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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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