Incorrect length parameter was passed to memset (lenght of array in
elements instead of length in bytes, which is 8 times more since int64
is used).
Bug: none
Change-Id: I9100d1986377a8b3b9e475d1fbc215f4a1dedfb1
Reviewed-on: https://webrtc-review.googlesource.com/44280
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21771}
This CL changes some types in MultiplexImage and MultiplexImageComponent. Also,
adds unit test coverage in TestMultiplexAdapter for these structs.
Bug: webrtc:7671
Change-Id: I832d0466dc67d3b6b7fa0d3fb76f02c0190e474f
Reviewed-on: https://webrtc-review.googlesource.com/44081
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Qiang Chen <qiangchen@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21770}
This CL only does the rename from"stereo" to multiplex". With this we have a
better name that doesn't clash with audio's usage of stereo.
Bug: webrtc:7671
Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
Reviewed-on: https://webrtc-review.googlesource.com/43242
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21769}
This is in preparation for
https://webrtc-review.googlesource.com/c/src/+/36340
With these changes we can avoid some strange #ifdefs in the code
that uses temporal layers.
Bug: webrtc:7925
Change-Id: I472210738ccc9f73812b8863951befeabec56f15
Reviewed-on: https://webrtc-review.googlesource.com/41280
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21759}
This CL is a followup to https://webrtc-review.googlesource.com/c/src/+/38481.
With the new approach we can just use the generic RTP packetizer to pass frames
over the wire as the specific info is contained within the bitstream. This makes
the new codec more modular and reduces its footprint.
Bug: webrtc:7671
Change-Id: Ib07f72a9d338e3cbfdbf39cba9891a959b5f7552
Reviewed-on: https://webrtc-review.googlesource.com/43220
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21753}
This reverts commit d756fd06fe.
Original change's description:
> Revert "Wrap Alpha and YUV frame into one EncodedImage for transmission"
>
> This reverts commit 5670c86aec.
>
> Reason for revert: Breaks downstream build. Need to add "#include <cstring>" to stereo_encoder_adapter.cc to use std::memcpy.
>
> Original change's description:
> > Wrap Alpha and YUV frame into one EncodedImage for transmission
> >
> > With alpha channel, we observe the artifacts on the receiver side, and
> > the reason is that when YUV channel has a key frame, it gives frame_buffer2
> > a chance to drop some previous frames. Then it is possible that some alpha
> > frames got dropped, which break the alpha frame dependence chain.
> >
> > In this CL, we pack the YUV frame and alpha encoded frame together as one
> > entity to solve the issue.
> >
> > Bug: webrtc:8773
> > Change-Id: Ibe746a46cb41fd92b399a7069e1d89f02f292af7
> > Reviewed-on: https://webrtc-review.googlesource.com/38481
> > Commit-Queue: Qiang Chen <qiangchen@chromium.org>
> > Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21737}
>
> TBR=qiangchen@chromium.org,emircan@webrtc.org
>
> Change-Id: I11eff814ce093bf6db327ebcd21b1b71a1929849
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8773
> Reviewed-on: https://webrtc-review.googlesource.com/43260
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21739}
TBR=deadbeef@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org
Change-Id: I0d64b7e7a62e4f35aa012270d3826a23b3fb2337
Bug: webrtc:8773
Reviewed-on: https://webrtc-review.googlesource.com/43440
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Reviewed-by: Qiang Chen <qiangchen@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21749}
Each simulcast stream requires dedicated decoder for decoding. SVC
can be decoded by single decoder. But in prod each receiver has its
decoder. We want to replicate this and also use one decoder per
spatial layer.
Also we create one frame writer per simulcast/spatial layer to dump
encoded/decoded frames of different layers to separate files.
Note that videoprocessor is still initialized with single
decoder/writer. It will be updated in next CL and start using
separate decoder/writer per layer.
Bug: webrtc:8524
Change-Id: I3bb3de77f97d51138b8b7675dd01bc281a078b2f
Reviewed-on: https://webrtc-review.googlesource.com/43280
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21744}
These parameters allow to configure number of simulcast/spatial layers
in video codec tests.
