Commit graph

73 commits

Author SHA1 Message Date
“Michael
3147e29c4e Refactor encoder-complexity param in VideoCodec w/backward compatibility
Move complexity parameter to the main VideoCodec class to enable
additional video codecs to use the parameter without creating a new
codec-specific structure.

Bug: webrtc:13694
Change-Id: Icb7cf640b178875d799f39ade8b5084e3222bb1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251921
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Michael Horowitz <mhoro@google.com>
Cr-Commit-Position: refs/heads/main@{#36040}
2022-02-21 19:40:44 +00:00
Sergey Silkin
984cf9b837 Explicitly set encoder and decoder format in codec tests.
This allows to differentiate and test codecs of the same type but
different implementations/settings.

Bug: none
Change-Id: I74f799b36411e63387513133ffc19a7f0c45d550
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238165
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35396}
2021-11-22 08:18:25 +00:00
Artem Titov
d3251968d1 Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.

Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
2021-11-15 21:44:59 +00:00
Ying Wang
d7dd0aa9ee Move sendtask after decoded frame writer is initialized.
Bug: webrtc:13293
Change-Id: Ic71f92a5204715480e207f908f70ffff63e31279
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235580
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35235}
2021-10-19 11:32:33 +00:00
Artem Titov
dcd7fc7ea8 Use backticks not vertical bars to denote variables in comments for /modules/video_coding
Bug: webrtc:12338
Change-Id: Ia8a9adea291d594e4f59a6a1203a7bfb0758adac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227165
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34684}
2021-08-09 15:26:22 +00:00
Sergey Silkin
d6afbead2d Correctly set number of reference buffers in H264 encoder
iNumRefFrame specifies total number of reference buffers to allocate.
For N temporal layers we need at least (N - 1) buffers to store last
encoded frames of all reference temporal layers.

There is no API in OpenH254 encoder to specify exact set of references
to be used to prediction of a given frame. Encoder can theoretically
use all available references.

Note that there is logic in OpenH264 which overrides iNumRefFrame to
max(iNumRefFrame, N - 1): https://source.chromium.org/chromium/chromium/src/+/main:third_party/openh264/src/codec/encoder/core/src/au_set.cpp;drc=8e90a2775c5b9448324fe8fef11d177cb65f36cc;l=122.
I.e., this change has no real effect. It only makes setup more clear.

Bug: none
Change-Id: If4b4970007e1cc55d8f052ea05213ab2e89a878f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225480
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34445}
2021-07-09 13:49:41 +00:00
Johannes Kron
c3fcee7c3a Move h264_profile_level_id and vp9_profile to api/video_codecs
This is a refactor to simplify a follow-up CL of adding
SdpVideoFormat::IsSameCodec.

The original files media/base/h264_profile_level_id.* and
media/base/vp9_profile.h must be kept until downstream projects
stop using them.

Bug: chroimium:1187565
Change-Id: Ib39eca095a3d61939a914d9bffaf4b891ddd222f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215236
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33782}
2021-04-20 09:42:05 +00:00
Sergey Silkin
db0b4a8935 Do not crash if codec is not available
Check if codec was successfully created and exit from RunTest if not
before creating VideoProcessor.

Bug: none
Change-Id: Ia6d7171650dbc9824fb78f4a8e2851f755cfd63b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209362
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33372}
2021-03-03 13:56:17 +00:00
Erik Språng
c12f625938 Adds VideoDecoder::GetDecoderInfo()
This adds a new way to poll decoder metadata.
A default implementation still delegates to the old methods.
Root call site is updates to not use the olds methods.

Follow-ups will dismantle usage of the olds methods in wrappers.

Bug: webrtc:12271
Change-Id: Id0fa6863c96ff9e3b849da452d6540e7c5da4512
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196520
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32976}
2021-01-14 13:33:22 +00:00
Erik Språng
ebe5acb27a VideoCodecTextFixture and YuvFrameReader improvements.
Adds ability to specify desired frame size separate from actual clip
resolution, as well as clip and desired fps.
This allows e.g. reading an HD clip but running benchmarks in VGA, and
to specify e.g. 60fps for the clip but 30for encoding where frame
dropping kicks in so that motion is actually correct rather than just
plaing the clip slowly.

Bug: webrtc:12229
Change-Id: I4ad4fcc335611a449dc2723ffafbec6731e89f55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195324
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32839}
2020-12-15 23:18:06 +00:00
Sergey Silkin
b72cc6d670 Analyze quality of dropped frames in VideoProcessor.
Calculate quality metrics for dropped frames by comparing original
frame against last decoded one.

