- Use NtpTime instead of pair of uint32_t to represent ntp time
- Increase precision estimate with NtpTime precision instead of ms precision
- Hide helper structs as private types
- Modernize interface to prefer return values over output parameters
- embed LinearRegression helper into the only user: UpdateParameters
Bug: webrtc:13757
Change-Id: I0a62a03e2869b2ae1eacaa15253accc43ba0a598
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254780
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36232}
WebRTC’s Audio Video sync can go in unbounded loop and keep on increasing audio delay if audio packets stop coming in.
The issue happens, if StreamSynchronization::ComputeDelays has:
1. relative_delay_ms = some positive value which causes avg_diff_ms_ > 30ms
2. current_audio_delay_ms < current_video_delay_ms
3. audio_delay_.extra_ms > 0 and video_delay_.extra_ms = 0
To compensate for relative delay, audio_delay_.extra_ms gets incremented every time StreamSynchronization::ComputeDelays is called by RtpStreamsSynchronizer::Process(), which happens every 1sec
RtpStreamsSynchronizer::Process() will try to set the new delay to audio stream by calling syncable_audio_->SetMinimumPlayoutDelay(target_audio_delay_ms);
This ends up calling DelayManager::SetMinimumDelay and update minimum_delay_ms_
But this update has no impact on the value returned by NetEqImpl::FilteredCurrentDelayMs (as there are no audio packets flowing in, hence neteq is not running) which is called next time RtpStreamsSynchronizer::Process(), runs and tried to compute the new audio delay (audio_info→current_delay_ms)
This causes audio delay to be increased in every iteration and it grows unbounded. I guess it will stop growing above 10sec as that is hardcoded max delay in NetEQ.
To avoid this added a check to not adjust delays when no new audio stream has come in.
Bug: webrtc:11894
Change-Id: If648f9227e43c351f887d054876cb119cc1a917e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183340
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Shyam Sadhwani <shyamsadhwani@fb.com>
Cr-Commit-Position: refs/heads/master@{#32106}
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280.
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.
Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25830}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/video/stream_synchronization_unittest.cc (Browse further)