Bug: webrtc:8524
Change-Id: Iad1332732758a8297abcf740c24c483e5fccec9a
Reviewed-on: https://webrtc-review.googlesource.com/43020
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21741}
This reverts commit 5670c86aec.
Reason for revert: Breaks downstream build. Need to add "#include <cstring>" to stereo_encoder_adapter.cc to use std::memcpy.
Original change's description:
> Wrap Alpha and YUV frame into one EncodedImage for transmission
>
> With alpha channel, we observe the artifacts on the receiver side, and
> the reason is that when YUV channel has a key frame, it gives frame_buffer2
> a chance to drop some previous frames. Then it is possible that some alpha
> frames got dropped, which break the alpha frame dependence chain.
>
> In this CL, we pack the YUV frame and alpha encoded frame together as one
> entity to solve the issue.
>
> Bug: webrtc:8773
> Change-Id: Ibe746a46cb41fd92b399a7069e1d89f02f292af7
> Reviewed-on: https://webrtc-review.googlesource.com/38481
> Commit-Queue: Qiang Chen <qiangchen@chromium.org>
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21737}
TBR=qiangchen@chromium.org,emircan@webrtc.org
Change-Id: I11eff814ce093bf6db327ebcd21b1b71a1929849
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8773
Reviewed-on: https://webrtc-review.googlesource.com/43260
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21739}
With alpha channel, we observe the artifacts on the receiver side, and
the reason is that when YUV channel has a key frame, it gives frame_buffer2
a chance to drop some previous frames. Then it is possible that some alpha
frames got dropped, which break the alpha frame dependence chain.
In this CL, we pack the YUV frame and alpha encoded frame together as one
entity to solve the issue.
Bug: webrtc:8773
Change-Id: Ibe746a46cb41fd92b399a7069e1d89f02f292af7
Reviewed-on: https://webrtc-review.googlesource.com/38481
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21737}
Try to use frame timestamps first if they look reasonable, otherwise
use realtime clock.
Also, lower limit from 90% of target to 85%.
Bug: webrtc:4172, chromium:802290
Change-Id: Iad489be7c7cf637345be4795e5089936ab9fab07
Reviewed-on: https://webrtc-review.googlesource.com/41041
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21729}
This feature is not needed in video codec testing framework. In WebRTC
video codecs never deal with packet loss. Packet loss is handled by
jitter buffer which prevents passing of incomplete frames to decoder.
Bug: webrtc:8768
Change-Id: I211cf51d913bec6a1f935e30691661d428ebd3b6
Reviewed-on: https://webrtc-review.googlesource.com/40740
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21722}
No stats is logged in this format any longer.
Bug: none
Change-Id: I5f91e93636b6d03ebd91c3b2518857275fb94de7
Reviewed-on: https://webrtc-review.googlesource.com/40700
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21690}
This is a reland of 18c4261339
Original change's description:
> Enables/disables simulcast streams by allocating a bitrate of 0 to the spatial layer.
>
> Creates VideoStreams & VideoCodec.simulcastStreams with an active field, and then allocates 0 bitrate to simulcast streams that are inactive. This turns off the encoder for specific simulcast streams.
>
> Bug: webrtc:8653
> Change-Id: Id93b03dcd8d1191a7d3300bd77882c8af96ee469
> Reviewed-on: https://webrtc-review.googlesource.com/37740
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21646}
TBR=sprang@webrtc.org,stefan@webrtc.org,deadbeef@webrtc.org
Bug: webrtc:8630
Change-Id: Ib3df6f9b7158bff362a7ec66fc57e368682c5846
Reviewed-on: https://webrtc-review.googlesource.com/40980
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21688}
This is a reland of 1880c7162b
Original change's description:
> Updated analysis in videoprocessor.
>
> - Run analysis after all frames are processed. Before part of it was
> done at bitrate change points;
> - Analysis is done for whole stream as well as for each rate update
> interval;
> - Changed units from number of frames to time units for some metrics
> and thresholds. E.g. 'num frames to hit tagret bitrate' is changed to
> 'time to reach target bitrate, sec';
> - Changed data type of FrameStatistic::max_nalu_length (renamed to
> max_nalu_size_bytes) from rtc::Optional to size_t. There it no need to
> use such advanced data type in such low level data structure.
