85a0496b8c
BUG=webrtc:4690 Review URL: https://codereview.webrtc.org/1414743004 Cr-Commit-Position: refs/heads/master@{#10424}
227 lines
8.5 KiB
C++
227 lines
8.5 KiB
C++
/*
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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 "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/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
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#include "webrtc/system_wrappers/interface/tick_util.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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namespace webrtc {
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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 << ", 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 << ", receive_transport: "
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<< (receive_transport ? "(Transport)" : "nullptr");
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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 << ", combined_audio_video_bwe: "
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<< (combined_audio_video_bwe ? "true" : "false");
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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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RemoteBitrateEstimator* remote_bitrate_estimator,
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const webrtc::AudioReceiveStream::Config& config,
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VoiceEngine* voice_engine)
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: remote_bitrate_estimator_(remote_bitrate_estimator),
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config_(config),
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voice_engine_(voice_engine),
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voe_base_(voice_engine),
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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(config.voe_channel_id != -1);
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RTC_DCHECK(remote_bitrate_estimator_ != nullptr);
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RTC_DCHECK(voice_engine_ != nullptr);
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RTC_DCHECK(rtp_header_parser_ != nullptr);
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for (const auto& ext : config.rtp.extensions) {
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// One-byte-extension local identifiers are in the range 1-14 inclusive.
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RTC_DCHECK_GE(ext.id, 1);
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RTC_DCHECK_LE(ext.id, 14);
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if (ext.name == RtpExtension::kAudioLevel) {
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RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
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kRtpExtensionAudioLevel, ext.id));
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} else if (ext.name == RtpExtension::kAbsSendTime) {
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RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
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kRtpExtensionAbsoluteSendTime, ext.id));
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} else if (ext.name == RtpExtension::kTransportSequenceNumber) {
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RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
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kRtpExtensionTransportSequenceNumber, ext.id));
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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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}
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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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}
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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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ScopedVoEInterface<VoENetEqStats> neteq(voice_engine_);
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ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine_);
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ScopedVoEInterface<VoEVideoSync> sync(voice_engine_);
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ScopedVoEInterface<VoEVolumeControl> volume(voice_engine_);
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unsigned int ssrc = 0;
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webrtc::CallStatistics call_stats = {0};
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webrtc::CodecInst codec_inst = {0};
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// Only collect stats if we have seen some traffic with the SSRC.
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if (rtp->GetRemoteSSRC(config_.voe_channel_id, ssrc) == -1 ||
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rtp->GetRTCPStatistics(config_.voe_channel_id, call_stats) == -1 ||
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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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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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{
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int jitter_buffer_delay_ms = 0;
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int playout_buffer_delay_ms = 0;
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sync->GetDelayEstimate(config_.voe_channel_id, &jitter_buffer_delay_ms,
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&playout_buffer_delay_ms);
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stats.delay_estimate_ms =
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jitter_buffer_delay_ms + playout_buffer_delay_ms;
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}
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{
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unsigned int level = 0;
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if (volume->GetSpeechOutputLevelFullRange(config_.voe_channel_id, level)
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!= -1) {
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stats.audio_level = static_cast<int32_t>(level);
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}
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}
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webrtc::NetworkStatistics ns = {0};
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if (neteq->GetNetworkStatistics(config_.voe_channel_id, ns) != -1) {
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// Get jitter buffer and total delay (alg + jitter + playout) stats.
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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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}
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webrtc::AudioDecodingCallStats ds;
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if (neteq->GetDecodingCallStatistics(config_.voe_channel_id, &ds) != -1) {
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stats.decoding_calls_to_silence_generator =
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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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}
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stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_;
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return stats;
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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::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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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 false;
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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 absolute sender time. RTP timestamps
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// may have different rates for audio and video and shouldn't be mixed.
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if (config_.combined_audio_video_bwe &&
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header.extension.hasAbsoluteSendTime) {
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int64_t arrival_time_ms = TickTime::MillisecondTimestamp();
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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, false);
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}
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return true;
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}
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} // namespace internal
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} // namespace webrtc
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