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test
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De-flake NonSenderRttStats and make it faster to run on average.
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2023-03-10 13:25:34 +00:00 |
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utility
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Remove dependency on rtc_base_approved from most targets
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2022-04-25 12:15:30 +00:00 |
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voip
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Implement support for Chrome task origin tracing. #3.5/4
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2023-03-01 11:11:37 +00:00 |
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audio_level.cc
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Migrate audio/ to use webrtc::Mutex
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2020-07-06 14:21:38 +00:00 |
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audio_level.h
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Migrate audio/ to use webrtc::Mutex
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2020-07-06 14:21:38 +00:00 |
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audio_receive_stream.cc
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Remove rtp header extension from config of Call audio and video receivers
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2023-01-31 11:58:43 +00:00 |
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audio_receive_stream.h
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Remove rtp header extension from config of Call audio and video receivers
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2023-01-31 11:58:43 +00:00 |
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audio_receive_stream_unittest.cc
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Remove rtp header extension from config of Call audio and video receivers
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2023-01-31 11:58:43 +00:00 |
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audio_send_stream.cc
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Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead
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2022-11-30 20:19:36 +00:00 |
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audio_send_stream.h
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Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead
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2022-11-30 20:19:36 +00:00 |
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audio_send_stream_tests.cc
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CallTest: migrate timeouts to TimeDelta.
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2022-08-16 12:06:54 +00:00 |
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audio_send_stream_unittest.cc
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pc: Add asynchronous RtpSender::SetParameters() call
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2022-11-15 15:31:40 +00:00 |
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audio_state.cc
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Rewrite AudioState null poller to use TaskQueueBase interface
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2022-08-16 13:16:24 +00:00 |
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audio_state.h
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Rewrite AudioState null poller to use TaskQueueBase interface
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2022-08-16 13:16:24 +00:00 |
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audio_state_unittest.cc
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Implement support for Chrome task origin tracing. #3.5/4
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2023-03-01 11:11:37 +00:00 |
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audio_transport_impl.cc
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Make capture timestamp optional in ADM.
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2023-01-23 17:29:06 +00:00 |
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audio_transport_impl.h
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Make capture timestamp optional in ADM.
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2023-01-23 17:29:06 +00:00 |
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BUILD.gn
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De-flake NonSenderRttStats and make it faster to run on average.
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2023-03-10 13:25:34 +00:00 |
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channel_receive.cc
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Delete RtpRtcpInterface::RemoteNtp as redundant to GetSenderReportStats
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2023-02-28 13:55:27 +00:00 |
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channel_receive.h
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stats: use uint64_t for RTCSentRtpStreamStats.packetsSent
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2023-03-16 06:46:19 +00:00 |
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channel_receive_frame_transformer_delegate.cc
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Add GetContributionSources to TransformableIncomingAudioFrame
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2022-10-11 12:52:21 +00:00 |
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channel_receive_frame_transformer_delegate.h
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Use backticks not vertical bars to denote variables in comments for /audio
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2021-07-27 15:36:40 +00:00 |
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channel_receive_frame_transformer_delegate_unittest.cc
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Move rtc::make_ref_counted to api/
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2022-06-15 09:47:38 +00:00 |
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channel_receive_unittest.cc
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Create unit test for the population of capture_start_ntp_time
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2023-02-06 14:00:39 +00:00 |
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channel_send.cc
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Break apart AudioCodingModule and AcmReceiver
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2023-02-01 16:09:26 +00:00 |
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channel_send.h
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[Stats] Expose totalPacketSendDelay for audio as well.
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2022-10-27 10:33:16 +00:00 |
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channel_send_frame_transformer_delegate.cc
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Add a clone method to the audio frame transformer API.
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2023-03-06 08:22:25 +00:00 |
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channel_send_frame_transformer_delegate.h
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Add a clone method to the audio frame transformer API.
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2023-03-06 08:22:25 +00:00 |
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channel_send_frame_transformer_delegate_unittest.cc
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Move rtc::make_ref_counted to api/
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2022-06-15 09:47:38 +00:00 |
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channel_send_unittest.cc
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Update RTP timestamp based on capture timestamp when audio send stream is resumed.
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2023-01-27 15:46:32 +00:00 |
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conversion.h
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Make header files self contained.
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2022-10-08 08:38:36 +00:00 |
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DEPS
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pc: Add asynchronous RtpSender::SetParameters() call
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2022-11-15 15:31:40 +00:00 |
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mock_voe_channel_proxy.h
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Implement RTCOutboundRtpStreamStats.targetBitrate for audio.
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2021-11-12 09:24:34 +00:00 |
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OWNERS
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Add alessiob@webrtc.org in audio/OWNERS
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2022-09-09 07:33:11 +00:00 |
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remix_resample.cc
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Reland "Rename FATAL() into RTC_FATAL()."
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2020-11-18 20:49:08 +00:00 |
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remix_resample.h
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Use backticks not vertical bars to denote variables in comments for /audio
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2021-07-27 15:36:40 +00:00 |
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remix_resample_unittest.cc
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Clarify and extend test support for certain sample rates in audio processing
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2022-08-03 14:26:36 +00:00 |
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