Reason for revert:
Breaks webrtc.linux
Original issue's description:
> Add Jpeg frame writer for test support.
>
> Also, use it to save worst psnr frame in video quality tests. It is indented that these saved frames from perfbots will be uploaded to the cloud and will be available in chrome perf dashboard. Because of that size of the saved frame is somewhat an issue. Also, y4m is not convenient to view.
>
> BUG=webrtc:8030
>
> Review-Url: https://codereview.webrtc.org/2990563002
> Cr-Commit-Position: refs/heads/master@{#19414}
> Committed: 26e5cbd6bbTBR=stefan@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:8030
Review-Url: https://codereview.webrtc.org/2998133002
Cr-Commit-Position: refs/heads/master@{#19419}
Also, use it to save worst psnr frame in video quality tests. It is indented that these saved frames from perfbots will be uploaded to the cloud and will be available in chrome perf dashboard. Because of that size of the saved frame is somewhat an issue. Also, y4m is not convenient to view.
BUG=webrtc:8030
Review-Url: https://codereview.webrtc.org/2990563002
Cr-Commit-Position: refs/heads/master@{#19414}
Instead explicitly ignore only the flags we know should be ignored.
BUG=webrtc:7568
Review-Url: https://codereview.webrtc.org/2968003003
Cr-Commit-Position: refs/heads/master@{#19412}
I want to publish an API for iSAC in webrtc/api/, and I want to use
the class names Audio{De,En}coderIsac{Fix,Float}.
BUG=webrtc:7835, webrtc:7841
Review-Url: https://codereview.webrtc.org/2996593002
Cr-Commit-Position: refs/heads/master@{#19381}
This CL brings us one step closer to removing CodecDatabase and
GenericEncoder, by removing the static VCM::Codec(). Codec specific
methods are moved to video_encoder.cc (they already belonged to this
class) and getting default generic codec settings has been moved to a
test specific file.
This CL also makes video_encoder.h pass style guide and lint checks,
since these checks are triggered with the new video_encoder.cc file.
BUG=webrtc:8064
Review-Url: https://codereview.webrtc.org/2993923002
Cr-Commit-Position: refs/heads/master@{#19303}
[This CL is work in progress.]
Wire up the rtp keep-alive in webrtc::Call::Config using new
SetRtpTransportParameters() method on RtpTransportInterface.
BUG=webrtc:7907
Review-Url: https://codereview.webrtc.org/2981513002
Cr-Commit-Position: refs/heads/master@{#19287}
This CL:
- Renames the ViEEncoder class to VideoStreamEncoder, according to discussions.
- Renames variables 'vie_encode' to 'video_stream_encoder'.
- Formatting to match style guide.
- No other changes.
BUG=webrtc:8064
Review-Url: https://codereview.webrtc.org/2995433002
Cr-Commit-Position: refs/heads/master@{#19237}
The destructor was not explicitly marked as "virtual" nor as "override".
BUG=None
Review-Url: https://codereview.webrtc.org/2986363002
Cr-Commit-Position: refs/heads/master@{#19236}
In preparation of making RTP packet demuxing many-to-one (one SSRC goes to one sink, but one sink may have multiple SSRCs), we need to remove FlexFEC's dependence on being able to register itself with the demuxer. Instead, we register FlexFEC streams with the streams they protect; when those streams get packets, they'll forward them to their associated FlexFEC streams, too.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2974453002
Cr-Commit-Position: refs/heads/master@{#19219}
This is the first in a series of CLs to add support for media
identification as part of unified plan SDP.
Bug: webrtc:4050
Change-Id: I0eb5639d240a9a1412c2b047a33d5112e4901f26
Reviewed-on: https://chromium-review.googlesource.com/576374
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19111}
All downstream code have been updated to the new location.
In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS
Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
BUG=webrtc:7634
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2976293002
Cr-Commit-Position: refs/heads/master@{#19094}
Defaults are consistent with these used in CallTest.
BUG=none
Review-Url: https://codereview.webrtc.org/2972423002
Cr-Commit-Position: refs/heads/master@{#18961}
Since the keep-alive payload type is not registered in the payload type
map of FakeNetworkPipe, it will cause a DCHECK to trigger unless we're
able to destroy the call before that.
