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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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112 } | 112 } |
113 | 113 |
114 webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { | 114 webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { |
115 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 115 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
116 webrtc::AudioSendStream::Stats stats; | 116 webrtc::AudioSendStream::Stats stats; |
117 stats.local_ssrc = config_.rtp.ssrc; | 117 stats.local_ssrc = config_.rtp.ssrc; |
118 ScopedVoEInterface<VoEAudioProcessing> processing(voice_engine()); | 118 ScopedVoEInterface<VoEAudioProcessing> processing(voice_engine()); |
119 ScopedVoEInterface<VoECodec> codec(voice_engine()); | 119 ScopedVoEInterface<VoECodec> codec(voice_engine()); |
120 ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine()); | 120 ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine()); |
121 ScopedVoEInterface<VoEVolumeControl> volume(voice_engine()); | 121 ScopedVoEInterface<VoEVolumeControl> volume(voice_engine()); |
122 unsigned int ssrc = 0; | 122 |
123 webrtc::CallStatistics call_stats = {0}; | 123 webrtc::CallStatistics call_stats = {0}; |
124 // TODO(solenberg): Change error code checking to RTC_CHECK_EQ(..., -1), if | 124 int error = rtp->GetRTCPStatistics(config_.voe_channel_id, call_stats); |
125 // possible... | 125 RTC_DCHECK_EQ(0, error); |
126 if (rtp->GetLocalSSRC(config_.voe_channel_id, ssrc) == -1 || | |
127 rtp->GetRTCPStatistics(config_.voe_channel_id, call_stats) == -1) { | |
128 return stats; | |
129 } | |
130 | |
131 stats.bytes_sent = call_stats.bytesSent; | 126 stats.bytes_sent = call_stats.bytesSent; |
132 stats.packets_sent = call_stats.packetsSent; | 127 stats.packets_sent = call_stats.packetsSent; |
| 128 // RTT isn't known until a RTCP report is received. Until then, VoiceEngine |
| 129 // returns 0 to indicate an error value. |
| 130 if (call_stats.rttMs > 0) { |
| 131 stats.rtt_ms = call_stats.rttMs; |
| 132 } |
| 133 // TODO(solenberg): [was ajm]: Re-enable this metric once we have a reliable |
| 134 // implementation. |
| 135 stats.aec_quality_min = -1; |
133 | 136 |
134 webrtc::CodecInst codec_inst = {0}; | 137 webrtc::CodecInst codec_inst = {0}; |
135 if (codec->GetSendCodec(config_.voe_channel_id, codec_inst) != -1) { | 138 if (codec->GetSendCodec(config_.voe_channel_id, codec_inst) != -1) { |
136 RTC_DCHECK_NE(codec_inst.pltype, -1); | 139 RTC_DCHECK_NE(codec_inst.pltype, -1); |
137 stats.codec_name = codec_inst.plname; | 140 stats.codec_name = codec_inst.plname; |
138 | 141 |
139 // Get data from the last remote RTCP report. | 142 // Get data from the last remote RTCP report. |
140 std::vector<webrtc::ReportBlock> blocks; | 143 std::vector<webrtc::ReportBlock> blocks; |
141 if (rtp->GetRemoteRTCPReportBlocks(config_.voe_channel_id, &blocks) != -1) { | 144 error = rtp->GetRemoteRTCPReportBlocks(config_.voe_channel_id, &blocks); |
142 for (const webrtc::ReportBlock& block : blocks) { | 145 RTC_DCHECK_EQ(0, error); |
143 // Lookup report for send ssrc only. | 146 for (const webrtc::ReportBlock& block : blocks) { |
144 if (block.source_SSRC == stats.local_ssrc) { | 147 // Lookup report for send ssrc only. |
145 stats.packets_lost = block.cumulative_num_packets_lost; | 148 if (block.source_SSRC == stats.local_ssrc) { |
146 stats.fraction_lost = Q8ToFloat(block.fraction_lost); | 149 stats.packets_lost = block.cumulative_num_packets_lost; |
147 stats.ext_seqnum = block.extended_highest_sequence_number; | 150 stats.fraction_lost = Q8ToFloat(block.fraction_lost); |
148 // Convert samples to milliseconds. | 151 stats.ext_seqnum = block.extended_highest_sequence_number; |
149 if (codec_inst.plfreq / 1000 > 0) { | 152 // Convert samples to milliseconds. |
150 stats.jitter_ms = | 153 if (codec_inst.plfreq / 1000 > 0) { |
151 block.interarrival_jitter / (codec_inst.plfreq / 1000); | 154 stats.jitter_ms = |
152 } | 155 block.interarrival_jitter / (codec_inst.plfreq / 1000); |
153 break; | |
154 } | 156 } |
| 157 break; |
155 } | 158 } |
156 } | 159 } |