>
> Bug: webrtc:8524
> Change-Id: Ic9f6eab5b15ee12a80324b1f9c101de1bf3c702f
> Reviewed-on: https://webrtc-review.googlesource.com/31901
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21653}
TBR=brandtr@webrtc.org, stefan@webrtc.org
Bug: webrtc:8524
Change-Id: Ie0ad7790689422ffa61da294967fc492a13b75ae
Reviewed-on: https://webrtc-review.googlesource.com/40202
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21668}
This reverts commit 1880c7162b.
Reason for revert: breaks internal tests
Original change's description:
> Updated analysis in videoprocessor.
>
> - Run analysis after all frames are processed. Before part of it was
> done at bitrate change points;
> - Analysis is done for whole stream as well as for each rate update
> interval;
> - Changed units from number of frames to time units for some metrics
> and thresholds. E.g. 'num frames to hit tagret bitrate' is changed to
> 'time to reach target bitrate, sec';
> - Changed data type of FrameStatistic::max_nalu_length (renamed to
> max_nalu_size_bytes) from rtc::Optional to size_t. There it no need to
> use such advanced data type in such low level data structure.
>
> Bug: webrtc:8524
> Change-Id: Ic9f6eab5b15ee12a80324b1f9c101de1bf3c702f
> Reviewed-on: https://webrtc-review.googlesource.com/31901
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21653}
TBR=brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,ssilkin@webrtc.org
Change-Id: Id0b7d387bbba02e71637b229aeed6f6cf012af46
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8524
Reviewed-on: https://webrtc-review.googlesource.com/40220
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21656}
This reverts commit 16beea7c90.
Reason for revert: breaks internal tests.
Original change's description:
> Set encodedWidth/encodedHeight to actual coded resolution.
>
> This fixes mismatch between resolution of coded frame indicated by VP9
> encoder wrapper and actual coded resolution. If internal resize is
> enabled VP9 encoder might downscale input frame when bitrate is too low
> to keep good spatial quality. Before this fix VP9 wrapper always set
> coded resolution equal to input resolution. Now it sets it to actual
> coded resolution which it reads from frame pkt.
>
> Bug: webrtc:5749
> Change-Id: I7dc8ba89947e99213a3b4c3cd4d974b662f090c4
> Reviewed-on: https://webrtc-review.googlesource.com/39661
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21651}
TBR=brandtr@webrtc.org,asapersson@webrtc.org,ssilkin@webrtc.org
Change-Id: I32cb1271c863cf435f3fc3b465e11232cbd6a888
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:5749
Reviewed-on: https://webrtc-review.googlesource.com/40161
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21655}
- Run analysis after all frames are processed. Before part of it was
done at bitrate change points;
- Analysis is done for whole stream as well as for each rate update
interval;
- Changed units from number of frames to time units for some metrics
and thresholds. E.g. 'num frames to hit tagret bitrate' is changed to
'time to reach target bitrate, sec';
- Changed data type of FrameStatistic::max_nalu_length (renamed to
max_nalu_size_bytes) from rtc::Optional to size_t. There it no need to
use such advanced data type in such low level data structure.
Bug: webrtc:8524
Change-Id: Ic9f6eab5b15ee12a80324b1f9c101de1bf3c702f
Reviewed-on: https://webrtc-review.googlesource.com/31901
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21653}
This fixes mismatch between resolution of coded frame indicated by VP9
encoder wrapper and actual coded resolution. If internal resize is
enabled VP9 encoder might downscale input frame when bitrate is too low
to keep good spatial quality. Before this fix VP9 wrapper always set
coded resolution equal to input resolution. Now it sets it to actual
coded resolution which it reads from frame pkt.
Bug: webrtc:5749
Change-Id: I7dc8ba89947e99213a3b4c3cd4d974b662f090c4
Reviewed-on: https://webrtc-review.googlesource.com/39661
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21651}
This reverts commit 18c4261339.