This feature makes comparison of encoders which do/don't drop frames
more fair.

The feature is controlled by analyze_quality_of_dropped_frames flag
and is disabled by default.

Bug: none
Change-Id: Ifab8df92d0b76e743ff3193c05d7c8dbd14921c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190660
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32518}
2020-10-29 08:23:49 +00:00
Erik Språng
ceb44959ca Reland: Wires up WebrtcKeyValueBasedConfig in media engines.
This is a reland of
https://webrtc-review.googlesource.com/c/src/+/174261

Patchset 1 contains the old cl (plus a merge conflict fix).
Later patchets are bufixes: A PeerConnection can be created without a
Call instance (in the case of DataChannel only), so we can't always
use that to fetch the current trials.

Old CL descritpion:

This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.

// Since re-land is otherwise unchanged, setting previous reviewers as TBR
TBR=kthelgason@webrtc.org,mbonadei@webrtc.org,stefan@webrtc.org,srte@webrtc.org

Bug: webrtc:11926
Change-Id: I57a9e8c3454f226f77fb93215bcac83da65034b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185003
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32163}
2020-09-22 16:08:22 +00:00
Artem Titov
5956a17ed6 Revert "Wires up WebrtcKeyValueBasedConfig in media engines."
This reverts commit 591b2ab82e.

Reason for revert: Breaks downstream project

Original change's description:
> Wires up WebrtcKeyValueBasedConfig in media engines.
> 
> This replaces field_trial:: -based functions from system_wrappers.
> Field trials are still used as fallback, but injectable trials are now
> possible.
> 
> Bug: webrtc:11926
> Change-Id: I70f28c4fbabf6d9e55052342000e38612b46682c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174261
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32129}

TBR=mbonadei@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,perkj@webrtc.org

Change-Id: I3e169149a8b787aa6366bb357abb71794534c63a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11926
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184507
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32132}
2020-09-17 20:17:38 +00:00
Erik Språng
591b2ab82e Wires up WebrtcKeyValueBasedConfig in media engines.
This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.

Bug: webrtc:11926
Change-Id: I70f28c4fbabf6d9e55052342000e38612b46682c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174261
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32129}
2020-09-17 16:24:10 +00:00
Niels Möller
5b69aa6613 Move definition of SpatialLayer to api/video_codecs/spatial_layer.h
Bug: webrtc:7660
Change-Id: I54009ebc5f65b6875a8c079ab5264e0c5ce9f654
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181500
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31942}
2020-08-17 09:45:19 +00:00
Jerome Jiang
7f7fb830ba Reland "Add av1 test running real video clips."
This reverts commit 6958d2c6f0.

Disable the test on iOS.

Bug: None
Change-Id: Ie42fada10a92bd4a802c6c79caeb4965410ddf6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176461
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Jerome Jiang <jianj@google.com>
Cr-Commit-Position: refs/heads/master@{#31437}
2020-06-04 06:32:46 +00:00
Ying Wang
6958d2c6f0 Revert "Add av1 test running real video clips."
This reverts commit 3a2be87b80.

Reason for revert: break internal test

Original change's description:
> Add av1 test running real video clips.
> 
> Bug: None
> Change-Id: I93bb8b3bf15d607d061aa74ad9e34609ffb2ef0a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175821
> Commit-Queue: Jerome Jiang <jianj@google.com>
> Commit-Queue: Stefan Holmer <holmer@google.com>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31401}

TBR=danilchap@webrtc.org,jianj@google.com,stefan@webrtc.org,holmer@google.com,marpan@webrtc.org

Change-Id: I2689ab4f7f26af6e26a4a188a2aa0b4f90a1a92f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176374
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31405}
2020-06-02 10:40:38 +00:00
Jerome Jiang
3a2be87b80 Add av1 test running real video clips.
Bug: None
Change-Id: I93bb8b3bf15d607d061aa74ad9e34609ffb2ef0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175821
Commit-Queue: Jerome Jiang <jianj@google.com>
Commit-Queue: Stefan Holmer <holmer@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31401}
2020-06-02 07:36:20 +00:00
Ilya Nikolaevskiy
39fb817efd [Video, Svc] Remove inactive spatial layers in codec initializer
This is more logical way to remove inactive lower layers.
Current way is to notify the encoder that the layer is inactive,
then renumber layers at the packatization level.

This Cl will allow to simplify libvpx vp9 encoder, svcRateAllocator and
vp9 packetizer.

