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test
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Move call/simulated_network to test/network
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2024-04-29 09:55:06 +00:00 |
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utility
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Cleanup expired field trial WebRTC-VoIPChannelRemixingAdjustmentKillSwitch
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2024-07-23 13:23:26 +00:00 |
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voip
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Migrate absl::optional to std::optional
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2024-09-02 12:16:47 +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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m130 merge fixes
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2024-10-23 20:01:09 -07:00 |
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audio_receive_stream.h
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Merge remote-tracking branch 'upstream/branch-heads/6723'
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2024-10-17 09:03:43 -07:00 |
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audio_receive_stream_unittest.cc
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Move thread handling from source tracker.
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2024-09-05 08:45:11 +00:00 |
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audio_send_stream.cc
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m130 merge fixes
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2024-10-23 20:01:09 -07:00 |
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audio_send_stream.h
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Merge remote-tracking branch 'upstream/branch-heads/6723'
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2024-10-17 09:03:43 -07:00 |
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audio_send_stream_tests.cc
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Expose AudioLevel as an absl::optional struct in api/rtp_headers.h
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2024-03-22 10:07:47 +00:00 |
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audio_send_stream_unittest.cc
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Migrate absl::optional to std::optional
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2024-09-02 12:16:47 +00:00 |
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audio_state.cc
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Merge remote-tracking branch 'google/branch-heads/6478'
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2024-06-21 16:31:45 -07:00 |
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audio_state.h
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Disable audio and media flow by default
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2023-09-12 18:03:28 -07:00 |
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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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Migrate absl::optional to std::optional
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2024-09-02 12:16:47 +00:00 |
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audio_transport_impl.h
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Migrate absl::optional to std::optional
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2024-09-02 12:16:47 +00:00 |
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BUILD.gn
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Remove use of AcmReceiver in ChannelReceive
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2024-09-06 12:47:36 +00:00 |
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channel_receive.cc
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Merge remote-tracking branch 'upstream/branch-heads/6723'
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2024-10-17 09:03:43 -07:00 |
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channel_receive.h
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Merge remote-tracking branch 'upstream/branch-heads/6723'
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2024-10-17 09:03:43 -07:00 |
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channel_receive_frame_transformer_delegate.cc
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Migrate absl::optional to std::optional
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2024-09-02 12:16:47 +00:00 |
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channel_receive_frame_transformer_delegate.h
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Pass receive_time through frame transformer
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2024-08-02 07:01:33 +00:00 |
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channel_receive_frame_transformer_delegate_unittest.cc
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Pass receive_time through frame transformer
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2024-08-02 07:01:33 +00:00 |
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channel_receive_unittest.cc
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Merge remote-tracking branch 'upstream/branch-heads/6723'
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2024-10-17 09:03:43 -07:00 |
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channel_send.cc
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Ensure the AudioCodingModule is reset when sending is stopped.
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2024-09-12 22:47:11 +00:00 |
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channel_send.h
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Migrate absl::optional to std::optional
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2024-09-02 12:16:47 +00:00 |
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channel_send_frame_transformer_delegate.cc
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Migrate absl::optional to std::optional
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2024-09-02 12:16:47 +00:00 |
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channel_send_frame_transformer_delegate.h
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Migrate absl::optional to std::optional
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2024-09-02 12:16:47 +00:00 |
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channel_send_frame_transformer_delegate_unittest.cc
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Migrate absl::optional to std::optional
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2024-09-02 12:16:47 +00:00 |
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channel_send_unittest.cc
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Ensure the AudioCodingModule is reset when sending is stopped.
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2024-09-12 22:47:11 +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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Move thread handling from source tracker.
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2024-09-05 08:45:11 +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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Remove PushResampler<T>::InitializeIfNeeded
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2024-07-04 10:33:21 +00:00 |
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remix_resample.h
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Update RemixAndResample to use audio views
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2024-07-03 09:52:24 +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 |
