diff options
Diffstat (limited to 'chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine.cc')
| -rw-r--r-- | chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine.cc | 707 |
1 files changed, 485 insertions, 222 deletions
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine.cc b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine.cc index 51da9ac6b14..f1460a6cb8a 100644 --- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine.cc +++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine.cc @@ -107,12 +107,6 @@ static const int kDefaultSoundclipDeviceId = -2; static const int kDefaultAudioDeviceId = 0; #endif -// extension header for audio levels, as defined in -// http://tools.ietf.org/html/draft-ietf-avtext-client-to-mixer-audio-level-03 -static const char kRtpAudioLevelHeaderExtension[] = - "urn:ietf:params:rtp-hdrext:ssrc-audio-level"; -static const int kRtpAudioLevelHeaderExtensionId = 1; - static const char kIsacCodecName[] = "ISAC"; static const char kL16CodecName[] = "L16"; // Codec parameters for Opus. @@ -194,6 +188,18 @@ static bool IsCodecMultiRate(const webrtc::CodecInst& codec) { return false; } +static bool IsTelephoneEventCodec(const std::string& name) { + return _stricmp(name.c_str(), "telephone-event") == 0; +} + +static bool IsCNCodec(const std::string& name) { + return _stricmp(name.c_str(), "CN") == 0; +} + +static bool IsRedCodec(const std::string& name) { + return _stricmp(name.c_str(), "red") == 0; +} + static bool FindCodec(const std::vector<AudioCodec>& codecs, const AudioCodec& codec, AudioCodec* found_codec) { @@ -229,8 +235,9 @@ static AudioOptions GetDefaultEngineOptions() { options.adjust_agc_delta.Set(0); options.experimental_agc.Set(false); options.experimental_aec.Set(false); + options.experimental_ns.Set(false); options.aec_dump.Set(false); - options.experimental_acm.Set(false); + options.opus_fec.Set(false); return options; } @@ -334,7 +341,6 @@ WebRtcVoiceEngine::WebRtcVoiceEngine() log_filter_(SeverityToFilter(kDefaultLogSeverity)), is_dumping_aec_(false), desired_local_monitor_enable_(false), - use_experimental_acm_(false), tx_processor_ssrc_(0), rx_processor_ssrc_(0) { Construct(); @@ -352,7 +358,6 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(VoEWrapper* voe_wrapper, log_filter_(SeverityToFilter(kDefaultLogSeverity)), is_dumping_aec_(false), desired_local_monitor_enable_(false), - use_experimental_acm_(false), tx_processor_ssrc_(0), rx_processor_ssrc_(0) { Construct(); @@ -378,12 +383,11 @@ void WebRtcVoiceEngine::Construct() { // Load our RTP Header extensions. rtp_header_extensions_.push_back( RtpHeaderExtension(kRtpAudioLevelHeaderExtension, - kRtpAudioLevelHeaderExtensionId)); + kRtpAudioLevelHeaderExtensionDefaultId)); + rtp_header_extensions_.push_back( + RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension, + kRtpAbsoluteSenderTimeHeaderExtensionDefaultId)); options_ = GetDefaultEngineOptions(); - - // Initialize the VoE Configuration to the default ACM. - voe_config_.Set<webrtc::AudioCodingModuleFactory>( - new webrtc::AudioCodingModuleFactory); } static bool IsOpus(const AudioCodec& codec) { @@ -396,12 +400,8 @@ static bool IsIsac(const AudioCodec& codec) { // True if params["stereo"] == "1" static bool IsOpusStereoEnabled(const AudioCodec& codec) { - CodecParameterMap::const_iterator param = - codec.params.find(kCodecParamStereo); - if (param == codec.params.end()) { - return false; - } - return param->second == kParamValueTrue; + int value; + return codec.GetParam(kCodecParamStereo, &value) && value == 1; } static bool IsValidOpusBitrate(int bitrate) { @@ -423,6 +423,22 @@ static int GetOpusBitrateFromParams(const AudioCodec& codec) { return bitrate; } +// Return true params[kCodecParamUseInbandFec] == kParamValueTrue, false +// otherwise. +static bool IsOpusFecEnabled(const AudioCodec& codec) { + int value; + return codec.GetParam(kCodecParamUseInbandFec, &value) && value == 1; +} + +// Set params[kCodecParamUseInbandFec]. Caller should make sure codec is Opus. +static void SetOpusFec(AudioCodec *codec, bool opus_fec) { + if (opus_fec) { + codec->params[kCodecParamUseInbandFec] = kParamValueTrue; + } else { + codec->params.erase(kCodecParamUseInbandFec); + } +} + void WebRtcVoiceEngine::ConstructCodecs() { LOG(LS_INFO) << "WebRtc VoiceEngine codecs:"; int ncodecs = voe_wrapper_->codec()->NumOfCodecs(); @@ -467,6 +483,7 @@ void WebRtcVoiceEngine::ConstructCodecs() { } // TODO(hellner): Add ptime, sprop-stereo, stereo and useinbandfec // when they can be set to values other than the default. + SetOpusFec(&codec, false); } codecs_.push_back(codec); } else { @@ -717,16 +734,11 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { options.typing_detection.Set(false); options.experimental_agc.Set(false); options.experimental_aec.Set(false); + options.experimental_ns.Set(false); #endif LOG(LS_INFO) << "Applying audio options: " << options.ToString(); - // Configure whether ACM1 or ACM2 is used. - bool enable_acm2 = false; - if (options.experimental_acm.Get(&enable_acm2)) { - EnableExperimentalAcm(enable_acm2); - } - webrtc::VoEAudioProcessing* voep = voe_wrapper_->processing(); bool echo_cancellation; @@ -804,8 +816,26 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { } } + bool experimental_ns; + if (options.experimental_ns.Get(&experimental_ns)) { + webrtc::AudioProcessing* audioproc = + voe_wrapper_->base()->audio_processing(); + // We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine + // returns NULL on audio_processing(). + if (audioproc) { + if (audioproc->EnableExperimentalNs(experimental_ns) == -1) { + LOG_RTCERR1(EnableExperimentalNs, experimental_ns); + return false; + } + } else { + LOG(LS_VERBOSE) << "Experimental noise suppression set to " + << experimental_ns; + } + } + bool highpass_filter; if (options.highpass_filter.Get(&highpass_filter)) { + LOG(LS_INFO) << "High pass filter enabled? " << highpass_filter; if (voep->EnableHighPassFilter(highpass_filter) == -1) { LOG_RTCERR1(SetHighpassFilterStatus, highpass_filter); return false; @@ -814,6 +844,7 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { bool stereo_swapping; if (options.stereo_swapping.Get(&stereo_swapping)) { + LOG(LS_INFO) << "Stereo swapping enabled? " << stereo_swapping; voep->EnableStereoChannelSwapping(stereo_swapping); if (voep->IsStereoChannelSwappingEnabled() != stereo_swapping) { LOG_RTCERR1(EnableStereoChannelSwapping, stereo_swapping); @@ -823,6 +854,7 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { bool typing_detection; if (options.typing_detection.Get(&typing_detection)) { + LOG(LS_INFO) << "Typing detection is enabled? " << typing_detection; if (voep->SetTypingDetectionStatus(typing_detection) == -1) { // In case of error, log the info and continue LOG_RTCERR1(SetTypingDetectionStatus, typing_detection); @@ -831,6 +863,7 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { int adjust_agc_delta; if (options.adjust_agc_delta.Get(&adjust_agc_delta)) { + LOG(LS_INFO) << "Adjust agc delta is " << adjust_agc_delta; if (!AdjustAgcLevel(adjust_agc_delta)) { return false; } @@ -838,6 +871,7 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { bool aec_dump; if (options.aec_dump.Get(&aec_dump)) { + LOG(LS_INFO) << "Aec dump is enabled? " << aec_dump; if (aec_dump) StartAecDump(kAecDumpByAudioOptionFilename); else @@ -846,6 +880,7 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { bool experimental_aec; if (options.experimental_aec.Get(&experimental_aec)) { + LOG(LS_INFO) << "Experimental aec is " << experimental_aec; webrtc::AudioProcessing* audioproc = voe_wrapper_->base()->audio_processing(); // We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine @@ -860,6 +895,7 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { uint32 recording_sample_rate; if (options.recording_sample_rate.Get(&recording_sample_rate)) { + LOG(LS_INFO) << "Recording sample rate is " << recording_sample_rate; if (voe_wrapper_->hw()->SetRecordingSampleRate(recording_sample_rate)) { LOG_RTCERR1(SetRecordingSampleRate, recording_sample_rate); } @@ -867,11 +903,21 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { uint32 playout_sample_rate; if (options.playout_sample_rate.Get(&playout_sample_rate)) { + LOG(LS_INFO) << "Playout sample rate is " << playout_sample_rate; if (voe_wrapper_->hw()->SetPlayoutSampleRate(playout_sample_rate)) { LOG_RTCERR1(SetPlayoutSampleRate, playout_sample_rate); } } + bool opus_fec = false; + if (options.opus_fec.Get(&opus_fec)) { + LOG(LS_INFO) << "Opus FEC is enabled? " << opus_fec; + for (std::vector<AudioCodec>::iterator it = codecs_.begin(); + it != codecs_.end(); ++it) { + if (IsOpus(*it)) + SetOpusFec(&(*it), opus_fec); + } + } return true; } @@ -1261,21 +1307,6 @@ bool WebRtcVoiceEngine::ShouldIgnoreTrace(const std::string& trace) { return false; } -void WebRtcVoiceEngine::EnableExperimentalAcm(bool enable) { - if (enable == use_experimental_acm_) - return; - if (enable) { - LOG(LS_INFO) << "VoiceEngine is set to use new ACM (ACM2 + NetEq4)."; - voe_config_.Set<webrtc::AudioCodingModuleFactory>( - new webrtc::NewAudioCodingModuleFactory()); - } else { - LOG(LS_INFO) << "VoiceEngine is set to use legacy ACM (ACM1 + Neteq3)."; - voe_config_.Set<webrtc::AudioCodingModuleFactory>( - new webrtc::AudioCodingModuleFactory()); - } - use_experimental_acm_ = enable; -} - void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace, int length) { talk_base::LoggingSeverity sev = talk_base::LS_VERBOSE; @@ -1433,6 +1464,25 @@ bool WebRtcVoiceEngine::SetAudioDeviceModule(webrtc::AudioDeviceModule* adm, return true; } +bool WebRtcVoiceEngine::StartAecDump(talk_base::PlatformFile file) { + FILE* aec_dump_file_stream = talk_base::FdopenPlatformFileForWriting(file); + if (!aec_dump_file_stream) { + LOG(LS_ERROR) << "Could not open AEC dump file stream."; + if (!talk_base::ClosePlatformFile(file)) + LOG(LS_WARNING) << "Could not close file."; + return false; + } + StopAecDump(); + if (voe_wrapper_->processing()->StartDebugRecording(aec_dump_file_stream) != + webrtc::AudioProcessing::kNoError) { + LOG_RTCERR0(StartDebugRecording); + fclose(aec_dump_file_stream); + return false; + } + is_dumping_aec_ = true; + return true; +} + bool WebRtcVoiceEngine::RegisterProcessor( uint32 ssrc, VoiceProcessor* voice_processor, @@ -1590,7 +1640,7 @@ void WebRtcVoiceEngine::StartAecDump(const std::string& filename) { // Start dumping AEC when we are not dumping. if (voe_wrapper_->processing()->StartDebugRecording( filename.c_str()) != webrtc::AudioProcessing::kNoError) { - LOG_RTCERR0(StartDebugRecording); + LOG_RTCERR1(StartDebugRecording, filename.c_str()); } else { is_dumping_aec_ = true; } @@ -1620,17 +1670,89 @@ int WebRtcVoiceEngine::CreateSoundclipVoiceChannel() { return CreateVoiceChannel(voe_wrapper_sc_.get()); } -// This struct relies on the generated copy constructor and assignment operator -// since it is used in an stl::map. -struct WebRtcVoiceMediaChannel::WebRtcVoiceChannelInfo { - WebRtcVoiceChannelInfo() : channel(-1), renderer(NULL) {} - WebRtcVoiceChannelInfo(int