WebRTC internal code should always use include paths that start
from the root of the project and that clearly identify the header file.
This allows 'gn check' to actually keep dependencies under control
because 'gn check' cannot enforce anything if the include path
is not fully qualified (starting from the root of the project).
Bug: webrtc:8815
Change-Id: I6c345c38fd990f66bc1a8129e7f7cee7d161e926
Reviewed-on: https://webrtc-review.googlesource.com/47120
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21848}
WebRTC internal code should always used include paths that starts
from the root of the project and that clearly identify the header file.
This allows 'gn check' to actually keep dependencies under control
because 'gn check' cannot enforce anything if the include path
is not fully qualified (starting from the root of the project).
Bug: webrtc:8815
Change-Id: I23fb4fed0c27a4d98bea360315b959af843587bc
Reviewed-on: https://webrtc-review.googlesource.com/46101
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21847}
Encoded frames are preserved and decoded after all layers are
encoded.
Each spatial layer is decoded with separate decoder.
For quality evaluation of lowres layers original input frame is
downscaled with bilinear interpolation.
Encoded and decoded frames are dumped into separate files.
For async codecs encoded frames are passed to decoder in encode
callback, as before.
Bug: webrtc:8524
Change-Id: Idb0c92c7274c1915cff9a011a2794f1cf4bc8cb1
Reviewed-on: https://webrtc-review.googlesource.com/43381
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21844}
This reverts commit 8fb22e71ee9bd77676838c5723f7e89a74a64aa9.
Reason for revert: breaks downstream projects
Original change's description:
> Updates tests for turning simulcast streams on/off.
>
> Due to libvpx we were restricted to always turning the low simulcast
> stream on, or else the encoder would always label the active streams'
> encoded frames as key frames. Now that libvpx has been updated and
> rolled in, this change updates tests to reflect that it is working.
>
> Bug: webrtc:8653
> Change-Id: I065ef817ace2292605e27e135802cf4a3e90647e
> Reviewed-on: https://webrtc-review.googlesource.com/46340
> 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@{#21831}
TBR=deadbeef@webrtc.org,sprang@webrtc.org,shampson@webrtc.org
Change-Id: If14074a7fc56c83b75584d8e9a6a913a40514bad
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8653
Reviewed-on: https://webrtc-review.googlesource.com/46840
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21832}
Due to libvpx we were restricted to always turning the low simulcast
stream on, or else the encoder would always label the active streams'
encoded frames as key frames. Now that libvpx has been updated and
rolled in, this change updates tests to reflect that it is working.
Bug: webrtc:8653
Change-Id: I065ef817ace2292605e27e135802cf4a3e90647e
Reviewed-on: https://webrtc-review.googlesource.com/46340
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@{#21831}
Due to the growing number of arguments, these functions are being replaced by the AudioProcessingBuilder class.
Bug: webrtc:8668
Change-Id: Ic3936fbd47d92eac22a857a678dca5fd8c029d8b
Reviewed-on: https://webrtc-review.googlesource.com/46241
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21826}
This fix the mismatch of resolution VP9 wrapper set for coded frame with
its actual resolution.
Bug: webm:1485, webrtc:5749
Change-Id: Ie1225d8f3a3d00e66229a1a79858d0a89b3d5fae
Reviewed-on: https://webrtc-review.googlesource.com/46040
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@{#21819}
This is a temporary measure until the synchronization method
used in the class, gets fixed.
Bug: chromium:796889, chromium:795340
Change-Id: Ie3d394ae42f005e8e0f353d04ea9c1d053ea9fd2
Reviewed-on: https://webrtc-review.googlesource.com/40460
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21812}
The original rtc_event_log_api is refactored to a pure API target plus
multiple targets coupled with WebRTC implementations.
Bug: None
Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f
Reviewed-on: https://webrtc-review.googlesource.com/43247
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#21811}
This is a reland of bbdabe50db0cf09f6007dda12a6476dc4602b174.
Original change's description:
> Rename stereo video codec to multiplex
>
> 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}
TBR=niklas.enbom@webrtc.org
Bug: webrtc:7671
Change-Id: I6f38dc46126f279f334d52b56339b40acdc30511
Reviewed-on: https://webrtc-review.googlesource.com/45820
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21794}
Delete the subdirectory modules/media_file, and all references to it.
