Starting from https://webrtc-review.googlesource.com/c/src/+/18140
(which includes 3296e11b37)
//third_party/protobuf starts to depend on //third_party/zlib.
To fix the Chromium Roll WebRTC has add the license file of
//third_party/zlib to its generate_licenses.py script in order to add
it to markdown generated license file.
Bug: None
Change-Id: If504ef00b166fdbcbe22acb0a2721bfb55624d3e
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/18244
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20566}
This reverts commit 47836b4ebb.
Reason for revert: This breaks internal tests, reverting to check if they recover.
Original change's description:
> Keep spatial_idx=kNoSpatialIdx(255) if there is no layer indices.
>
> spatial_idx is not present in RTP header if there is no temporal or
> spatial layering. But the parser sets spatial_idx to 0 in this case.
> When reflector repacketizes such packets it writes layering indices
> into outgoing packets. When packets arrive to receiver it thinks that
> it deals with multi layer stream and passes it through special path
> in Vp9 reference frame finder which never outputs inter frames.
>
> I modified the parser such that it keeps spatial_idx=kNoSpatialIdx(255)
> when there is no layer indices in RTP header. Related unit tests have
> been modified as well.
>
> Bug: none
> Change-Id: I14498cafb4e57797577dc873298c35b243479f88
> Reviewed-on: https://webrtc-review.googlesource.com/17980
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20560}
TBR=brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,ssilkin@webrtc.org
Change-Id: I67d083cf769974d8df8bd5d70942af97db578db9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/20501
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20565}
The old value was 170, but experiments have shown that 70 is better.
This will let the AGC reduce the gain further when input clipping is
detected. The effect should be less clipping, but sometimes slightly
lower signals.
In Chrome, the value 70 has already been used since June (see
https://codereview.chromium.org/2928133002).
Bug: webrtc:6622, chromium:672476
Change-Id: Ie5a60bb875eef71f303b28e096b22a8cd4b449d4
Reviewed-on: https://webrtc-review.googlesource.com/20222
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20563}
The new SurfaceEglRenderer helper class extends EglRenderer and
implements rendering on a SurfaceView.
Bug: webrtc:8242
Change-Id: Ic532fe487755d3b54c6bd03f239d714e1ecb10ad
Reviewed-on: https://webrtc-review.googlesource.com/2940
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20562}
This can occur if there are callbacks in-flight when the compression
session is destroyed. Has been observed but is rare.
Bug: webrtc:8489
Change-Id: I5d4b35c555f6ff68af48edfcc7acf53395fa86fe
Reviewed-on: https://webrtc-review.googlesource.com/18220
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20561}
spatial_idx is not present in RTP header if there is no temporal or
spatial layering. But the parser sets spatial_idx to 0 in this case.
When reflector repacketizes such packets it writes layering indices
into outgoing packets. When packets arrive to receiver it thinks that
it deals with multi layer stream and passes it through special path
in Vp9 reference frame finder which never outputs inter frames.
I modified the parser such that it keeps spatial_idx=kNoSpatialIdx(255)
when there is no layer indices in RTP header. Related unit tests have
been modified as well.
Bug: none
Change-Id: I14498cafb4e57797577dc873298c35b243479f88
Reviewed-on: https://webrtc-review.googlesource.com/17980
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20560}
The UMA metric will log the same information that goes into the
googCurrentDelayMs stat.
Bug: webrtc:8488
Change-Id: I26abb3d86a07e8c0ddb4168540a8e2458115f004
Reviewed-on: https://webrtc-review.googlesource.com/18201
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20557}
Starting from https://chromium-review.googlesource.com/750645,
Chromium started to depend on //third-party/auto in order to use
an errorprone plugin that disallows the usage of synchronized public
methods.
Bug: webrtc:8491
Change-Id: Ie9bb70520fc713dc294050c8a536ce5091b8339c
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/20002
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20556}
We need to get rid of the ones that don't take audio codec factory
arguments in order to eliminate the dependency on audio codec
implementations.
BUG=webrtc:8396
Change-Id: Id0c1c3b70c2b3479da81ba1056cc69e857e454bd
Reviewed-on: https://webrtc-review.googlesource.com/12281
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20555}
Also put Baseline profile in front of Constrained Baseline profile. The
reason is that the HW encoders are mostly BP, and we want this to be the
first codec in the list so that HW is preferred by default.
The H264 tests in chromium needs to be updated again with this change,
which was changed here: https://codereview.chromium.org/2985263002/.
Bug: webrtc:8317
Change-Id: Ief75683962b79b6664143d73b9259729c66ce082
Reviewed-on: https://webrtc-review.googlesource.com/17780
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20554}
So that we don't have to be capable of creating one ourselves, which
requires a dependency on the audio decoders.
In the process, remove the default values for the VoEBase::Init() arguments,
since there isn't a sensible default value for the audio decoder factory
anymore.
BUG=webrtc:8396
Change-Id: Idb433efa49e1a68e8206d369d27b3c255185777a
Reviewed-on: https://webrtc-review.googlesource.com/18200
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20552}
The length of the generated comfort noise is measured with a
counter. A bug in the implementation caused the counter to be reset
not only when a new packet was decoded, but also when NetEq asked the
decoder for more comfort noise without giving it a new packet to
decode. This means that the counter was reset once every 20 ms (in the
case of Opus), and it would never match the gap in timestamps that is
the exit criterion for CNG. This would have resulted in perpetual CNG,
but there is a stop-gap in NetEq. If the buffer level exceeds 4 times
the target level, CNG mode is exited anyway. This is what happens at
the end of every silence period.