Bug: webrtc:11319
Change-Id: Idf0bb30b729f5ecc97e31454b32934546b681aa2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173182
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31058}
2020-04-14 09:37:44 +00:00
Ilya Nikolaevskiy
03d909634b Ensure that the first active layer isn't disabled by too low input resolution
If e.g. CPU adaptation reduces input video size too much, video pipeline would
reduce the number of used simulcast streams/spatial layers. This may result in
disabled video if some streams are disabled by Rtp encoding parameters API.

Bug: webrtc:11319
Change-Id: Id7f157255599dcb6f494129b83477cda4bea982a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168480
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30498}
2020-02-11 14:57:51 +00:00
Sergey Silkin
df8fd28d0b Add output_path to VideoCodecTestFixture::Config.
This lets test to set output path explicitly.

Bug: none
Change-Id: I756484775f4c7f44cd1bb904c89d9215ffa48fe1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158798
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29702}
2019-11-06 08:48:52 +00:00
Danil Chapovalov
eb90e6ffe3 Merge SendTask implementation for SingleThreadedTaskQueueForTesting and TaskQueueForTest
That allows to use SingleThreadedTaskQueueForTesting via TaskQueueBase interface
but still have access to test-only SendTask function.

Bug: webrtc:10933
Change-Id: I3cc397e55ea2f1ed9e5d885d6a2ccda412beb826
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156002
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29480}
2019-10-15 09:17:36 +00:00
Mirko Bonadei
317a1f09ed Use std::make_unique instead of absl::make_unique.
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 15:47:29 +00:00
Marin Kišić
9a91161b9f Fixing way of printing logs because RTC_LOG() on Android has limit on printing 1024-60 characters in line.
Bug: webrtc:10885
Change-Id: I42c365555b682f3352644330167e2a4331ba0527
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149180
Commit-Queue: Marin Kišić <kisicmar@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29151}
2019-09-11 10:44:07 +00:00
Sergey Silkin
44cec0b5bd Handle non-integer frame rates in video codec tests.
Encoder API accepts non-integer frame rate since
https://webrtc-review.googlesource.com/c/src/+/131949.

Bug: webrtc:10812
Change-Id: I5fc9c5dfac4b182b84a735218a2946a95cc2b93c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143483
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28548}
2019-07-12 07:37:43 +00:00
Florent Castelli
668ce0c7fa Remove trial WebRTC-SimulcastMaxLayers and make its behavior default
Also cleans up the unused parameters from GetSimulcastConfig.

Bug: webrtc:8785, webrtc:8486
Change-Id: I1aea8f6c9e6590211ec5ee5cafc0ec2a2100d68f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144627
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28496}
2019-07-05 14:55:46 +00:00
Rasmus Brandt
315de596b0 Switch to RTC_LOG(LS_INFO) for non-perf VideoCodecTest text output.
This allows picking up the output in Android tests, where stdout/stderr
is lost but RTC_LOGs are picked up by the org.webrtc.Logging utility.

Tested: Downstream Android tests.
Bug: webrtc:10349
Change-Id: I1379f4303640dbc9621c64d9c88cf61bc8447ab6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132704
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27616}
2019-04-15 12:08:15 +00:00
Rasmus Brandt
70c961f965 Delete unused members of VideoCodecH264.
profile-level-id for H.264 comes in through the SdpVideoFormat,
rather than through these members.

Bug: None
Change-Id: I9c4ea8873346ca16174aecf5f90a649cbaf913dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132545
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27571}
2019-04-11 15:32:48 +00:00
Rasmus Brandt
fd720b2406 Switch to SendTask instead of manually waiting for event.
Bug: webrtc:10349
Change-Id: I128856d2baf221d67e957ce0614b075ecef3c5fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131140
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27439}
2019-04-03 12:03:14 +00:00
Niels Möller
9d8eaac4ee Delete unneeded direct includes of common_types.h
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:

api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/

There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.

Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
2019-04-01 07:18:13 +00:00
Rasmus Brandt
001c782ff2 Save encoded ivf files separately for different TLs.
This allows offline visualization of the different TL.

For now, there is no need to do the same for the decoded frames.

