244 Commits

Author SHA1 Message Date
Danil Chapovalov
2549f174b5 Remove RTPFragmentationHeader creation and propagation through webrtc
Bug: webrtc:6471
Change-Id: I5cb1e10088aaecb5981888082b87ae9957bbaaef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181541
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31955}
2020-08-17 16:37:33 +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
Markus Handell
6deec38ede Migrate modules/video_coding to webrtc::Mutex.
Bug: webrtc:11567
Change-Id: I8023fbe7595f7ba8ae7c7db3583fc2e560ec3df2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178803
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31644}
2020-07-07 11:41:21 +00:00
Danil Chapovalov
54544ec92a In av1 encoder set bitrate per layer when scalability is used.
Bug: webrtc:11404
Change-Id: If779c16ffb55d28d21a0900439cbe2614507e557
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177015
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31570}
2020-06-26 09:26:05 +00:00
Danil Chapovalov
90dc5dae2c Factor encoded frame generation code into own target
Show it can make vp9 tests cleaner too.

Bug: None
Change-Id: I8333a61dec1ef90ade9faffea94e1555ccbfcfaa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177013
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31523}
2020-06-15 12:25:06 +00:00
Jerome Jiang
7f7fb830ba Reland "Add av1 test running real video clips."
This reverts commit 6958d2c6f0ce5267bdc4120d88680a4be9ed5e59.

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 3a2be87b80759a659901776024609de7a19fbaca.

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
Mirko Bonadei
a81e9c82fc Wrap WebRTC OBJC API types with RTC_OBJC_TYPE.
This CL introduced 2 new macros that affect the WebRTC OBJC API symbols:

- RTC_OBJC_TYPE_PREFIX:
  Macro used to prepend a prefix to the API types that are exported with
  RTC_OBJC_EXPORT.

  Clients can patch the definition of this macro locally and build
  WebRTC.framework with their own prefix in case symbol clashing is a
  problem.

  This macro must only be defined by changing the value in
  sdk/objc/base/RTCMacros.h  and not on via compiler flag to ensure
  it has a unique value.

- RCT_OBJC_TYPE:
  Macro used internally to reference API types. Declaring an API type
  without using this macro will not include the declared type in the
  set of types that will be affected by the configurable
  RTC_OBJC_TYPE_PREFIX.

Manual changes:
https://webrtc-review.googlesource.com/c/src/+/173781/5..10

The auto-generated changes in PS#5 have been done with:
https://webrtc-review.googlesource.com/c/src/+/174061.

Bug: None
Change-Id: I0d54ca94db764fb3b6cb4365873f79e14cd879b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173781
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31153}
2020-05-04 15:01:26 +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
Jerome Jiang
c93abcb341 VP9 test: change threshold to allow resizing for twice
Recent change in libvpx allows a second resize for low resolution.

Bug: None
Change-Id: I45a7ce376b274778b2fa183346de1993ef43bde7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169941
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30719}
2020-03-09 09:56:05 +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
Artem Titov
33f9d2b383 Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class
Bug: webrtc:10138
Change-Id: If85290581a72f81cf60181de7a7134cc9db7716e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161327
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30033}
2019-12-07 00:54:26 +00:00
Artem Titov
5256d8bc4b Refactor FrameGenerator to return VideoFrameBuffer with VideoFrame::UpdateRect
Bug: webrtc:10138
Change-Id: I22079e2630bb1f3bb27472795fe923f9143b3401
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161010
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29973}
2019-12-02 17:11:37 +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
Florent Castelli
8bbdb5b9bd Update VideoBitrateAllocator allocate to take a struct with more fields
We want to evaluate more data in order to make better choices in the
bitrate allocators.
In order to freely update the parameter list without
breaking the API many times for projects customizing them, we'll use a
struct instead.

Bug: webrtc:10126
Change-Id: I443f86781c5134950294cdd1e3197a47447cf973
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141418
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28748}
2019-08-02 13:52:54 +00:00
Sebastian Jansson
ed0febf573 Add k prefix to FrameGenerator::OutputType enum values
This prepares for using VideoFrameBuffer::Type as
FrameGenerator::OutputType, which will reduce the
number of redundant enums in the code.

