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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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Finish resolving merge conflicts
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2022-11-11 19:10:59 -05:00 |
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voip
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Ensure RtpSenderEgress run on worker queue
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2023-06-09 13:40:35 +00:00 |
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audio_level.cc
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Update to WebRTC 4389 (e7d9f74)
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2021-04-16 13:26:31 -07:00 |
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audio_level.h
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Update to WebRTC 4389 (e7d9f74)
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2021-04-16 13:26:31 -07:00 |
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audio_receive_stream.cc
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Merge branch 'm116' into 5845
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2023-08-09 14:40:20 -05:00 |
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audio_receive_stream.h
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Merge branch 'm116' into 5845
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2023-08-09 14:40:20 -05:00 |
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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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Merge branch 'm116' into 5845
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2023-08-09 14:40:20 -05:00 |
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audio_send_stream.h
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Merge branch 'm116' into 5845
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2023-08-09 14:40:20 -05:00 |
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audio_send_stream_tests.cc
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Reland "Remove dependency of video_replay on TestADM."
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2023-04-25 09:39:22 +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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Merge branch 'm116' into 5845
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2023-08-09 14:40:20 -05:00 |
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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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Merge branch 'm116' into 5845
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2023-08-09 14:40:20 -05:00 |
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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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Make all encodedaudioframes inherit from TransformableAudioFrameI'face
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2023-06-19 18:54:47 +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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Pass rtcp message to RtpTransportController through newer interface
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2023-05-17 17:19:23 +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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Make all encodedaudioframes inherit from TransformableAudioFrameI'face
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2023-06-19 18:54:47 +00:00 |
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channel_send_frame_transformer_delegate.h
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Make all encodedaudioframes inherit from TransformableAudioFrameI'face
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2023-06-19 18:54:47 +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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Simplify handling rtcp messages in audio send channel
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2023-05-17 06:32:12 +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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Update to WebRTC 4389 (e7d9f74)
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2021-04-16 13:26:31 -07:00 |
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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 |
