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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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117 if (extension.uri == RtpExtension::kAudioLevelUri) { | 117 if (extension.uri == RtpExtension::kAudioLevelUri) { |
118 channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id); | 118 channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id); |
119 bool registered = rtp_header_parser_->RegisterRtpHeaderExtension( | 119 bool registered = rtp_header_parser_->RegisterRtpHeaderExtension( |
120 kRtpExtensionAudioLevel, extension.id); | 120 kRtpExtensionAudioLevel, extension.id); |
121 RTC_DCHECK(registered); | 121 RTC_DCHECK(registered); |
122 } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) { | 122 } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) { |
123 channel_proxy_->EnableReceiveTransportSequenceNumber(extension.id); | 123 channel_proxy_->EnableReceiveTransportSequenceNumber(extension.id); |
124 bool registered = rtp_header_parser_->RegisterRtpHeaderExtension( | 124 bool registered = rtp_header_parser_->RegisterRtpHeaderExtension( |
125 kRtpExtensionTransportSequenceNumber, extension.id); | 125 kRtpExtensionTransportSequenceNumber, extension.id); |
126 RTC_DCHECK(registered); | 126 RTC_DCHECK(registered); |
127 } else if (extension.uri == RtpExtension::kAbsSendTimeUri) { | |
128 LOG(LS_WARNING) << RtpExtension::kAbsSendTimeUri | |
129 << " is no longer supported for audio."; | |
130 } else { | 127 } else { |
131 RTC_NOTREACHED() << "Unsupported RTP extension."; | 128 RTC_NOTREACHED() << "Unsupported RTP extension."; |
132 } | 129 } |
133 } | 130 } |
134 // Configure bandwidth estimation. | 131 // Configure bandwidth estimation. |
135 channel_proxy_->RegisterReceiverCongestionControlObjects( | 132 channel_proxy_->RegisterReceiverCongestionControlObjects( |
136 congestion_controller->packet_router()); | 133 congestion_controller->packet_router()); |
137 if (UseSendSideBwe(config)) { | 134 if (UseSendSideBwe(config)) { |
138 remote_bitrate_estimator_ = | 135 remote_bitrate_estimator_ = |
139 congestion_controller->GetRemoteBitrateEstimator(true); | 136 congestion_controller->GetRemoteBitrateEstimator(true); |
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300 | 297 |
301 VoiceEngine* AudioReceiveStream::voice_engine() const { | 298 VoiceEngine* AudioReceiveStream::voice_engine() const { |
302 internal::AudioState* audio_state = | 299 internal::AudioState* audio_state = |
303 static_cast<internal::AudioState*>(audio_state_.get()); | 300 static_cast<internal::AudioState*>(audio_state_.get()); |
304 VoiceEngine* voice_engine = audio_state->voice_engine(); | 301 VoiceEngine* voice_engine = audio_state->voice_engine(); |
305 RTC_DCHECK(voice_engine); | 302 RTC_DCHECK(voice_engine); |
306 return voice_engine; | 303 return voice_engine; |
307 } | 304 } |
308 } // namespace internal | 305 } // namespace internal |
309 } // namespace webrtc | 306 } // namespace webrtc |
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