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webrtc_m130/audio
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Alessio Bazzica c92338a13d Remove CallReceiveStatistics::rttMs 
Bug: webrtc:10739
Change-Id: I747ef1d4bf8980755e7c6dcac22e5ed129f6f9cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274580
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38041}
2022-09-09 08:35:41 +00:00
..
test
CallTest: migrate timeouts to TimeDelta.
2022-08-16 12:06:54 +00:00
utility
Remove dependency on rtc_base_approved from most targets
2022-04-25 12:15:30 +00:00
voip
Update rtc::Event::Wait call sites to use TimeDelta.
2022-08-19 10:07:28 +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
Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay
2022-07-20 09:14:03 +00:00
audio_receive_stream.cc
Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay
2022-07-20 09:14:03 +00:00
audio_receive_stream.h
Add SetTransportCc to ReceiveStreamInterface.
2022-05-30 14:07:04 +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
Remove rtc::Location from SendTask test helper
2022-08-11 12:55:32 +00:00
audio_send_stream.cc
Update audio/, media/, and video/ to not use implicit conversion
2022-04-21 09:00:14 +00:00
audio_send_stream.h
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +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
Reland "Reland "Remove unused APM voice activity detection sub-module""
2022-02-16 08:41:30 +00:00
audio_transport_impl.h
Remove typing detection
2022-03-23 10:23:54 +00:00
BUILD.gn
Rewrite AudioState null poller to use TaskQueueBase interface
2022-08-16 13:16:24 +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
Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable
2022-07-20 08:15:08 +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
Remove CallReceiveStatistics::rttMs
2022-09-09 08:35:41 +00:00
channel_receive.h
Remove CallReceiveStatistics::rttMs
2022-09-09 08:35:41 +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
Rewrite AudioState null poller to use TaskQueueBase interface
2022-08-16 13:16:24 +00:00
channel_send.h
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
conversion.h
…
DEPS
Cleanup of bwe_defines.h
2020-11-26 12:26:02 +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