Bug: webrtc:10349
Tested: 1) ninja -C out/Debug; and out/Debug/modules_tests --gtest_filter="*MultiresVP8*:*SvcVP9*". 2) Downstream tests.
Change-Id: Iaf5ab19ee681488706d8777a5adba78efd5cc1ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128861
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27240}
2019-03-22 13:15:54 +00:00
Niels Möller
8f7ce222e7 Make VideoFrameType an enum class, and move to separate file and target
Bug: webrtc:5876, webrtc:6883
Change-Id: I1435cfa9e8e54c4ba2978261048ff3fbb993ce0e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126225
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27239}
2019-03-22 12:44:51 +00:00
Rasmus Brandt
c528410d2c Improve VideoCodecTest perf stats output.
- Output verbose send stats before verbose recv stats.
- Add |rate_profile_idx| to output names.
- Only report encode framerate and keyframe size for the entire stream.
- Add encoded bitrate/framerate stats per layer. Remove # dropped frames.
- Add U/V quality stats (mainly to compare to HW codecs)

Bug: webrtc:10349
Change-Id: I8f0d05e0fdf96ea998a06732462a080245b61221
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128614
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27223}
2019-03-21 12:17:09 +00:00
Danil Chapovalov
d26a916a80 Avoid using GlobalTaskQueueFactory for TaskQueueForTest
To remove global task factory, rtc::TaskQueue need to loose it's convenient constructor
TaskQueueForTest can be used instead in tests and keep the convenient constructor.

Also cleanup the TaskQueueForTest a bit:
move the class to webrtc namespace
add default constructor
disallow copy using language construct instead of macro
cleanup build dependencies
rename build target (to match move out of the rtc namespace)

Bug: webrtc:10284
Change-Id: I17fddf3f8d4f363df7d495c28a5b0a28abda1ba7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127571
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27193}
2019-03-19 18:11:52 +00:00
Rasmus Brandt
6f0aafa531 Add PrintResults to VideoCodecTest.
These are used by the test runner to pick up perf values
to be shown in the perf dashboard.

Bug: webrtc:10349
Change-Id: Ib3b2479f7a20b66192751bee8237d757f5870bd2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126220
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27020}
2019-03-07 15:12:40 +00:00
Rasmus Brandt
7b3f4a2035 Remove unused |keyframe_interval| from codec tests.
Bug: webrtc:10349
Change-Id: Iada8c8a1824f6e5424f503bb67b00382069b1dbd
Reviewed-on: https://webrtc-review.googlesource.com/c/124486
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26866}
2019-02-27 07:26:30 +00:00
Sergey Silkin
a0b1fb9ac7 Pass H264 profile/level settings to codec.
Bug: none
Change-Id: I0587a3d7c12a779a968b8c392c3dfa91b4ad040a
Reviewed-on: https://webrtc-review.googlesource.com/c/123180
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26735}
2019-02-18 13:06:35 +00:00
Sergey Silkin
6e1402b25f Skip SSIM calculation in real time mode.
SSIM calculation is not optimized. It takes ~100ms to process 720p frame
on Galaxy S8.

Bug: none
Change-Id: I51cc26d81124f06b2dfb27814edf2e4ae58141ce
Reviewed-on: https://webrtc-review.googlesource.com/c/121762
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26730}
2019-02-18 08:49:47 +00:00
Niels Möller
b7edf69e9a Delete rtc::File, usage replaced with FileWrapper
Bug: webrtc:6463
Change-Id: Ia0767a2e6bbacc43e63c30ed3bd3edb10ff6e645
Reviewed-on: https://webrtc-review.googlesource.com/c/121943
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26613}
2019-02-08 16:23:53 +00:00
Niels Möller
24871e4cbe Rename EncodedImage::_buffer --> buffer_, and make private
Bug: webrtc:9378
Change-Id: I0a0636077b270a7c73bafafb958132fa648aca70
Reviewed-on: https://webrtc-review.googlesource.com/c/117722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26294}
2019-01-17 12:38:15 +00:00
Niels Möller
77536a2b81 Rename EncodedImage::_length --> size_, and make private.
Use size() accessor function. Also replace most nearby uses of _buffer
with data().

Bug: webrtc:9378
Change-Id: I1ac3459612f7c6151bd057d05448da1c4e1c6e3d
Reviewed-on: https://webrtc-review.googlesource.com/c/116783
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26273}
2019-01-16 07:40:47 +00:00
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
Sergey Silkin
d716fb9ecb Reland "Refactor rate profile update."
This is a reland of b6cdfdc165

Original change's description:
> Refactor rate profile update.
>
> RateProfile::frame_num specifies frame at which this rate profile
> should be applied.
>
> Bug: none
> Change-Id: I003ee43f44299a49d83f547558284817bfaeacc0
> Reviewed-on: https://webrtc-review.googlesource.com/c/115242
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26080}

TBR=ilnik@webrtc.org,shampson@webrtc.org

Bug: none
Change-Id: I6ccbb32efe3d52c97e73e248ce5f06d672c9fba5
Reviewed-on: https://webrtc-review.googlesource.com/c/116286
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26155}
2019-01-08 10:35:42 +00:00
Sergey Silkin
08223c1576 Revert "Reland "Refactor rate profile update.""
This reverts commit 77aedaee69.

