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webrtc_m130/audio
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Harald Alvestrand afaae4e38a Remove remaining .cc files from rtc_media_base 
Also remove all dependencies on rtc_media_base except for a few
that are suspected of being linker directives.

Bug: webrtc:14775
Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41886}
2024-03-12 14:09:38 +00:00
..
test
Remove low_bandwidth_audio_test.
2023-06-01 07:20:38 +00:00
utility
Remove dependency on rtc_base_approved from most targets
2022-04-25 12:15:30 +00:00
voip
Cleanup usage of the rtc::TaskQueue in audio/
2024-01-18 12:24:14 +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
[SourceTracker] Move state to the worker thread, remove mutex.
2023-04-25 08:18:42 +00:00
audio_receive_stream.cc
Log audio stream start/stop.
2023-12-12 10:43:47 +00:00
audio_receive_stream.h
Use SequenceChecker(SequenceChecker::kDetached) in a few places.
2023-03-24 07:44:18 +00:00
audio_send_stream_tests.cc
Rename AudioLevel to AudioLevelExtension in rtp_header_extensions.h
2024-02-22 23:12:52 +00:00
audio_send_stream_unittest.cc
Add option for the audio encoder to allocate a bitrate range.
2024-02-07 09:47:16 +00:00
audio_send_stream.cc
Add option for the audio encoder to allocate a bitrate range.
2024-02-07 09:47:16 +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
Use SequenceChecker(SequenceChecker::kDetached) in a few places.
2023-03-24 07:44:18 +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
Make capture timestamp optional in ADM.
2023-01-23 17:29:06 +00:00
audio_transport_impl.h
Make capture timestamp optional in ADM.
2023-01-23 17:29:06 +00:00
BUILD.gn
Remove remaining .cc files from rtc_media_base
2024-03-12 14:09:38 +00:00
channel_receive_frame_transformer_delegate_unittest.cc
Consolidate encoded transform mocks into api/test/
2024-01-26 12:46:34 +00:00
channel_receive_frame_transformer_delegate.cc
Remove DCHECK on setting audio rcvr encoded transform twice
2023-11-27 10:53:24 +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
Consolidate encoded transform mocks into api/test/
2024-01-26 12:46:34 +00:00
channel_receive.cc
Remove extraneous partial re-initialization of NetEq in the ChannelReceive ctor
2024-02-08 11:33:25 +00:00
channel_receive.h
Propagate time of the last received packet with Timestamp type
2023-06-02 14:29:19 +00:00
channel_send_frame_transformer_delegate_unittest.cc
Change vector<const T> with const vector<T>
2024-02-08 11:42:40 +00:00
channel_send_frame_transformer_delegate.cc
Propagate sequence number to cloned encoded audio frames
2024-01-29 14:29:44 +00:00
channel_send_frame_transformer_delegate.h
Replace rtc::TaskQueue* with TaskQueueBase* in audio channel send frame transformer
2024-01-18 09:39:55 +00:00
channel_send_unittest.cc
Consolidate encoded transform mocks into api/test/
2024-01-26 12:46:34 +00:00
channel_send.cc
Rename AudioLevel to AudioLevelExtension in rtp_header_extensions.h
2024-02-22 23:12:52 +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
Expose audio mimeType for insertable streams
2023-11-03 09:49:12 +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
Reland "Rename FATAL() into RTC_FATAL()."
2020-11-18 20:49:08 +00:00
remix_resample.h
Use backticks not vertical bars to denote variables in comments for /audio
2021-07-27 15:36:40 +00:00