WebRTC standalone tests show 24% speed up on foreman_cif, 16% speed up
on a 240p clip and 11% speed up on Bridge_r180.
Bug: None
Change-Id: I433b7a8841bd9df2402575f72dd1984cc5e011a9
Reviewed-on: https://webrtc-review.googlesource.com/29260
Commit-Queue: Jerome Jiang <jianj@google.com>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21096}
A lot of WebRTC targets were depending on //third_party/libyuv using
public_deps instead of deps. This causes issues because a the
inclusion of libyuv headers is not declared to the build system and
this creates hidden dependencies that put the modularity of the project
at risk.
Bug: webrtc:8603
Change-Id: Ide0ceb84eb5640ae664dc782f3a722b55c3b601a
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/28120
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21039}
This will also cause us to use the new Android HardwareVideoEncoder,
instead of the deprecated MediaCodecVideoEncoderFactory. Unfortunately,
the new HW encoder does not seem to work as good as the old (or the new
encoder is more strict with return values or something). I don't think
it adds much value to continue testing the deprecated encoder, so I
filed a bug for fixing the new encoder, and in this CL I disabled the
tests on Android. I want to remove as many places as possible where we
use the old WebRtcVideoEncoderFactory interface, because it makes it
more difficult to migrate to the new interface.
Bug: webrtc:7925
Change-Id: If8e34752148a5e5139944d2dfbe7e231fe58aeb9
Reviewed-on: https://webrtc-review.googlesource.com/27540
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21037}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=sprang@webrtc.org
Bug: None
Change-Id: Ic429f28a8610ca798e29c45ec1f64604d6f9687f
Reviewed-on: https://webrtc-review.googlesource.com/23603
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20955}
This is a reland of 20f2133d5d
Original change's description:
> Add stereo codec header and pass it through RTP
>
> - Defines CodecSpecificInfoStereo that carries stereo specific header info from
> encoded image.
> - Defines RTPVideoHeaderStereo that carries the above info to packetizer,
> see module_common_types.h.
> - Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
> header.
> - Uses new data containers in StereoAdapter classes.
>
> This CL is the step 3 for adding alpha channel support over the wire in webrtc.
> See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
> CL that gives an idea about how it will come together.
> Design Doc: https://goo.gl/sFeSUT
>
> Bug: webrtc:7671
> Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
> Reviewed-on: https://webrtc-review.googlesource.com/22900
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20920}
TBR=danilchap@webrtc.org, sprang@webrtc.org, niklas.enbom@webrtc.org
Bug: webrtc:7671
Change-Id: If8f0c7e6e3a2a704f19161f0e8bf1880906e7fe0
Reviewed-on: https://webrtc-review.googlesource.com/27160
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20946}
This reverts commit 20f2133d5d.
Reason for revert: Breaks downstream project.
Original change's description:
> Add stereo codec header and pass it through RTP
>
> - Defines CodecSpecificInfoStereo that carries stereo specific header info from
> encoded image.
> - Defines RTPVideoHeaderStereo that carries the above info to packetizer,
> see module_common_types.h.
> - Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
> header.
> - Uses new data containers in StereoAdapter classes.
>
> This CL is the step 3 for adding alpha channel support over the wire in webrtc.
> See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
> CL that gives an idea about how it will come together.
> Design Doc: https://goo.gl/sFeSUT
>
> Bug: webrtc:7671
> Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
> Reviewed-on: https://webrtc-review.googlesource.com/22900
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20920}
TBR=danilchap@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,niklas.enbom@webrtc.org,emircan@webrtc.org
Change-Id: I57f3172ca3c60a84537d577a574dc8018e12d634
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/26940
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20931}
- Defines CodecSpecificInfoStereo that carries stereo specific header info from
encoded image.
- Defines RTPVideoHeaderStereo that carries the above info to packetizer,
see module_common_types.h.
- Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
header.
- Uses new data containers in StereoAdapter classes.
