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
UseSendSideBwe(const webrtc::AudioReceiveStream::Config & config)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
ToString() const49 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
ToString() const65 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 {
AudioReceiveStream(CongestionController * congestion_controller,const webrtc::AudioReceiveStream::Config & config,const rtc::scoped_refptr<webrtc::AudioState> & audio_state)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
~AudioReceiveStream()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
Start()142 void AudioReceiveStream::Start() {
143 RTC_DCHECK(thread_checker_.CalledOnValidThread());
144 }
145
Stop()146 void AudioReceiveStream::Stop() {
147 RTC_DCHECK(thread_checker_.CalledOnValidThread());
148 }
149
SignalNetworkState(NetworkState state)150 void AudioReceiveStream::SignalNetworkState(NetworkState state) {
151 RTC_DCHECK(thread_checker_.CalledOnValidThread());
152 }
153
DeliverRtcp(const uint8_t * packet,size_t length)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
DeliverRtp(const uint8_t * packet,size_t length,const PacketTime & packet_time)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
GetStats() const190 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
SetSink(rtc::scoped_ptr<AudioSinkInterface> sink)238 void AudioReceiveStream::SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) {
239 RTC_DCHECK(thread_checker_.CalledOnValidThread());
240 channel_proxy_->SetSink(std::move(sink));
241 }
242
config() const243 const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
244 RTC_DCHECK(thread_checker_.CalledOnValidThread());
245 return config_;
246 }
247
voice_engine() const248 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