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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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827 " shutdown"); | 827 " shutdown"); |
828 } | 828 } |
829 | 829 |
830 Channel::Channel(int32_t channelId, | 830 Channel::Channel(int32_t channelId, |
831 uint32_t instanceId, | 831 uint32_t instanceId, |
832 const VoEBase::ChannelConfig& config) | 832 const VoEBase::ChannelConfig& config) |
833 : _instanceId(instanceId), | 833 : _instanceId(instanceId), |
834 _channelId(channelId), | 834 _channelId(channelId), |
835 event_log_proxy_(new RtcEventLogProxy()), | 835 event_log_proxy_(new RtcEventLogProxy()), |
836 rtp_header_parser_(RtpHeaderParser::Create()), | 836 rtp_header_parser_(RtpHeaderParser::Create()), |
837 rtp_payload_registry_(new RTPPayloadRegistry()), | 837 rtp_payload_registry_( |
| 838 new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))), |
838 rtp_receive_statistics_( | 839 rtp_receive_statistics_( |
839 ReceiveStatistics::Create(Clock::GetRealTimeClock())), | 840 ReceiveStatistics::Create(Clock::GetRealTimeClock())), |
840 rtp_receiver_( | 841 rtp_receiver_( |
841 RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(), | 842 RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(), |
842 this, | 843 this, |
843 this, | 844 this, |
844 rtp_payload_registry_.get())), | 845 rtp_payload_registry_.get())), |
845 telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()), | 846 telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()), |
846 _outputAudioLevel(), | 847 _outputAudioLevel(), |
847 _externalTransport(false), | 848 _externalTransport(false), |
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3227 int64_t min_rtt = 0; | 3228 int64_t min_rtt = 0; |
3228 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != | 3229 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != |
3229 0) { | 3230 0) { |
3230 return 0; | 3231 return 0; |
3231 } | 3232 } |
3232 return rtt; | 3233 return rtt; |
3233 } | 3234 } |
3234 | 3235 |
3235 } // namespace voe | 3236 } // namespace voe |
3236 } // namespace webrtc | 3237 } // namespace webrtc |
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