| /* |
| * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "webrtc/audio/audio_receive_stream.h" |
| |
| #include <string> |
| #include <utility> |
| |
| #include "webrtc/api/call/audio_sink.h" |
| #include "webrtc/audio/audio_state.h" |
| #include "webrtc/audio/conversion.h" |
| #include "webrtc/base/checks.h" |
| #include "webrtc/base/logging.h" |
| #include "webrtc/base/timeutils.h" |
| #include "webrtc/modules/congestion_controller/include/congestion_controller.h" |
| #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" |
| #include "webrtc/voice_engine/channel_proxy.h" |
| #include "webrtc/voice_engine/include/voe_base.h" |
| #include "webrtc/voice_engine/include/voe_codec.h" |
| #include "webrtc/voice_engine/include/voe_neteq_stats.h" |
| #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" |
| #include "webrtc/voice_engine/include/voe_video_sync.h" |
| #include "webrtc/voice_engine/include/voe_volume_control.h" |
| #include "webrtc/voice_engine/voice_engine_impl.h" |
| |
| namespace webrtc { |
| namespace { |
| |
| bool UseSendSideBwe(const webrtc::AudioReceiveStream::Config& config) { |
| if (!config.rtp.transport_cc) { |
| return false; |
| } |
| for (const auto& extension : config.rtp.extensions) { |
| if (extension.uri == RtpExtension::kTransportSequenceNumberUri) { |
| return true; |
| } |
| } |
| return false; |
| } |
| } // namespace |
| |
| std::string AudioReceiveStream::Config::Rtp::ToString() const { |
| std::stringstream ss; |
| ss << "{remote_ssrc: " << remote_ssrc; |
| ss << ", local_ssrc: " << local_ssrc; |
| ss << ", transport_cc: " << (transport_cc ? "on" : "off"); |
| ss << ", nack: " << nack.ToString(); |
| ss << ", extensions: ["; |
| for (size_t i = 0; i < extensions.size(); ++i) { |
| ss << extensions[i].ToString(); |
| if (i != extensions.size() - 1) { |
| ss << ", "; |
| } |
| } |
| ss << ']'; |
| ss << '}'; |
| return ss.str(); |
| } |
| |
| std::string AudioReceiveStream::Config::ToString() const { |
| std::stringstream ss; |
| ss << "{rtp: " << rtp.ToString(); |
| ss << ", rtcp_send_transport: " |
| << (rtcp_send_transport ? "(Transport)" : "nullptr"); |
| ss << ", voe_channel_id: " << voe_channel_id; |
| if (!sync_group.empty()) { |
| ss << ", sync_group: " << sync_group; |
| } |
| ss << '}'; |
| return ss.str(); |
| } |
| |
| namespace internal { |
| AudioReceiveStream::AudioReceiveStream( |
| CongestionController* congestion_controller, |
| const webrtc::AudioReceiveStream::Config& config, |
| const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
| webrtc::RtcEventLog* event_log) |
| : config_(config), |
| audio_state_(audio_state), |
| rtp_header_parser_(RtpHeaderParser::Create()) { |
| LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString(); |
| RTC_DCHECK_NE(config_.voe_channel_id, -1); |
| RTC_DCHECK(audio_state_.get()); |
| RTC_DCHECK(congestion_controller); |
| RTC_DCHECK(rtp_header_parser_); |
| |
| VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine()); |
| channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id); |
| channel_proxy_->SetRtcEventLog(event_log); |
| channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc); |
| // TODO(solenberg): Config NACK history window (which is a packet count), |
| // using the actual packet size for the configured codec. |
| channel_proxy_->SetNACKStatus(config_.rtp.nack.rtp_history_ms != 0, |
| config_.rtp.nack.rtp_history_ms / 20); |
| |
| // TODO(ossu): This is where we'd like to set the decoder factory to |
| // use. However, since it needs to be included when constructing Channel, we |
| // cannot do that until we're able to move Channel ownership into the |
| // Audio{Send,Receive}Streams. The best we can do is check that we're not |
| // trying to use two different factories using the different interfaces. |
| RTC_CHECK(config.decoder_factory); |
| RTC_CHECK_EQ(config.decoder_factory, |
| channel_proxy_->GetAudioDecoderFactory()); |
| |
| channel_proxy_->RegisterExternalTransport(config.rtcp_send_transport); |
| |
| for (const auto& extension : config.rtp.extensions) { |
| if (extension.uri == RtpExtension::kAudioLevelUri) { |
| channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id); |
| bool registered = rtp_header_parser_->RegisterRtpHeaderExtension( |
| kRtpExtensionAudioLevel, extension.id); |
| RTC_DCHECK(registered); |
| } else if (extension.uri == RtpExtension::kAbsSendTimeUri) { |
| channel_proxy_->SetReceiveAbsoluteSenderTimeStatus(true, extension.id); |
| bool registered = rtp_header_parser_->RegisterRtpHeaderExtension( |
| kRtpExtensionAbsoluteSendTime, extension.id); |
| RTC_DCHECK(registered); |
| } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) { |
| channel_proxy_->EnableReceiveTransportSequenceNumber(extension.id); |
| bool registered = rtp_header_parser_->RegisterRtpHeaderExtension( |
| kRtpExtensionTransportSequenceNumber, extension.id); |
| RTC_DCHECK(registered); |
| } else { |
| RTC_NOTREACHED() << "Unsupported RTP extension."; |
| } |
| } |
| // Configure bandwidth estimation. |
| channel_proxy_->RegisterReceiverCongestionControlObjects( |
| congestion_controller->packet_router()); |
| if (UseSendSideBwe(config)) { |
| remote_bitrate_estimator_ = |
| congestion_controller->GetRemoteBitrateEstimator(true); |
| } |
| } |
| |
| AudioReceiveStream::~AudioReceiveStream() { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString(); |
