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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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28 #include "webrtc/modules/audio_device/include/audio_device.h" | 28 #include "webrtc/modules/audio_device/include/audio_device.h" |
29 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 29 #include "webrtc/modules/audio_processing/include/audio_processing.h" |
30 #include "webrtc/modules/include/module_common_types.h" | 30 #include "webrtc/modules/include/module_common_types.h" |
31 #include "webrtc/modules/pacing/packet_router.h" | 31 #include "webrtc/modules/pacing/packet_router.h" |
32 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" | 32 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" |
33 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" | 33 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" |
34 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" | 34 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" |
35 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h" | 35 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h" |
36 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" | 36 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" |
37 #include "webrtc/modules/utility/include/process_thread.h" | 37 #include "webrtc/modules/utility/include/process_thread.h" |
| 38 #include "webrtc/system_wrappers/include/field_trial.h" |
38 #include "webrtc/system_wrappers/include/trace.h" | 39 #include "webrtc/system_wrappers/include/trace.h" |
39 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" | 40 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" |
40 #include "webrtc/voice_engine/output_mixer.h" | 41 #include "webrtc/voice_engine/output_mixer.h" |
41 #include "webrtc/voice_engine/statistics.h" | 42 #include "webrtc/voice_engine/statistics.h" |
42 #include "webrtc/voice_engine/utility.h" | 43 #include "webrtc/voice_engine/utility.h" |
43 | 44 |
44 namespace webrtc { | 45 namespace webrtc { |
45 namespace voe { | 46 namespace voe { |
46 | 47 |
47 namespace { | 48 namespace { |
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897 restored_packet_in_use_(false), | 898 restored_packet_in_use_(false), |
898 rtcp_observer_(new VoERtcpObserver(this)), | 899 rtcp_observer_(new VoERtcpObserver(this)), |
899 associate_send_channel_(ChannelOwner(nullptr)), | 900 associate_send_channel_(ChannelOwner(nullptr)), |
900 pacing_enabled_(config.enable_voice_pacing), | 901 pacing_enabled_(config.enable_voice_pacing), |
901 feedback_observer_proxy_(new TransportFeedbackProxy()), | 902 feedback_observer_proxy_(new TransportFeedbackProxy()), |
902 seq_num_allocator_proxy_(new TransportSequenceNumberProxy()), | 903 seq_num_allocator_proxy_(new TransportSequenceNumberProxy()), |
903 rtp_packet_sender_proxy_(new RtpPacketSenderProxy()), | 904 rtp_packet_sender_proxy_(new RtpPacketSenderProxy()), |
904 retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(), | 905 retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(), |
905 kMaxRetransmissionWindowMs)), | 906 kMaxRetransmissionWindowMs)), |
906 decoder_factory_(config.acm_config.decoder_factory), | 907 decoder_factory_(config.acm_config.decoder_factory), |
907 // TODO(elad.alon): Subsequent CL experiments with PLR source. | 908 use_twcc_plr_for_ana_( |
908 use_twcc_plr_for_ana_(false) { | 909 webrtc::field_trial::FindFullName("UseTwccPlrForAna") == "Enabled") { |
909 WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId), | 910 WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId), |
910 "Channel::Channel() - ctor"); | 911 "Channel::Channel() - ctor"); |
911 AudioCodingModule::Config acm_config(config.acm_config); | 912 AudioCodingModule::Config acm_config(config.acm_config); |
912 acm_config.id = VoEModuleId(instanceId, channelId); | 913 acm_config.id = VoEModuleId(instanceId, channelId); |
913 acm_config.neteq_config.enable_muted_state = true; | 914 acm_config.neteq_config.enable_muted_state = true; |
914 audio_coding_.reset(AudioCodingModule::Create(acm_config)); | 915 audio_coding_.reset(AudioCodingModule::Create(acm_config)); |
915 | 916 |
916 _outputAudioLevel.Clear(); | 917 _outputAudioLevel.Clear(); |
917 | 918 |
918 RtpRtcp::Configuration configuration; | 919 RtpRtcp::Configuration configuration; |
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3033 int64_t min_rtt = 0; | 3034 int64_t min_rtt = 0; |
3034 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != | 3035 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != |
3035 0) { | 3036 0) { |
3036 return 0; | 3037 return 0; |
3037 } | 3038 } |
3038 return rtt; | 3039 return rtt; |
3039 } | 3040 } |
3040 | 3041 |
3041 } // namespace voe | 3042 } // namespace voe |
3042 } // namespace webrtc | 3043 } // namespace webrtc |
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