| /* |
| * 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_send_stream.h" |
| |
| #include <string> |
| |
| #include "webrtc/audio/audio_state.h" |
| #include "webrtc/audio/conversion.h" |
| #include "webrtc/audio/scoped_voe_interface.h" |
| #include "webrtc/base/checks.h" |
| #include "webrtc/base/event.h" |
| #include "webrtc/base/logging.h" |
| #include "webrtc/base/task_queue.h" |
| #include "webrtc/modules/congestion_controller/include/congestion_controller.h" |
| #include "webrtc/modules/pacing/paced_sender.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "webrtc/voice_engine/channel_proxy.h" |
| #include "webrtc/voice_engine/include/voe_audio_processing.h" |
| #include "webrtc/voice_engine/include/voe_codec.h" |
| #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" |
| #include "webrtc/voice_engine/include/voe_volume_control.h" |
| #include "webrtc/voice_engine/voice_engine_impl.h" |
| |
| namespace webrtc { |
| std::string AudioSendStream::Config::Rtp::ToString() const { |
| std::stringstream ss; |
| ss << "{ssrc: " << ssrc; |
| ss << ", extensions: ["; |
| for (size_t i = 0; i < extensions.size(); ++i) { |
| ss << extensions[i].ToString(); |
| if (i != extensions.size() - 1) { |
| ss << ", "; |
| } |
| } |
| ss << ']'; |
| ss << ", nack: " << nack.ToString(); |
| ss << ", c_name: " << c_name; |
| ss << '}'; |
| return ss.str(); |
| } |
| |
| std::string AudioSendStream::Config::ToString() const { |
| std::stringstream ss; |
| ss << "{rtp: " << rtp.ToString(); |
| ss << ", voe_channel_id: " << voe_channel_id; |
| // TODO(solenberg): Encoder config. |
| ss << ", cng_payload_type: " << cng_payload_type; |
| ss << '}'; |
| return ss.str(); |
| } |
| |
| namespace internal { |
| AudioSendStream::AudioSendStream( |
| const webrtc::AudioSendStream::Config& config, |
| const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
| rtc::TaskQueue* worker_queue, |
| CongestionController* congestion_controller, |
| BitrateAllocator* bitrate_allocator, |
| RtcEventLog* event_log) |
| : worker_queue_(worker_queue), |
| config_(config), |
| audio_state_(audio_state), |
| bitrate_allocator_(bitrate_allocator) { |
| LOG(LS_INFO) << "AudioSendStream: " << config_.ToString(); |
| RTC_DCHECK_NE(config_.voe_channel_id, -1); |
| RTC_DCHECK(audio_state_.get()); |
| RTC_DCHECK(congestion_controller); |
| |
| VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine()); |
| channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id); |
| channel_proxy_->SetRtcEventLog(event_log); |
| channel_proxy_->RegisterSenderCongestionControlObjects( |
| congestion_controller->pacer(), |
| congestion_controller->GetTransportFeedbackObserver(), |
| congestion_controller->packet_router()); |
| channel_proxy_->SetRTCPStatus(true); |
| channel_proxy_->SetLocalSSRC(config.rtp.ssrc); |
| channel_proxy_->SetRTCP_CNAME(config.rtp.c_name); |
| // 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); |
| |
| channel_proxy_->RegisterExternalTransport(config.send_transport); |
| |
| for (const auto& extension : config.rtp.extensions) { |
| if (extension.uri == RtpExtension::kAbsSendTimeUri) { |
| channel_proxy_->SetSendAbsoluteSenderTimeStatus(true, extension.id); |
| } else if (extension.uri == RtpExtension::kAudioLevelUri) { |
| channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id); |
| } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) { |
| channel_proxy_->EnableSendTransportSequenceNumber(extension.id); |
| } else { |
| RTC_NOTREACHED() << "Registering unsupported RTP extension."; |
| } |
| } |
| } |
| |
| AudioSendStream::~AudioSendStream() { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString(); |
| channel_proxy_->DeRegisterExternalTransport(); |
| channel_proxy_->ResetCongestionControlObjects(); |
| channel_proxy_->SetRtcEventLog(nullptr); |
| } |
| |
| void AudioSendStream::Start() { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| if (config_.min_bitrate_kbps != -1 && config_.max_bitrate_kbps != -1) { |
| RTC_DCHECK_GE(config_.max_bitrate_kbps, config_.min_bitrate_kbps); |
| rtc::Event thread_sync_event(false /* manual_reset */, false); |
| worker_queue_->PostTask([this, &thread_sync_event] { |
| bitrate_allocator_->AddObserver(this, config_.min_bitrate_kbps * 1000, |
| config_.max_bitrate_kbps * 1000, 0, true); |
| thread_sync_event.Set(); |
| }); |
| thread_sync_event.Wait(rtc::Event::kForever); |
| } |
| |
| ScopedVoEInterface<VoEBase> base(voice_engine()); |
| int error = base->StartSend(config_.voe_channel_id); |
| if (error != 0) { |
| LOG(LS_ERROR) << "AudioSendStream::Start failed with error: " << error; |
| } |
| } |
| |
| void AudioSendStream::Stop() { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| rtc::Event thread_sync_event(false /* manual_reset */, false); |
| worker_queue_->PostTask([this, &thread_sync_event] { |
