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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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