This CL is expected to lower goog_max_decode_ms and total_delay_incl_network/receiver_time for screenshare.
Reason for revert:
This CL did not cause the unexpected goog_encode_usage_percent and goog_avg_encode_ms perf changes.
Original issue's description:
> Revert of VCMCodecTimer: Change filter from max to 95th percentile (patchset #5 id:180001 of https://codereview.webrtc.org/1742323002/ )
>
> Reason for revert:
> Caused unexpected perf stats changes, see http://crbug/594575.
>
> Original issue's description:
> > VCMCodecTimer: Change filter from max to 95th percentile
> >
> > The purpose with this change is to make the filter more robust against anomalies. googMaxDecodeMs is expected to drop a litte by this.
> >
> > BUG=b/27306053
> >
> > Committed: https://crrev.com/4bf0c717740d1834e810ea5f32b3c4306c64235f
> > Cr-Commit-Position: refs/heads/master@{#11952}
>
> TBR=stefan@webrtc.org,philipel@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=594575,b/27306053
>
> Committed: https://crrev.com/c4a74e95b545f4752d4e72961ac03c1380d4bc1f
> Cr-Commit-Position: refs/heads/master@{#12018}
TBR=stefan@webrtc.org,philipel@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=594575,b/27306053
Review URL: https://codereview.webrtc.org/1824763003
Cr-Commit-Position: refs/heads/master@{#12087}
From an earlier CL, we start to feed UpdateLevel() with power instead of energy. I found that UpdateLevel() is still taking the input as energy and normalize it. This CL fixes this.
The earlier CL is
https://codereview.webrtc.org/1542573002/
BUG=
Review URL: https://codereview.webrtc.org/1810773003
Cr-Commit-Position: refs/heads/master@{#12084}
Applications can choose to decide when to give up control of the
AVAudioSession to WebRTC. Otherwise, behavior should be
unchanged.
Adds a toggle to AppRTCDemo so developers can see the different
paths.
BUG=
R=haysc@webrtc.org
Review URL: https://codereview.webrtc.org/1822543002 .
Cr-Commit-Position: refs/heads/master@{#12080}
ScreenCapturerMac wasn't handling the following two cases properly
which could cause crashes:
1. CGDisplayCreateImage() returns image with depth other than 32-bit
2. CGDisplayCreateImage() returns image with dimensions different
from expected (e.g. when screen resolution is being changed).
I suspect that (2) was causing the linked bug.
BUG=crbug.com/504927
R=jiayl@webrtc.org
Review URL: https://codereview.webrtc.org/1816723002 .
Cr-Commit-Position: refs/heads/master@{#12077}
Removes code duplication and use of the dangerous public destructor in
RefCountImpl.
Also making wider use of scoped_refptr and fixing various leaks in the
process.
BUG=webrtc:5229
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1477013005 .
Cr-Commit-Position: refs/heads/master@{#12075}
Reason for revert:
The tests in the CL are failing on the bots in the Webrtc Waterfall (allthough they did not fail on the commit bots). I will therefore revise and reland the test.
Original issue's description:
> Added a bitexactness test for the echo canceller in the audio processing module.
>
> BUG=webrtc:5337
>
> Committed: https://crrev.com/7c448e1a384224aa16a21715e83574f3f553b730
> Cr-Commit-Position: refs/heads/master@{#12068}
TBR=henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5337
Review URL: https://codereview.webrtc.org/1824583003
Cr-Commit-Position: refs/heads/master@{#12072}
Reason for revert:
This needs to be reverted as a previous CL which needs to be reverted causes a merge clash with this CL.
Original issue's description:
> Added a bitexactness test for the echo control mobile in the audio processing module
>
> BUG=webrtc:5663
>
> Committed: https://crrev.com/105831ef4a38ac49e5e2c1ce207884da0a26c773
> Cr-Commit-Position: refs/heads/master@{#12069}
TBR=henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5663
Review URL: https://codereview.webrtc.org/1819803002
Cr-Commit-Position: refs/heads/master@{#12071}
This CL also extracts part of the functionality used
in the bitexactness test for the high-pass filter into
a separate file in order to be able to reuse that
functionality in bitexactness tests for the other
submodules in APM (including the bitexactness test for
the noise suppressor).
BUG=wertc:5336
Review URL: https://codereview.webrtc.org/1783203002
Cr-Commit-Position: refs/heads/master@{#12061}
WebRtcIsacfix_PitchFilterCore requires indW32 >= PITCH_FRACORDER - 2;
otherwise, it will read from entries of ubufQQ that haven't been
written yet. (The problem of indW32 being too small has only been seen
in fuzzer tests, not in real life.)
BUG=chromium:581901
Review URL: https://codereview.webrtc.org/1811453002
Cr-Commit-Position: refs/heads/master@{#12047}
Except in places where this would break out-of-tree code,
such as Chromium.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1785173002
Cr-Commit-Position: refs/heads/master@{#12037}
It would be good to have a dedicated DebugDumpReplayer. There is one but it hides itself in DebugDumpTest.
This CL is to separate it out.
BUG=
Review URL: https://codereview.webrtc.org/1810463002
Cr-Commit-Position: refs/heads/master@{#12029}
Reason for revert:
Caused unexpected perf stats changes, see http://crbug/594575.