With this CL, the bug should be fixed. The fix is wrapped in an
experiment, to allow verifying the fix and the impact of it with real
world data.
Bug: webrtc:8488
Change-Id: Idfc24df780eb2c55dbf08de840e6644e8557a0af
Reviewed-on: https://webrtc-review.googlesource.com/18181
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20551}
Previously input buffers would be filled incorrectly for sparsely
packed buffers where stride is not equal to the plane width.
Bug: webrtc:8478
Change-Id: I080fa3c354a27982bb996be8c1e41b103384e4bc
Reviewed-on: https://webrtc-review.googlesource.com/17321
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20550}
Follow-up on https://webrtc-review.googlesource.com/c/src/+/17784.
Adds a new PC integration test using the newly added StartRecording API.
Bug: webrtc:7313
Change-Id: Ibd59910ca5d8f8ac96cfb891f41039759a18b6f6
Reviewed-on: https://webrtc-review.googlesource.com/17940
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20549}
This is a reland of ba78b5a905
Original change's description:
> Android: Generate JNI code for VideoSink and VideoEncoder
>
> This is the first CL to start generating JNI code. It has updated two of
> the most recent classes to use JNI code generation.
>
> Bug: webrtc:8278
> Change-Id: I1b19ee78c273346ceeaa0401dbdf8696803f16c7
> Reviewed-on: https://webrtc-review.googlesource.com/3820
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#19994}
Bug: webrtc:8278
Change-Id: Id3e6513736eb87d7c234be3b0d13c5d30435201c
Reviewed-on: https://webrtc-review.googlesource.com/4500
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20548}
This ensures that on service outages we get a clear early error message instead of something cryptic later down the line
Bug: None
Change-Id: Ib637ed97144284e3744aaa948f594f5795fa9c72
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/18040
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20545}
When cinnamon is used, it always wraps the application window with its own
window. Instead of (0, 0), the DesktopRect from XWindowAttributes starts from
(10, 36).
So this change considers this difference when translating the DesktopRect in
GetWindowRect() function.
Bug: chromium:778035
Change-Id: I4944b2d1e13a4c379e114fd1749d74e95a63003b
Reviewed-on: https://webrtc-review.googlesource.com/17660
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#20538}
This class will be used for filtering remote clock offset in rtp streams.
It is a separate wrapper around PercentileFilter because it will be used
in that form in several places.
Bug: webrtc:8468
Change-Id: If1f6c38ac1ffa02232c1aed5512b92878b1c346a
Reviewed-on: https://webrtc-review.googlesource.com/17841
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20531}
At the beginning of the call, when rtt is not yet estimated, SR packets
are not used for estimation. Yet, it may happen that on some non-SR
RTCP packet RTT would become available. At that time an old SR will be
used for remote clock estimation. This will lead to remote clock offset
to the past too much.
Bug: webrtc:8468
Change-Id: I1bdbd56a7bab1c28e73987e5fb307f8e7382b045
Reviewed-on: https://webrtc-review.googlesource.com/16840
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20528}
Second attempt to land https://webrtc-review.googlesource.com/c/src/+/16180
Now removes voice_engine dependency from peerconnection and fixes a minor
const issue in NullAudioPoller.
TBR=solenberg
Bug: webrtc:7313
Change-Id: Ibfddbdc76118581e4a4dc64575203f84c1659e5c
Reviewed-on: https://webrtc-review.googlesource.com/17784
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20526}
Using layoutSubviews is a simpler solution to achieveing the desired
effect. Plus this will get rid of warrnings on iOS 11
Bug: webrtc:8467
Change-Id: Idaa041b7a0ed889905d97f645408fb3437154e73
Reviewed-on: https://webrtc-review.googlesource.com/17380
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20523}
WebRTC rolls into Chromium are failing, we should fix it ASAP.
Log:
FAILED:
obj/third_party/webrtc/modules/video_coding/webrtc_stereo/stereo_encoder_adapter.obj
ninja -t msvc -e environment.x64 -- E:\b\c\goma_client/gomacc.exe
"e:\b\c\win_toolchain\vs_files\88c3b62e1eb0893b8cd57e3f4859c3af27907f64\vc\tools\msvc\14.11.25503\bin\hostx64\x64/cl.exe"
/nologo /showIncludes
@obj/third_party/webrtc/modules/video_coding/webrtc_stereo/stereo_encoder_adapter.obj.rsp
/c
../../third_party/webrtc/modules/video_coding/codecs/stereo/stereo_encoder_adapter.cc
/Foobj/third_party/webrtc/modules/video_coding/webrtc_stereo/stereo_encoder_adapter.obj
/Fd"obj/third_party/webrtc/modules/video_coding/webrtc_stereo_cc.pdb"
../../third_party/webrtc/modules/video_coding/codecs/stereo/stereo_encoder_adapter.cc(134):
error C2220: warning treated as error - no 'object' file generated
../../third_party/webrtc/modules/video_coding/codecs/stereo/stereo_encoder_adapter.cc(134):
warning C4267: 'argument': conversion from 'size_t' to 'uint32_t',
possible loss of data
Bug: chromium:780411
Change-Id: Ia80f4551d0efeebc6d084e951f5c25e8b9401250
Reviewed-on: https://webrtc-review.googlesource.com/17781
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Max Morin <maxmorin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20522}
There is a crash happening in this neighbourhood, so adding
CHECKs to tease it out explicitly.
BUG=webrtc:8481
Change-Id: I79a2ec8fd838f4a4735a04496e363b72975919ec
Reviewed-on: https://webrtc-review.googlesource.com/17361
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20520}