Reason for revert: Broke internal tests
Original change's description:
> Enables/disables simulcast streams by allocating a bitrate of 0 to the spatial layer.
>
> Creates VideoStreams & VideoCodec.simulcastStreams with an active field, and then allocates 0 bitrate to simulcast streams that are inactive. This turns off the encoder for specific simulcast streams.
>
> Bug: webrtc:8653
> Change-Id: Id93b03dcd8d1191a7d3300bd77882c8af96ee469
> Reviewed-on: https://webrtc-review.googlesource.com/37740
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21646}
TBR=deadbeef@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,shampson@webrtc.org
Change-Id: I0aeb743cbd2e8d564aa732c937587c25a4c49b09
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8653
Reviewed-on: https://webrtc-review.googlesource.com/39883
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21647}
Creates VideoStreams & VideoCodec.simulcastStreams with an active field, and then allocates 0 bitrate to simulcast streams that are inactive. This turns off the encoder for specific simulcast streams.
Bug: webrtc:8653
Change-Id: Id93b03dcd8d1191a7d3300bd77882c8af96ee469
Reviewed-on: https://webrtc-review.googlesource.com/37740
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21646}
Because they usually happen for all or for no frames in a stream.
Bug: chrome:801327
Change-Id: Ie09b93a0822e821076ff2743cf1223c29fcf44a6
Reviewed-on: https://webrtc-review.googlesource.com/39785
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21644}
These headers contain public defines for the module, so it makes
sense it's visible.
Bug: webrtc:7620
Change-Id: I4c2604dcfb3beb4bfa0803a15d0e1a89c374e4cf
Reviewed-on: https://webrtc-review.googlesource.com/39261
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21616}
It should not be used anymore - see
https://webrtc-review.googlesource.com/39503
but it can be removed only after the DEPS roll.
Bug: webrtc:8745
Change-Id: I1ee29948e99be6cb4a700b67d37f72a0747a9fc5
Reviewed-on: https://webrtc-review.googlesource.com/39505
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21612}
This is a reland of https://webrtc-review.googlesource.com/34380
The main problem with that CL was that we used frame timestamps as basis
for frame dropping, but those might not be continuous or even populated
in some circumstances.
Additionally, I found that the bitrate was off since the encoder does
not not take the dropped frames into account, so if we drop every other
frame continiusoly, the bitrate sent will be around half of the target.
Patch set 1 is the original CL, subsequent patch sets cotains fixes.
Bug: webrtc:4172
Change-Id: I8ec8dddcebf4ce44f28dd9055cf9c46bbd68e4a6
Reviewed-on: https://webrtc-review.googlesource.com/39201
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21601}
This replaces most of the existing dependencies on the application
limited region(ALR) detector. This is to achieve a greater separation of
concerns and will make further refactoring regarding the ALR Detector
less invasive on other parts of the code base.
Bug: webrtc:8415
Change-Id: I92912254c6d02285cce6a88f6789f0ac94794c88
Reviewed-on: https://webrtc-review.googlesource.com/37560
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21598}
This reverts commit c73e1f4378.
Reason for revert:
The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660
Original change's description:
> Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
>
> This reverts commit 588c548657.
>
> Reason for revert:
>
> Breaks Chrome FYI:
>
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
> -> returned 1
> ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
> static_library(target_name) {
> ^----------------------------
> The item //content/renderer:renderer
> can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
> because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
> //third_party/webrtc/*
> //third_party/webrtc_overrides/*
> ]
>
> https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
>
> Original change's description:
> > GN rtc_* templates: Set default visibility to webrtc_root + "/*"
> >
> > This means that by default, targets are visible to everything under
> > the WebRTC root, but not visible to anything else.
> >
> > API targets are manually tagged with visibility "*", so that targets
> > outside the WebRTC tree can see them.
> >
> > BUG=webrtc:8254
> >
> > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> > Reviewed-on: https://webrtc-review.googlesource.com/24140
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21548}
>
> TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
>
> Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8254
> Reviewed-on: https://webrtc-review.googlesource.com/38760
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21555}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org
Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38860
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21558}
This reverts commit 588c548657.
