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webrtc/audio
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Jakob Ivarsson db208317eb Update RTP timestamp based on capture timestamp when audio send stream is resumed. 
This removes the previous approach where we continued to update the timestamp when the capturer is running but the send stream is stopped in favor of a more general approach that also works when the capturer is paused.

Some assumptions for this change to be correct: input sample rate and frame size will be the same before/after the stream is paused.

Bug: webrtc:12397
Change-Id: I3b03741cd6d3285cbc9aee3893800729852e6cfa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291526
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39213}
2023-01-27 15:46:32 +00:00
..
test Delete unused Audio Bwe integration test. 2023-01-26 09:31:44 +00:00
utility Remove dependency on rtc_base_approved from most targets 2022-04-25 12:15:30 +00:00
voip [Unwrap] Use RtpTimestampUnwrapper in audio_ingress 2023-01-09 14:47:22 +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 Delete unnecssary AudioReceiveStreamInterface::GetRtpExtensions 2023-01-18 14:06:33 +00:00
audio_receive_stream.h Delete unnecssary AudioReceiveStreamInterface::GetRtpExtensions 2023-01-18 14:06:33 +00:00
audio_receive_stream_unittest.cc Reland "Remove use of ReceiveStreamRtpConfig:transport_cc" 2023-01-04 11:35:19 +00:00
audio_send_stream.cc Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead 2022-11-30 20:19:36 +00:00
audio_send_stream.h Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead 2022-11-30 20:19:36 +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 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 Reset encoder when audio send stream is stopped. 2023-01-26 15:20:02 +00:00
channel_receive.cc Add RtcEvent to store when MinimumSetDelay is set on NetEq 2023-01-13 17:15:48 +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_receive_unittest.cc Add a basic unittest for webrtc::voe::ChannelReceive 2023-01-25 20:06:26 +00:00
channel_send.cc Update RTP timestamp based on capture timestamp when audio send stream is resumed. 2023-01-27 15:46:32 +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
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
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