Just register it in the fake network as media type "any", it will be
discarded early on the receive side anyway.
BUG=webrt:7964
Review-Url: https://codereview.webrtc.org/2979543002
Cr-Commit-Position: refs/heads/master@{#18953}
All downstream code have been updated to the new location.
In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS
Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
BUG=webrtc:7634
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2973183002
Cr-Commit-Position: refs/heads/master@{#18948}
Some frames are already marked as 'timing frames' via video-timing RTP header extension. Timestamps along full WebRTC pipeline are gathered for these frames. This CL implements reporting of these timestamps for a single
timing frame since the last GetStats(). The frame with the longest end-to-end delay between two consecutive GetStats calls is reported.
The purpose of this timing information is not to provide a realtime statistics but to provide debugging information as it will help identify problematic places in video pipeline for outliers (frames which took longest to process).
BUG=webrtc:7594
Review-Url: https://codereview.webrtc.org/2946413002
Cr-Commit-Position: refs/heads/master@{#18909}
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
(This got reverted because of a problem with the Opus encoder parts.
Re-landing without changes.)
BUG=webrtc:7837
Review-Url: https://codereview.webrtc.org/2950453002
Cr-Commit-Position: refs/heads/master@{#18855}
[This CL is a rebase of an original CL by solenberg@:
https://codereview.webrtc.org/2948763002/ which in turn was a
rebase of an original CL by peah@:
https://chromium-review.googlesource.com/c/527032/]
Allow an external audio processing module to be used in WebRTC
This CL adds support for optionally using an externally created audio
processing module in a peerconnection. The ownership is shared
between the peerconnection and the external creator of the module.
As part of this the internal ownership of the audio processing module
is moved from VoiceEngine to WebRtcVoiceEngine.
BUG=webrtc:7775
Review-Url: https://codereview.webrtc.org/2961723004
Cr-Commit-Position: refs/heads/master@{#18837}
Reland the base->rtc_base without adding stub headers (will be
done in follow-up CL). This preserves git blame history of all files.
BUG=webrtc:7634
NOTRY=True
TBR=kwiberg@webrtc.org
Change-Id: Iea3bb6f3f67b8374c96337b63e8f5aa3e6181012
Reviewed-on: https://chromium-review.googlesource.com/554611
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18821}
Will reland in two different commits to preserve git blame history.
BUG=webrtc:7634
NOTRY=True
TBR=kwiberg@webrtc.org
Change-Id: I550da8525aeb9c5b8f96338fcf1c9714f3dcdab1
Reviewed-on: https://chromium-review.googlesource.com/554610
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18820}
This refactoring takes a careful approach to avoid rushing the change:
* stub headers are left in all the old locations of webrtc/base
* existing GN targets are kept and now just forward to the moved ones
using public_deps.
The only exception to the above is the base_java target and its .java files,
which were moved to webrtc/rtc_base right away since it's not possible
to use public_deps for android_library.
To avoid breaking builds, a temporary Dummy.java file was added to
the new intermediate target in webrtc/rtc_base:base_java as well to avoid
hitting a GN assert in the android_library template.
The above approach should make the transition smooth without breaking
downstream.
A helper script was created (https://codereview.webrtc.org/2879203002/)
and was run like this:
stub-headers.py -s webrtc/base -d webrtc/rtc_base -i 7634
stub-headers.py -s webrtc/base/numerics -d webrtc/rtc_base/numerics -i 7634
Fixed invalid header guards in the following files:
webrtc/base/base64.h
webrtc/base/cryptstring.h
webrtc/base/event.h
webrtc/base/flags.h
webrtc/base/httpbase.h
webrtc/base/httpcommon-inl.h
webrtc/base/httpcommon.h
webrtc/base/httpserver.h
webrtc/base/logsinks.h
webrtc/base/macutils.h
webrtc/base/nattypes.h
webrtc/base/openssladapter.h
webrtc/base/opensslstreamadapter.h
webrtc/base/pathutils.h
webrtc/base/physicalsocketserver.h
webrtc/base/proxyinfo.h
webrtc/base/sigslot.h
webrtc/base/sigslotrepeater.h
webrtc/base/socket.h
webrtc/base/socketaddresspair.h
webrtc/base/socketfactory.h
webrtc/base/stringutils.h
webrtc/base/testbase64.h
webrtc/base/testutils.h
webrtc/base/transformadapter.h
webrtc/base/win32filesystem.h
Added new header guards to:
sslroots.h
testbase64.h
BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True
R=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2877023002 .