Reason for revert:
Breaks Chrome FYI:
/b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
-> returned 1
ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
static_library(target_name) {
^----------------------------
The item //content/renderer:renderer
can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
//third_party/webrtc/*
//third_party/webrtc_overrides/*
]
https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
Original change's description:
> GN rtc_* templates: Set default visibility to webrtc_root + "/*"
>
> This means that by default, targets are visible to everything under
> the WebRTC root, but not visible to anything else.
>
> API targets are manually tagged with visibility "*", so that targets
> outside the WebRTC tree can see them.
>
> BUG=webrtc:8254
>
> Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> Reviewed-on: https://webrtc-review.googlesource.com/24140
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21548}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38760
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21555}
This means that by default, targets are visible to everything under
the WebRTC root, but not visible to anything else.
API targets are manually tagged with visibility "*", so that targets
outside the WebRTC tree can see them.
BUG=webrtc:8254
Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
Reviewed-on: https://webrtc-review.googlesource.com/24140
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21548}
When Chromium hooks up with the stereo codec, then it has difficulty
communicating with a google chrome without stereo codec. By design, we
do allow codec choice for the standalone codecs, but the problem is
that we do not handle the payload correctly, and thus the existence
of stereo codec will remove the payload registry of the standalone
version of its associated codec. (For example, stereo codec on top of
VP9 will remove the payload registry of standalone VP9 codec.)
This CL fixes the issue. When generating payload data, we should use
"stereo" as payload name, instead of its associated codecs.
Bug: webrtc:8657
Change-Id: I9e0b54de6bd41d370b9353f9553c998e4049789f
Reviewed-on: https://webrtc-review.googlesource.com/33122
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21523}
This creates a new target for pure defines and interfaces. I think
that makes sense (though include/ makes it harder to see when .cc and
.h files should live together).
Bug: webrtc:7620
Change-Id: Ifb0f50faf99166202836c0446feed3443eb52c6e
Reviewed-on: https://webrtc-review.googlesource.com/34657
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21516}
This is a reland of 727b7d0470
Original change's description:
> Reland "Reland "Put internal video codec factories into separate target""
>
> This is a reland of 0efd1e8b7e
> Original change's description:
> > Reland "Put internal video codec factories into separate target"
> >
> > This is a reland of 51698aefd4
> > Original change's description:
> > > Put internal video codec factories into separate target
> > >
> > > The purpose is to start splitting out the dependencies to the built-in
> > > SW video codecs, so that clients can decide to not depend on them and
> > > get a reduction in binary size.
> > >
> > > Replaces https://webrtc-review.googlesource.com/c/src/+/29101
> > >
> > > Bug: webrtc:7925
> > > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c
> > > Reviewed-on: https://webrtc-review.googlesource.com/33420
> > > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#21381}
> >
> > Bug: webrtc:7925
> > Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842
> > Reviewed-on: https://webrtc-review.googlesource.com/35261
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21389}
>
> Bug: webrtc:7925
> Change-Id: Id1c7f270676e9e4ca57ca8aa1305cf5554290754
> Reviewed-on: https://webrtc-review.googlesource.com/35501
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21464}
Bug: webrtc:7925
Change-Id: I0b3b5e03d29dadbcbe13cb7ce5369299bb6c0454
Reviewed-on: https://webrtc-review.googlesource.com/37000
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21513}
This reverts commit 727b7d0470.
Reason for revert: Breaks build
Original change's description:
> Reland "Reland "Put internal video codec factories into separate target""
>
> This is a reland of 0efd1e8b7e
> Original change's description:
> > Reland "Put internal video codec factories into separate target"
> >
> > This is a reland of 51698aefd4
> > Original change's description:
> > > Put internal video codec factories into separate target
> > >
> > > The purpose is to start splitting out the dependencies to the built-in
> > > SW video codecs, so that clients can decide to not depend on them and
> > > get a reduction in binary size.
