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webrtc/audio
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Henrik Boström 0c126ed47a De-flake NonSenderRttStats and make it faster to run on average. 
It takes several seconds until we get an RTT measurement because that
requires RTCP packets to be received and those are not sent very often.

This CL makes the test faster on average by unblocking it as soon as
we see an RTT measurement (as opposed to always blocking for 10
seconds), this usually unblocks after around 5 seconds.

But to de-flake those rare instances where the test takes more than 10s
to run, the maximum timeout is extended to 20 seconds.

Patch Set 4: also fix use-of-uninitialized value.

Bug: webrtc:14981
Change-Id: Ieca94c90dfb52c3b17584a06660ff66c6462aa8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296822
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39531}
2023-03-10 13:25:34 +00:00
..
test De-flake NonSenderRttStats and make it faster to run on average. 2023-03-10 13:25:34 +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.cc Remove rtp header extension from config of Call audio and video receivers 2023-01-31 11:58:43 +00:00
audio_receive_stream.h Remove rtp header extension from config of Call audio and video receivers 2023-01-31 11:58:43 +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_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 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 De-flake NonSenderRttStats and make it faster to run on average. 2023-03-10 13:25:34 +00:00
channel_receive.cc Delete RtpRtcpInterface::RemoteNtp as redundant to GetSenderReportStats 2023-02-28 13:55:27 +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 Create unit test for the population of capture_start_ntp_time 2023-02-06 14:00:39 +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
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_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