Some death tests for AEC3 cause memory leaks on trybots. This CL
temporarily disables BlockProcessor.VerifyRenderBlockSizeCheck.
Bug: webrtc:8449,webrtc:6985
Change-Id: I2900a73f7c7d5bf0e8b58a20f9a40bd5d654629a
Reviewed-on: https://webrtc-review.googlesource.com/15500
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20431}
This CL replaces 5 left shifts where the shifted value may be
negative. The shifts are replaced with equivalent multiplications.
Bug: chromium:777231, chromium:776719, chromium:776624, chromium:776286
Change-Id: Ifb27d5506eac779e60f238432bdf9e4bc5b2da4c
Reviewed-on: https://webrtc-review.googlesource.com/14800
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20430}
When audio_processing_fuzzer runs with 'DCHECK_ALWAYS_ON', it crashes
when both AEC and AECM is enabled at the same time. This change
detects that case and fixes
https://clusterfuzz.com/v2/testcase-detail/6389429496446976.
It also removes an unnecessary safeguard that didn't allow fuzzing
with 8kHz input signals.
Bug: chromium:776358
Change-Id: I33c18a2a235e50ae410f7be24637872823e432eb
Reviewed-on: https://webrtc-review.googlesource.com/15320
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20428}
Unlike all the other API directories, api/ is the root of an entire
tree of directories that are also API directories.
BUG=webrtc:8445
Change-Id: I218befe6fb6113b95599512f062ebe63abc98889
Reviewed-on: https://webrtc-review.googlesource.com/15321
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20427}
We've done this previously with the other audio encoders, but Opus had
to wait until all external users had been updated.
BUG=webrtc:7847
Change-Id: I70422d7b6c715f32a43bee88febcf6b6155e18b3
Reviewed-on: https://webrtc-review.googlesource.com/8000
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20424}
Add fine grained dropped video frames counters on sending side
4 new counters added to SendStatisticsProxy and reported to UMA and logs.
Bug: webrtc:8355
Change-Id: I1f9bdfea9cbf17cf38b3cb2f55d406ffdb06614f
Reviewed-on: https://webrtc-review.googlesource.com/14580
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20421}
Now decision between using SimulcastEncoderAdapter and using VP8 encoder
is postponed before codec is initialized for VP8 internal codecs. This is done
be new VP8EncoderProxy class. New error code for codec initialization is used
to signal that simulcast parameters are not supported.
Bug: webrtc:7925
Change-Id: I3a82c21bf5dfaaa7fa25350986830523f02c39d8
Reviewed-on: https://webrtc-review.googlesource.com/13980
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20419}
This CL balances the NLP tradeoff in AEC3 to properly handle the cases
when the echo path is so strong that it saturates the echo and when it
is so weak that the echo is very low compared to nearend.
Bug: webrtc:8411, webrtc:8412, chromium:775653
Change-Id: I5aff74dfadd51cac1ce71b1cb935d68a5be6918d
Reviewed-on: https://webrtc-review.googlesource.com/14120
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20418}
system_wrapper/Atomic32 has been deprecated (which is already just a
wrapper of std::atomic) in favor of platform-independent std::atomic
from C++11. This CL replaces all use of Atomic32 in video/
Bug: webrtc:8428
Change-Id: If4dab4909df06944c009e7b70141f58daef7be10
Reviewed-on: https://webrtc-review.googlesource.com/14720
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Yuwei Huang <yuweih@google.com>
Cr-Commit-Position: refs/heads/master@{#20417}
This is a reland of Id63f0deb7b335690157ab157c35177b7836688da.
Original change's description:
> Enable the clang style plugin in rtc_base/
>
> Enabled the plugin and cleaned up all issues it found.
