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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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| 933 AudioCodingModule::Config acm_config(config.acm_config); | 933 AudioCodingModule::Config acm_config(config.acm_config); |
| 934 acm_config.id = VoEModuleId(instanceId, channelId); | 934 acm_config.id = VoEModuleId(instanceId, channelId); |
| 935 acm_config.neteq_config.enable_muted_state = true; | 935 acm_config.neteq_config.enable_muted_state = true; |
| 936 audio_coding_.reset(AudioCodingModule::Create(acm_config)); | 936 audio_coding_.reset(AudioCodingModule::Create(acm_config)); |
| 937 | 937 |
| 938 _outputAudioLevel.Clear(); | 938 _outputAudioLevel.Clear(); |
| 939 | 939 |
| 940 RtpRtcp::Configuration configuration; | 940 RtpRtcp::Configuration configuration; |
| 941 configuration.audio = true; | 941 configuration.audio = true; |
| 942 configuration.outgoing_transport = this; | 942 configuration.outgoing_transport = this; |
| 943 configuration.overhead_observer = this; |
| 943 configuration.receive_statistics = rtp_receive_statistics_.get(); | 944 configuration.receive_statistics = rtp_receive_statistics_.get(); |
| 944 configuration.bandwidth_callback = rtcp_observer_.get(); | 945 configuration.bandwidth_callback = rtcp_observer_.get(); |
| 945 if (pacing_enabled_) { | 946 if (pacing_enabled_) { |
| 946 configuration.paced_sender = rtp_packet_sender_proxy_.get(); | 947 configuration.paced_sender = rtp_packet_sender_proxy_.get(); |
| 947 configuration.transport_sequence_number_allocator = | 948 configuration.transport_sequence_number_allocator = |
| 948 seq_num_allocator_proxy_.get(); | 949 seq_num_allocator_proxy_.get(); |
| 949 configuration.transport_feedback_callback = feedback_observer_proxy_.get(); | 950 configuration.transport_feedback_callback = feedback_observer_proxy_.get(); |
| 950 } | 951 } |
| 951 configuration.event_log = &(*event_log_proxy_); | 952 configuration.event_log = &(*event_log_proxy_); |
| 952 configuration.rtt_stats = &(*rtcp_rtt_stats_proxy_); | 953 configuration.rtt_stats = &(*rtcp_rtt_stats_proxy_); |
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| 2874 } | 2875 } |
| 2875 | 2876 |
| 2876 void Channel::SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) { | 2877 void Channel::SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) { |
| 2877 rtcp_rtt_stats_proxy_->SetRtcpRttStats(rtcp_rtt_stats); | 2878 rtcp_rtt_stats_proxy_->SetRtcpRttStats(rtcp_rtt_stats); |
| 2878 } | 2879 } |
| 2879 | 2880 |
| 2880 void Channel::SetTransportOverhead(int transport_overhead_per_packet) { | 2881 void Channel::SetTransportOverhead(int transport_overhead_per_packet) { |
| 2881 _rtpRtcpModule->SetTransportOverhead(transport_overhead_per_packet); | 2882 _rtpRtcpModule->SetTransportOverhead(transport_overhead_per_packet); |
| 2882 } | 2883 } |
| 2883 | 2884 |
| 2885 void Channel::OnOverheadChanged(size_t overhead_bytes_per_packet) { |
| 2886 audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
| 2887 if (*encoder) { |
| 2888 (*encoder)->OnReceivedOverhead(overhead_bytes_per_packet); |
| 2889 } |
| 2890 }); |
| 2891 } |
| 2892 |
| 2884 int Channel::RegisterExternalMediaProcessing(ProcessingTypes type, | 2893 int Channel::RegisterExternalMediaProcessing(ProcessingTypes type, |
| 2885 VoEMediaProcess& processObject) { | 2894 VoEMediaProcess& processObject) { |
| 2886 WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), | 2895 WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 2887 "Channel::RegisterExternalMediaProcessing()"); | 2896 "Channel::RegisterExternalMediaProcessing()"); |
| 2888 | 2897 |
| 2889 rtc::CritScope cs(&_callbackCritSect); | 2898 rtc::CritScope cs(&_callbackCritSect); |
| 2890 | 2899 |
| 2891 if (kPlaybackPerChannel == type) { | 2900 if (kPlaybackPerChannel == type) { |
| 2892 if (_outputExternalMediaCallbackPtr) { | 2901 if (_outputExternalMediaCallbackPtr) { |
| 2893 _engineStatisticsPtr->SetLastError( | 2902 _engineStatisticsPtr->SetLastError( |
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| 3279 int64_t min_rtt = 0; | 3288 int64_t min_rtt = 0; |
| 3280 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != | 3289 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != |
| 3281 0) { | 3290 0) { |
| 3282 return 0; | 3291 return 0; |
| 3283 } | 3292 } |
| 3284 return rtt; | 3293 return rtt; |
| 3285 } | 3294 } |
| 3286 | 3295 |
| 3287 } // namespace voe | 3296 } // namespace voe |
| 3288 } // namespace webrtc | 3297 } // namespace webrtc |
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