|
|
void setAudioInputDevice(std::string id) {
|
|
_threads->getWorkerThread()->BlockingCall([&]() {
|
|
SetAudioInputDeviceById(_audioDeviceModule.get(), id);
|
|
});
|
|
}
|
|
|
|
void setAudioOutputDevice(std::string id) {
|
|
_threads->getWorkerThread()->BlockingCall([&]() {
|
|
SetAudioOutputDeviceById(_audioDeviceModule.get(), id);
|
|
});
|
|
}
|
|
|
|
void setIsLowBatteryLevel(bool isLowBatteryLevel) {
|
|
if (_isBatteryLow != isLowBatteryLevel) {
|
|
_isBatteryLow = isLowBatteryLevel;
|
|
sendMediaState();
|
|
}
|
|
}
|
|
|
|
void stop(std::function<void(FinalState)> completion) {
|
|
FinalState finalState;
|
|
|
|
json11::Json::object statsLog;
|
|
|
|
for (int i = (int)_networkStateLogRecords.size() - 1; i >= 1; i--) {
|
|
// coalesce events within 5ms
|
|
if (_networkStateLogRecords[i].timestamp - _networkStateLogRecords[i - 1].timestamp < 5) {
|
|
_networkStateLogRecords.erase(_networkStateLogRecords.begin() + i - 1);
|
|
}
|
|
}
|
|
|
|
json11::Json::array jsonNetworkStateLogRecords;
|
|
int64_t baseTimestamp = 0;
|
|
for (const auto &record : _networkStateLogRecords) {
|
|
json11::Json::object jsonRecord;
|
|
|
|
std::ostringstream timestampString;
|
|
|
|
if (baseTimestamp == 0) {
|
|
baseTimestamp = record.timestamp;
|
|
}
|
|
timestampString << (record.timestamp - baseTimestamp);
|
|
|
|
jsonRecord.insert(std::make_pair("t", json11::Json(timestampString.str())));
|
|
jsonRecord.insert(std::make_pair("c", json11::Json(record.record.isConnected ? 1 : 0)));
|
|
if (record.record.route) {
|
|
jsonRecord.insert(std::make_pair("local", json11::Json(record.record.route->localDescription)));
|
|
jsonRecord.insert(std::make_pair("remote", json11::Json(record.record.route->remoteDescription)));
|
|
}
|
|
if (record.record.connection) {
|
|
json11::Json::object jsonConnection;
|
|
|
|
auto serializeCandidate = [](NativeNetworkingImpl::ConnectionDescription::CandidateDescription const &candidate) -> json11::Json::object {
|
|
json11::Json::object jsonCandidate;
|
|
|
|
jsonCandidate.insert(std::make_pair("type", json11::Json(candidate.type)));
|
|
jsonCandidate.insert(std::make_pair("protocol", json11::Json(candidate.protocol)));
|
|
jsonCandidate.insert(std::make_pair("address", json11::Json(candidate.address)));
|
|
|
|
return jsonCandidate;
|
|
};
|
|
|
|
jsonConnection.insert(std::make_pair("local", serializeCandidate(record.record.connection->local)));
|
|
jsonConnection.insert(std::make_pair("remote", serializeCandidate(record.record.connection->remote)));
|
|
|
|
jsonRecord.insert(std::make_pair("network", std::move(jsonConnection)));
|
|
}
|
|
if (record.record.isFailed) {
|
|
jsonRecord.insert(std::make_pair("failed", json11::Json(1)));
|
|
}
|
|
|
|
jsonNetworkStateLogRecords.push_back(std::move(jsonRecord));
|
|
}
|
|
statsLog.insert(std::make_pair("network", std::move(jsonNetworkStateLogRecords)));
|
|
|
|
json11::Json::array jsonNetworkBitrateLogRecords;
|
|
for (const auto &record : _networkBitrateLogRecords) {
|
|
json11::Json::object jsonRecord;
|
|
|
|