>
> Bug: webrtc:163
> Change-Id: Id63f0deb7b335690157ab157c35177b7836688da
> Reviewed-on: https://webrtc-review.googlesource.com/14660
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20401}
Bug: webrtc:163
Change-Id: I861a5fe741215115b0e7a2be9c0786836ff5376e
Reviewed-on: https://webrtc-review.googlesource.com/15040
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20416}
The DataChannel OPEN message is sent in-band in this test, and waiting
for the signaling state to change is not sufficient to guarantee that the
callee received this message
Bug: webrtc:8443
Change-Id: I76fa6348b6f8e1e70fb41a4e644aee805b2ef4de
Reviewed-on: https://webrtc-review.googlesource.com/15060
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20415}
RtcpTransceiver name reserved for thread-safe version that planned to
be wrapper of the RtcpTransceiverImpl
BUG=webrtc:8239
Change-Id: If8a3092eb1b8e4175e3efd23b52e1043cdabf19f
Reviewed-on: https://webrtc-review.googlesource.com/7920
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20414}
WebRtcSession is being merged into PeerConnection, and to make the
code review easier this is the first step towards achieving that.
Bug: webrtc:8323
Change-Id: I33778e46f20cb14089dff4328947868e207476bd
Reviewed-on: https://webrtc-review.googlesource.com/8760
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20413}
Originally, the idea was to implement QUIC data channels as a
PeerConnection API. Now, the effort has shifted to implementing it as a
part of ORTC which will live in Chromium. Since this code has not been
maintained and is not currently being used, remove it to reduce
maintenance overhead while a copy will be retained in the Git history.
Bug: webrtc:8385
Change-Id: I2719c007a0de0118b67d41a425f900b66c52f65a
Reviewed-on: https://webrtc-review.googlesource.com/14100
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20412}
This CL adds an EncodedFrameChecker interface which can be used by users
of the VideoProcessor to inject customized per-frame checks to the
encoding/decoding pipeline. This currently has two uses:
- Verifying that the QP parser works correctly for VP8 and VP9, by comparing the
parsed QP to that produced by libvpx.
- Verifying that our H.264 encoders always produce SPS/PPS/IDR in tandem.
TESTED=Galaxy S8, Pixel 2 XL, iPhone 7.
BUG=webrtc:8423
Change-Id: Ic3e401546e239a9ffaf2ed2907689cebb1127805
Reviewed-on: https://webrtc-review.googlesource.com/14559
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20409}
This is protected behind a field trial, for controlled rollout.
TESTED=MediaCodec (Qualcomm + Exynos) and VideoToolbox senders.
BUG=webrtc:8423
Change-Id: Ibccefb3d374e4a44461d33e77eff754d8d752666
Reviewed-on: https://webrtc-review.googlesource.com/13863
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20408}
In the legacy C part of AGC, an audio level 'cur_level' is represented as
(1+frac) * 2^(31 - zeros)
The 'zeros' exponent part is used for looking up a gain value in a
table, and 'frac' is used for interpolating between two nearby table
values. Code snippet below:
zeros = WebRtcSpl_NormU32((uint32_t)cur_level);
tmp32 = (cur_level << zeros) & 0x7FFFFFFF;
frac = (int16_t)(tmp32 >> 19);
In the second line, 'cur_level' is shifted upwards so that the leading
bit is '1', after which the leading bit is cleared. The result is
'frac' in Q31.
The compiler type of 'cur_level << zeros' is 'int32_t'. This is a
fuzzer error 'Left shift cannot be represented in int32_t',
because the leading sign bit is 1. This CL changes the compiler type to
uint32_t.
Bug: chromium:776286
Change-Id: Ie29552b75e690057bd76fc88e747841b531e3802
Reviewed-on: https://webrtc-review.googlesource.com/14841
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20405}
Alternative VAD based on the existing one in WebRTC.
It is used to extract VAD annotations in APM-QA.
TBR=
Bug: webrtc:7494
Change-Id: I6af412742f804631ad4f3ba3ccf71a30d74de984
Reviewed-on: https://webrtc-review.googlesource.com/14553
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20404}
In SrtpTransport::SetRtcpParams, send_rtcp_session_ should really call
SetSend rather than SetRecv.
Modified the LOG message in SrtpTransport::SetRtpParams.
Bug: webrtc:8436
Change-Id: Iccbfbc5ef2d4f4ebd5f876c3f6dcc81671fdc631
Reviewed-on: https://webrtc-review.googlesource.com/14562
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20403}
This reverts commit af551a1956d2ec3a388cf7e0e88c7ee2c2b61291.