Bug: none
Change-Id: I19d86420a7d1d51cb6174c914a90484918106c5a
Reviewed-on: https://webrtc-review.googlesource.com/40540
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21790}
This change makes the fix for too long delays during Opus DTX periods
permanent. The fix has up until now been under an experiment, named
WebRTC-NetEqOpusDtxDelayFix.
Bug: webrtc:8488,chromium:780849
Change-Id: I006abb67f96d9d7880bf2215d7d6b52db6cbbfbc
Reviewed-on: https://webrtc-review.googlesource.com/44420
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21786}
This reverts commit 4954a77cf81e6793245f52d485834acd3e6eab1c.
Reason for revert: Breaks downstream build which was depending on the name "kVideoCodecStereo". Will need to do some sort of trickery to make this change without breaking the relevant code. Sorry. :(
Original change's description:
> Reland "Rename stereo video codec to multiplex"
>
> This is a reland of bbdabe50db0cf09f6007dda12a6476dc4602b174.
> This was reverted because of breaking internal build. I contacted sheriff
> and looked at logs but cannot find anything related to this CL. This was landed
> with #3850 build which caused exception, but 3847-3855 seem to all have failed.
> I am relanding to see if it will work this time or it will give some related
> error message that can guide me.
>
> Original change's description:
> > Rename stereo video codec to multiplex
> >
> > 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}
>
> TBR=niklas.enbom@webrtc.org
>
> Bug: webrtc:7671
> Change-Id: I5934abad1ce28acf02842ea8ee2af7768a826eb8
> Reviewed-on: https://webrtc-review.googlesource.com/44520
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21780}
TBR=sprang@webrtc.org,niklas.enbom@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org
Change-Id: I0a71327c2ddfdd030b1e058cd6a41b1689836719
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/44621
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21783}
This is a reland of bbdabe50db0cf09f6007dda12a6476dc4602b174.
This was reverted because of breaking internal build. I contacted sheriff
and looked at logs but cannot find anything related to this CL. This was landed
with #3850 build which caused exception, but 3847-3855 seem to all have failed.
I am relanding to see if it will work this time or it will give some related
error message that can guide me.
Original change's description:
> Rename stereo video codec to multiplex
>
> 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}
TBR=niklas.enbom@webrtc.org
Bug: webrtc:7671
Change-Id: I5934abad1ce28acf02842ea8ee2af7768a826eb8
Reviewed-on: https://webrtc-review.googlesource.com/44520
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21780}
If a WebRTC build target requires gmock it has to include
test/gmock.h and just depend on //test:test_support.
Unfortunately //testtest_support was a leaky abstraction because it
wasn't propagating the correct -I compiler flag. To make everything
work, all the targets that use gmock started also to depend on
//testing/gmock (even if they were not including any gmock header
directly).
This CL makes //testtest_support propagate the include path up in the
dependency chain so it is possible to remove unused dependencies.
Note: all_dependent_configs should probably be used in the original
gmock target. There is an ongoing discussion about it. This CL solves
the problem on WebRTC side and it is forward compatible.
TBR=phoglund@webrtc.org
Bug: webrtc:8603
Change-Id: If08daf2ce9a6431a6e881a236743b4ec33b59ea7
Reviewed-on: https://webrtc-review.googlesource.com/44340
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21776}
Various files in webrtc codebase use scanf/sscanf function without
including stdio.h header file which is supposed to define it. This
somehow works when using glibc, but fails with uClibc.
Bug: webrtc:8641
Change-Id: Ie4ae17af32b32ed8cea567166b6b0e5193966995
Reviewed-on: https://webrtc-review.googlesource.com/32261
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21775}
This reverts commit bbdabe50db0cf09f6007dda12a6476dc4602b174.
Reason for revert: This breaks the internal build.
Original change's description:
> Rename stereo video codec to multiplex
>
> 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}
TBR=sprang@webrtc.org,niklas.enbom@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org
Change-Id: Icf019cb09e07de45821d31d7d8ea7707d01346ee
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/44360
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21774}
This reverts commit 4dc891f5e3a4bcad4db31e1af0ad45b6c471eef2.
