mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-18 08:07:56 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions
webrtc/audio
History
Florent Castelli acabb3641b pc: Add asynchronous RtpSender::SetParameters() call 
As the synchronous version only posts a task to recreate the encoder
later, it is not possible to catch errors and state changes that
could appear then.
The asynchronous version of SetParameters() aims to solve this by
providing a callback to wait for the completion of the encoder
reconfiguration, allowing any error to be propagate and subsequent
getParameters() call to have up to date information.

Bug: webrtc:11607
Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38627}
2022-11-15 15:31:40 +00:00
..
test Add a new PeerConnectionE2EQualityTestFixture::AddPeer method. 2022-11-10 16:54:19 +00:00
utility Remove dependency on rtc_base_approved from most targets 2022-04-25 12:15:30 +00:00
voip Replace Thread::Invoke with Thread::BlockingCall 2022-09-09 10:44:17 +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 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_receive_stream_unittest.cc Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay 2022-07-20 09:14:03 +00:00
audio_send_stream.cc pc: Add asynchronous RtpSender::SetParameters() call 2022-11-15 15:31:40 +00:00
audio_send_stream.h pc: Add asynchronous RtpSender::SetParameters() call 2022-11-15 15:31:40 +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_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_state_unittest.cc Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable 2022-07-20 08:15:08 +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 pc: Add asynchronous RtpSender::SetParameters() call 2022-11-15 15:31:40 +00:00
channel_receive.cc More audio stack traces 2022-10-04 14:31:52 +00:00
channel_receive.h audio: make packets lost a signed integer 2022-11-01 11:46:49 +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_frame_transformer_delegate_unittest.cc Move rtc::make_ref_counted to api/ 2022-06-15 09:47:38 +00:00
channel_send.cc [Stats] Expose totalPacketSendDelay for audio as well. 2022-10-27 10:33:16 +00:00
channel_send.h [Stats] Expose totalPacketSendDelay for audio as well. 2022-10-27 10:33:16 +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_frame_transformer_delegate_unittest.cc Move rtc::make_ref_counted to api/ 2022-06-15 09:47:38 +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.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