1 /* 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #include "webrtc/audio/audio_receive_stream.h" 12 13 #include <string> 14 #include <utility> 15 16 #include "webrtc/audio/audio_sink.h" 17 #include "webrtc/audio/audio_state.h" 18 #include "webrtc/audio/conversion.h" 19 #include "webrtc/base/checks.h" 20 #include "webrtc/base/logging.h" 21 #include "webrtc/call/congestion_controller.h" 22 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" 23 #include "webrtc/system_wrappers/include/tick_util.h" 24 #include "webrtc/voice_engine/channel_proxy.h" 25 #include "webrtc/voice_engine/include/voe_base.h" 26 #include "webrtc/voice_engine/include/voe_codec.h" 27 #include "webrtc/voice_engine/include/voe_neteq_stats.h" 28 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" 29 #include "webrtc/voice_engine/include/voe_video_sync.h" 30 #include "webrtc/voice_engine/include/voe_volume_control.h" 31 #include "webrtc/voice_engine/voice_engine_impl.h" 32 33 namespace webrtc { 34 namespace { 35 36 bool UseSendSideBwe(const webrtc::AudioReceiveStream::Config& config) { 37 if (!config.rtp.transport_cc) { 38 return false; 39 } 40 for (const auto& extension : config.rtp.extensions) { 41 if (extension.name == RtpExtension::kTransportSequenceNumber) { 42 return true; 43 } 44 } 45 return false; 46 } 47 } // namespace 48 49 std::string AudioReceiveStream::Config::Rtp::ToString() const { 50 std::stringstream ss; 51 ss << "{remote_ssrc: " << remote_ssrc; 52 ss << ", local_ssrc: " << local_ssrc; 53 ss << ", extensions: ["; 54 for (size_t i = 0; i < extensions.size(); ++i) { 55 ss << extensions[i].ToString(); 56 if (i != extensions.size() - 1) { 57 ss << ", "; 58 } 59 } 60 ss << ']'; 61 ss << '}'; 62 return ss.str(); 63 } 64 65 std::string AudioReceiveStream::Config::ToString() const { 66 std::stringstream ss; 67 ss << "{rtp: " << rtp.ToString(); 68 ss << ", receive_transport: " 69 << (receive_transport ? "(Transport)" : "nullptr"); 70 ss << ", rtcp_send_transport: " 71 << (rtcp_send_transport ? "(Transport)" : "nullptr"); 72 ss << ", voe_channel_id: " << voe_channel_id; 73 if (!sync_group.empty()) { 74 ss << ", sync_group: " << sync_group; 75 } 76 ss << ", combined_audio_video_bwe: " 77 << (combined_audio_video_bwe ? "true" : "false"); 78 ss << '}'; 79 return ss.str(); 80 } 81 82 namespace internal { 83 AudioReceiveStream::AudioReceiveStream( 84 CongestionController* congestion_controller, 85 const webrtc::AudioReceiveStream::Config& config, 86 const rtc::scoped_refptr<webrtc::AudioState>& audio_state) 87 : config_(config), 88 audio_state_(audio_state), 89 rtp_header_parser_(RtpHeaderParser::Create()) { 90 LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString(); 91 RTC_DCHECK_NE(config_.voe_channel_id, -1); 92 RTC_DCHECK(audio_state_.get()); 93 RTC_DCHECK(congestion_controller); 94 RTC_DCHECK(rtp_header_parser_); 95 96 VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine()); 97 channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id); 98 channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc); 99 for (const auto& extension : config.rtp.extensions) { 100 if (extension.name == RtpExtension::kAudioLevel) { 101 channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id); 102 bool registered = rtp_header_parser_->RegisterRtpHeaderExtension( 103 kRtpExtensionAudioLevel, extension.id); 104 RTC_DCHECK(registered); 105 } else if (extension.name == RtpExtension::kAbsSendTime) { 106 channel_proxy_->SetReceiveAbsoluteSenderTimeStatus(true, extension.id); 107 bool registered = rtp_header_parser_->RegisterRtpHeaderExtension( 108 kRtpExtensionAbsoluteSendTime, extension.id); 109 RTC_DCHECK(registered); 110 } else if (extension.name == RtpExtension::kTransportSequenceNumber) { 111 bool registered = rtp_header_parser_->RegisterRtpHeaderExtension( 112 kRtpExtensionTransportSequenceNumber, extension.id); 113 RTC_DCHECK(registered); 114 } else { 115 RTC_NOTREACHED() << "Unsupported RTP extension."; 116 } 117 } 118 // Configure bandwidth estimation. 119 channel_proxy_->SetCongestionControlObjects( 120 nullptr, nullptr, congestion_controller->packet_router()); 121 if (config.combined_audio_video_bwe) { 122 if (UseSendSideBwe(config)) { 123 remote_bitrate_estimator_ = 124 congestion_controller->GetRemoteBitrateEstimator(true); 125 } else { 126 remote_bitrate_estimator_ = 127 congestion_controller->GetRemoteBitrateEstimator(false); 128 } 129 RTC_DCHECK(remote_bitrate_estimator_); 130 } 131 } 132 133 AudioReceiveStream::~AudioReceiveStream() { 134 RTC_DCHECK(thread_checker_.CalledOnValidThread()); 135 LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString(); 136 channel_proxy_->SetCongestionControlObjects(nullptr, nullptr, nullptr); 137 if (remote_bitrate_estimator_) { 138 remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc); 139 } 140 } 141 142 void AudioReceiveStream::Start() { 143 RTC_DCHECK(thread_checker_.CalledOnValidThread()); 144 } 145 146 void AudioReceiveStream::Stop() { 147 RTC_DCHECK(thread_checker_.CalledOnValidThread()); 148 } 149 150 void AudioReceiveStream::SignalNetworkState(NetworkState state) { 151 RTC_DCHECK(thread_checker_.CalledOnValidThread()); 152 } 153 154 bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) { 155 // TODO(solenberg): Tests call this function on a network thread, libjingle 156 // calls on the worker thread. We should move towards always using a network 157 // thread. Then this check can be enabled. 