It was included twice. In both of these targets:
webrtc/modules/audio_processing:{audio_processing_tests,
audio_processing_unittests}
In audio_processing_tests, the new unit tests were added to
public_deps, which (we think) somehow caused webrtc:webrtc_tests
to list the AecDump tests, to much confusion.
Bug: webrtc:7404
Change-Id: I5788d93fef00d30a523312f317dde90eb64db8de
Reviewed-on: https://chromium-review.googlesource.com/543120
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18715}
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}
We'd like to remove all occurrences of WEBRTC_TRACE and delete the
macro! One logging mechanism is enough.
NOTRY=True
Bug: webrtc:5118
Change-Id: Ic226318e0aebe3a71785fcb4ce07371872ab7128
Reviewed-on: https://chromium-review.googlesource.com/518133
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18712}
There is a dependency chain from Chromium windows main_dll to Opus
which should never exist. We used to rely on rtc_static_library
to break this chain. So this CL replaced some rtc_source_set
with rtc_static_library.
libvpx fix (https://chromium-review.googlesource.com/c/544107/) for
ios-simulator linking issue is landed and this CL can be sumbitted once the new
Chromium is rolled into WebRTC.
BUG=chromium:734631
Review-Url: https://codereview.webrtc.org/2947273002
Cr-Commit-Position: refs/heads/master@{#18709}
It can; this CL makes it a lot closer, if not all the way to a merge.
Performance from video_loopback and screenshare_loopback should now
match what we're seeing in FullStackTest, which will make debugging and
assesment of quality differences much easier.
It also adds the ability to view all of the simulcast streams at once,
in separate windows.
BUG=webrtc:7694
Review-Url: https://codereview.webrtc.org/2946893003
Cr-Commit-Position: refs/heads/master@{#18703}
Reenable pacer_exit timestamp updates for the timing frames and
exclude timing-frames carrying packets from the FEC.
BUG=webrtc:7859
Review-Url: https://codereview.webrtc.org/2947133002
Cr-Commit-Position: refs/heads/master@{#18702}
Add webrtc/rtc_base in preparation for the larger rename
in https://codereview.webrtc.org/2877023002.
This is needed in order to make downstream projects update
their dependencies (since it's not possible to use public_deps
for android_library).
BUG=webrtc:7634
NOTRY=True
Change-Id: I90fa72c9ebbc4e69bb3133413c3258432a5a72b1
Reviewed-on: https://chromium-review.googlesource.com/541215
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18699}
When building the WebRTC project for iOS, the build will fail on Xcode 9
because of a missing framework-header (AVFoundation). This pull-request
will add the missing "#import <AVFoundation/AVFoundation.h>" line to the
"RTCCameraVideoCapturer" class.
BUG=webrtc:7846
Review-Url: https://codereview.webrtc.org/2944753002
Cr-Commit-Position: refs/heads/master@{#18698}
This reverts commit bd2220a9c496ef2e8567b68d4be9435a110bdc34.
Reason for revert: Broke external clients
Original change's description:
> Support more formats in RTCVideoFrame
>
> Implement Obj-C version of webrtc::VideoFrameBuffer and use that in
> RTCVideoFrame.
>
> Bug: webrtc:7785
> Change-Id: I49f42bcf451dd6769b3a79a65fe7b400dce22677
> Reviewed-on: https://chromium-review.googlesource.com/536773
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#18691}
TBR=magjed@webrtc.org,andersc@webrtc.org
Change-Id: Id765dd9543ed0613a6b2de108b268c3501025fcd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7785
Reviewed-on: https://chromium-review.googlesource.com/542837
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18697}
And implementation class RtpStreamReceiverController.
It's responsible for demuxing, and acts as factory for
RtpStreamReceiverInterface.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2886993005
Cr-Commit-Position: refs/heads/master@{#18696}
Reason for revert:
Relanding the orginal CL. The breakage would be a flakey build.
Original issue's description:
> Revert of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2945233002/ )
>
> Reason for revert:
> The Android 32 (more config) bot is broken.
>
> Original issue's description:
> > Try to fix the binary size increase issue on Chromium.