ch, AudioRenderer* r) - : channel(ch), - renderer(r) {} - ~WebRtcVoiceChannelInfo() {} +class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer + : public AudioRenderer::Sink { + public: + WebRtcVoiceChannelRenderer(int ch, + webrtc::AudioTransport* voe_audio_transport) + : channel_(ch), + voe_audio_transport_(voe_audio_transport), + renderer_(NULL) { + } + virtual ~WebRtcVoiceChannelRenderer() { + Stop(); + } + + // Starts the rendering by setting a sink to the renderer to get data + // callback. + // This method is called on the libjingle worker thread. + // TODO(xians): Make sure Start() is called only once. + void Start(AudioRenderer* renderer) { + talk_base::CritScope lock(&lock_); + ASSERT(renderer != NULL); + if (renderer_ != NULL) { + ASSERT(renderer_ == renderer); + return; + } + + // TODO(xians): Remove AddChannel() call after Chrome turns on APM + // in getUserMedia by default. + renderer->AddChannel(channel_); + renderer->SetSink(this); + renderer_ = renderer; + } + + // Stops rendering by setting the sink of the renderer to NULL. No data + // callback will be received after this method. + // This method is called on the libjingle worker thread. + void Stop() { + talk_base::CritScope lock(&lock_); + if (renderer_ == NULL) + return; + + renderer_->RemoveChannel(channel_); + renderer_->SetSink(NULL); + renderer_ = NULL; + } + + // AudioRenderer::Sink implementation. + // This method is called on the audio thread. + virtual void OnData(const void* audio_data, + int bits_per_sample, + int sample_rate, + int number_of_channels, + int number_of_frames) OVERRIDE { + voe_audio_transport_->OnData(channel_, + audio_data, + bits_per_sample, + sample_rate, + number_of_channels, + number_of_frames); + } + + // Callback from the |renderer_| when it is going away. In case Start() has + // never been called, this callback won't be triggered. + virtual void OnClose() OVERRIDE { + talk_base::CritScope lock(&lock_); + // Set |renderer_| to NULL to make sure no more callback will get into + // the renderer. + renderer_ = NULL; + } + + // Accessor to the VoE channel ID. + int channel() const { return channel_; } + + private: + const int channel_; + webrtc::AudioTransport* const voe_audio_transport_; + + // Raw pointer to AudioRenderer owned by LocalAudioTrackHandler. + // PeerConnection will make sure invalidating the pointer before the object + // goes away. + AudioRenderer* renderer_; - int channel; - AudioRenderer* renderer; + // Protects |renderer_| in Start(), Stop() and OnClose(). + talk_base::CriticalSection lock_; }; // WebRtcVoiceMediaChannel @@ -1639,7 +1761,6 @@ WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine) engine, engine->CreateMediaVoiceChannel()), send_bw_setting_(false), - send_autobw_(false), send_bw_bps_(0), options_(), dtmf_allowed_(false), @@ -1753,7 +1874,7 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) { } if (dscp_option_changed) { talk_base::DiffServCodePoint dscp = talk_base::DSCP_DEFAULT; - if (options.dscp.GetWithDefaultIfUnset(false)) + if (options_.dscp.GetWithDefaultIfUnset(false)) dscp = kAudioDscpValue; if (MediaChannel::SetDscp(dscp) != 0) { LOG(LS_WARNING) << "Failed to set DSCP settings for audio channel"; @@ -1823,8 +1944,8 @@ bool WebRtcVoiceMediaChannel::SetRecvCodecs( for (ChannelMap::iterator it = receive_channels_.begin(); it != receive_channels_.end() && ret; ++it) { if (engine()->voe()->codec()->SetRecPayloadType( - it->second.channel, voe_codec) == -1) { - LOG_RTCERR2(SetRecPayloadType, it->second.channel, + it->second->channel(), voe_codec) == -1) { + LOG_RTCERR2(SetRecPayloadType, it->second->channel(), ToString(voe_codec)); ret = false; } @@ -1846,24 +1967,38 @@ bool WebRtcVoiceMediaChannel::SetRecvCodecs( bool WebRtcVoiceMediaChannel::SetSendCodecs( int channel, const std::vector<AudioCodec>& codecs) { - // Disable VAD, and FEC unless we know the other side wants them. + // Disable VAD, FEC, and RED unless we know the other side wants them. engine()->voe()->codec()->SetVADStatus(channel, false); engine()->voe()->rtp()->SetNACKStatus(channel, false, 0); +#ifdef USE_WEBRTC_DEV_BRANCH + engine()->voe()->rtp()->SetREDStatus(channel, false); + engine()->voe()->codec()->SetFECStatus(channel, false); +#else + // TODO(minyue): Remove code under #else case after new WebRTC roll. engine()->voe()->rtp()->SetFECStatus(channel, false); +#endif // USE_WEBRTC_DEV_BRANCH // Scan through the list to figure out the codec to use for sending, along // with the proper configuration for VAD and DTMF. - bool first = true; + bool found_send_codec = false; webrtc::CodecInst send_codec; memset(&send_codec, 0, sizeof(send_codec)); + bool nack_enabled = nack_enabled_; + + // Set send codec (the first non-telephone-event/CN codec) for (std::vector<AudioCodec>::const_iterator it = codecs.begin(); it != codecs.end(); ++it) { // Ignore codecs we don't know about. The negotiation step should prevent // this, but double-check to be sure. webrtc::CodecInst voe_codec; if (!engine()->FindWebRtcCodec(*it, &voe_codec)) { - LOG(LS_WARNING) << "Unknown codec " << ToString(voe_codec); + LOG(LS_WARNING) << "Unknown codec " << ToString(*it); + continue; + } + + if (IsTelephoneEventCodec(it->name) || IsCNCodec(it->name)) { + // Skip telephone-event/CN codec, which will be handled later. continue; } @@ -1900,23 +2035,99 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs( if (bitrate_from_params != 0) { voe_codec.rate = bitrate_from_params; } + + // For Opus, we also enable inband FEC if it is