This CL is the step 3 for adding alpha channel support over the wire in webrtc.
See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
CL that gives an idea about how it will come together.
Design Doc: https://goo.gl/sFeSUT
Bug: webrtc:7671
Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
Reviewed-on: https://webrtc-review.googlesource.com/22900
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20920}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
Bug: None
Change-Id: Iedebf4dc56a973306e7d7e7649525879808dc72b
Reviewed-on: https://webrtc-review.googlesource.com/23578
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20878}
Calculation of quality metrics required writing of decoded video
to file. There were two drawbacks with that approach. First, frame
drops significantly affected metrics because comparison was done
against the last decoded frame. Second, simulcast/SVC required
writing of multiple files. This might be too much data to dump.
On-fly metrics calculation is done in frame decoded callback.
Calculation time is excluded from encoding/decoding time. If CPU
usage measurement is enabled metrics calculation is disabled since
it affects CPU usage. The results are reported in Stats::PrintSummary.
Bug: webrtc:8524
Change-Id: Id54fb21f2f95deeb93757afaf46bde7d7ae18dac
Reviewed-on: https://webrtc-review.googlesource.com/22560
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20798}
Because some hardware encoders may not preserve capture timestamp, use
rtp timestamp as a key to capture time storage and restore corrupted
capture timestamps whenever possible.
Bug: webrtc:8497
Change-Id: Ib3449313c65e864a89cd645a3f5dec99aa9cf8e4
Reviewed-on: https://webrtc-review.googlesource.com/23620
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20732}
Ran into these when trying a newer libstdc++
Bug: None
Change-Id: Ie3ce0ae1ae1e6da1a15476fbf942b48b37adc9fa
Reviewed-on: https://webrtc-review.googlesource.com/23501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20701}
VideoEncoderSoftwareFallbackWrapper is updated to take a VideoEncoder as
argument instead relying on built-in SW codecs. The purpose is to make
VideoEncoderSoftwareFallbackWrapper more modular and not depend on
built-in SW encoders.
Bug: webrtc:7925
Change-Id: I99896f0751cfb77e01efd29c97d3bd07bdb2c7c0
Reviewed-on: https://webrtc-review.googlesource.com/22320
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20671}
Conditional visibility is complex to maintain and it is not well
supported by other build systems.
This CL removes it and falls back on the more relaxed visibility value
("*" in this case).
It is not a problem because the targets that are using conditional
visibility are all marked as "testonly" and this is probably enough to
keep the build graph clean.
Bug: None
Change-Id: I2d2b5ac9a02d08c2863950116db455976ee1459c
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/14902
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20658}
This means we can properly declare the dependency between
libjingle_peerconnection_api and video_frame_api. i420
pulls in system_wrappers, which can't be a dependency of
the public API.
Plan:
1) Land this CL + send out PSA
2) Make all direct users of i420_buffer depend on the
new video_frame_api_i420 target
3) Move i420_buffer.cc to the new target
4) Make libjingle_peerconnection_api depend on
video_frame_api, since it no longer contains i420 code
Bug: webrtc:7504
Change-Id: I30d90f2ac7af53748859bbde8aed92386d5501f9
Reviewed-on: https://webrtc-review.googlesource.com/9382
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20656}
- Add alpha accessors to PlanarYuvBuffer interface, null by defualt.
- Add WrapI420ABuffer() that creates a container which implements these
accessors.
- Show the use via StereoDecoderAdapter.
This CL is the step 2 for adding alpha channel support over the wire in webrtc.
See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
CL that gives an idea about how it will come together.
Design Doc: https://goo.gl/sFeSUT
Bug: webrtc:7671
Change-Id: Id5691cde00088ec811b63d89080d33ad2d6e3939
Reviewed-on: https://webrtc-review.googlesource.com/21130
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20635}
This excludes these frames from being counted as dropped by encoder.
Also fix bitrate projected distribution for vp9 svc for outliers
detection.