Original issue's description:
> VCMCodecTimer: Change filter from max to 95th percentile
>
> The purpose with this change is to make the filter more robust against anomalies. googMaxDecodeMs is expected to drop a litte by this.
>
> BUG=b/27306053
>
> Committed: https://crrev.com/4bf0c717740d1834e810ea5f32b3c4306c64235f
> Cr-Commit-Position: refs/heads/master@{#11952}
TBR=stefan@webrtc.org,philipel@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=594575,b/27306053
Review URL: https://codereview.webrtc.org/1808693002
Cr-Commit-Position: refs/heads/master@{#12018}
Reason for revert:
Revert because it breaks downstream code.
Original issue's description:
> Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code.
>
> BUG=webrtc:4690
>
> Committed: https://crrev.com/69a81999ace08e40e2b2ec526b0e111aa11b9538
> Cr-Commit-Position: refs/heads/master@{#12015}
TBR=henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1812453002
Cr-Commit-Position: refs/heads/master@{#12016}
Reason for revert:
The openmax_dl include change breaks downstream projects.
Original issue's description:
> Add check_deps rules in DEPS files.
>
> Add fine-grained check_deps rules for all of WebRTC.
> This will help both maintaining sane dependencies and provides a way
> to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.
>
> Example:
> buildtools/checkdeps/graphdeps.py --root=. --format=png \
> --out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
> --excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'
>
> will produce a neat webrtc.png image showcasing the dependencies
> (according to the DEPS file) for the bitrate_controller module.
> Some dependencies are filtered out for readability.
>
> BUG=webrtc:5623
> TESTED=Passing runs using:
> buildtools/checkdeps/checkdeps.py --root=. talk
> buildtools/checkdeps/checkdeps.py --root=. webrtc
>
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/086f851b7b9b4bcbd4fe507c3bf83b760bd7f4d9
> Cr-Commit-Position: refs/heads/master@{#12008}
TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5623
Review URL: https://codereview.webrtc.org/1808573002
Cr-Commit-Position: refs/heads/master@{#12009}
Add fine-grained check_deps rules for all of WebRTC.
This will help both maintaining sane dependencies and provides a way
to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.
Example:
buildtools/checkdeps/graphdeps.py --root=. --format=png \
--out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
--excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'
will produce a neat webrtc.png image showcasing the dependencies
(according to the DEPS file) for the bitrate_controller module.
Some dependencies are filtered out for readability.
BUG=webrtc:5623
TESTED=Passing runs using:
buildtools/checkdeps/checkdeps.py --root=. talk
buildtools/checkdeps/checkdeps.py --root=. webrtc
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1796413002 .
Cr-Commit-Position: refs/heads/master@{#12008}
Part 3 of refactor. Also:
- better weak pointer delegate storage + tests
- we now ignore route changes when we're interrupted
- fixed bug where preferred sample rate wasn't set if audio session
wasn't active
BUG=
Review URL: https://codereview.webrtc.org/1796983004
Cr-Commit-Position: refs/heads/master@{#12007}
Add WEBRTC_HAS_NEON and __SSE2__, otherwise, CPU detection fails in some cases.
Review URL: https://codereview.webrtc.org/1784323005
Cr-Commit-Position: refs/heads/master@{#12001}
this class replace and extend RTCPUtility::RtcpCommonHeader structure and RTCPUtility::RtcpParseCommonHeader function.
In addition to header fields, payload pointer is stored because rtcp header without payload is rarely useful.
Sample usage can be checked in 'RTCP Parser sketched' CL: https://codereview.webrtc.org/1555683002/
BUG=webrtc:5260
R=asapersson@webrtc.org, sprang@webrtc.org
Review URL: https://codereview.webrtc.org/1575413002 .
Cr-Commit-Position: refs/heads/master@{#11999}
This CL removes the dependency of AudioProcessing in
EchoCancellerImpl. It is breaking the public APM API by
having a different error code behavior so please review it
carefully. I made a comment about the API breaking change
in the code section of this CL.
BUG=webrtc:5337
Review URL: https://codereview.webrtc.org/1770823002
Cr-Commit-Position: refs/heads/master@{#11998}
"WebRTC.Video.AVSyncOffsetInMs"
The absolute value of the sync offset between a rendered video frame and the latest played audio frame is measured per video frame. The average offset per received video stream is recorded when a stream is removed.
Updated sync tests in call_perf_tests.cc to use this implementation.
BUG=webrtc:5493
Review URL: https://codereview.webrtc.org/1756193005
Cr-Commit-Position: refs/heads/master@{#11993}
- Clean up unused methods in voe::Channel following removal of VoEDtmf APIs.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1785643006
Cr-Commit-Position: refs/heads/master@{#11976}
In addition:
- Introduces RTCAudioSessionTest
- iOS/Mac gtests now have an autoreleasepool
- Moves ScopedAutoreleasePool to rtc_base_approved
- Introduces route change button in AppRTCDemo
BUG=webrtc:5649
Review URL: https://codereview.webrtc.org/1782363002
Cr-Commit-Position: refs/heads/master@{#11971}
Testing the nack module by implementing it into the current jitter buffer
under the experiment WebRTC-NewVideoJitterBuffer.
BUG=webrtc:5514
Review URL: https://codereview.webrtc.org/1778503002
Cr-Commit-Position: refs/heads/master@{#11969}