requested. + if (IsOpusFecEnabled(*it)) { + LOG(LS_INFO) << "Enabling Opus FEC on channel " << channel; +#ifdef USE_WEBRTC_DEV_BRANCH + if (engine()->voe()->codec()->SetFECStatus(channel, true) == -1) { + // Enable in-band FEC of the Opus codec. Treat any failure as a fatal + // internal error. + LOG_RTCERR2(SetFECStatus, channel, true); + return false; + } +#endif // USE_WEBRTC_DEV_BRANCH + } + } + + // We'll use the first codec in the list to actually send audio data. + // Be sure to use the payload type requested by the remote side. + // "red", for RED audio, is a special case where the actual codec to be + // used is specified in params. + if (IsRedCodec(it->name)) { + // Parse out the RED parameters. If we fail, just ignore RED; + // we don't support all possible params/usage scenarios. + if (!GetRedSendCodec(*it, codecs, &send_codec)) { + continue; + } + + // Enable redundant encoding of the specified codec. Treat any + // failure as a fatal internal error. +#ifdef USE_WEBRTC_DEV_BRANCH + LOG(LS_INFO) << "Enabling RED on channel " << channel; + if (engine()->voe()->rtp()->SetREDStatus(channel, true, it->id) == -1) { + LOG_RTCERR3(SetREDStatus, channel, true, it->id); +#else + // TODO(minyue): Remove code under #else case after new WebRTC roll. + LOG(LS_INFO) << "Enabling FEC"; + if (engine()->voe()->rtp()->SetFECStatus(channel, true, it->id) == -1) { + LOG_RTCERR3(SetFECStatus, channel, true, it->id); +#endif // USE_WEBRTC_DEV_BRANCH + return false; + } + } else { + send_codec = voe_codec; + nack_enabled = IsNackEnabled(*it); + } + found_send_codec = true; + break; + } + + if (nack_enabled_ != nack_enabled) { + SetNack(channel, nack_enabled); + nack_enabled_ = nack_enabled; + } + + if (!found_send_codec) { + LOG(LS_WARNING) << "Received empty list of codecs."; + return false; + } + + // Set the codec immediately, since SetVADStatus() depends on whether + // the current codec is mono or stereo. + if (!SetSendCodec(channel, send_codec)) + return false; + + // Always update the |send_codec_| to the currently set send codec. + send_codec_.reset(new webrtc::CodecInst(send_codec)); + + if (send_bw_setting_) { + SetSendBandwidthInternal(send_bw_bps_); + } + + // Loop through the codecs list again to config the telephone-event/CN codec. + for (std::vector<AudioCodec>::const_iterator it = codecs.begin(); + it != codecs.end(); ++it) { + // Ignore codecs we don't know about. The negotiation step should prevent + // this, but double-check to be sure. + webrtc::CodecInst voe_codec; + if (!engine()->FindWebRtcCodec(*it, &voe_codec)) { + LOG(LS_WARNING) << "Unknown codec " << ToString(*it); + continue; } // Find the DTMF telephone event "codec" and tell VoiceEngine channels // about it. - if (_stricmp(it->name.c_str(), "telephone-event") == 0 || - _stricmp(it->name.c_str(), "audio/telephone-event") == 0) { + if (IsTelephoneEventCodec(it->name)) { if (engine()->voe()->dtmf()->SetSendTelephoneEventPayloadType( channel, it->id) == -1) { LOG_RTCERR2(SetSendTelephoneEventPayloadType, channel, it->id); return false; } - } - - // Turn voice activity detection/comfort noise on if supported. - // Set the wideband CN payload type appropriately. - // (narrowband always uses the static payload type 13). - if (_stricmp(it->name.c_str(), "CN") == 0) { + } else if (IsCNCodec(it->name)) { + // Turn voice activity detection/comfort noise on if supported. + // Set the wideband CN payload type appropriately. + // (narrowband always uses the static payload type 13). webrtc::PayloadFrequencies cn_freq; switch (it->clockrate) { case 8000: @@ -1949,7 +2160,6 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs( // send the offer. } } - // Only turn on VAD if we have a CN payload type that matches the // clockrate for the codec we are going to use. if (it->clockrate == send_codec.plfreq) { @@ -1960,56 +2170,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs( } } } - - // We'll use the first codec in the list to actually send audio data. - // Be sure to use the payload type requested by the remote side. - // "red", for FEC audio, is a special case where the actual codec to be - // used is specified in params. - if (first) { - if (_stricmp(it->name.c_str(), "red") == 0) { - // Parse out the RED parameters. If we fail, just ignore RED; - // we don't support all possible params/usage scenarios. - if (!GetRedSendCodec(*it, codecs, &send_codec)) { - continue; - } - - // Enable redundant encoding of the specified codec. Treat any - // failure as a fatal internal error. - LOG(LS_INFO) << "Enabling FEC"; - if (engine()->voe()->rtp()->SetFECStatus(channel, true, it->id) == -1) { - LOG_RTCERR3(SetFECStatus, channel, true, it->id); - return false; - } - } else { - send_codec = voe_codec; - nack_enabled_ = IsNackEnabled(*it); - SetNack(channel, nack_enabled_); - } - first = false; - // Set the codec immediately, since SetVADStatus() depends on whether - // the current codec is mono or stereo. - if (!SetSendCodec(channel, send_codec)) - return false; - } } - - // If we're being asked to set an empty list of codecs, due to a buggy client, - // choose the most common format: PCMU - if (first) { - LOG(LS_WARNING) << "Received empty list of codecs; using PCMU/8000"; - AudioCodec codec(0, "PCMU", 8000, 0, 1, 0); - engine()->FindWebRtcCodec(codec, &send_codec); - if (!SetSendCodec(channel, send_codec)) - return false; - } - - // Always update the |send_codec_| to the currently set send codec. - send_codec_.reset(new webrtc::CodecInst(send_codec)); - - if (send_bw_setting_) { - SetSendBandwidthInternal(send_autobw_, send_bw_bps_); - } - return true; } @@ -2029,13 +2190,13 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs( send_codecs_ = codecs; for (ChannelMap::iterator iter = send_channels_.begin(); iter != send_channels_.end(); ++iter) { - if (!SetSendCodecs(iter->second.channel, codecs)) { + if (!SetSendCodecs(iter->second->channel(), codecs)) { return false; } } + // Set nack status on receive channels and update |nack_enabled_|. SetNack(receive_channels_, nack_enabled_); - return true; } @@ -2043,7 +2204,7 @@ void WebRtcVoiceMediaChannel::SetNack(const ChannelMap& channels, bool nack_enabled) { for (ChannelMap::const_iterator it = channels.begin(); it != channels.end(); ++it) { - SetNack(it->second.channel, nack_enabled); + SetNack(it->second->channel(), nack_enabled); } } @@ -2063,7 +2224,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodec( << ", bitrate=" << send_codec.rate; for (ChannelMap::iterator iter = send_channels_.begin(); iter != send_channels_.end(); ++iter) { - if (!SetSendCodec(iter->second.channel, send_codec)) + if (!SetSendCodec(iter->second->channel(), send_codec)) return false; } @@ -2075,6 +2236,13 @@ bool WebRtcVoiceMediaChannel::SetSendCodec( LOG(LS_INFO) << "Send channel " << channel << " selected voice codec " << ToString(send_codec) << ", bitrate=" << send_codec.rate; + webrtc::CodecInst current_codec; + if (engine()->voe()->codec()->GetSendCodec(channel, current_codec) == 0 && + (send_codec == current_codec)) { + // Codec is already configured, we can return without setting it again. + return true; + } + if (engine()->voe()->codec()->SetSendCodec(channel, send_codec) == -1) { LOG_RTCERR2(SetSendCodec, channel, ToString(send_codec)); return false; @@ -2084,43 +2252,96 @@ bool WebRtcVoiceMediaChannel::SetSendCodec( bool WebRtcVoiceMediaChannel::SetRecvRtpHeaderExtensions( const std::vector<RtpHeaderExtension>& extensions) { - // We don't support any incoming extensions headers right now. + if (receive_extensions_ == extensions) { + return true; + } + + // The default channel may or may not be in |receive_channels_|. Set the rtp + // header extensions for default channel regardless. + if (!SetChannelRecvRtpHeaderExtensions(voe_channel(), extensions)) { + return false; + } + + // Loop through all receive channels and enable/disable the extensions. + for (ChannelMap::const_iterator channel_it = receive_channels_.begin(); + channel_it != receive_channels_.end(); ++channel_it) { + if (!SetChannelRecvRtpHeaderExtensions(channel_it->second->channel(), + extensions)) { + return false; + } + } + + receive_extensions_ = extensions; + return true; +} + +bool WebRtcVoiceMediaChannel::SetChannelRecvRtpHeaderExtensions( + int channel_id, const std::vector<RtpHeaderExtension>& extensions) { +#ifdef USE_WEBRTC_DEV_BRANCH + const RtpHeaderExtension* audio_level_extension = + FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension); + if (!SetHeaderExtension( + &webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus, channel_id, + audio_level_extension)) { + return false; + } +#endif // USE_WEBRTC_DEV_BRANCH + + const RtpHeaderExtension* send_time_extension = + FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension); + if (!SetHeaderExtension( + &webrtc::VoERTP_RTCP::SetReceiveAbsoluteSenderTimeStatus, channel_id, + send_time_extension)) { + return false; + } return true; } bool WebRtcVoiceMediaChannel::SetSendRtpHeaderExtensions( const std::vector<RtpHeaderExtension>& extensions) { - // Enable the audio level extension header if requested. - std::vector<RtpHeaderExtension>::const_iterator it; - for (it = extensions.begin(); it != extensions.end(); ++it) { - if (it->uri == kRtpAudioLevelHeaderExtension) { - break; - } + if (send_extensions_ == extensions) { + return true; } - bool enable = (it != extensions.end()); - int id = 0; + // The default channel may or may not be in |send_channels_|. Set the rtp + // header extensions for default channel regardless. - if (enable) { - id = it->id; - if (id < kMinRtpHeaderExtensionId || - id > kMaxRtpHeaderExtensionId) { - LOG(LS_WARNING) << "Invalid RTP header extension id " << id; - return false; - } + if (!SetChannelSendRtpHeaderExtensions(voe_channel(), extensions)) { + return false; } - LOG(LS_INFO) << "Enabling audio level header extension with ID " << id; - for (ChannelMap::const_iterator iter = send_channels_.begin(); - iter != send_channels_.end(); ++iter) { - if (engine()->voe()->rtp()->SetRTPAudioLevelIndicationStatus( - iter->second.channel, enable, id) == -1) { - LOG_RTCERR3(SetRTPAudioLevelIndicationStatus, - iter->second.channel, enable, id); + // Loop through all send channels and enable/disable the extensions. + for (ChannelMap::const_iterator channel_it = send_channels_.begin(); + channel_it != send_channels_.end(); ++channel_it) { + if (!SetChannelSendRtpHeaderExtensions(channel_it->second->channel(), + extensions)) { return false; } } + send_extensions_ = extensions; + return true; +} + +bool WebRtcVoiceMediaChannel::SetChannelSendRtpHeaderExtensions( + int channel_id, const std::vector<RtpHeaderExtension>& extensions) { + const RtpHeaderExtension* audio_level_extension = + FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension); + + if (!SetHeaderExtension( + &webrtc::VoERTP_RTCP::SetSendAudioLevelIndicationStatus, channel_id, + audio_level_extension)) { + return false; + } + + const RtpHeaderExtension* send_time_extension = + FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension); + if (!SetHeaderExtension( + &webrtc::VoERTP_RTCP::SetSendAbsoluteSenderTimeStatus, channel_id, + send_time_extension)) { + return false; + } + return true; } @@ -2150,9 +2371,9 @@ bool WebRtcVoiceMediaChannel::ChangePlayout(bool playout) { } for (ChannelMap::iterator it = receive_channels_.begin(); it != receive_channels_.end() && result; ++it) { - if (!SetPlayout(it->second.channel, playout)) { + if (!SetPlayout(it->second->channel(), playout)) { LOG(LS_ERROR) << "SetPlayout " << playout << " on channel " - << it->second.channel << " failed"; + << it->second->channel() << " failed"; result = false; } } @@ -2190,7 +2411,7 @@ bool WebRtcVoiceMediaChannel::ChangeSend(SendFlags send) { // Change the settings on each send channel. for (ChannelMap::iterator iter = send_channels_.begin(); iter != send_channels_.end(); ++iter) { - if (!ChangeSend(iter->second.channel, send)) + if (!ChangeSend(iter->second->channel(), send)) return false; } @@ -2224,6 +2445,7 @@ bool WebRtcVoiceMediaChannel::ChangeSend(int channel, SendFlags send) { return true; } +// TODO(ronghuawu): Change this method to return bool. void WebRtcVoiceMediaChannel::ConfigureSendChannel(int channel) { if (engine()->voe()->network()->RegisterExternalTransport( channel, *this) == -1) { @@ -2235,6 +2457,9 @@ void WebRtcVoiceMediaChannel::ConfigureSendChannel(int channel) { // Reset all recv codecs; they will be enabled via SetRecvCodecs. ResetRecvCodecs(channel); + + // Set RTP header extension for the new channel. + SetChannelSendRtpHeaderExtensions(channel, send_extensions_); } bool WebRtcVoiceMediaChannel::DeleteChannel(int channel) { @@ -2262,7 +2487,7 @@ bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) { bool default_channel_is_available = true; for (ChannelMap::const_iterator iter = send_channels_.begin(); iter != send_channels_.end(); ++iter) { - if (IsDefaultChannel(iter->second.channel)) { + if (IsDefaultChannel(iter->second->channel())) { default_channel_is_available = false; break; } @@ -2282,7 +2507,11 @@ bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) { // Save the channel to send_channels_, so that RemoveSendStream() can still // delete the channel in case failure happens below. - send_channels_[sp.first_ssrc()] = WebRtcVoiceChannelInfo(channel, NULL); + webrtc::AudioTransport* audio_transport = + engine()->voe()->base()->audio_transport(); + send_channels_.insert(std::make_pair( + sp.first_ssrc(), + new WebRtcVoiceChannelRenderer(channel, audio_transport))); // Set the send (local) SSRC. // If there are multiple send SSRCs, we can only set the first one here, and @@ -2301,10 +2530,10 @@ bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) { for (ChannelMap::const_iterator it = receive_channels_.begin(); it != receive_channels_.end(); ++it) { // Only update the SSRC for non-default channels. - if (!IsDefaultChannel(it->second.channel)) { - if (engine()->voe()->rtp()->SetLocalSSRC(it->second.channel, + if (!IsDefaultChannel(it->second->channel())) { + if (engine()->voe()->rtp()->SetLocalSSRC(it->second->channel(), sp.first_ssrc()) != 0) { - LOG_RTCERR2(SetLocalSSRC, it->second.channel, sp.first_ssrc()); + LOG_RTCERR2(SetLocalSSRC, it->second->channel(), sp.first_ssrc()); return false; } } @@ -2331,12 +2560,13 @@ bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32 ssrc) { return false; } - int channel = it->second.channel; + int channel = it->second->channel(); ChangeSend(channel, SEND_NOTHING); - // Notify the audio renderer that the send channel is going away. - if (it->second.renderer) - it->second.renderer->RemoveChannel(channel); + // Delete the WebRtcVoiceChannelRenderer object connected to the channel, + // this will disconnect the audio renderer with the send channel. + delete it->second; + send_channels_.erase(it); if (IsDefaultChannel(channel)) { // Do not delete the default channel since the receive channels depend on @@ -2350,7 +2580,6 @@ bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32 ssrc) { return false; } - send_channels_.erase(it); if (send_channels_.empty()) ChangeSend(SEND_NOTHING); @@ -2376,12 +2605,15 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) { // Reuse default channel for recv stream in non-conference mode call // when the default channel is not being used. + webrtc::AudioTransport* audio_transport = + engine()->voe()->base()->audio_transport(); if (!InConferenceMode() && default_receive_ssrc_ == 0) { LOG(LS_INFO) << "Recv stream " << sp.first_ssrc() << " reuse default channel"; default_receive_ssrc_ = sp.first_ssrc(); receive_channels_.insert(std::make_pair( - default_receive_ssrc_, WebRtcVoiceChannelInfo(voe_channel(), NULL))); + default_receive_ssrc_, + new WebRtcVoiceChannelRenderer(voe_channel(), audio_transport))); return SetPlayout(voe_channel(), playout_); } @@ -2398,7 +2630,8 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) { } receive_channels_.insert( - std::make_pair(ssrc, WebRtcVoiceChannelInfo(channel, NULL))); + std::make_pair( + ssrc, new WebRtcVoiceChannelRenderer(channel, audio_transport))); LOG(LS_INFO) << "New audio stream " << ssrc << " registered to VoiceEngine channel #" @@ -2415,14 +2648,14 @@ bool WebRtcVoiceMediaChannel::ConfigureRecvChannel(int channel) { } // Use the same SSRC as our default channel (so the RTCP reports are correct). - unsigned int send_ssrc; + unsigned int send_ssrc = 0; webrtc::VoERTP_RTCP* rtp = engine()->voe()->rtp(); if (rtp->GetLocalSSRC(voe_channel(), send_ssrc) == -1) { - LOG_RTCERR2(GetSendSSRC, channel, send_ssrc); + LOG_RTCERR1(GetSendSSRC, channel); return