Bug: webrtc:8497
Change-Id: Id37487456170c61e2323a660668f0c319ea5831d
Reviewed-on: https://webrtc-review.googlesource.com/21223
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20632}
Even if we're not going to transmit any timing info over the wire.
Bug: webrtc:8504
Change-Id: Id54192a10e6b2a6a2cb57a2ff6b28fc0d16e471d
Reviewed-on: https://webrtc-review.googlesource.com/21160
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20628}
This reverts commit 267d84baf0.
Reason for reland: Fix the bug; decoder is not allowed to ever be null and we need to use a
NullVideoDecoder that ignores calls instead.
Original change's description:
> Revert "Update internal video decoder factory to new interface"
>
> This reverts commit b2fc9b1b10.
>
> Reason for revert: Suspected to cause failures on Android bots on webrtc.fyi, see https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/builds/21051
>
> Original change's description:
> > Update internal video decoder factory to new interface
> >
> > We want to move away from cricket::WebRtcVideoDecoderFactory and this CL
> > updates the internal factory. Also, VideoDecoderSoftwareFallbackWrapper
> > is updated to take a VideoDecoder as argument instead of a factory so it
> > can be used with external SW decoders.
> >
> > Bug: webrtc:7925
> > Change-Id: Ie6dc6c24f8610a2129620c6e2b42e3cebb2ddef7
> > Reviewed-on: https://webrtc-review.googlesource.com/7301
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20597}
>
> TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org
>
> Change-Id: I0a12c98fdc30f00d58c85ee7e088f50160d39724
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:7925
> Reviewed-on: https://webrtc-review.googlesource.com/21420
> Reviewed-by: Christian Fremerey <chfremer@webrtc.org>
> Commit-Queue: Christian Fremerey <chfremer@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20605}
TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org,chfremer@webrtc.org,chfremer@google.com
Change-Id: I6cf5794dc3fadfa86809a94da80b69dbb4c56f52
No-Try: true
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/21541
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20623}
This reverts commit b2fc9b1b10.
Reason for revert: Suspected to cause failures on Android bots on webrtc.fyi, see https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/builds/21051
Original change's description:
> Update internal video decoder factory to new interface
>
> We want to move away from cricket::WebRtcVideoDecoderFactory and this CL
> updates the internal factory. Also, VideoDecoderSoftwareFallbackWrapper
> is updated to take a VideoDecoder as argument instead of a factory so it
> can be used with external SW decoders.
>
> Bug: webrtc:7925
> Change-Id: Ie6dc6c24f8610a2129620c6e2b42e3cebb2ddef7
> Reviewed-on: https://webrtc-review.googlesource.com/7301
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20597}
TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org
Change-Id: I0a12c98fdc30f00d58c85ee7e088f50160d39724
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/21420
Reviewed-by: Christian Fremerey <chfremer@webrtc.org>
Commit-Queue: Christian Fremerey <chfremer@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20605}
Deprecated avcodec_decode_video2 is replaced with
avcodec_send_packet and avcodec_receive_frame.
https://www.ffmpeg.org/doxygen/3.1/group__lavc__encdec.html
Setting av_context_->refcounted_frames=1 is removed. This
is automatically enabled if avcodec_receive_frame us used.
Bug: webrtc:8493
Change-Id: Id1d2b1315717f4553fb7fc182197f9429bd31daf
Reviewed-on: https://webrtc-review.googlesource.com/18462
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20601}
We want to move away from cricket::WebRtcVideoDecoderFactory and this CL
updates the internal factory. Also, VideoDecoderSoftwareFallbackWrapper
is updated to take a VideoDecoder as argument instead of a factory so it
can be used with external SW decoders.
Bug: webrtc:7925
Change-Id: Ie6dc6c24f8610a2129620c6e2b42e3cebb2ddef7
Reviewed-on: https://webrtc-review.googlesource.com/7301
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20597}
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}
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}
This CL is the step 1 for adding alpha channel support over the wire in webrtc.
