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Per Kjellander 50b0a76ee7 Reland "[WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream" 
This reverts commit 73f048daf07b157961c43e7dbc9d2c378e6457d8.

Reason for revert: Real culprit fixed here: https://chromium-review.googlesource.com/c/chromium/src/+/4417639

Original change's description:
> Revert "[WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream"
>
> This reverts commit dd557fdb1e300068c62c870d9dc5273b48c7b79d.
>
> Reason for revert: Looks like the Chromium FYI builders are failing.
>
> Original change's description:
> > [WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream
> >
> > This remove use of MaybeWorkerThread* rtp_transport_queue_ from
> > AudioSendStream.  The worker queue is alwauys assumed ot be used where
> > rtp_transport_queue_ was used.
> >
> > Bug: webrtc:14502
> > Change-Id: Ia516ce7340d712671e0ecb301bba9d66e7216973
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300400
> > Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
> > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
> > Commit-Queue: Per Kjellander <perkj@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39816}
>
> Bug: webrtc:14502
> Change-Id: I0547548032756fc579b76b6bb362f576aa06b8f7
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301020
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#39820}

Bug: webrtc:14502
Change-Id: I4db2560de3b21ee0c5c7c579af1891b2c7b2815f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300866
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39828}
2023-04-12 13:39:05 +00:00
..
test
Remove AudioConfig::Mode.
2023-04-04 08:44:23 +00:00
utility
Remove dependency on rtc_base_approved from most targets
2022-04-25 12:15:30 +00:00
voip
Implement support for Chrome task origin tracing. #3.5/4
2023-03-01 11:11:37 +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
Remove rtp header extension from config of Call audio and video receivers
2023-01-31 11:58:43 +00:00
audio_receive_stream.cc
Use SequenceChecker(SequenceChecker::kDetached) in a few places.
2023-03-24 07:44:18 +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
CallTest: migrate timeouts to TimeDelta.
2022-08-16 12:06:54 +00:00
audio_send_stream_unittest.cc
Reland "[WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream"
2023-04-12 13:39:05 +00:00
audio_send_stream.cc
Reland "[WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream"
2023-04-12 13:39:05 +00:00
audio_send_stream.h
Reland "[WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream"
2023-04-12 13:39:05 +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
Reland "[WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream"
2023-04-12 13:39:05 +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_unittest.cc
Create unit test for the population of capture_start_ntp_time
2023-02-06 14:00:39 +00:00
channel_receive.cc
Delete RtpRtcpInterface::RemoteNtp as redundant to GetSenderReportStats
2023-02-28 13:55:27 +00:00
channel_receive.h
stats: use uint64_t for RTCSentRtpStreamStats.packetsSent
2023-03-16 06:46:19 +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
Add a clone method to the audio frame transformer API.
2023-03-06 08:22:25 +00:00
channel_send_frame_transformer_delegate.h
Add a clone method to the audio frame transformer API.
2023-03-06 08:22:25 +00:00
channel_send_unittest.cc
Update RTP timestamp based on capture timestamp when audio send stream is resumed.
2023-01-27 15:46:32 +00:00
channel_send.cc
Break apart AudioCodingModule and AcmReceiver
2023-02-01 16:09:26 +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