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test
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Remove low_bandwidth_audio_test.
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2023-06-01 07:20:38 +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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audio: fix some typos
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2023-09-21 05:42:29 +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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Propagate time of the last received packet with Timestamp type
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2023-06-02 14:29:19 +00:00 |
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audio_receive_stream.h
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Use SequenceChecker(SequenceChecker::kDetached) in a few places.
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2023-03-24 07:44:18 +00:00 |
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audio_receive_stream_unittest.cc
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[SourceTracker] Move state to the worker thread, remove mutex.
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2023-04-25 08:18:42 +00:00 |
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audio_send_stream.cc
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Simplify handling rtcp messages in audio send channel
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2023-05-17 06:32:12 +00:00 |
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audio_send_stream.h
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Simplify handling rtcp messages in audio send channel
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2023-05-17 06:32:12 +00:00 |
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audio_send_stream_tests.cc
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Remove internal overrides using old SendRtp and SendRtcp interfaces.
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2023-08-15 13:20:21 +00:00 |
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audio_send_stream_unittest.cc
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Pass rtcp message to RtpTransportController through newer interface
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2023-05-17 17:19:23 +00:00 |
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audio_state.cc
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Use SequenceChecker(SequenceChecker::kDetached) in a few places.
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2023-03-24 07:44:18 +00:00 |
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audio_state.h
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Use SequenceChecker(SequenceChecker::kDetached) in a few places.
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2023-03-24 07:44:18 +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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Remove low_bandwidth_audio_test.
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2023-06-01 07:20:38 +00:00 |
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channel_receive.cc
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Change ChannelReceive::GetAudioFrameWithInfo to use new Converts method
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2023-09-15 10:46:02 +00:00 |
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channel_receive.h
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Propagate time of the last received packet with Timestamp type
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2023-06-02 14:29:19 +00:00 |
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channel_receive_frame_transformer_delegate.cc
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Support receiving cloned encoded audio frames
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2023-08-21 09:43:52 +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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Support sending IncomingFrames in audio
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2023-08-28 13:52:20 +00:00 |
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channel_receive_unittest.cc
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Remove internal overrides using old SendRtp and SendRtcp interfaces.
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2023-08-15 13:20:21 +00:00 |
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channel_send.cc
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Make field trial string DisableRtxRateLimiter enabled by default.
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2023-10-27 12:33:58 +00:00 |
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channel_send.h
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Simplify handling rtcp messages in audio send channel
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2023-05-17 06:32:12 +00:00 |
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channel_send_frame_transformer_delegate.cc
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Pass the correct abs_capture_timestamp while cloning audio frame
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2023-09-26 15:57:52 +00:00 |
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channel_send_frame_transformer_delegate.h
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Move RTPTimestamp offset handling out of encoded transform delegate
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2023-08-29 12:28:41 +00:00 |
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channel_send_frame_transformer_delegate_unittest.cc
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Move RTPTimestamp offset handling out of encoded transform delegate
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2023-08-29 12:28:41 +00:00 |
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channel_send_unittest.cc
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Move RTPTimestamp offset handling out of encoded transform delegate
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2023-08-29 12:28:41 +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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Simplify handling rtcp messages in audio send channel
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2023-05-17 06:32:12 +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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