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