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webrtc_m130/audio
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Jakob Ivarsson 6d5fa001df Flush buffers when stopping audio receive stream. 
Bug: chromium:1400642
Change-Id: I19f22ca2fcf04d5e973d0e49fda841c9d40b12a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290723
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39045}
2023-01-09 19:16:26 +00:00
..
test
Reland "Remove use of ReceiveStreamRtpConfig:transport_cc"
2023-01-04 11:35:19 +00:00
utility
Remove dependency on rtc_base_approved from most targets
2022-04-25 12:15:30 +00:00
voip
[Unwrap] Use RtpTimestampUnwrapper in audio_ingress
2023-01-09 14:47:22 +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
Reland "Remove use of ReceiveStreamRtpConfig:transport_cc"
2023-01-04 11:35:19 +00:00
audio_receive_stream.cc
Reland "Remove use of ReceiveStreamRtpConfig:transport_cc"
2023-01-04 11:35:19 +00:00
audio_receive_stream.h
Reland "Remove use of ReceiveStreamRtpConfig:transport_cc"
2023-01-04 11:35:19 +00:00
audio_send_stream_tests.cc
CallTest: migrate timeouts to TimeDelta.
2022-08-16 12:06:54 +00:00
audio_send_stream_unittest.cc
pc: Add asynchronous RtpSender::SetParameters() call
2022-11-15 15:31:40 +00:00
audio_send_stream.cc
Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead
2022-11-30 20:19:36 +00:00
audio_send_stream.h
Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead
2022-11-30 20:19:36 +00:00
audio_state_unittest.cc
Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable
2022-07-20 08:15:08 +00:00
audio_state.cc
Rewrite AudioState null poller to use TaskQueueBase interface
2022-08-16 13:16:24 +00:00
audio_state.h
Rewrite AudioState null poller to use TaskQueueBase interface
2022-08-16 13:16:24 +00:00
audio_transport_impl.cc
More audio stack traces
2022-10-04 14:31:52 +00:00
audio_transport_impl.h
Remove typing detection
2022-03-23 10:23:54 +00:00
BUILD.gn
Split media_channel and its dependencies from the rtc_media_base target
2022-12-16 12:15:22 +00:00
channel_receive_frame_transformer_delegate_unittest.cc
Move rtc::make_ref_counted to api/
2022-06-15 09:47:38 +00:00
channel_receive_frame_transformer_delegate.cc
Add GetContributionSources to TransformableIncomingAudioFrame
2022-10-11 12:52:21 +00:00
channel_receive_frame_transformer_delegate.h
Use backticks not vertical bars to denote variables in comments for /audio
2021-07-27 15:36:40 +00:00
channel_receive.cc
Flush buffers when stopping audio receive stream.
2023-01-09 19:16:26 +00:00
channel_receive.h
audio: make packets lost a signed integer
2022-11-01 11:46:49 +00:00
channel_send_frame_transformer_delegate_unittest.cc
Move rtc::make_ref_counted to api/
2022-06-15 09:47:38 +00:00
channel_send_frame_transformer_delegate.cc
Update audio code to not use implicit T* --> scoped_refptr<T> conversion
2022-01-13 15:49:49 +00:00
channel_send_frame_transformer_delegate.h
Use backticks not vertical bars to denote variables in comments for /audio
2021-07-27 15:36:40 +00:00
channel_send.cc
Remove pending packets from the pacer when an RTP module is removed.
2022-12-13 11:32:58 +00:00
channel_send.h
[Stats] Expose totalPacketSendDelay for audio as well.
2022-10-27 10:33:16 +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
Implement RTCOutboundRtpStreamStats.targetBitrate for audio.
2021-11-12 09:24:34 +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