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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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802 rtp_header_parser_(RtpHeaderParser::Create()), | 802 rtp_header_parser_(RtpHeaderParser::Create()), |
803 rtp_payload_registry_( | 803 rtp_payload_registry_( |
804 new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))), | 804 new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))), |
805 rtp_receive_statistics_( | 805 rtp_receive_statistics_( |
806 ReceiveStatistics::Create(Clock::GetRealTimeClock())), | 806 ReceiveStatistics::Create(Clock::GetRealTimeClock())), |
807 rtp_receiver_( | 807 rtp_receiver_( |
808 RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(), | 808 RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(), |
809 this, | 809 this, |
810 this, | 810 this, |
811 rtp_payload_registry_.get())), | 811 rtp_payload_registry_.get())), |
812 telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()), | |
813 _outputAudioLevel(), | 812 _outputAudioLevel(), |
814 _externalTransport(false), | 813 _externalTransport(false), |
815 // Avoid conflict with other channels by adding 1024 - 1026, | 814 // Avoid conflict with other channels by adding 1024 - 1026, |
816 // won't use as much as 1024 channels. | 815 // won't use as much as 1024 channels. |
817 _inputFilePlayerId(VoEModuleId(instanceId, channelId) + 1024), | 816 _inputFilePlayerId(VoEModuleId(instanceId, channelId) + 1024), |
818 _outputFilePlayerId(VoEModuleId(instanceId, channelId) + 1025), | 817 _outputFilePlayerId(VoEModuleId(instanceId, channelId) + 1025), |
819 _outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026), | 818 _outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026), |
820 _outputFileRecording(false), | 819 _outputFileRecording(false), |
821 _outputExternalMedia(false), | 820 _outputExternalMedia(false), |
822 _inputExternalMediaCallbackPtr(NULL), | 821 _inputExternalMediaCallbackPtr(NULL), |
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972 return -1; | 971 return -1; |
973 } | 972 } |
974 | 973 |
975 // --- RTP/RTCP module initialization | 974 // --- RTP/RTCP module initialization |
976 | 975 |
977 // Ensure that RTCP is enabled by default for the created channel. | 976 // Ensure that RTCP is enabled by default for the created channel. |
978 // Note that, the module will keep generating RTCP until it is explicitly | 977 // Note that, the module will keep generating RTCP until it is explicitly |
979 // disabled by the user. | 978 // disabled by the user. |
980 // After StopListen (when no sockets exists), RTCP packets will no longer | 979 // After StopListen (when no sockets exists), RTCP packets will no longer |
981 // be transmitted since the Transport object will then be invalid. | 980 // be transmitted since the Transport object will then be invalid. |
982 telephone_event_handler_->SetTelephoneEventForwardToDecoder(true); | |
983 // RTCP is enabled by default. | 981 // RTCP is enabled by default. |
984 _rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound); | 982 _rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound); |
985 // --- Register all permanent callbacks | 983 // --- Register all permanent callbacks |
986 const bool fail = (audio_coding_->RegisterTransportCallback(this) == -1) || | 984 const bool fail = (audio_coding_->RegisterTransportCallback(this) == -1) || |
987 (audio_coding_->RegisterVADCallback(this) == -1); | 985 (audio_coding_->RegisterVADCallback(this) == -1); |
988 | 986 |
989 if (fail) { | 987 if (fail) { |
990 _engineStatisticsPtr->SetLastError( | 988 _engineStatisticsPtr->SetLastError( |
991 VE_CANNOT_INIT_CHANNEL, kTraceError, | 989 VE_CANNOT_INIT_CHANNEL, kTraceError, |
992 "Channel::Init() callbacks not registered"); | 990 "Channel::Init() callbacks not registered"); |
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3205 int64_t min_rtt = 0; | 3203 int64_t min_rtt = 0; |
3206 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != | 3204 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != |
3207 0) { | 3205 0) { |
3208 return 0; | 3206 return 0; |
3209 } | 3207 } |
3210 return rtt; | 3208 return rtt; |
3211 } | 3209 } |
3212 | 3210 |
3213 } // namespace voe | 3211 } // namespace voe |
3214 } // namespace webrtc | 3212 } // namespace webrtc |
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