> >
> > The target common_video used to depend on rtc_media_base which introduces
> > the dependency on p2p. This probably causes the binary size increase on Win
> > Chromium. Some chromium targets like src/media/gpu:gpu depends on common_video directly.
> >
> > BUG=chromium:734631
> >
> > Review-Url: https://codereview.webrtc.org/2945233002
> > Cr-Commit-Position: refs/heads/master@{#18693}
> > Committed: 9ed1609737
>
> TBR=kjellander@webrtc.org,deadbeef@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=chromium:734631
>
> Review-Url: https://codereview.webrtc.org/2949953003
> Cr-Commit-Position: refs/heads/master@{#18694}
> Committed: c2e208a924TBR=kjellander@webrtc.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:734631
Review-Url: https://codereview.webrtc.org/2949883003
Cr-Commit-Position: refs/heads/master@{#18695}
Reason for revert:
The Android 32 (more config) bot is broken.
Original issue's description:
> Try to fix the binary size increase issue on Chromium.
>
> The target common_video used to depend on rtc_media_base which introduces
> the dependency on p2p. This probably causes the binary size increase on Win
> Chromium. Some chromium targets like src/media/gpu:gpu depends on common_video directly.
>
> BUG=chromium:734631
>
> Review-Url: https://codereview.webrtc.org/2945233002
> Cr-Commit-Position: refs/heads/master@{#18693}
> Committed: 9ed1609737TBR=kjellander@webrtc.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:734631
Review-Url: https://codereview.webrtc.org/2949953003
Cr-Commit-Position: refs/heads/master@{#18694}
The target common_video used to depend on rtc_media_base which introduces
the dependency on p2p. This probably causes the binary size increase on Win
Chromium. Some chromium targets like src/media/gpu:gpu depends on common_video directly.
BUG=chromium:734631
Review-Url: https://codereview.webrtc.org/2945233002
Cr-Commit-Position: refs/heads/master@{#18693}
Implement Obj-C version of webrtc::VideoFrameBuffer and use that in
RTCVideoFrame.
Bug: webrtc:7785
Change-Id: I49f42bcf451dd6769b3a79a65fe7b400dce22677
Reviewed-on: https://chromium-review.googlesource.com/536773
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18691}
The implementation creates an Android hardware video decoder. It is built
around the same patterns as the HardwareVideoEncoderFactory.
This change pulls some shared code and constants into a common "utils" class.
Finally, adds an instrumentation test for the HardwareVideoDecoder.
BUG=webrtc:7760
Change-Id: Iea6eaae7727925743cb54f7c3153a6c07d62f55d
Reviewed-on: https://chromium-review.googlesource.com/536254
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18686}
Floating-point calculations are not guaranteed to happen at compile time
unless you force the issue with constexpr. This initializer was found
by running tools\win\static_initializers on a canary build
chrome_child.dll. constexpr was added to kSilenceRms for consistency.
BUG=chromium:341941
Review-Url: https://codereview.webrtc.org/2943833002
Cr-Commit-Position: refs/heads/master@{#18684}
The network stats used to be polled from the NetEq object once at the
very end of the simulation. With this change, the stats are polled
once every second, and then aggregated at the end of the run. This
leads to more meaningful numbers.
Bug: webrtc:2692
Change-Id: I9e0f4ddada2f9e42fb9234970deb1af235fffc8c
Reviewed-on: https://chromium-review.googlesource.com/541218
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18682}
This change adds an option to have neteq_rtpplay generate a Matlab
script. When executed in Matlab, the script will generate graphs with
the timing information from the test run.
The script is generated when the flag --matlabplot is passed to
neteq_rtpplay.
The CL also adds better checking and reporting about packets discarded
in the process of finding out the initial sampling rate.
Bug: webrtc:2692, webrtc:7467
Change-Id: I805e7c83b82533142b6e74bf065506e3d60a8170
Reviewed-on: https://chromium-review.googlesource.com/541276
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18680}
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}
* Change decoder thread to use new thread function type.
* Reduce the time of when video_receiver_ receives callbacks on the process thread to match with Start/Stop of the decoder.
* Not triggering shutdown unless the thread is running.
BUG=webrtc:7361
Review-Url: https://codereview.webrtc.org/2944033003
Cr-Commit-Position: refs/heads/master@{#18675}
