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_send_stream.h"
12
13 #include <string>
14
15 #include "webrtc/audio/audio_state.h"
16 #include "webrtc/audio/conversion.h"
17 #include "webrtc/audio/scoped_voe_interface.h"
18 #include "webrtc/base/checks.h"
19 #include "webrtc/base/logging.h"
20 #include "webrtc/call/congestion_controller.h"
21 #include "webrtc/modules/pacing/paced_sender.h"
22 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
23 #include "webrtc/voice_engine/channel_proxy.h"
24 #include "webrtc/voice_engine/include/voe_audio_processing.h"
25 #include "webrtc/voice_engine/include/voe_codec.h"
26 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
27 #include "webrtc/voice_engine/include/voe_volume_control.h"
28 #include "webrtc/voice_engine/voice_engine_impl.h"
29
30 namespace webrtc {
ToString() const31 std::string AudioSendStream::Config::Rtp::ToString() const {
32 std::stringstream ss;
33 ss << "{ssrc: " << ssrc;
34 ss << ", extensions: [";
35 for (size_t i = 0; i < extensions.size(); ++i) {
36 ss << extensions[i].ToString();
37 if (i != extensions.size() - 1) {
38 ss << ", ";
39 }
40 }
41 ss << ']';
42 ss << ", c_name: " << c_name;
43 ss << '}';
44 return ss.str();
45 }
46
ToString() const47 std::string AudioSendStream::Config::ToString() const {
48 std::stringstream ss;
49 ss << "{rtp: " << rtp.ToString();
50 ss << ", voe_channel_id: " << voe_channel_id;
51 // TODO(solenberg): Encoder config.
52 ss << ", cng_payload_type: " << cng_payload_type;
53 ss << ", red_payload_type: " << red_payload_type;
54 ss << '}';
55 return ss.str();
56 }
57
58 namespace internal {
AudioSendStream(const webrtc::AudioSendStream::Config & config,const rtc::scoped_refptr<webrtc::AudioState> & audio_state,CongestionController * congestion_controller)59 AudioSendStream::AudioSendStream(
60 const webrtc::AudioSendStream::Config& config,
61 const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
62 CongestionController* congestion_controller)
63 : config_(config), audio_state_(audio_state) {
64 LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
65 RTC_DCHECK_NE(config_.voe_channel_id, -1);
66 RTC_DCHECK(audio_state_.get());
67 RTC_DCHECK(congestion_controller);
68
69 VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
70 channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
71 channel_proxy_->SetCongestionControlObjects(
72 congestion_controller->pacer(),
73 congestion_controller->GetTransportFeedbackObserver(),
74 congestion_controller->packet_router());
75 channel_proxy_->SetRTCPStatus(true);
76 channel_proxy_->SetLocalSSRC(config.rtp.ssrc);
77 channel_proxy_->SetRTCP_CNAME(config.rtp.c_name);
78
79 for (const auto& extension : config.rtp.extensions) {
80 if (extension.name == RtpExtension::kAbsSendTime) {
81 channel_proxy_->SetSendAbsoluteSenderTimeStatus(true, extension.id);
82 } else if (extension.name == RtpExtension::kAudioLevel) {
83 channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id);
84 } else if (extension.name == RtpExtension::kTransportSequenceNumber) {
85 channel_proxy_->EnableSendTransportSequenceNumber(extension.id);
86 } else {
87 RTC_NOTREACHED() << "Registering unsupported RTP extension.";
88 }
89 }
90 }
91
~AudioSendStream()92 AudioSendStream::~AudioSendStream() {
93 RTC_DCHECK(thread_checker_.CalledOnValidThread());
94 LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
95 channel_proxy_->SetCongestionControlObjects(nullptr, nullptr, nullptr);
96 }
97
Start()98 void AudioSendStream::Start() {
99 RTC_DCHECK(thread_checker_.CalledOnValidThread());
100 }
101
Stop()102 void AudioSendStream::Stop() {
103 RTC_DCHECK(thread_checker_.CalledOnValidThread());
104 }
105
SignalNetworkState(NetworkState state)106 void AudioSendStream::SignalNetworkState(NetworkState state) {
107 RTC_DCHECK(thread_checker_.CalledOnValidThread());
108 }
109
DeliverRtcp(const uint8_t * packet,size_t length)110 bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
111 // TODO(solenberg): Tests call this function on a network thread, libjingle
112 // calls on the worker thread. We should move towards always using a network
113 // thread. Then this check can be enabled.