Cr-Commit-Position: refs/heads/master@{#18816}
Patch set 1 is a reland + trivial rebase.
Patch set >= 2 contains bug fixes.
> Original issue's description:
> > Fix bug in vie_encoder.cc which caused channel parameters not to be updated at regular intervals, as it was intended.
> >
> > That however exposes a bunch of failed test, so this CL also fixed a few other things:
> > * FakeEncoder should trust the configured FPS value rather than guesstimating itself based on the realtime clock, so as not to completely undershoot targets in offline mode. Also, compensate for key-frame overshoots when outputting delta frames.
> > * FrameDropper should not assuming incoming frame rate is 0 if no frames have been seen.
> > * Fix a bunch of test cases that started failing because they were relying on the fake encoder undershooting.
> > * Fix test
> >
> > BUG=7664
> >
> > Review-Url: https://codereview.webrtc.org/2883963002
> > Cr-Commit-Position: refs/heads/master@{#18473}
> > Committed: 6431e21da6
BUG=webrtc:7664
Review-Url: https://codereview.webrtc.org/2953053002
Cr-Commit-Position: refs/heads/master@{#18782}
An automatic undef checker noticed an error in the new
audio_processing_fuzzer. An array was default-initialized, meaning its
contents had indeterminate values. This array was later filtered for
forbidden values, which happened or did not happen based on the
uninitialized values. Hence the error.
TBR=henrik.lundin@webrtc.org
NOTRY=true
Bug: 735718, webrtc:7820
Change-Id: Ib50a6f7a5f054472fef07a27d37e02cea432a8ca
Reviewed-on: https://chromium-review.googlesource.com/544937
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18713}
This is a high-level fuzzer that creates an AudioProcessing instance.
All possible combinations of publicly visible components are
fuzzed. Input and output sample rate, call order and use of the
float/fix interface is fuzzed. Sample rate may change between calls.
To fuzz floating point numbers, raw data is converted to floats, and
filtered for special values like 'inf', 'nan' and very large
values. Note that the default use case of APM is to only allow values
between +/- 2^15.
BUG=webrtc:7820
Review-Url: https://codereview.webrtc.org/2876793002
Cr-Commit-Position: refs/heads/master@{#18678}
Timing information is gathered in EncodedImage,
starting at encoders. Then it's sent using RTP header extension. In the
end, it's gathered at the GenericDecoder. Actual reporting and tests
will be in the next CLs.
BUG=webrtc:7594
Review-Url: https://codereview.webrtc.org/2911193002
Cr-Commit-Position: refs/heads/master@{#18659}
Reason for revert:
breaking downstream projects
Original issue's description:
> Opus implementation of the AudioDecoderFactoryTemplate API
>
> BUG=webrtc:7837
>
> Review-Url: https://codereview.webrtc.org/2942733003
> Cr-Commit-Position: refs/heads/master@{#18646}
> Committed: d053fe4ab3TBR=ossu@webrtc.org,solenberg@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7837
Review-Url: https://codereview.webrtc.org/2944763002
Cr-Commit-Position: refs/heads/master@{#18648}
This is to make it possible to override the rtc_task_queue target only.
BUG=none
Review-Url: https://codereview.webrtc.org/2931273002
Cr-Commit-Position: refs/heads/master@{#18534}
Fix issue where running tests on iOS would get killed after a certain
time had passed. This seems to be due to springboard killing apps
that don't have a GUI running. Creating a UIApplication to wrap
the test suite seems to solve this problem in chromium.