> > >
> > > Replaces https://webrtc-review.googlesource.com/c/src/+/29101
> > >
> > > Bug: webrtc:7925
> > > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c
> > > Reviewed-on: https://webrtc-review.googlesource.com/33420
> > > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#21381}
> >
> > Bug: webrtc:7925
> > Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842
> > Reviewed-on: https://webrtc-review.googlesource.com/35261
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21389}
>
> Bug: webrtc:7925
> Change-Id: Id1c7f270676e9e4ca57ca8aa1305cf5554290754
> Reviewed-on: https://webrtc-review.googlesource.com/35501
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21464}
TBR=magjed@webrtc.org,andersc@webrtc.org
Change-Id: I8a0621eb91f9ce4835f012e74b6a1da9bf740963
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/36940
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21465}
This is a reland of 0efd1e8b7e
Original change's description:
> Reland "Put internal video codec factories into separate target"
>
> This is a reland of 51698aefd4
> Original change's description:
> > Put internal video codec factories into separate target
> >
> > The purpose is to start splitting out the dependencies to the built-in
> > SW video codecs, so that clients can decide to not depend on them and
> > get a reduction in binary size.
> >
> > Replaces https://webrtc-review.googlesource.com/c/src/+/29101
> >
> > Bug: webrtc:7925
> > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c
> > Reviewed-on: https://webrtc-review.googlesource.com/33420
> > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21381}
>
> Bug: webrtc:7925
> Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842
> Reviewed-on: https://webrtc-review.googlesource.com/35261
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21389}
Bug: webrtc:7925
Change-Id: Id1c7f270676e9e4ca57ca8aa1305cf5554290754
Reviewed-on: https://webrtc-review.googlesource.com/35501
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21464}
This reverts commit 718d8631b0.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Revert "Revert "Add ProtectionBitrateCalculator as an abstract class. ProtectionBitrateCalculatorDefault implements ProtectionBitrateCalculator. Register VideoSendStream to packet feedback""
>
> This reverts commit 53d901332c.
>
> Reason for revert: root cause has been found and will be addressed in the patch.The root cause was protection_bitrate_calculator_ is now destructed before worker_queue_, and worker_queue_ may contain tasks which involves protection_bitrate_calculator_, so they need to be destructed in the opposite order.
> That was not an issue since before this cl we didn't allocate protection_bitrate_calculator_ on the heap.
>
> Original change's description:
> > Revert "Add ProtectionBitrateCalculator as an abstract class. ProtectionBitrateCalculatorDefault implements ProtectionBitrateCalculator. Register VideoSendStream to packet feedback"
> >
> > This reverts commit e58e91b6d1.
> >
> > Reason for revert: Breaks downstream project b/70848177
> >
> > Original change's description:
> > > Add ProtectionBitrateCalculator as an abstract class. ProtectionBitrateCalculatorDefault implements ProtectionBitrateCalculator. Register VideoSendStream to packet feedback
> > >
> > > Bug: webrtc:8656
> > > Change-Id: Iab4f6ab8997cb082762218afc8580e9985ac2522
> > > Reviewed-on: https://webrtc-review.googlesource.com/33010
> > > Commit-Queue: Ying Wang <yinwa@webrtc.org>
> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#21348}
> >
> > TBR=stefan@webrtc.org,philipel@webrtc.org,yinwa@webrtc.org
> >
> > Change-Id: Ic186ba78be429bd1046ceac15051a3382b6ffc4f
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:8656
> > Reviewed-on: https://webrtc-review.googlesource.com/35080
> > Commit-Queue: Lu Liu <lliuu@webrtc.org>
> > Reviewed-by: Lu Liu <lliuu@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21374}
>
> TBR=stefan@webrtc.org,philipel@webrtc.org,lliuu@webrtc.org,yujo@chromium.org,yinwa@webrtc.org
>
> Change-Id: Ie2b5a2a2ead0f20ac67c1ea9b8d192af66bddf8d
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8656
> Reviewed-on: https://webrtc-review.googlesource.com/35320
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Ying Wang <yinwa@webrtc.org>
> Commit-Queue: Ying Wang <yinwa@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21409}
TBR=stefan@webrtc.org,philipel@webrtc.org,lliuu@webrtc.org,yujo@chromium.org,yinwa@webrtc.org
Change-Id: I9773aaa942054dcfbab6002a5d713ab3526b0534
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8656
Reviewed-on: https://webrtc-review.googlesource.com/35700
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21410}
This reverts commit 53d901332c.