jsonRecord.insert(std::make_pair("b", json11::Json(record.record.bitrate)));
|
|
|
|
jsonNetworkBitrateLogRecords.push_back(std::move(jsonRecord));
|
|
}
|
|
statsLog.insert(std::make_pair("bitrate", std::move(jsonNetworkBitrateLogRecords)));
|
|
|
|
auto jsonStatsLog = json11::Json(std::move(statsLog));
|
|
|
|
if (!_statsLogPath.data.empty()) {
|
|
std::ofstream file;
|
|
file.open(_statsLogPath.data);
|
|
|
|
file << jsonStatsLog.dump();
|
|
|
|
file.close();
|
|
}
|
|
|
|
completion(finalState);
|
|
}
|
|
|
|
void adjustBitratePreferences(bool resetStartBitrate) {
|
|
if (_outgoingAudioChannel) {
|
|
_outgoingAudioChannel->setMaxBitrate(32 * 1024);
|
|
}
|
|
if (_outgoingVideoChannel) {
|
|
_outgoingVideoChannel->setMaxBitrate(1000 * 1024);
|
|
}
|
|
}
|
|
|
|
private:
|
|
rtc::scoped_refptr<webrtc::AudioDeviceModule> createAudioDeviceModule() {
|
|
const auto create = [&](webrtc::AudioDeviceModule::AudioLayer layer) {
|
|
#ifdef WEBRTC_IOS
|
|
return rtc::make_ref_counted<webrtc::tgcalls_ios_adm::AudioDeviceModuleIOS>(false, false, 1);
|
|
#else
|
|
return webrtc::AudioDeviceModule::Create(
|
|
layer,
|
|
_taskQueueFactory.get());
|
|
#endif
|
|
};
|
|
const auto check = [&](const rtc::scoped_refptr<webrtc::AudioDeviceModule> &result) {
|
|
return (result && result->Init() == 0) ? result : nullptr;
|
|
};
|
|
if (_createAudioDeviceModule) {
|
|
if (const auto result = check(_createAudioDeviceModule(_taskQueueFactory.get()))) {
|
|
return result;
|
|
}
|
|
}
|
|
return check(create(webrtc::AudioDeviceModule::kPlatformDefaultAudio));
|
|
}
|
|
|
|
private:
|
|
SignalingProtocolVersion _signalingProtocolVersion;
|
|
std::shared_ptr<Threads> _threads;
|
|
std::vector<RtcServer> _rtcServers;
|
|
std::unique_ptr<Proxy> _proxy;
|
|
bool _enableP2P = false;
|
|
EncryptionKey _encryptionKey;
|
|
std::function<void(State)> _stateUpdated;
|
|
std::function<void(int)> _signalBarsUpdated;
|
|
std::function<void(float, float)> _audioLevelsUpdated;
|
|
std::function<void(bool)> _remoteBatteryLevelIsLowUpdated;
|
|
std::function<void(AudioState, VideoState)> _remoteMediaStateUpdated;
|
|
std::function<void(float)> _remotePrefferedAspectRatioUpdated;
|
|
std::function<void(const std::vector<uint8_t> &)> _signalingDataEmitted;
|
|
std::function<rtc::scoped_refptr<webrtc::AudioDeviceModule>(webrtc::TaskQueueFactory*)> _createAudioDeviceModule;
|
|
FilePath _statsLogPath;
|
|
|
|
std::unique_ptr<SignalingConnection> _signalingConnection;
|
|
std::unique_ptr<EncryptedConnection> _signalingEncryptedConnection;
|
|
|
|
int64_t _startTimestamp = 0;
|
|
|
|
absl::optional<NetworkStateLogRecord> _currentNetworkStateLogRecord;
|
|
std::vector<StateLogRecord<NetworkStateLogRecord>> _networkStateLogRecords;
|
|
std::vector<StateLogRecord<NetworkBitrateLogRecord>> _networkBitrateLogRecords;
|
|
|
|
absl::optional<NativeNetworkingImpl::State> _networkState;
|
|
|
|
bool _handshakeCompleted = false;
|
|
std::vector<cricket::Candidate> _pendingIceCandidates;
|
|
bool _isDataChannelOpen = false;
|
|
|
|
std::unique_ptr<webrtc::RtcEventLogNull> _eventLog;
|
|
