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webrtc_m130/audio
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Manashi Sarkar 0121ff40da Revert "Propagate arrival time inside NetEq" 
This reverts commit 5237cbbe6851a6728c5d78add47e3d0cb80142ec.

Reason for revert: Breaks build.

Original change's description:
> Propagate arrival time inside NetEq
>
> Bug: webrtc:341266986
> Change-Id: I313ded76b884e9ee0f00f43541c8e9aebc406001
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351340
> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
> Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42381}

Bug: webrtc:341266986
Change-Id: I3c067b95055a8b3e7208cc6e45a5b581f8d65be6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351541
Commit-Queue: Manashi Sarkar <manashi@google.com>
Reviewed-by: Lionel Koenig <lionelk@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Lionel Koenig Gélas <lionelk@google.com>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42387}
2024-05-27 17:17:04 +00:00
..
test
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
utility
Add audio view classes
2024-05-24 18:08:37 +00:00
voip
Mass removal of absl_deps in all BUILD.gn files
2024-05-23 15:09:46 +00:00
audio_level.cc
Migrate audio/ to use webrtc::Mutex
2020-07-06 14:21:38 +00:00
audio_level.h
Migrate audio/ to use webrtc::Mutex
2020-07-06 14:21:38 +00:00
audio_receive_stream_unittest.cc
RTCP: implement reduced size RTCP for audio
2024-05-16 18:24:10 +00:00
audio_receive_stream.cc
RTCP: implement reduced size RTCP for audio
2024-05-16 18:24:10 +00:00
audio_receive_stream.h
RTCP: implement reduced size RTCP for audio
2024-05-16 18:24:10 +00:00
audio_send_stream_tests.cc
Expose AudioLevel as an absl::optional struct in api/rtp_headers.h
2024-03-22 10:07:47 +00:00
audio_send_stream_unittest.cc
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
audio_send_stream.cc
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
audio_send_stream.h
Add option for the audio encoder to allocate a bitrate range.
2024-02-07 09:47:16 +00:00
audio_state_unittest.cc
Implement support for Chrome task origin tracing. #3.5/4
2023-03-01 11:11:37 +00:00
audio_state.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
audio_state.h
Use SequenceChecker(SequenceChecker::kDetached) in a few places.
2023-03-24 07:44:18 +00:00
audio_transport_impl.cc
Add audio view classes
2024-05-24 18:08:37 +00:00
audio_transport_impl.h
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
BUILD.gn
Mass removal of absl_deps in all BUILD.gn files
2024-05-23 15:09:46 +00:00
channel_receive_frame_transformer_delegate_unittest.cc
Calculate the audio level of audio packets before encoded transforms
2024-04-29 15:14:25 +00:00
channel_receive_frame_transformer_delegate.cc
Add passkey to TransformableFrameInterface to prevent external impls
2024-05-16 13:12:51 +00:00
channel_receive_frame_transformer_delegate.h
Cleanup usage of the rtc::TaskQueue in audio/
2024-01-18 12:24:14 +00:00
channel_receive_unittest.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
channel_receive.cc
Revert "Propagate arrival time inside NetEq"
2024-05-27 17:17:04 +00:00
channel_receive.h
RTCP: implement reduced size RTCP for audio
2024-05-16 18:24:10 +00:00
channel_send_frame_transformer_delegate_unittest.cc
Calculate the audio level of audio packets before encoded transforms
2024-04-29 15:14:25 +00:00
channel_send_frame_transformer_delegate.cc
Add passkey to TransformableFrameInterface to prevent external impls
2024-05-16 13:12:51 +00:00
channel_send_frame_transformer_delegate.h
Calculate the audio level of audio packets before encoded transforms
2024-04-29 15:14:25 +00:00
channel_send_unittest.cc
Calculate the audio level of audio packets before encoded transforms
2024-04-29 15:14:25 +00:00
channel_send.cc
Pass the absolute capture timestamp to rtcp
2024-05-14 08:35:44 +00:00
channel_send.h
Expose audio mimeType for insertable streams
2023-11-03 09:49:12 +00:00
conversion.h
Make header files self contained.
2022-10-08 08:38:36 +00:00
DEPS
pc: Add asynchronous RtpSender::SetParameters() call
2022-11-15 15:31:40 +00:00
mock_voe_channel_proxy.h
RTCP: implement reduced size RTCP for audio
2024-05-16 18:24:10 +00:00
OWNERS
Add alessiob@webrtc.org in audio/OWNERS
2022-09-09 07:33:11 +00:00
remix_resample_unittest.cc
Clarify and extend test support for certain sample rates in audio processing
2022-08-03 14:26:36 +00:00
remix_resample.cc
Add audio view classes
2024-05-24 18:08:37 +00:00
remix_resample.h
Start using ArrayView in AudioFrame, update PushResampler
2024-04-30 15:33:08 +00:00