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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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116 if (extension.uri == RtpExtension::kAudioLevelUri) { | 116 if (extension.uri == RtpExtension::kAudioLevelUri) { |
117 channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id); | 117 channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id); |
118 bool registered = rtp_header_parser_->RegisterRtpHeaderExtension( | 118 bool registered = rtp_header_parser_->RegisterRtpHeaderExtension( |
119 kRtpExtensionAudioLevel, extension.id); | 119 kRtpExtensionAudioLevel, extension.id); |
120 RTC_DCHECK(registered); | 120 RTC_DCHECK(registered); |
121 } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) { | 121 } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) { |
122 channel_proxy_->EnableReceiveTransportSequenceNumber(extension.id); | 122 channel_proxy_->EnableReceiveTransportSequenceNumber(extension.id); |
123 bool registered = rtp_header_parser_->RegisterRtpHeaderExtension( | 123 bool registered = rtp_header_parser_->RegisterRtpHeaderExtension( |
124 kRtpExtensionTransportSequenceNumber, extension.id); | 124 kRtpExtensionTransportSequenceNumber, extension.id); |
125 RTC_DCHECK(registered); | 125 RTC_DCHECK(registered); |
| 126 } else if (extension.uri == RtpExtension::kAbsSendTimeUri) { |
| 127 LOG(LS_WARNING) << RtpExtension::kAbsSendTimeUri |
| 128 << " is no longer supported for audio."; |
126 } else { | 129 } else { |
127 RTC_NOTREACHED() << "Unsupported RTP extension."; | 130 RTC_NOTREACHED() << "Unsupported RTP extension."; |
128 } | 131 } |
129 } | 132 } |
130 // Configure bandwidth estimation. | 133 // Configure bandwidth estimation. |
131 channel_proxy_->RegisterReceiverCongestionControlObjects( | 134 channel_proxy_->RegisterReceiverCongestionControlObjects( |
132 congestion_controller->packet_router()); | 135 congestion_controller->packet_router()); |
133 if (UseSendSideBwe(config)) { | 136 if (UseSendSideBwe(config)) { |
134 remote_bitrate_estimator_ = | 137 remote_bitrate_estimator_ = |
135 congestion_controller->GetRemoteBitrateEstimator(true); | 138 congestion_controller->GetRemoteBitrateEstimator(true); |
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283 | 286 |
284 VoiceEngine* AudioReceiveStream::voice_engine() const { | 287 VoiceEngine* AudioReceiveStream::voice_engine() const { |
285 internal::AudioState* audio_state = | 288 internal::AudioState* audio_state = |
286 static_cast<internal::AudioState*>(audio_state_.get()); | 289 static_cast<internal::AudioState*>(audio_state_.get()); |
287 VoiceEngine* voice_engine = audio_state->voice_engine(); | 290 VoiceEngine* voice_engine = audio_state->voice_engine(); |
288 RTC_DCHECK(voice_engine); | 291 RTC_DCHECK(voice_engine); |
289 return voice_engine; | 292 return voice_engine; |
290 } | 293 } |
291 } // namespace internal | 294 } // namespace internal |
292 } // namespace webrtc | 295 } // namespace webrtc |
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