Reason for revert: Fails to compile on Chromium FYI bot: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/21375
Original change's description:
> Enable the clang style plugin in rtc_base/
>
> Enabled the plugin and cleaned up all issues it found.
>
> Bug: webrtc:163
> Change-Id: Id63f0deb7b335690157ab157c35177b7836688da
> Reviewed-on: https://webrtc-review.googlesource.com/14660
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20401}
TBR=steveanton@webrtc.org,kwiberg@webrtc.org
Change-Id: Iafdf4bc1744a981b5d7d38e4a0c5b2d88753f00a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:163
Reviewed-on: https://webrtc-review.googlesource.com/14740
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20402}
Enabled the plugin and cleaned up all issues it found.
Bug: webrtc:163
Change-Id: Id63f0deb7b335690157ab157c35177b7836688da
Reviewed-on: https://webrtc-review.googlesource.com/14660
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20401}
This is a reland of b49b66109ea8a0a33a3192ebccf91366af2e49ae.
Original change's description:
> Rewrite WebRtcSession BUNDLE tests as PeerConnection tests
>
> Bug: webrtc:8222
> Change-Id: Id47e4544dc073564ad7e63d02865ca80dd5a85ff
> Reviewed-on: https://webrtc-review.googlesource.com/8280
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20365}
Bug: webrtc:8222
Change-Id: If3dcd8090875c641881e2b9e92fc1db387ba1de5
Reviewed-on: https://webrtc-review.googlesource.com/14400
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20397}
In addition to NV12 frames, also support cropping/scaling RGB frames and
converting RGB frames to i420.
This CL also removes the hardcoding of pixel format in
RTCCameraVideoCapturer. Instead, use the first available format for the
output device that our pipeline supports.
Bug: webrtc:8351
Change-Id: If479b4934c47cd2994936913f55e60fbbee3893b
Reviewed-on: https://webrtc-review.googlesource.com/8920
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20396}
This CL includes the patch from oprypin@webrtc.org, which is also applied
to the MIPS code (also affected), and the protobuf for ApmTest.Process
(audio_processing_unittest.cc), which used when WEBRTC_AUDIOPROC_FIXED_PROFILE
is set.
This change has been tested on mobile platforms.
Bug: webrtc:8200
Change-Id: Ic50a5ab57c16551397756b1fb473e1067b8e7ece
Reviewed-on: https://webrtc-review.googlesource.com/10811
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20394}
Eliminates manual reference counting logic, and one of the few
remaining uses of the Atomic32 type.
Bug: webrtc:8270
Change-Id: Ibbbf227c710f7e8f68a98bac46d3e24b7cb5ee2f
Reviewed-on: https://webrtc-review.googlesource.com/14600
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20393}
The 'parametricNoise' field is never initialized in the
'WebRtcNs_InitCore' function that initializes a 'NoiseSuppressionC'
struct.
This leads to use of unititialized value, which may affect the audio
output and result of the noise suppressor.
The issue was found by the Chrome fuzzer:
https://clusterfuzz.com/v2/testcase-detail/4749034115039232
Bug: chromium:776673
Change-Id: I1c3fd80cff178f2d5917064ad07f88c7b9a29e7d
Reviewed-on: https://webrtc-review.googlesource.com/14556
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20388}
UBSan will trigger when shifting a negative value. This change avoids
that by replacing "x << 8" with "x * (1 << 8)".
Bug: chromium:666877
Change-Id: Ic89bd98e5a3feff35075df96b104b386cb4d8803
Reviewed-on: https://webrtc-review.googlesource.com/14552
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20387}
To make testing easier all of PacketQueues functions have been made virtual,
and PacketQueue2 now inherits PacketQueue. This change was made to minimize
changes in PacedSender.
Bug: webrtc:8287, webrtc:8288
Change-Id: I2593340e7cc7da617370b0a33e7b9deeb46d9487
Reviewed-on: https://webrtc-review.googlesource.com/9380
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20385}
It was used only in tests.
Bug: webrtc:8422
Change-Id: I67b58663c171202240d1c5a7c230d6cd4cd6149b
Reviewed-on: https://webrtc-review.googlesource.com/13102
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20382}