- Add the footprint for adapter classes that wraps actual codecs.
- This CL does not add a webrtc::VideoFrame container that can carry alpha to
make the CL shorter for an easier review. Therefore, it exercises a code path
for when we receive no alpha input, just regular I420 frames.
- Unittest sends a video frame for encode/decode through these adapters and
checks the output PSNR.
- See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
CL that gives an idea about how it will come together.
Design Doc: https://goo.gl/sFeSUT
Bug: webrtc:7671
Change-Id: I9d3be13647a0a958feceb8d7a9aa93852fc6a1fa
Reviewed-on: https://webrtc-review.googlesource.com/11841
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20490}
Always enabling verbose mode means about 100% more text is printed,
but this should not be a problem as the only time that we explicitly
look at the logs is when the bots are failing, or when we want to save
all output for plotting.
BUG=webrtc:8448
Change-Id: Ia5feab5220d047440d15cddb7d3fbca1c5a4aaf5
Reviewed-on: https://webrtc-review.googlesource.com/16140
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20461}
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 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}
4 new counters added to SendStatisticsProxy and reported to UMA and logs.
Bug: webrtc:8355
Change-Id: Idf9b8dfc295c92821e058a97cb3894dc6a446082
Reviewed-on: https://webrtc-review.googlesource.com/12260
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20347}
TBR=sprang@webrtc.org
This is a reland of af721b72cc
Original change's description:
> Remove sent framerate and bitrate calculations from MediaOptimization.
>
> Add RateTracker for sent framerate and bitrate in SendStatisticsProxy.
>
> Store sent frame info in map to solve potential issue where sent framerate statistics could be
> incorrect.
>
> Bug: webrtc:8375
> Change-Id: I4a6e3956013438a711b8c2e73a8cd90c52dd1210
> Reviewed-on: https://webrtc-review.googlesource.com/7880
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20225}
Bug: webrtc:8375
Change-Id: I06ea90ae8646ba11ddd8ddceb82ea82d75ae2109
Reviewed-on: https://webrtc-review.googlesource.com/11320
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20308}
Since WEBRTC_ANDROID is defined by WebRTC while ANDROID is defined by
Chromium we should stop using ANDROID in WebRTC source code.
Bug: webrtc:8400
Change-Id: I1d59caaabd8af2423e86476b72e0e9185e6c7a3a
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/10805
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20306}
This reverts commit af721b72cc.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Remove sent framerate and bitrate calculations from MediaOptimization.
>
> Add RateTracker for sent framerate and bitrate in SendStatisticsProxy.
>
> Store sent frame info in map to solve potential issue where sent framerate statistics could be
> incorrect.
>
> Bug: webrtc:8375
> Change-Id: I4a6e3956013438a711b8c2e73a8cd90c52dd1210
> Reviewed-on: https://webrtc-review.googlesource.com/7880
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20225}
TBR=asapersson@webrtc.org,sprang@webrtc.org
Change-Id: Ic914f03ff7065ac410ae06b6f82b56a935399b66
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8375
Reviewed-on: https://webrtc-review.googlesource.com/8480
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20248}
Add RateTracker for sent framerate and bitrate in SendStatisticsProxy.
Store sent frame info in map to solve potential issue where sent framerate statistics could be
incorrect.
Bug: webrtc:8375
Change-Id: I4a6e3956013438a711b8c2e73a8cd90c52dd1210
Reviewed-on: https://webrtc-review.googlesource.com/7880
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20225}
Care should be taken when landing this, because it will affect users of
WebRTC. I'm thinking primarily of Chromium. Chromium will start to
support High profile and Baseline profile using SW codecs with this CL.
Clients who do SDP munging without looking at the H264 profile might
switch from Constrained Baseline to High profile with this change.