std::unique_ptr<webrtc::TaskQueueFactory> _taskQueueFactory;
|
|
std::unique_ptr<webrtc::Call> _call;
|
|
webrtc::LocalAudioSinkAdapter _audioSource;
|
|
rtc::scoped_refptr<webrtc::AudioDeviceModule> _audioDeviceModule;
|
|
|
|
std::unique_ptr<rtc::UniqueRandomIdGenerator> _uniqueRandomIdGenerator;
|
|
webrtc::RtpTransport *_rtpTransport = nullptr;
|
|
std::unique_ptr<ChannelManager> _channelManager;
|
|
std::unique_ptr<webrtc::VideoBitrateAllocatorFactory> _videoBitrateAllocatorFactory;
|
|
std::string _initialInputDeviceId;
|
|
std::string _initialOutputDeviceId;
|
|
|
|
std::unique_ptr<ContentNegotiationContext> _contentNegotiationContext;
|
|
|
|
std::shared_ptr<ThreadLocalObject<NativeNetworkingImpl>> _networking;
|
|
|
|
absl::optional<std::string> _outgoingAudioChannelId;
|
|
std::unique_ptr<OutgoingAudioChannel> _outgoingAudioChannel;
|
|
bool _isMicrophoneMuted = false;
|
|
|
|
std::vector<webrtc::SdpVideoFormat> _availableVideoFormats;
|
|
|
|
absl::optional<std::string> _outgoingVideoChannelId;
|
|
std::shared_ptr<OutgoingVideoChannel> _outgoingVideoChannel;
|
|
absl::optional<std::string> _outgoingScreencastChannelId;
|
|
std::shared_ptr<OutgoingVideoChannel> _outgoingScreencastChannel;
|
|
|
|
bool _isBatteryLow = false;
|
|
|
|
std::unique_ptr<IncomingV2AudioChannel> _incomingAudioChannel;
|
|
std::unique_ptr<IncomingV2VideoChannel> _incomingVideoChannel;
|
|
std::unique_ptr<IncomingV2VideoChannel> _incomingScreencastChannel;
|
|
|
|
std::weak_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> _currentSink;
|
|
|
|
std::shared_ptr<VideoCaptureInterface> _videoCapture;
|
|
std::shared_ptr<VideoCaptureInterface> _screencastCapture;
|
|
std::shared_ptr<PlatformContext> _platformContext;
|
|
};
|
|
|
|
InstanceV2Impl::InstanceV2Impl(Descriptor &&descriptor) {
|
|
if (descriptor.config.logPath.data.size() != 0) {
|
|
_logSink = std::make_unique<LogSinkImpl>(descriptor.config.logPath);
|
|
}
|
|
rtc::LogMessage::LogToDebug(rtc::LS_INFO);
|
|
rtc::LogMessage::SetLogToStderr(false);
|
|
if (_logSink) {
|
|
rtc::LogMessage::AddLogToStream(_logSink.get(), rtc::LS_INFO);
|
|
}
|
|
|
|
_threads = StaticThreads::getThreads();
|
|
_internal.reset(new ThreadLocalObject<InstanceV2ImplInternal>(_threads->getMediaThread(), [descriptor = std::move(descriptor), threads = _threads]() mutable {
|
|
return new InstanceV2ImplInternal(std::move(descriptor), threads);
|
|
}));
|
|
_internal->perform([](InstanceV2ImplInternal *internal) {
|
|
internal->start();
|
|
});
|
|
}
|
|
|
|
InstanceV2Impl::~InstanceV2Impl() {
|
|
rtc::LogMessage::RemoveLogToStream(_logSink.get());
|
|
}
|
|
|
|
void InstanceV2Impl::receiveSignalingData(const std::vector<uint8_t> &data) {
|
|
_internal->perform([data](InstanceV2ImplInternal *internal) {
|
|
internal->receiveSignalingData(data);
|
|
});
|
|
}
|
|
|
|
void InstanceV2Impl::setVideoCapture(std::shared_ptr<VideoCaptureInterface> videoCapture) {
|
|
_internal->perform([videoCapture](InstanceV2ImplInternal *internal) {
|
|
internal->setVideoCapture(videoCapture);
|
|
});
|
|
}
|
|
|
|