| bitrate_allocator_->RemoveObserver(this); |
| thread_sync_event.Set(); |
| }); |
| thread_sync_event.Wait(rtc::Event::kForever); |
| |
| ScopedVoEInterface<VoEBase> base(voice_engine()); |
| int error = base->StopSend(config_.voe_channel_id); |
| if (error != 0) { |
| LOG(LS_ERROR) << "AudioSendStream::Stop failed with error: " << error; |
| } |
| } |
| |
| bool AudioSendStream::SendTelephoneEvent(int payload_type, int event, |
| int duration_ms) { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type) && |
| channel_proxy_->SendTelephoneEventOutband(event, duration_ms); |
| } |
| |
| void AudioSendStream::SetMuted(bool muted) { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| channel_proxy_->SetInputMute(muted); |
| } |
| |
| webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| webrtc::AudioSendStream::Stats stats; |
| stats.local_ssrc = config_.rtp.ssrc; |
| ScopedVoEInterface<VoEAudioProcessing> processing(voice_engine()); |
| ScopedVoEInterface<VoECodec> codec(voice_engine()); |
| ScopedVoEInterface<VoEVolumeControl> volume(voice_engine()); |
| |
| webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics(); |
| stats.bytes_sent = call_stats.bytesSent; |
| stats.packets_sent = call_stats.packetsSent; |
| // RTT isn't known until a RTCP report is received. Until then, VoiceEngine |
| // returns 0 to indicate an error value. |
| if (call_stats.rttMs > 0) { |
| stats.rtt_ms = call_stats.rttMs; |
| } |
| // TODO(solenberg): [was ajm]: Re-enable this metric once we have a reliable |
| // implementation. |
| stats.aec_quality_min = -1; |
| |
| webrtc::CodecInst codec_inst = {0}; |
| if (codec->GetSendCodec(config_.voe_channel_id, codec_inst) != -1) { |
| RTC_DCHECK_NE(codec_inst.pltype, -1); |
| stats.codec_name = codec_inst.plname; |
| |
| // Get data from the last remote RTCP report. |
| for (const auto& block : channel_proxy_->GetRemoteRTCPReportBlocks()) { |
| // Lookup report for send ssrc only. |
| if (block.source_SSRC == stats.local_ssrc) { |
| stats.packets_lost = block.cumulative_num_packets_lost; |
| stats.fraction_lost = Q8ToFloat(block.fraction_lost); |
| stats.ext_seqnum = block.extended_highest_sequence_number; |
| // Convert samples to milliseconds. |
| if (codec_inst.plfreq / 1000 > 0) { |
| stats.jitter_ms = |
| block.interarrival_jitter / (codec_inst.plfreq / 1000); |
| } |
| break; |
| } |
| } |
| } |
| |
| // Local speech level. |
| { |
| unsigned int level = 0; |
| int error = volume->GetSpeechInputLevelFullRange(level); |
| RTC_DCHECK_EQ(0, error); |
| stats.audio_level = static_cast<int32_t>(level); |
| } |
| |
| bool echo_metrics_on = false; |
| int error = processing->GetEcMetricsStatus(echo_metrics_on); |
| RTC_DCHECK_EQ(0, error); |
| if (echo_metrics_on) { |
| // These can also be negative, but in practice -1 is only used to signal |
| // insufficient data, since the resolution is limited to multiples of 4 ms. |
| int median = -1; |
| int std = -1; |
| float dummy = 0.0f; |
| error = processing->GetEcDelayMetrics(median, std, dummy); |
| RTC_DCHECK_EQ(0, error); |
| stats.echo_delay_median_ms = median; |
| stats.echo_delay_std_ms = std; |
| |
| // These can take on valid negative values, so use the lowest possible level |
| // as default rather than -1. |
| int erl = -100; |
| int erle = -100; |
| int dummy1 = 0; |
| int dummy2 = 0; |
| error = processing->GetEchoMetrics(erl, erle, dummy1, dummy2); |
| RTC_DCHECK_EQ(0, error); |
| stats.echo_return_loss = erl; |
| stats.echo_return_loss_enhancement = erle; |
| } |
| |
| internal::AudioState* audio_state = |
| static_cast<internal::AudioState*>(audio_state_.get()); |
| stats.typing_noise_detected = audio_state->typing_noise_detected(); |
| |
| return stats; |
| } |
| |
| void AudioSendStream::SignalNetworkState(NetworkState state) { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| } |
| |
| bool AudioSendStream::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); |
| } |
| |
| uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps, |
| uint8_t fraction_loss, |
| int64_t rtt) { |
| RTC_DCHECK_GE(bitrate_bps, |
| static_cast<uint32_t>(config_.min_bitrate_kbps * 1000)); |
| // The bitrate allocator might allocate an higher than max configured bitrate |
| // if there is room, to allow for, as example, extra FEC. Ignore that for now. |
| const uint32_t max_bitrate_bps = config_.max_bitrate_kbps * 1000; |
| if (bitrate_bps > max_bitrate_bps) |
| bitrate_bps = max_bitrate_bps; |
| |
| channel_proxy_->SetBitrate(bitrate_bps); |
| |
| // The amount of audio protection is not exposed by the encoder, hence |
| // always returning 0. |
| return 0; |
| } |
| |
| const webrtc::AudioSendStream::Config& AudioSendStream::config() const { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| return config_; |
| } |
| |
| VoiceEngine* AudioSendStream::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 |