Bug: webrtc:8317
Change-Id: Idca3a6b761a66d9e521b913b850c6ae14381f1f4
Reviewed-on: https://webrtc-review.googlesource.com/6341
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20190}
This is in preparation for deleting the include in rtc_base/refcount.h,
but that change has to wait for some downstream applications to
not rely in the indirect include.
Partial reland of "Make rtc_base/refcount.h self contained, not including refcountedobject.h."
This is a reland of b7239a9dc8
Original change's description:
> Make rtc_base/refcount.h self contained, not including refcountedobject.h.
>
> The refcount.h file doesn't depend on anything from
> refcountedobject.h. The motivation of this change to make it possible
> to add additional declarations to refcount.h, and include it from
> refcountedobject.h.
>
> Bug: webrtc:8270
> Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
> Reviewed-on: https://webrtc-review.googlesource.com/5760
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20106}
Bug: webrtc:8270
Change-Id: I63a42712f6c1ec83823c629d1a954fd1a04d4a6c
Reviewed-on: https://webrtc-review.googlesource.com/7281
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20185}
This is a reland of b7239a9dc8
Original change's description:
> Make rtc_base/refcount.h self contained, not including refcountedobject.h.
>
> The refcount.h file doesn't depend on anything from
> refcountedobject.h. The motivation of this change to make it possible
> to add additional declarations to refcount.h, and include it from
> refcountedobject.h.
>
> Bug: webrtc:8270
> Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
> Reviewed-on: https://webrtc-review.googlesource.com/5760
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20106}
Bug: webrtc:8270
Change-Id: I9738f6680ab52d0f43639a1a39175fdba5957681
Reviewed-on: https://webrtc-review.googlesource.com/5840
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20180}
This is expected to result in a slight loss of overall quality, but
should offset by quicker switching between temporal layers with flaky
connections.
Bug: webrtc:7694
Change-Id: Ib605802bb59f12773652324ac66cdf4774ae6bb6
Reviewed-on: https://webrtc-review.googlesource.com/6881
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20160}
This reverts commit b7239a9dc8.
Reason for revert: Broke chromium mac build, compilation failures on content/renderer/media/webrtc/webrtc_video_frame_adapter.h.
Original change's description:
> Make rtc_base/refcount.h self contained, not including refcountedobject.h.
>
> The refcount.h file doesn't depend on anything from
> refcountedobject.h. The motivation of this change to make it possible
> to add additional declarations to refcount.h, and include it from
> refcountedobject.h.
>
> Bug: webrtc:8270
> Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
> Reviewed-on: https://webrtc-review.googlesource.com/5760
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20106}
TBR=kwiberg@webrtc.org,nisse@webrtc.org
Change-Id: I7334597cc8979ba9cfaff526967084349ef27f3c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8270
Reviewed-on: https://webrtc-review.googlesource.com/5800
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20107}
The refcount.h file doesn't depend on anything from
refcountedobject.h. The motivation of this change to make it possible
to add additional declarations to refcount.h, and include it from
refcountedobject.h.
Bug: webrtc:8270
Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
Reviewed-on: https://webrtc-review.googlesource.com/5760
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20106}
This CL is a clean-up to prepare for adding more supported codecs for the internal H264 SW codec.
Bug: webrtc:8317
Change-Id: If483d05c01c40bbc81cbd1a6aad89961689714ef
Reviewed-on: https://webrtc-review.googlesource.com/4940
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20105}
All frames are checked against hard-coded dependency graph
using new helper class. It's invoked in RTC_DCHECK(). Only
DefaultTemporalLayers are fully implemented in this CL, checker
for ScreenshareLayers is not doing anything for now.
Bug: none
Change-Id: I3ec017176d8c25f7572c8f161e52f2ebfac8220f
Reviewed-on: https://webrtc-review.googlesource.com/3740
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20066}
Test enables single-nalu mode, sets limit for nalu lenght and verifies
that encoder follows that limit.