void InstanceV2Impl::setRequestedVideoAspect(float aspect) {
|
|
_internal->perform([aspect](InstanceV2ImplInternal *internal) {
|
|
internal->setRequestedVideoAspect(aspect);
|
|
});
|
|
}
|
|
|
|
void InstanceV2Impl::setNetworkType(NetworkType networkType) {
|
|
_internal->perform([networkType](InstanceV2ImplInternal *internal) {
|
|
internal->setNetworkType(networkType);
|
|
});
|
|
}
|
|
|
|
void InstanceV2Impl::setMuteMicrophone(bool muteMicrophone) {
|
|
_internal->perform([muteMicrophone](InstanceV2ImplInternal *internal) {
|
|
internal->setMuteMicrophone(muteMicrophone);
|
|
});
|
|
}
|
|
|
|
void InstanceV2Impl::setIncomingVideoOutput(std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) {
|
|
_internal->perform([sink](InstanceV2ImplInternal *internal) {
|
|
internal->setIncomingVideoOutput(sink);
|
|
});
|
|
}
|
|
|
|
void InstanceV2Impl::setAudioInputDevice(std::string id) {
|
|
_internal->perform([id](InstanceV2ImplInternal *internal) {
|
|
internal->setAudioInputDevice(id);
|
|
});
|
|
}
|
|
|
|
void InstanceV2Impl::setAudioOutputDevice(std::string id) {
|
|
_internal->perform([id](InstanceV2ImplInternal *internal) {
|
|
internal->setAudioOutputDevice(id);
|
|
});
|
|
}
|
|
|
|
void InstanceV2Impl::setIsLowBatteryLevel(bool isLowBatteryLevel) {
|
|
_internal->perform([isLowBatteryLevel](InstanceV2ImplInternal *internal) {
|
|
internal->setIsLowBatteryLevel(isLowBatteryLevel);
|
|
});
|
|
}
|
|
|
|
void InstanceV2Impl::setInputVolume(float level) {
|
|
}
|
|
|
|
void InstanceV2Impl::setOutputVolume(float level) {
|
|
}
|
|
|
|
void InstanceV2Impl::setAudioOutputDuckingEnabled(bool enabled) {
|
|
}
|
|
|
|
void InstanceV2Impl::setAudioOutputGainControlEnabled(bool enabled) {
|
|
}
|
|
|
|
void InstanceV2Impl::setEchoCancellationStrength(int strength) {
|
|
}
|
|
|
|
std::vector<std::string> InstanceV2Impl::GetVersions() {
|
|
std::vector<std::string> result;
|
|
result.push_back("7.0.0");
|
|
result.push_back("8.0.0");
|
|
result.push_back("9.0.0");
|
|
return result;
|
|
}
|
|
|
|
int InstanceV2Impl::GetConnectionMaxLayer() {
|
|
return 92;
|
|
}
|
|
|
|
std::string InstanceV2Impl::getLastError() {
|
|
return "";
|
|
}
|
|
|
|
std::string InstanceV2Impl::getDebugInfo() {
|
|
return "";
|
|
}
|
|
|
|
int64_t InstanceV2Impl::getPreferredRelayId() {
|
|
return 0;
|
|
}
|
|
|
|
TrafficStats InstanceV2Impl::getTrafficStats() {
|
|
return {};
|
|
}
|
|
|
|
PersistentState InstanceV2Impl::getPersistentState() {
|
|
return {};
|
|
}
|
|
|
|
void InstanceV2Impl::stop(std::function<void(FinalState)> completion) {
|
|
std::string debugLog;
|
|
if (_logSink) {
|
|
debugLog = _logSink->result();
|
|
}
|
|
_internal->perform([completion, debugLog = std::move(debugLog)](InstanceV2ImplInternal *internal) mutable {
|
|
internal->stop([completion, debugLog = std::move(debugLog)](FinalState finalState) mutable {
|
|
finalState.debugLog = debugLog;
|
|
completion(finalState);
|
|
});
|
|
});
|
|
}
|
|
|
|
template <>
|
|
bool Register<InstanceV2Impl>() {
|
|
return Meta::RegisterOne<InstanceV2Impl>();
|
|
}
|
|
|
|
} // namespace tgcalls
|