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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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29 #include "webrtc/modules/audio_device/include/audio_device.h" | 29 #include "webrtc/modules/audio_device/include/audio_device.h" |
30 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 30 #include "webrtc/modules/audio_processing/include/audio_processing.h" |
31 #include "webrtc/modules/include/module_common_types.h" | 31 #include "webrtc/modules/include/module_common_types.h" |
32 #include "webrtc/modules/pacing/packet_router.h" | 32 #include "webrtc/modules/pacing/packet_router.h" |
33 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" | 33 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" |
34 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" | 34 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" |
35 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" | 35 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" |
36 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h" | 36 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h" |
37 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" | 37 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" |
38 #include "webrtc/modules/utility/include/process_thread.h" | 38 #include "webrtc/modules/utility/include/process_thread.h" |
| 39 #include "webrtc/system_wrappers/include/field_trial.h" |
39 #include "webrtc/system_wrappers/include/trace.h" | 40 #include "webrtc/system_wrappers/include/trace.h" |
40 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" | 41 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" |
41 #include "webrtc/voice_engine/output_mixer.h" | 42 #include "webrtc/voice_engine/output_mixer.h" |
42 #include "webrtc/voice_engine/statistics.h" | 43 #include "webrtc/voice_engine/statistics.h" |
43 #include "webrtc/voice_engine/utility.h" | 44 #include "webrtc/voice_engine/utility.h" |
44 | 45 |
45 namespace webrtc { | 46 namespace webrtc { |
46 namespace voe { | 47 namespace voe { |
47 | 48 |
48 namespace { | 49 namespace { |
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898 restored_packet_in_use_(false), | 899 restored_packet_in_use_(false), |
899 rtcp_observer_(new VoERtcpObserver(this)), | 900 rtcp_observer_(new VoERtcpObserver(this)), |
900 associate_send_channel_(ChannelOwner(nullptr)), | 901 associate_send_channel_(ChannelOwner(nullptr)), |
901 pacing_enabled_(config.enable_voice_pacing), | 902 pacing_enabled_(config.enable_voice_pacing), |
902 feedback_observer_proxy_(new TransportFeedbackProxy()), | 903 feedback_observer_proxy_(new TransportFeedbackProxy()), |
903 seq_num_allocator_proxy_(new TransportSequenceNumberProxy()), | 904 seq_num_allocator_proxy_(new TransportSequenceNumberProxy()), |
904 rtp_packet_sender_proxy_(new RtpPacketSenderProxy()), | 905 rtp_packet_sender_proxy_(new RtpPacketSenderProxy()), |
905 retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(), | 906 retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(), |
906 kMaxRetransmissionWindowMs)), | 907 kMaxRetransmissionWindowMs)), |
907 decoder_factory_(config.acm_config.decoder_factory), | 908 decoder_factory_(config.acm_config.decoder_factory), |
908 // TODO(elad.alon): Subsequent CL experiments with PLR source. | 909 use_twcc_plr_for_ana_( |
909 use_twcc_plr_for_ana_(false) { | 910 webrtc::field_trial::FindFullName("UseTwccPlrForAna") == "Enabled") { |
910 WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId), | 911 WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId), |
911 "Channel::Channel() - ctor"); | 912 "Channel::Channel() - ctor"); |
912 AudioCodingModule::Config acm_config(config.acm_config); | 913 AudioCodingModule::Config acm_config(config.acm_config); |
913 acm_config.id = VoEModuleId(instanceId, channelId); | 914 acm_config.id = VoEModuleId(instanceId, channelId); |
914 acm_config.neteq_config.enable_muted_state = true; | 915 acm_config.neteq_config.enable_muted_state = true; |
915 audio_coding_.reset(AudioCodingModule::Create(acm_config)); | 916 audio_coding_.reset(AudioCodingModule::Create(acm_config)); |
916 | 917 |
917 _outputAudioLevel.Clear(); | 918 _outputAudioLevel.Clear(); |
918 | 919 |
919 RtpRtcp::Configuration configuration; | 920 RtpRtcp::Configuration configuration; |
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3061 int64_t min_rtt = 0; | 3062 int64_t min_rtt = 0; |
3062 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != | 3063 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != |
3063 0) { | 3064 0) { |
3064 return 0; | 3065 return 0; |
3065 } | 3066 } |
3066 return rtt; | 3067 return rtt; |
3067 } | 3068 } |
3068 | 3069 |
3069 } // namespace voe | 3070 } // namespace voe |
3070 } // namespace webrtc | 3071 } // namespace webrtc |
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