Reason for revert: Reverting this CL to make it possible to revert https://webrtc-review.googlesource.com/c/src/+/43242
Original change's description:
> Add unit tests covering MultiplexImageComponent
>
> 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}
TBR=qiangchen@chromium.org,emircan@webrtc.org
Change-Id: I9cce6ed5f2990a2f443e04a9e5913cbd296242e4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/44341
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21773}
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}
The AecMobile struct contains a ::farendOld field. It's type is 'short [2][80]'.
The field was initialized by
memset(&aecm->farendOld[0][0], 0, 160);
But sizeof(short) is not guaranteed to be 1. This causes use of
unititialized memory on some platforms. According to MSAN, it can
affect the output of the echo canceller.
The issue was found by the MSAN fuzzer.
This change initializes the array properly.
Bug: chromium:805396
Change-Id: Ibcaca2185cfa153e8fd826e9addfc04d7b65e417
Reviewed-on: https://webrtc-review.googlesource.com/43860
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21764}
A coherence vector cohxd is computed in
WebRtcAec_ComputeCoherence. The coherence values should theoretically
be 0 <= x <= 1. Due to the way they are computed that is not always
the case.
The coherence values are used to update an error signal
estimate hNl in webrtc::EchoSuppression. 'hNl[i]' should contain an
error magnitude for frequency 'i'.
The error magnitudes are used as a basis for exponentiation. If a
magnitude is negative, the result is NaN.
The NaNs will then spread to the output signal.
This change caps the hNl values at 0. I considered capping the
coherence values at 1. The coherence values are calculated differently
for MIPS, NEON and SSE. Therefore it's simpler to cap the hNl values
instead.
The issue was found by the AudioProcessing fuzzer.
Bug: chromium:804634
Change-Id: I8ebaa441d77c3f79d9c194a850cb2b9eed1c2024
Reviewed-on: https://webrtc-review.googlesource.com/43740
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21761}
This change
* replaces a left shift with multiplication, because the shiftee can
be negative.
* replaces a right shift (a >> b) with the expression (b >= 32 ? 0 : a >> b)
because a is a 32-bit value, and b can be >= 32.
cppreference quote relating to the second change:
"In any case, if the value of the right operand is
negative or is greater or equal to the number of bits in the promoted
left operand, the behavior is undefined."
Bug: chromium:805832 chromium:803078
Change-Id: I67db0c3fedb0af197b2205d424414a84f8fde474
Reviewed-on: https://webrtc-review.googlesource.com/43761
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21760}
The faster AEC3 alignment introduced recently may in
cases cause the alignment (and the AEC3) to repeatedly
reset. This CL avoids these resets by handling buffer
issues (which are triggering the resets) separately
during the initial coarse alignment phase.
Change-Id: Idf5e2ffda2591906da8060d03ec8ca73cdaedf53
Bug: webrtc:8798,chromium:805815
Reviewed-on: https://webrtc-review.googlesource.com/43480
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21758}
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 d756fd06fed1b6c65dcb263cbd8f00ca23d72f3b.
Original change's description:
> Revert "Wrap Alpha and YUV frame into one EncodedImage for transmission"
>
> This reverts commit 5670c86aeccc9bc1191725431de7998d21b73c07.
>
> 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}
This change handles a special case in NoiseSuppression. The special
case was found by the AudioProcessing fuzzer.
A const copy of the capture audio stream is sent to
NoiseSuppression::AnalyzeCaptureAudio. Then audio undergoes processing
by e.g. the echo canceller. Then it's processed by
NoiseSuppression::ProcessCaptureAudio.
The special case is when the following conditions are all satisfied:
* All stream samples are constantly zero in the call to
AnalyzeCaptureAudio
* a processing component modifies it to be nonzero before the call to
ProcessCaptureAudio
* The array NoiseSuppressionC::magnPrevAnalyze is filled with
zeros. This holds after initialization.
In this case, there is a division by zero in WebRtcNs_ProcessCore. The
resulting NaN values pollute the output signal. They are only detected
several submodules later in the process chain. The NaN values cause
the EchoDetector to crash in debug mode.
There is special handling of the case when the signal is constant zero
in ProcessCore. This change avoids zero division by handling this
issue the same way.
Bug: chromium:803810 chromium:804634
Change-Id: I6d698dd0cd27e6d550b42085124300ce58533125
Reviewed-on: https://webrtc-review.googlesource.com/41282
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21745}
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}