158 // RTC_DCHECK(!thread_checker_.CalledOnValidThread()); 159 return false; 160 } 161 162 bool AudioReceiveStream::DeliverRtp(const uint8_t* packet, 163 size_t length, 164 const PacketTime& packet_time) { 165 // TODO(solenberg): Tests call this function on a network thread, libjingle 166 // calls on the worker thread. We should move towards always using a network 167 // thread. Then this check can be enabled. 168 // RTC_DCHECK(!thread_checker_.CalledOnValidThread()); 169 RTPHeader header; 170 if (!rtp_header_parser_->Parse(packet, length, &header)) { 171 return false; 172 } 173 174 // Only forward if the parsed header has one of the headers necessary for 175 // bandwidth estimation. RTP timestamps has different rates for audio and 176 // video and shouldn't be mixed. 177 if (remote_bitrate_estimator_ && 178 (header.extension.hasAbsoluteSendTime || 179 header.extension.hasTransportSequenceNumber)) { 180 int64_t arrival_time_ms = TickTime::MillisecondTimestamp(); 181 if (packet_time.timestamp >= 0) 182 arrival_time_ms = (packet_time.timestamp + 500) / 1000; 183 size_t payload_size = length - header.headerLength; 184 remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size, 185 header, false); 186 } 187 return true; 188 } 189 190 webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const { 191 RTC_DCHECK(thread_checker_.CalledOnValidThread()); 192 webrtc::AudioReceiveStream::Stats stats; 193 stats.remote_ssrc = config_.rtp.remote_ssrc; 194 ScopedVoEInterface<VoECodec> codec(voice_engine()); 195 196 webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics(); 197 webrtc::CodecInst codec_inst = {0}; 198 if (codec->GetRecCodec(config_.voe_channel_id, codec_inst) == -1) { 199 return stats; 200 } 201 202 stats.bytes_rcvd = call_stats.bytesReceived; 203 stats.packets_rcvd = call_stats.packetsReceived; 204 stats.packets_lost = call_stats.cumulativeLost; 205 stats.fraction_lost = Q8ToFloat(call_stats.fractionLost); 206 stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_; 207 if (codec_inst.pltype != -1) { 208 stats.codec_name = codec_inst.plname; 209 } 210 stats.ext_seqnum = call_stats.extendedMax; 211 if (codec_inst.plfreq / 1000 > 0) { 212 stats.jitter_ms = call_stats.jitterSamples / (codec_inst.plfreq / 1000); 213 } 214 stats.delay_estimate_ms = channel_proxy_->GetDelayEstimate(); 215 stats.audio_level = channel_proxy_->GetSpeechOutputLevelFullRange(); 216 217 // Get jitter buffer and total delay (alg + jitter + playout) stats. 218 auto ns = channel_proxy_->GetNetworkStatistics(); 219 stats.jitter_buffer_ms = ns.currentBufferSize; 220 stats.jitter_buffer_preferred_ms = ns.preferredBufferSize; 221 stats.expand_rate = Q14ToFloat(ns.currentExpandRate); 222 stats.speech_expand_rate = Q14ToFloat(ns.currentSpeechExpandRate); 223 stats.secondary_decoded_rate = Q14ToFloat(ns.currentSecondaryDecodedRate); 224 stats.accelerate_rate = Q14ToFloat(ns.currentAccelerateRate); 225 stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate); 226 227 auto ds = channel_proxy_->GetDecodingCallStatistics(); 228 stats.decoding_calls_to_silence_generator = ds.calls_to_silence_generator; 229 stats.decoding_calls_to_neteq = ds.calls_to_neteq; 230 stats.decoding_normal = ds.decoded_normal; 231 stats.decoding_plc = ds.decoded_plc; 232 stats.decoding_cng = ds.decoded_cng; 233 stats.decoding_plc_cng = ds.decoded_plc_cng; 234 235 return stats; 236 } 237 238 void AudioReceiveStream::SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) { 239 RTC_DCHECK(thread_checker_.CalledOnValidThread()); 240 channel_proxy_->SetSink(std::move(sink)); 241 } 242 243 const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const { 244 RTC_DCHECK(thread_checker_.CalledOnValidThread()); 245 return config_; 246 } 247 248 VoiceEngine* AudioReceiveStream::voice_engine() const { 249 internal::AudioState* audio_state = 250 static_cast<internal::AudioState*>(audio_state_.get()); 251 VoiceEngine* voice_engine = audio_state->voice_engine(); 252 RTC_DCHECK(voice_engine); 253 return voice_engine; 254 } 255 } // namespace internal 256 } // namespace webrtc 257