I found that QP jumps significantly when the mode is enabled. In result
encoder might produce 4kbyte and 0.4kbyte frames back-to-back. But it
seems that happens only to couple of frames in the beginning. This
caused test to fail with default RC thresholds. To bypass this I
increased frame size mismatch threshold from 20 to 30%. This should be
Ok considering single-nalu mode is rare.
BUG=webrtc:8070
Review-Url: https://codereview.webrtc.org/3014623002
Cr-Commit-Position: refs/heads/master@{#20023}
Test was Android-only, so it was disabled completely.
TBR=brandtr@webrtc.org
Bug: webrtc:8280
Change-Id: Id45eedac90fb892f5a380e5c2614037e01ee8c76
Reviewed-on: https://webrtc-review.googlesource.com/3460
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19954}
Reason for revert:
Fixes has landed.
Original issue's description:
> Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
>
> Reason for revert:
> We are not certain this is the behavior we want.
>
> Original issue's description:
> > Fix the video buffer size should take rtt into consideration
> >
> > BUG=webrtc:8010
> >
> > Review-Url: https://codereview.webrtc.org/2980413002
> > Cr-Commit-Position: refs/heads/master@{#19285}
> > Committed: f1e08d0b58
>
> TBR=sprang@webrtc.org,gustavogb@gmail.com
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:8010
>
> Review-Url: https://codereview.webrtc.org/3002033002
> Cr-Commit-Position: refs/heads/master@{#19442}
> Committed: bdbc8895f3TBR=sprang@webrtc.org,gustavogb@gmail.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:8010
Review-Url: https://codereview.webrtc.org/3016633002
Cr-Commit-Position: refs/heads/master@{#19944}
- Group member variables into two structs: target rates/actual rates.
- Split verify and print of rate control metrics into separate functions.
- Rename member variables.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/3009423002
Cr-Commit-Position: refs/heads/master@{#19925}
* Add audio_ prefix to CallTest::{en,de}coder_factory_.
* Let VideoQualityTest only instantiate encoders using encoder factories.
* Add HW encoder factories to VideoQualityTest.
* Add full stack tests:
- sqcif7 at 30 kbps: MediaCodec and libvpx.
- 240p10 at 100 kbps: MediaCodec, libvpx, and MediaCodec+libvpx.
BUG=webrtc:8219
Review-Url: https://codereview.webrtc.org/3005253002
Cr-Commit-Position: refs/heads/master@{#19923}
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.
The cpplint complaint is:
Include the directory when naming .h files [build/include] [4]
This CL disables the error but we have to remove these two headers
from the root directory.
NOPRESUBMIT=true
Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
Making a long chain of interface changes to route a CodecSpecificInfo
struct from the video encoder function to the RTPSenderVideo. This
will be used to convey information needed by the RTP packetizer when
building the RTP headers.
Review URL: http://webrtc-codereview.appspot.com/56001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@140 4adac7df-926f-26a2-2b94-8c16560cd09d
Added metrics averaged over intervals of the loss/bandwidth reports, to be used for adjustment of robustness settings. Separated this set
from the (global) metrics used for resolution adaptation.
Some code cleanup in content_metrics.cc/.h.
Review URL: http://webrtc-codereview.appspot.com/52002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@125 4adac7df-926f-26a2-2b94-8c16560cd09d
Use get/set reference frames to realize a decoder cloning. Must
also inject the latest keyframe. Note: this CL does not work with
the Bali release of libvpx. Must apply the bug fix in commit fbea3728.
Review URL: http://webrtc-codereview.appspot.com/32004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@67 4adac7df-926f-26a2-2b94-8c16560cd09d
The Reset function was modified so that the encoder is destroyed
and recreated on reset. Initialization of the encoder and setting
of the encoder speed is now done in a private method, to avoid
code duplication. (It is used both in InitEncode and in Reset.)
This change is needed to make the unit tests pass with newer
versions of libvpx.
Review URL: http://webrtc-codereview.appspot.com/33004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@56 4adac7df-926f-26a2-2b94-8c16560cd09d