This CL adds a class for this purpose. Most of the code was copied
from chromium with bits taken out.
Bug: webrtc:7161, webrtc:7758
Change-Id: I10f9bc8914e73f2870a9b0a2703cde496af8db2f
Reviewed-on: https://chromium-review.googlesource.com/528173
Reviewed-by: Henrik Andreasson <henrika@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Kári Tristan Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18509}
Reason for revert:
Looks like there's still one failing perf test:
RampUpTest.UpDownUpTransportSequenceNumberPacketLoss
Original issue's description:
> Reland of Periodically update codec bit/frame rate settings. (patchset #1 id:1 of https://codereview.webrtc.org/2923993002/ )
>
> Reason for revert:
> Create reland cl that we can patch with fix.
>
> Original issue's description:
> > Revert of Periodically update codec bit/frame rate settings. (patchset #8 id:140001 of https://codereview.webrtc.org/2883963002/ )
> >
> > Reason for revert:
> > Breaks some Call perf tests that are not run by the try bots....
> >
> > Original issue's description:
> > > Fix bug in vie_encoder.cc which caused channel parameters not to be updated at regular intervals, as it was intended.
> > >
> > > That however exposes a bunch of failed test, so this CL also fixed a few other things:
> > > * FakeEncoder should trust the configured FPS value rather than guesstimating itself based on the realtime clock, so as not to completely undershoot targets in offline mode. Also, compensate for key-frame overshoots when outputting delta frames.
> > > * FrameDropper should not assuming incoming frame rate is 0 if no frames have been seen.
> > > * Fix a bunch of test cases that started failing because they were relying on the fake encoder undershooting.
> > > * Fix test
> > >
> > > BUG=7664
> > >
> > > Review-Url: https://codereview.webrtc.org/2883963002
> > > Cr-Commit-Position: refs/heads/master@{#18473}
> > > Committed: 6431e21da6
> >
> > TBR=stefan@webrtc.org,holmer@google.com
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=7664
> >
> > Review-Url: https://codereview.webrtc.org/2923993002
> > Cr-Commit-Position: refs/heads/master@{#18475}
> > Committed: 5390c4814d
>
> TBR=stefan@webrtc.org,holmer@google.com
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=7664
>
> Review-Url: https://codereview.webrtc.org/2924023002
> Cr-Commit-Position: refs/heads/master@{#18497}
> Committed: cdafeda1cbTBR=stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=7664
Review-Url: https://codereview.webrtc.org/2926283002
Cr-Commit-Position: refs/heads/master@{#18500}
Reason for revert:
Create reland cl that we can patch with fix.
Original issue's description:
> Revert of Periodically update codec bit/frame rate settings. (patchset #8 id:140001 of https://codereview.webrtc.org/2883963002/ )
>
> Reason for revert:
> Breaks some Call perf tests that are not run by the try bots....
>
> Original issue's description:
> > Fix bug in vie_encoder.cc which caused channel parameters not to be updated at regular intervals, as it was intended.
> >
> > That however exposes a bunch of failed test, so this CL also fixed a few other things:
> > * FakeEncoder should trust the configured FPS value rather than guesstimating itself based on the realtime clock, so as not to completely undershoot targets in offline mode. Also, compensate for key-frame overshoots when outputting delta frames.
> > * FrameDropper should not assuming incoming frame rate is 0 if no frames have been seen.
> > * Fix a bunch of test cases that started failing because they were relying on the fake encoder undershooting.
> > * Fix test
> >
> > BUG=7664
> >
> > Review-Url: https://codereview.webrtc.org/2883963002
> > Cr-Commit-Position: refs/heads/master@{#18473}
> > Committed: 6431e21da6
>
> TBR=stefan@webrtc.org,holmer@google.com
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=7664
>
> Review-Url: https://codereview.webrtc.org/2923993002
> Cr-Commit-Position: refs/heads/master@{#18475}
> Committed: 5390c4814dTBR=stefan@webrtc.org,holmer@google.com
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=7664
Review-Url: https://codereview.webrtc.org/2924023002
Cr-Commit-Position: refs/heads/master@{#18497}
