Per K 02af84064c PacketRouter directly notify RtpTransportControllerSender when sending ... With this cl RtpTransportControllerSend::OnAddPacket is instead directly invoked from PacketRouter::SendPacket instead of going via RTP module. Transport sequence numbers are instead of directly written to header extension, added to RtpPacketToSendMetaData and written to the extenion by RTP module. This is to allow transport sequence numbers without actually sending them in an extension. Bug: webrtc:15368 Change-Id: Idd03e02a4257dfc4d0f1898b2803345975d7dad2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344720 Reviewed-by: Erik Språng <sprang@google.com> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41974}		2024-03-28 09:27:43 +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.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_receive_stream_unittest.cc	[SourceTracker] Move state to the worker thread, remove mutex.	2023-04-25 08:18:42 +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_send_stream_tests.cc	Expose AudioLevel as an absl::optional struct in api/rtp_headers.h	2024-03-22 10:07:47 +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_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_state_unittest.cc	Implement support for Chrome task origin tracing. #3.5/4	2023-03-01 11:11:37 +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.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_receive_frame_transformer_delegate.cc	Expose AudioLevel as an absl::optional struct in api/rtp_headers.h	2024-03-22 10:07:47 +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_frame_transformer_delegate_unittest.cc	Consolidate encoded transform mocks into api/test/	2024-01-26 12:46:34 +00:00
channel_receive_unittest.cc	Consolidate encoded transform mocks into api/test/	2024-01-26 12:46:34 +00:00
channel_send.cc	PacketRouter directly notify RtpTransportControllerSender when sending	2024-03-28 09:27:43 +00:00
channel_send.h	Expose audio mimeType for insertable streams	2023-11-03 09:49:12 +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_frame_transformer_delegate_unittest.cc	Change vector<const T> with const vector<T>	2024-02-08 11:42:40 +00:00
channel_send_unittest.cc	Consolidate encoded transform mocks into api/test/	2024-01-26 12:46:34 +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.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
remix_resample_unittest.cc	Clarify and extend test support for certain sample rates in audio processing	2022-08-03 14:26:36 +00:00