114 // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
115 return false;
116 }
117
SendTelephoneEvent(int payload_type,uint8_t event,uint32_t duration_ms)118 bool AudioSendStream::SendTelephoneEvent(int payload_type, uint8_t event,
119 uint32_t duration_ms) {
120 RTC_DCHECK(thread_checker_.CalledOnValidThread());
121 return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type) &&
122 channel_proxy_->SendTelephoneEventOutband(event, duration_ms);
123 }
124
GetStats() const125 webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
126 RTC_DCHECK(thread_checker_.CalledOnValidThread());
127 webrtc::AudioSendStream::Stats stats;
128 stats.local_ssrc = config_.rtp.ssrc;
129 ScopedVoEInterface<VoEAudioProcessing> processing(voice_engine());
130 ScopedVoEInterface<VoECodec> codec(voice_engine());
131 ScopedVoEInterface<VoEVolumeControl> volume(voice_engine());
132
133 webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics();
134 stats.bytes_sent = call_stats.bytesSent;
135 stats.packets_sent = call_stats.packetsSent;
136 // RTT isn't known until a RTCP report is received. Until then, VoiceEngine
137 // returns 0 to indicate an error value.
138 if (call_stats.rttMs > 0) {
139 stats.rtt_ms = call_stats.rttMs;
140 }
141 // TODO(solenberg): [was ajm]: Re-enable this metric once we have a reliable
142 // implementation.
143 stats.aec_quality_min = -1;
144
145 webrtc::CodecInst codec_inst = {0};
146 if (codec->GetSendCodec(config_.voe_channel_id, codec_inst) != -1) {
147 RTC_DCHECK_NE(codec_inst.pltype, -1);
148 stats.codec_name = codec_inst.plname;
149
150 // Get data from the last remote RTCP report.
151 for (const auto& block : channel_proxy_->GetRemoteRTCPReportBlocks()) {
152 // Lookup report for send ssrc only.
153 if (block.source_SSRC == stats.local_ssrc) {
154 stats.packets_lost = block.cumulative_num_packets_lost;
155 stats.fraction_lost = Q8ToFloat(block.fraction_lost);
156 stats.ext_seqnum = block.extended_highest_sequence_number;
157 // Convert samples to milliseconds.
158 if (codec_inst.plfreq / 1000 > 0) {
159 stats.jitter_ms =
160 block.interarrival_jitter / (codec_inst.plfreq / 1000);
161 }
162 break;
163 }
164 }
165 }
166
167 // Local speech level.
168 {
169 unsigned int level = 0;
170 int error = volume->GetSpeechInputLevelFullRange(level);
171 RTC_DCHECK_EQ(0, error);
172 stats.audio_level = static_cast<int32_t>(level);
173 }
174
175 bool echo_metrics_on = false;
176 int error = processing->GetEcMetricsStatus(echo_metrics_on);
177 RTC_DCHECK_EQ(0, error);
178 if (echo_metrics_on) {
179 // These can also be negative, but in practice -1 is only used to signal
180 // insufficient data, since the resolution is limited to multiples of 4 ms.
181 int median = -1;
182 int std = -1;
183 float dummy = 0.0f;
184 error = processing->GetEcDelayMetrics(median, std, dummy);
185 RTC_DCHECK_EQ(0, error);
186 stats.echo_delay_median_ms = median;
187 stats.echo_delay_std_ms = std;
188
189 // These can take on valid negative values, so use the lowest possible level
190 // as default rather than -1.
191 int erl = -100;
192 int erle = -100;
193 int dummy1 = 0;
194 int dummy2 = 0;
195 error = processing->GetEchoMetrics(erl, erle, dummy1, dummy2);
196 RTC_DCHECK_EQ(0, error);
197 stats.echo_return_loss = erl;
198 stats.echo_return_loss_enhancement = erle;
199 }
200
201 internal::AudioState* audio_state =
202 static_cast<internal::AudioState*>(audio_state_.get());
203 stats.typing_noise_detected = audio_state->typing_noise_detected();
204
205 return stats;
206 }
207
config() const208 const webrtc::AudioSendStream::Config& AudioSendStream::config() const {
209 RTC_DCHECK(thread_checker_.CalledOnValidThread());
210 return config_;
211 }
212
voice_engine() const213 VoiceEngine* AudioSendStream::voice_engine() const {
214 internal::AudioState* audio_state =
215 static_cast<internal::AudioState*>(audio_state_.get());
216 VoiceEngine* voice_engine = audio_state->voice_engine();
217 RTC_DCHECK(voice_engine);
218 return voice_engine;
219 }
220 } // namespace internal
221 } // namespace webrtc
222