Reason for revert: root cause has been found and will be addressed in the patch.The root cause was protection_bitrate_calculator_ is now destructed before worker_queue_, and worker_queue_ may contain tasks which involves protection_bitrate_calculator_, so they need to be destructed in the opposite order.
That was not an issue since before this cl we didn't allocate protection_bitrate_calculator_ on the heap.
Original change's description:
> Revert "Add ProtectionBitrateCalculator as an abstract class. ProtectionBitrateCalculatorDefault implements ProtectionBitrateCalculator. Register VideoSendStream to packet feedback"
>
> This reverts commit e58e91b6d1.
>
> Reason for revert: Breaks downstream project b/70848177
>
> Original change's description:
> > Add ProtectionBitrateCalculator as an abstract class. ProtectionBitrateCalculatorDefault implements ProtectionBitrateCalculator. Register VideoSendStream to packet feedback
> >
> > Bug: webrtc:8656
> > Change-Id: Iab4f6ab8997cb082762218afc8580e9985ac2522
> > Reviewed-on: https://webrtc-review.googlesource.com/33010
> > Commit-Queue: Ying Wang <yinwa@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21348}
>
> TBR=stefan@webrtc.org,philipel@webrtc.org,yinwa@webrtc.org
>
> Change-Id: Ic186ba78be429bd1046ceac15051a3382b6ffc4f
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8656
> Reviewed-on: https://webrtc-review.googlesource.com/35080
> Commit-Queue: Lu Liu <lliuu@webrtc.org>
> Reviewed-by: Lu Liu <lliuu@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21374}
TBR=stefan@webrtc.org,philipel@webrtc.org,lliuu@webrtc.org,yujo@chromium.org,yinwa@webrtc.org
Change-Id: Ie2b5a2a2ead0f20ac67c1ea9b8d192af66bddf8d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8656
Reviewed-on: https://webrtc-review.googlesource.com/35320
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21409}
This reverts commit 0efd1e8b7e.
Reason for revert: Breaking internal builds
Original change's description:
> Reland "Put internal video codec factories into separate target"
>
> This is a reland of 51698aefd4
> Original change's description:
> > Put internal video codec factories into separate target
> >
> > The purpose is to start splitting out the dependencies to the built-in
> > SW video codecs, so that clients can decide to not depend on them and
> > get a reduction in binary size.
> >
> > Replaces https://webrtc-review.googlesource.com/c/src/+/29101
> >
> > Bug: webrtc:7925
> > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c
> > Reviewed-on: https://webrtc-review.googlesource.com/33420
> > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21381}
>
> Bug: webrtc:7925
> Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842
> Reviewed-on: https://webrtc-review.googlesource.com/35261
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21389}
TBR=magjed@webrtc.org,andersc@webrtc.org
Change-Id: I8d3b788cc9e43261b3ed6d3d52427b5e26bc827e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/35187
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21393}
This is a reland of 51698aefd4
Original change's description:
> Put internal video codec factories into separate target
>
> The purpose is to start splitting out the dependencies to the built-in
> SW video codecs, so that clients can decide to not depend on them and
> get a reduction in binary size.
>
> Replaces https://webrtc-review.googlesource.com/c/src/+/29101
>
> Bug: webrtc:7925
> Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c
> Reviewed-on: https://webrtc-review.googlesource.com/33420
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21381}
Bug: webrtc:7925
Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842
Reviewed-on: https://webrtc-review.googlesource.com/35261
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21389}
We currently use raw jobject in our code mixed with sporadic
ScopedLocalRefFrame. This CL moves every jobject into a scoped object,
either local, global, or a parameter. Also, this CL uses the JNI
generation script to generate declaration stubs for the Java->C++
functions so that it no longer becomes possible to mistype them
without getting compilation errors.
TBR=brandt@webrtc.org
Bug: webrtc:8278,webrtc:6969
Change-Id: Ic7bac74a89c11180177d65041086d7db1cdfb516
Reviewed-on: https://webrtc-review.googlesource.com/34655
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21387}