Reason for revert: breaks VideoCodecTestVideoToolbox tests.

Original change's description:
> Reland "Refactor rate profile update."
> 
> This is a reland of b6cdfdc165
> 
> Original change's description:
> > Refactor rate profile update.
> > 
> > RateProfile::frame_num specifies frame at which this rate profile
> > should be applied.
> > 
> > Bug: none
> > Change-Id: I003ee43f44299a49d83f547558284817bfaeacc0
> > Reviewed-on: https://webrtc-review.googlesource.com/c/115242
> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> > Reviewed-by: Seth Hampson <shampson@webrtc.org>
> > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#26080}
> 
> Bug: none
> Change-Id: I2604878d0bbee0f2182ad74e3cc29546310b76f3
> Reviewed-on: https://webrtc-review.googlesource.com/c/115401
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26145}

TBR=ilnik@webrtc.org,shampson@webrtc.org,ssilkin@webrtc.org

Change-Id: Ib53eae70c380eefa303ddb01441f23e32f06b3ad
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/116285
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26148}
2019-01-07 15:41:17 +00:00
Sergey Silkin
77aedaee69 Reland "Refactor rate profile update."
This is a reland of b6cdfdc165

Original change's description:
> Refactor rate profile update.
> 
> RateProfile::frame_num specifies frame at which this rate profile
> should be applied.
> 
> Bug: none
> Change-Id: I003ee43f44299a49d83f547558284817bfaeacc0
> Reviewed-on: https://webrtc-review.googlesource.com/c/115242
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26080}

Bug: none
Change-Id: I2604878d0bbee0f2182ad74e3cc29546310b76f3
Reviewed-on: https://webrtc-review.googlesource.com/c/115401
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26145}
2019-01-07 11:18:26 +00:00
Sergey Silkin
a1f78a4fa6 Revert "Refactor rate profile update."
This reverts commit b6cdfdc165.

Reason for revert: breaks downstream projects

Original change's description:
> Refactor rate profile update.
> 
> RateProfile::frame_num specifies frame at which this rate profile
> should be applied.
> 
> Bug: none
> Change-Id: I003ee43f44299a49d83f547558284817bfaeacc0
> Reviewed-on: https://webrtc-review.googlesource.com/c/115242
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26080}

TBR=ilnik@webrtc.org,shampson@webrtc.org,ssilkin@webrtc.org

Change-Id: I5957a0169841008436d1db70403d3694bf25d5cf
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/115400
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26081}
2018-12-21 09:05:01 +00:00
Sergey Silkin
b6cdfdc165 Refactor rate profile update.
RateProfile::frame_num specifies frame at which this rate profile
should be applied.

Bug: none
Change-Id: I003ee43f44299a49d83f547558284817bfaeacc0
Reviewed-on: https://webrtc-review.googlesource.com/c/115242
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26080}
2018-12-21 08:32:35 +00:00
Yves Gerey
3e70781361 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280.
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.

Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 18:25:07 +00:00
Niels Möller
c572ff3c71 Add default constructor for rtc::Event
Bug: webrtc:9962
Change-Id: Icaa91e657e6881fcb1553f354c07866109a0ea68
Reviewed-on: https://webrtc-review.googlesource.com/c/109500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25535}
2018-11-07 08:57:50 +00:00
Erik Språng
e2fd86a79c Move encoder metadata into EncoderInfo struct.
This deprecates the following methods in VideoEncoder:
  virtual ScalingSettings GetScalingSettings() const;
  virtual bool SupportsNativeHandle() const;
  virtual const char* ImplementationName() const;

Though they are not marked RTC_DEPRECATED since we still want to call
them from within the default GetEncoderInfo() until downstream
projects have been updated.

Furthmore, implementation name is changed from const char* to
std:string, which prevents some lifetime issues with dynamic encoder
names, and CodecSpecificInfo.codec_name is removed in favor of getting
the implementation name via GetEncoderInfo().

This CL removes calls to these deprecated methods, follow-ups will also
remove implementations of the methods and replace them with new
GetEncoderInfo() substitutions.

Bug: webrtc:9890
Change-Id: I6fd6e531480c0b952f53dbd5105e0b0adc3e3b0c
Reviewed-on: https://webrtc-review.googlesource.com/c/106905
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25351}
2018-10-25 08:51:53 +00:00