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}