| channel_proxy_->DeRegisterExternalTransport(); |
| channel_proxy_->ResetCongestionControlObjects(); |
| channel_proxy_->SetRtcEventLog(nullptr); |
| if (remote_bitrate_estimator_) { |
| remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc); |
| } |
| } |
| |
| void AudioReceiveStream::Start() { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| ScopedVoEInterface<VoEBase> base(voice_engine()); |
| int error = base->StartPlayout(config_.voe_channel_id); |
| if (error != 0) { |
| LOG(LS_ERROR) << "AudioReceiveStream::Start failed with error: " << error; |
| } |
| } |
| |
| void AudioReceiveStream::Stop() { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| ScopedVoEInterface<VoEBase> base(voice_engine()); |
| base->StopPlayout(config_.voe_channel_id); |
| } |
| |
| webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| webrtc::AudioReceiveStream::Stats stats; |
| stats.remote_ssrc = config_.rtp.remote_ssrc; |
| ScopedVoEInterface<VoECodec> codec(voice_engine()); |
| |
| webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics(); |
| webrtc::CodecInst codec_inst = {0}; |
| if (codec->GetRecCodec(config_.voe_channel_id, codec_inst) == -1) { |
| return stats; |
| } |
| |
| stats.bytes_rcvd = call_stats.bytesReceived; |
| stats.packets_rcvd = call_stats.packetsReceived; |
| stats.packets_lost = call_stats.cumulativeLost; |
| stats.fraction_lost = Q8ToFloat(call_stats.fractionLost); |
| stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_; |
| if (codec_inst.pltype != -1) { |
| stats.codec_name = codec_inst.plname; |
| } |
| stats.ext_seqnum = call_stats.extendedMax; |
| if (codec_inst.plfreq / 1000 > 0) { |
| stats.jitter_ms = call_stats.jitterSamples / (codec_inst.plfreq / 1000); |
| } |
| stats.delay_estimate_ms = channel_proxy_->GetDelayEstimate(); |
| stats.audio_level = channel_proxy_->GetSpeechOutputLevelFullRange(); |
| |
| // Get jitter buffer and total delay (alg + jitter + playout) stats. |
| auto ns = channel_proxy_->GetNetworkStatistics(); |
| stats.jitter_buffer_ms = ns.currentBufferSize; |
| stats.jitter_buffer_preferred_ms = ns.preferredBufferSize; |
| stats.expand_rate = Q14ToFloat(ns.currentExpandRate); |
| stats.speech_expand_rate = Q14ToFloat(ns.currentSpeechExpandRate); |
| stats.secondary_decoded_rate = Q14ToFloat(ns.currentSecondaryDecodedRate); |
| stats.accelerate_rate = Q14ToFloat(ns.currentAccelerateRate); |
| stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate); |
| |
| auto ds = channel_proxy_->GetDecodingCallStatistics(); |
| stats.decoding_calls_to_silence_generator = ds.calls_to_silence_generator; |
| stats.decoding_calls_to_neteq = ds.calls_to_neteq; |
| stats.decoding_normal = ds.decoded_normal; |
| stats.decoding_plc = ds.decoded_plc; |
| stats.decoding_cng = ds.decoded_cng; |
| stats.decoding_plc_cng = ds.decoded_plc_cng; |
| |
| return stats; |
| } |
| |
| void AudioReceiveStream::SetSink(std::unique_ptr<AudioSinkInterface> sink) { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| channel_proxy_->SetSink(std::move(sink)); |
| } |
| |
| void AudioReceiveStream::SetGain(float gain) { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| channel_proxy_->SetChannelOutputVolumeScaling(gain); |
| } |
| |
| const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| return config_; |
| } |
| |
| void AudioReceiveStream::SignalNetworkState(NetworkState state) { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| } |
| |
| bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) { |
| // TODO(solenberg): Tests call this function on a network thread, libjingle |
| // calls on the worker thread. We should move towards always using a network |
| // thread. Then this check can be enabled. |
| // RTC_DCHECK(!thread_checker_.CalledOnValidThread()); |
| return channel_proxy_->ReceivedRTCPPacket(packet, length); |
| } |
| |
| bool AudioReceiveStream::DeliverRtp(const uint8_t* packet, |
| size_t length, |
| const PacketTime& packet_time) { |
| // TODO(solenberg): Tests call this function on a network thread, libjingle |
| // calls on the worker thread. We should move towards always using a network |
| // thread. Then this check can be enabled. |
| // RTC_DCHECK(!thread_checker_.CalledOnValidThread()); |
| RTPHeader header; |
| if (!rtp_header_parser_->Parse(packet, length, &header)) { |
| return false; |
| } |
| |
| // Only forward if the parsed header has one of the headers necessary for |
| // bandwidth estimation. RTP timestamps has different rates for audio and |
| // video and shouldn't be mixed. |
| if (remote_bitrate_estimator_ && |
| header.extension.hasTransportSequenceNumber) { |
| int64_t arrival_time_ms = rtc::TimeMillis(); |
| if (packet_time.timestamp >= 0) |
| arrival_time_ms = (packet_time.timestamp + 500) / 1000; |
| size_t payload_size = length - header.headerLength; |
| remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size, |
| header); |
| } |
| |
| return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time); |
| } |
| |
| VoiceEngine* AudioReceiveStream::voice_engine() const { |
| internal::AudioState* audio_state = |
| static_cast<internal::AudioState*>(audio_state_.get()); |
| VoiceEngine* voice_engine = audio_state->voice_engine(); |
| RTC_DCHECK(voice_engine); |
| return voice_engine; |
| } |
| } // namespace internal |
| } // namespace webrtc |