false; } if (rtp->SetLocalSSRC(channel, send_ssrc) == -1) { - LOG_RTCERR2(SetSendSSRC, channel, send_ssrc); + LOG_RTCERR1(SetSendSSRC, channel); return false; } @@ -2463,6 +2696,11 @@ bool WebRtcVoiceMediaChannel::ConfigureRecvChannel(int channel) { } SetNack(channel, nack_enabled_); + // Set RTP header extension for the new channel. + if (!SetChannelRecvRtpHeaderExtensions(channel, receive_extensions_)) { + return false; + } + return SetPlayout(channel, playout_); } @@ -2475,34 +2713,28 @@ bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32 ssrc) { return false; } + // Delete the WebRtcVoiceChannelRenderer object connected to the channel, this + // will disconnect the audio renderer with the receive channel. + // Cache the channel before the deletion. + const int channel = it->second->channel(); + delete it->second; + receive_channels_.erase(it); + if (ssrc == default_receive_ssrc_) { - ASSERT(IsDefaultChannel(it->second.channel)); + ASSERT(IsDefaultChannel(channel)); // Recycle the default channel is for recv stream. if (playout_) SetPlayout(voe_channel(), false); - if (it->second.renderer) - it->second.renderer->RemoveChannel(voe_channel()); - default_receive_ssrc_ = 0; - receive_channels_.erase(it); return true; } - // Non default channel. - // Notify the renderer that channel is going away. - if (it->second.renderer) - it->second.renderer->RemoveChannel(it->second.channel); - LOG(LS_INFO) << "Removing audio stream " << ssrc - << " with VoiceEngine channel #" << it->second.channel << "."; - if (!DeleteChannel(it->second.channel)) { - // Erase the entry anyhow. - receive_channels_.erase(it); + << " with VoiceEngine channel #" << channel << "."; + if (!DeleteChannel(channel)) return false; - } - receive_channels_.erase(it); bool enable_default_channel_playout = false; if (receive_channels_.empty()) { // The last stream was removed. We can now enable the default @@ -2540,19 +2772,11 @@ bool WebRtcVoiceMediaChannel::SetRemoteRenderer(uint32 ssrc, return true; } - AudioRenderer* remote_renderer = it->second.renderer; - if (renderer) { - ASSERT(remote_renderer == NULL || remote_renderer == renderer); - if (!remote_renderer) { - renderer->AddChannel(it->second.channel); - } - } else if (remote_renderer) { - // |renderer| == NULL, remove the channel from the renderer. - remote_renderer->RemoveChannel(it->second.channel); - } + if (renderer) + it->second->Start(renderer); + else + it->second->Stop(); - // Assign the new value to the struct. - it->second.renderer = renderer; return true; } @@ -2570,17 +2794,11 @@ bool WebRtcVoiceMediaChannel::SetLocalRenderer(uint32 ssrc, return true; } - AudioRenderer* local_renderer = it->second.renderer; - if (renderer) { - ASSERT(local_renderer == NULL || local_renderer == renderer); - if (!local_renderer) - renderer->AddChannel(it->second.channel); - } else if (local_renderer) { - local_renderer->RemoveChannel(it->second.channel); - } + if (renderer) + it->second->Start(renderer); + else + it->second->Stop(); - // Assign the new value to the struct. - it->second.renderer = renderer; return true; } @@ -2591,7 +2809,7 @@ bool WebRtcVoiceMediaChannel::GetActiveStreams( actives->clear(); for (ChannelMap::iterator it = receive_channels_.begin(); it != receive_channels_.end(); ++it) { - int level = GetOutputLevel(it->second.channel); + int level = GetOutputLevel(it->second->channel()); if (level > 0) { actives->push_back(std::make_pair(it->first, level)); } @@ -2604,7 +2822,7 @@ int WebRtcVoiceMediaChannel::GetOutputLevel() { int highest = GetOutputLevel(voe_channel()); for (ChannelMap::iterator it = receive_channels_.begin(); it != receive_channels_.end(); ++it) { - int level = GetOutputLevel(it->second.channel); + int level = GetOutputLevel(it->second->channel()); highest = talk_base::_max(level, highest); } return highest; @@ -2647,7 +2865,7 @@ bool WebRtcVoiceMediaChannel::SetOutputScaling( channels.push_back(voe_channel()); for (ChannelMap::const_iterator it = receive_channels_.begin(); it != receive_channels_.end(); ++it) { - channels.push_back(it->second.channel); + channels.push_back(it->second->channel()); } } else { // Collect only the channel of the specified ssrc. int channel = GetReceiveChannelNum(ssrc); @@ -2783,7 +3001,7 @@ bool WebRtcVoiceMediaChannel::InsertDtmf(uint32 ssrc, int event, bool default_channel_is_inuse = false; for (ChannelMap::const_iterator iter = send_channels_.begin(); iter != send_channels_.end(); ++iter) { - if (IsDefaultChannel(iter->second.channel)) { + if (IsDefaultChannel(iter->second->channel())) { default_channel_is_inuse = true; break; } @@ -2791,7 +3009,7 @@ bool WebRtcVoiceMediaChannel::InsertDtmf(uint32 ssrc, int event, if (default_channel_is_inuse) { channel = voe_channel(); } else if (!send_channels_.empty()) { - channel = send_channels_.begin()->second.channel; + channel = send_channels_.begin()->second->channel(); } } else { channel = GetSendChannelNum(ssrc); @@ -2889,11 +3107,12 @@ void WebRtcVoiceMediaChannel::OnRtcpReceived( for (ChannelMap::iterator iter = send_channels_.begin(); iter != send_channels_.end(); ++iter) { // Make sure not sending the same packet to default channel more than once. - if (IsDefaultChannel(iter->second.channel) && has_sent_to_default_channel) + if (IsDefaultChannel(iter->second->channel()) && + has_sent_to_default_channel) continue; engine()->voe()->network()->ReceivedRTCPPacket( - iter->second.channel, + iter->second->channel(), packet->data(), static_cast<unsigned int>(packet->length())); } @@ -2912,18 +3131,23 @@ bool WebRtcVoiceMediaChannel::MuteStream(uint32 ssrc, bool muted) { return true; } -bool WebRtcVoiceMediaChannel::SetSendBandwidth(bool autobw, int bps) { - LOG(LS_INFO) << "WebRtcVoiceMediaChanne::SetSendBandwidth."; +bool WebRtcVoiceMediaChannel::SetStartSendBandwidth(int bps) { + // TODO(andresp): Add support for setting an independent start bandwidth when + // bandwidth estimation is enabled for voice engine. + return false; +} - send_bw_setting_ = true; - send_autobw_ = autobw; - send_bw_bps_ = bps; +bool WebRtcVoiceMediaChannel::SetMaxSendBandwidth(int bps) { + LOG(LS_INFO) << "WebRtcVoiceMediaChanne::SetSendBandwidth."; - return SetSendBandwidthInternal(send_autobw_, send_bw_bps_); + return SetSendBandwidthInternal(bps); } -bool WebRtcVoiceMediaChannel::SetSendBandwidthInternal(bool autobw, int bps) { - LOG(LS_INFO) << "WebRtcVoiceMediaChanne::SetSendBandwidthInternal."; +bool WebRtcVoiceMediaChannel::SetSendBandwidthInternal(int bps) { + LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendBandwidthInternal."; + + send_bw_setting_ = true; + send_bw_bps_ = bps; if (!send_codec_) { LOG(LS_INFO) << "The send codec has not been set up yet. " @@ -2932,7 +3156,9 @@ bool WebRtcVoiceMediaChannel::SetSendBandwidthInternal(bool autobw, int bps) { } // Bandwidth is auto by default. - if (autobw || bps <= 0) + // TODO(bemasc): Fix this so that if SetMaxSendBandwidth(50) is followed by + // SetMaxSendBandwith(0), the second call removes the previous limit. + if (bps <= 0) return true; webrtc::CodecInst codec = *send_codec_; @@ -2997,7 +3223,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) { for (ChannelMap::const_iterator channel_iter = send_channels_.begin(); channel_iter != send_channels_.end(); ++channel_iter) { - const int channel = channel_iter->second.channel; + const int channel = channel_iter->second->channel(); // Fill in the sender info, based on what we know, and what the // remote side told us it got from its RTCP report. @@ -3069,7 +3295,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) { std::vector<int> channels; for (ChannelMap::const_iterator it = receive_channels_.begin(); it != receive_channels_.end(); ++it) { - channels.push_back(it->second.channel); + channels.push_back(it->second->channel()); } if (channels.empty()) { channels.push_back(voe_channel()); @@ -3091,6 +3317,12 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) { rinfo.fraction_lost = static_cast<float>(cs.fractionLost) / (1 << 8); rinfo.packets_lost = cs.cumulativeLost; rinfo.ext_seqnum = cs.extendedMax; +#ifdef USE_WEBRTC_DEV_BRANCH + rinfo.capture_start_ntp_time_ms = cs.capture_start_ntp_time_ms_; +#endif + if (codec.pltype != -1) { + rinfo.codec_name = codec.plname; + } // Convert samples to milliseconds. if (codec.plfreq / 1000 > 0) { rinfo.jitter_ms = cs.jitterSamples / (codec.plfreq / 1000); @@ -3106,6 +3338,20 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) { rinfo.expand_rate = static_cast<float>(ns.currentExpandRate) / (1 << 14); } + + webrtc::AudioDecodingCallStats ds; + if (engine()->voe()->neteq() && + engine()->voe()->neteq()->GetDecodingCallStatistics( + *it, &ds) != -1) { + rinfo.decoding_calls_to_silence_generator = + ds.calls_to_silence_generator; + rinfo.decoding_calls_to_neteq = ds.calls_to_neteq; + rinfo.decoding_normal = ds.decoded_normal; + rinfo.decoding_plc = ds.decoded_plc; + rinfo.decoding_cng = ds.decoded_cng; + rinfo.decoding_plc_cng = ds.decoded_plc_cng; + } + if (engine()->voe()->sync()) { int jitter_buffer_delay_ms = 0; int playout_buffer_delay_ms = 0; @@ -3147,7 +3393,7 @@ bool WebRtcVoiceMediaChannel::FindSsrc(int channel_num, uint32* ssrc) { // Check whether this is a sending channel. for (ChannelMap::const_iterator it = send_channels_.begin(); it != send_channels_.end(); ++it) { - if (it->second.channel == channel_num) { + if (it->second->channel() == channel_num) { // This is a sending channel. uint32 local_ssrc = 0; if (engine()->voe()->rtp()->GetLocalSSRC( @@ -3161,7 +3407,7 @@ bool WebRtcVoiceMediaChannel::FindSsrc(int channel_num, uint32* ssrc) { // Check whether this is a receiving channel. for (ChannelMap::const_iterator it = receive_channels_.begin(); it != receive_channels_.end(); ++it) { - if (it->second.channel == channel_num) { + if (it->second->channel() == channel_num) { *ssrc = it->first; return true; } @@ -3189,14 +3435,14 @@ int WebRtcVoiceMediaChannel::GetOutputLevel(int channel) { int WebRtcVoiceMediaChannel::GetReceiveChannelNum(uint32 ssrc) { ChannelMap::iterator it = receive_channels_.find(ssrc); if (it != receive_channels_.end()) - return it->second.channel; + return it->second->channel(); return (ssrc == default_receive_ssrc_) ? voe_channel() : -1; } int WebRtcVoiceMediaChannel::GetSendChannelNum(uint32 ssrc) { ChannelMap::iterator it = send_channels_.find(ssrc); if (it != send_channels_.end()) - return it->second.channel; + return it->second->channel(); return -1; } @@ -3333,6 +3579,23 @@ VoiceMediaChannel::Error } } +bool WebRtcVoiceMediaChannel::SetHeaderExtension(ExtensionSetterFunction setter, + int channel_id, const RtpHeaderExtension* extension) { + bool enable = false; + int id = 0; + std::string uri; + if (extension) { + enable = true; + id = extension->id; + uri = extension->uri; + } + if ((engine()->voe()->rtp()->*setter)(channel_id, enable, id) != 0) { + LOG_RTCERR4(*setter, uri, channel_id, enable, id); + return false; + } + return true; +} + int WebRtcSoundclipStream::Read(void *buf, int len) { size_t res = 0; mem_.Read(buf, len, &res, NULL); |