157 } | 160 } |
158 | 161 |
159 // RTT isn't known until a RTCP report is received. Until then, VoiceEngine | |
160 // returns 0 to indicate an error value. | |
161 if (call_stats.rttMs > 0) { | |
162 stats.rtt_ms = call_stats.rttMs; | |
163 } | |
164 | |
165 // Local speech level. | 162 // Local speech level. |
166 { | 163 { |
167 unsigned int level = 0; | 164 unsigned int level = 0; |
168 if (volume->GetSpeechInputLevelFullRange(level) != -1) { | 165 error = volume->GetSpeechInputLevelFullRange(level); |
169 stats.audio_level = static_cast<int32_t>(level); | 166 RTC_DCHECK_EQ(0, error); |
170 } | 167 stats.audio_level = static_cast<int32_t>(level); |
171 } | 168 } |
172 | 169 |
173 // TODO(ajm): Re-enable this metric once we have a reliable implementation. | |
174 stats.aec_quality_min = -1; | |
175 | |
176 bool echo_metrics_on = false; | 170 bool echo_metrics_on = false; |
177 if (processing->GetEcMetricsStatus(echo_metrics_on) != -1 && | 171 error = processing->GetEcMetricsStatus(echo_metrics_on); |
178 echo_metrics_on) { | 172 RTC_DCHECK_EQ(0, error); |
| 173 if (echo_metrics_on) { |
179 // These can also be negative, but in practice -1 is only used to signal | 174 // 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. | 175 // insufficient data, since the resolution is limited to multiples of 4 ms. |
181 int median = -1; | 176 int median = -1; |
182 int std = -1; | 177 int std = -1; |
183 float dummy = 0.0f; | 178 float dummy = 0.0f; |
184 if (processing->GetEcDelayMetrics(median, std, dummy) != -1) { | 179 error = processing->GetEcDelayMetrics(median, std, dummy); |
185 stats.echo_delay_median_ms = median; | 180 RTC_DCHECK_EQ(0, error); |
186 stats.echo_delay_std_ms = std; | 181 stats.echo_delay_median_ms = median; |
187 } | 182 stats.echo_delay_std_ms = std; |
188 | 183 |
189 // These can take on valid negative values, so use the lowest possible level | 184 // These can take on valid negative values, so use the lowest possible level |
190 // as default rather than -1. | 185 // as default rather than -1. |
191 int erl = -100; | 186 int erl = -100; |
192 int erle = -100; | 187 int erle = -100; |
193 int dummy1 = 0; | 188 int dummy1 = 0; |
194 int dummy2 = 0; | 189 int dummy2 = 0; |
195 if (processing->GetEchoMetrics(erl, erle, dummy1, dummy2) != -1) { | 190 error = processing->GetEchoMetrics(erl, erle, dummy1, dummy2); |
196 stats.echo_return_loss = erl; | 191 RTC_DCHECK_EQ(0, error); |
197 stats.echo_return_loss_enhancement = erle; | 192 stats.echo_return_loss = erl; |
198 } | 193 stats.echo_return_loss_enhancement = erle; |
199 } | 194 } |
200 | 195 |
201 internal::AudioState* audio_state = | 196 internal::AudioState* audio_state = |
202 static_cast<internal::AudioState*>(audio_state_.get()); | 197 static_cast<internal::AudioState*>(audio_state_.get()); |
203 stats.typing_noise_detected = audio_state->typing_noise_detected(); | 198 stats.typing_noise_detected = audio_state->typing_noise_detected(); |
204 | 199 |
205 return stats; | 200 return stats; |
206 } | 201 } |
207 | 202 |
208 const webrtc::AudioSendStream::Config& AudioSendStream::config() const { | 203 const webrtc::AudioSendStream::Config& AudioSendStream::config() const { |
209 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | 204 RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
210 return config_; | 205 return config_; |
211 } | 206 } |
212 | 207 |
213 VoiceEngine* AudioSendStream::voice_engine() const { | 208 VoiceEngine* AudioSendStream::voice_engine() const { |
214 internal::AudioState* audio_state = | 209 internal::AudioState* audio_state = |
215 static_cast<internal::AudioState*>(audio_state_.get()); | 210 static_cast<internal::AudioState*>(audio_state_.get()); |
216 VoiceEngine* voice_engine = audio_state->voice_engine(); | 211 VoiceEngine* voice_engine = audio_state->voice_engine(); |
217 RTC_DCHECK(voice_engine); | 212 RTC_DCHECK(voice_engine); |
218 return voice_engine; | 213 return voice_engine; |
219 } | 214 } |
220 } // namespace internal | 215 } // namespace internal |
221 } // namespace webrtc | 216 } // namespace webrtc |
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