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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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915 RtpRtcp::Configuration configuration; | 915 RtpRtcp::Configuration configuration; |
916 configuration.audio = true; | 916 configuration.audio = true; |
917 configuration.outgoing_transport = this; | 917 configuration.outgoing_transport = this; |
918 configuration.audio_messages = this; | 918 configuration.audio_messages = this; |
919 configuration.receive_statistics = rtp_receive_statistics_.get(); | 919 configuration.receive_statistics = rtp_receive_statistics_.get(); |
920 configuration.bandwidth_callback = rtcp_observer_.get(); | 920 configuration.bandwidth_callback = rtcp_observer_.get(); |
921 configuration.paced_sender = rtp_packet_sender_proxy_.get(); | 921 configuration.paced_sender = rtp_packet_sender_proxy_.get(); |
922 configuration.transport_sequence_number_allocator = | 922 configuration.transport_sequence_number_allocator = |
923 seq_num_allocator_proxy_.get(); | 923 seq_num_allocator_proxy_.get(); |
924 configuration.transport_feedback_callback = feedback_observer_proxy_.get(); | 924 configuration.transport_feedback_callback = feedback_observer_proxy_.get(); |
| 925 configuration.event_log = event_log; |
925 | 926 |
926 _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration)); | 927 _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration)); |
927 | 928 |
928 statistics_proxy_.reset(new StatisticsProxy(_rtpRtcpModule->SSRC())); | 929 statistics_proxy_.reset(new StatisticsProxy(_rtpRtcpModule->SSRC())); |
929 rtp_receive_statistics_->RegisterRtcpStatisticsCallback( | 930 rtp_receive_statistics_->RegisterRtcpStatisticsCallback( |
930 statistics_proxy_.get()); | 931 statistics_proxy_.get()); |
931 | 932 |
932 Config audioproc_config; | 933 Config audioproc_config; |
933 audioproc_config.Set<ExperimentalAgc>(new ExperimentalAgc(false)); | 934 audioproc_config.Set<ExperimentalAgc>(new ExperimentalAgc(false)); |
934 rx_audioproc_.reset(AudioProcessing::Create(audioproc_config)); | 935 rx_audioproc_.reset(AudioProcessing::Create(audioproc_config)); |
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4109 int64_t min_rtt = 0; | 4110 int64_t min_rtt = 0; |
4110 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) | 4111 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) |
4111 != 0) { | 4112 != 0) { |
4112 return 0; | 4113 return 0; |
4113 } | 4114 } |
4114 return rtt; | 4115 return rtt; |
4115 } | 4116 } |
4116 | 4117 |
4117 } // namespace voe | 4118 } // namespace voe |
4118 } // namespace webrtc | 4119 } // namespace webrtc |
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