admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/audio
History
Jakob Ivarsson 340d6c0236 Remove packet overhead lock and cached bitrate constraints. 
These are no longer needed since the RTP transport runs on the worker
thread now.

Some tests that were too strict on ordering needed change.

Bug: none
Change-Id: I4265cb1a4fd3355208f19aefdbb7abeb45b6cadf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335700
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41599}
2024-01-23 08:50:12 +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
Remove internal overrides using old SendRtp and SendRtcp interfaces.
2023-08-15 13:20:21 +00:00
audio_send_stream_unittest.cc
Remove packet overhead lock and cached bitrate constraints.
2024-01-23 08:50:12 +00:00
audio_send_stream.cc
Remove packet overhead lock and cached bitrate constraints.
2024-01-23 08:50:12 +00:00
audio_send_stream.h
Remove packet overhead lock and cached bitrate constraints.
2024-01-23 08:50:12 +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
Record audio timestamps from iOS.
2024-01-19 12:35:53 +00:00
channel_receive_frame_transformer_delegate_unittest.cc
Support shortcircuiting encoded transforms
2023-11-17 13:03:27 +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
Remove DCHECK on setting audio rcvr encoded transform twice
2023-11-27 10:53:24 +00:00
channel_receive.cc
Remove DCHECK on setting audio rcvr encoded transform twice
2023-11-27 10:53:24 +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
Replace rtc::TaskQueue* with TaskQueueBase* in audio channel send frame transformer
2024-01-18 09:39:55 +00:00
channel_send_frame_transformer_delegate.cc
Replace rtc::TaskQueue* with TaskQueueBase* in audio channel send frame transformer
2024-01-18 09:39:55 +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
Use Environment in RtpTransportyControllerSend
2023-12-20 14:47:51 +00:00
channel_send.cc
Cleanup usage of the rtc::TaskQueue in audio/
2024-01-18 12:24:14 +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