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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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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887 _outputSpeechType(AudioFrame::kNormalSpeech), | 887 _outputSpeechType(AudioFrame::kNormalSpeech), |
888 restored_packet_in_use_(false), | 888 restored_packet_in_use_(false), |
889 rtcp_observer_(new VoERtcpObserver(this)), | 889 rtcp_observer_(new VoERtcpObserver(this)), |
890 associate_send_channel_(ChannelOwner(nullptr)), | 890 associate_send_channel_(ChannelOwner(nullptr)), |
891 pacing_enabled_(config.enable_voice_pacing), | 891 pacing_enabled_(config.enable_voice_pacing), |
892 feedback_observer_proxy_(new TransportFeedbackProxy()), | 892 feedback_observer_proxy_(new TransportFeedbackProxy()), |
893 seq_num_allocator_proxy_(new TransportSequenceNumberProxy()), | 893 seq_num_allocator_proxy_(new TransportSequenceNumberProxy()), |
894 rtp_packet_sender_proxy_(new RtpPacketSenderProxy()), | 894 rtp_packet_sender_proxy_(new RtpPacketSenderProxy()), |
895 retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(), | 895 retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(), |
896 kMaxRetransmissionWindowMs)), | 896 kMaxRetransmissionWindowMs)), |
897 decoder_factory_(config.acm_config.decoder_factory) { | 897 decoder_factory_(config.acm_config.decoder_factory), |
| 898 // Bitrate smoother can be initialized with arbitrary time constant |
| 899 // (0 used here). The actual time constant will be set in SetBitRate. |
| 900 bitrate_smoother_(0, Clock::GetRealTimeClock()) { |
898 WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId), | 901 WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId), |
899 "Channel::Channel() - ctor"); | 902 "Channel::Channel() - ctor"); |
900 AudioCodingModule::Config acm_config(config.acm_config); | 903 AudioCodingModule::Config acm_config(config.acm_config); |
901 acm_config.id = VoEModuleId(instanceId, channelId); | 904 acm_config.id = VoEModuleId(instanceId, channelId); |
902 acm_config.neteq_config.enable_muted_state = true; | 905 acm_config.neteq_config.enable_muted_state = true; |
903 audio_coding_.reset(AudioCodingModule::Create(acm_config)); | 906 audio_coding_.reset(AudioCodingModule::Create(acm_config)); |
904 | 907 |
905 _outputAudioLevel.Clear(); | 908 _outputAudioLevel.Clear(); |
906 | 909 |
907 RtpRtcp::Configuration configuration; | 910 RtpRtcp::Configuration configuration; |
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1299 } | 1302 } |
1300 | 1303 |
1301 void Channel::SetBitRate(int bitrate_bps) { | 1304 void Channel::SetBitRate(int bitrate_bps) { |
1302 WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), | 1305 WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
1303 "Channel::SetBitRate(bitrate_bps=%d)", bitrate_bps); | 1306 "Channel::SetBitRate(bitrate_bps=%d)", bitrate_bps); |
1304 audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { | 1307 audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
1305 if (*encoder) | 1308 if (*encoder) |
1306 (*encoder)->OnReceivedTargetAudioBitrate(bitrate_bps); | 1309 (*encoder)->OnReceivedTargetAudioBitrate(bitrate_bps); |
1307 }); | 1310 }); |
1308 retransmission_rate_limiter_->SetMaxRate(bitrate_bps); | 1311 retransmission_rate_limiter_->SetMaxRate(bitrate_bps); |
| 1312 |
| 1313 // We give smoothed bitrate allocation to audio network adaptor as |
| 1314 // the uplink bandwidth. |
| 1315 // TODO(michaelt) : Remove kDefaultBitrateSmoothingTimeConstantMs as soon as |
| 1316 // we pass the probing interval to this function. |
| 1317 constexpr int64_t kDefaultBitrateSmoothingTimeConstantMs = 20000; |
| 1318 bitrate_smoother_.SetTimeConstantMs(kDefaultBitrateSmoothingTimeConstantMs); |
| 1319 bitrate_smoother_.AddSample(bitrate_bps); |
| 1320 audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
| 1321 if (*encoder) { |
| 1322 (*encoder)->OnReceivedUplinkBandwidth( |
| 1323 static_cast<int>(*bitrate_smoother_.GetAverage())); |
| 1324 } |
| 1325 }); |
1309 } | 1326 } |
1310 | 1327 |
1311 void Channel::OnIncomingFractionLoss(int fraction_lost) { | 1328 void Channel::OnIncomingFractionLoss(int fraction_lost) { |
1312 audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { | 1329 audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
1313 if (*encoder) | 1330 if (*encoder) |
1314 (*encoder)->OnReceivedUplinkPacketLossFraction(fraction_lost / 255.0f); | 1331 (*encoder)->OnReceivedUplinkPacketLossFraction(fraction_lost / 255.0f); |
1315 }); | 1332 }); |
1316 } | 1333 } |
1317 | 1334 |
1318 int32_t Channel::SetVADStatus(bool enableVAD, | 1335 int32_t Channel::SetVADStatus(bool enableVAD, |
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3227 int64_t min_rtt = 0; | 3244 int64_t min_rtt = 0; |
3228 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != | 3245 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != |
3229 0) { | 3246 0) { |
3230 return 0; | 3247 return 0; |
3231 } | 3248 } |
3232 return rtt; | 3249 return rtt; |
3233 } | 3250 } |
3234 | 3251 |
3235 } // namespace voe | 3252 } // namespace voe |
3236 } // namespace webrtc | 3253 } // namespace webrtc |
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