Bug: webrtc:9883
Change-Id: I253f5f1ea7181e02a5cf1a92925f51da8ada6aa2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146982
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28696}
2019-07-29 09:41:31 +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
Jonas Olsson
a4d873786f Format almost everything.
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
2019-07-08 13:45:15 +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
Niels Möller
4d504c76cb New interface EncodedImageBufferInterface, replacing use of CopyOnWriteBuffer
Bug: webrtc:9378
Change-Id: I62b7adbd9dd539c545b5b1b1520721482a4623c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138820
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28317}
2019-06-19 07:02:34 +00:00
Elad Alon
370f93a34a Reland "Inform VideoEncoder of negotiated capabilities"
This is a reland of 11dfff0878c949f2e19d95a0ddc209cdad94b3b4

Now that I am sure that WebRTC code is not calling the obsolete
versions, I will just remove the NOT_REACHED and call the
new version from the old ones, so as not to trip up downstream
projects.

Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}

TBR=sakal@webrtc.org,kwiberg@webrtc.org,sprang@webrtc.org

Bug: webrtc:10720
Change-Id: I46c69e45c190805c07f7e51acbe277d7eebd1600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141412
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28236}
2019-06-11 14:49:37 +00:00
Philip Eliasson
49d661a7d3 Revert "Inform VideoEncoder of negotiated capabilities"
This reverts commit 11dfff0878c949f2e19d95a0ddc209cdad94b3b4.

Reason for revert: Downstream import failure.

Original change's description:
> Inform VideoEncoder of negotiated capabilities
> 
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
> 
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}

TBR=sakal@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org

Change-Id: I7f833055c67f1f879b01dd8c156ba7b8840e8747
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10720
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141411
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28225}
2019-06-11 11:56:04 +00:00
Elad Alon
11dfff0878 Inform VideoEncoder of negotiated capabilities
After this CL lands, an announcement will be made to
discuss-webrtc about the deprecation of one version
of InitEncode().

Bug: webrtc:10720
Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28224}
2019-06-11 11:32:13 +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
Yves Gerey
79e9f4b9c1 Replace test::Statistics by webrtc::RunningStatistics.
The former became redundant and didn't guarantee
numerical stability for variance computation.

Bug: webrtc:10412
Change-Id: Idc291abe9add41bde9e7734f179e5d6c65f2630b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132460
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27605}
2019-04-13 17:55:27 +00:00
Erik Språng
16cb8f5d74 Reland "Replace usage of old SetRates/SetRateAllocation methods"
This is a reland of 7ac0d5f348f0b956089c4ed65c46e65bac125508

Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
>
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
>
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}

TBR=brandtr@webrtc.org,sakal@webrtc.org,perkj@webrtc.org

Bug: webrtc:10481
Change-Id: I2978d5c527a18e885b7845c4e53a2424e8ad5b4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132551
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27593}
2019-04-12 13:37:32 +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
Niels Möller
7aacdd9515 Reland "Delete CodecSpecificInfo argument from VideoDecoder::Decode"
This is a reland of 39d3a7de02d63894d12e7332322e1d80cd7c0d40

Original change's description:
> Delete CodecSpecificInfo argument from VideoDecoder::Decode
>
> Bug: webrtc:10379
> Change-Id: I079b419604bf4e9c1994fe203d7db131a0ccddb6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125920
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27022}

Tbr: kwiberg@webrtc.org
Bug: webrtc:10379
Change-Id: I8197bebd2ae7dc460644a98795b8257b033c27c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126480
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27565}
2019-04-11 13:03:52 +00:00
Minyue Li
7ddef1af88 Revert "Replace usage of old SetRates/SetRateAllocation methods"
This reverts commit 7ac0d5f348f0b956089c4ed65c46e65bac125508.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
> 
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
> 
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}

TBR=brandtr@webrtc.org,sakal@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,perkj@webrtc.org

Change-Id: I576760b584e3f258013b0279c0c173c895bbb37e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10481
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132561
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27559}
2019-04-11 10:50:29 +00:00
Erik Språng
7ac0d5f348 Replace usage of old SetRates/SetRateAllocation methods
This rather large CL replaces all relevant usage of the old
VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
API is unchanged to allow downstream projects to update without
breakage.

Bug: webrtc:10481
Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27554}
2019-04-11 07:46:09 +00:00
Mirko Bonadei
6a489f22c7 Fully qualify googletest symbols.
Semi-automatically created with:

git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format

After this, two .cc files failed to compile and I have fixed them
manually.

Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
2019-04-09 17:18:20 +00:00
Sebastian Jansson
b55015e4e1 Replacing SequencedTaskChecker with SequenceChecker.
Bug: webrtc:9883
Change-Id: I5e3189da2a46e6f4ed1a3c5a5dfd2f7d75a16b5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130961
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27518}
2019-04-09 12:28:04 +00:00
philipel
a553c72659 Tune VideoCodecTestLibvpx.TemporalLayersVP8 thresholds.
After https://webrtc-review.googlesource.com/c/src/+/131141 there are some minor
changes to the encoding performance, hence the updated values.

Bug: none
Change-Id: I070a62ce725b0a79e5cb5c4679ab643de50c46c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131334
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27466}
2019-04-05 14:43:40 +00:00
philipel
59e875ce18 Tune VideoCodecTestLibvpx.MultiresVP8 thresholds.
After https://webrtc-review.googlesource.com/c/src/+/131141 there are some minor
changes to the encoding performance, hence the updated values.

Bug: none
Change-Id: Ifa661eea15a0d52f4760f4aac9294074faab757f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131382
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27463}
2019-04-05 13:15:34 +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
7d72d0fb39 Change VideoCodecTestStats API.
- Add GetFrameStatistics API:
  This is useful for downstream test users that want to read frame-level stats.
- Remove other APIs that are not used by downstream tests:
    * AddFrame
    * GetFrame
    * GetFrameWithTimestamp
    * SliceAndCalcAggregatedVideoStatistic
    * PrintFrameStatistics
    * Size
    * Clear
  The implementations, which are used by the fixture implementation, are kept.

Bug: webrtc:10349
Change-Id: Id2f6fa5a36b8341a5ccb365725f71ebe0c0f1570
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128779
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27306}
2019-03-27 09:50:54 +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
69008a8718 Avoid div-by-zero in VideoCodecTest stats calculation.
Bug: webrtc:10400
Change-Id: I82b1e86cc8f7d1547fc4863c08c0f8ab82801ac4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128086
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27145}
2019-03-15 13:15:02 +00:00
Niels Möller
009ab3c438 Delete EncodedImage::GetBufferPaddingBytes
For the ffmpeg H.264 decoder, rely on ffmpeg being configured with
CONFIG_SAFE_BITSTREAM_READER.

Bug: webrtc:9378
Change-Id: Ia7a46580d520808e36581252a95feeb5f9c57bf9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/119665
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27124}
2019-03-14 09:08:19 +00:00
Danil Chapovalov
471783fc87 Remove rtc::QueuedTask alias, use webrtc::QueuedTask directly
Use absl::WrapUnique/absl::make_unique to create the queued tasks.

Bug: webrtc:10191
Change-Id: I8f47a60cb326b0fc361c7f0e338b25373d39937c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126525
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27063}
2019-03-11 16:49:21 +00:00
Danil Chapovalov
ad89528051 Reland "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current"
This reverts commit 42d8c93ec351b68554825b58a3dc6525a7dc84da.

Reason for revert: Got Aliby for FEC test flakes

Original change's description:
> Revert "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current"
> 
> This reverts commit 304e9d2df347630d71fd4423f5971f30dac73e41.
> 
> Reason for revert: Breaks downstream projects.
> Seems to make VideoSendStreamTest.SupportsFlexfecSimulcastVp8 flaky.
> 
> Original change's description:
> > Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current
> > 
> > Bug: webrtc:10191
> > Change-Id: I506cc50a90c73a6a4f6a3de36de0999cca72f5ba
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126230
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27035}
> 
> TBR=danilchap@webrtc.org,kwiberg@webrtc.org
> 
> Change-Id: If98324f88e4b3d18bf2fe33597dfb9711867c243
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10191
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126484
> Reviewed-by: Yves Gerey <yvesg@webrtc.org>
> Commit-Queue: Yves Gerey <yvesg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27041}

TBR=danilchap@webrtc.org,kwiberg@webrtc.org,yvesg@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10191
Change-Id: Id87a17ae415142b8e0b11ba03ae7bad84a473fb0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126720
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27056}
2019-03-11 12:32:49 +00:00
Yves Gerey
42d8c93ec3 Revert "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current"
This reverts commit 304e9d2df347630d71fd4423f5971f30dac73e41.

Reason for revert: Breaks downstream projects.
Seems to make VideoSendStreamTest.SupportsFlexfecSimulcastVp8 flaky.

Original change's description:
> Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current
> 
> Bug: webrtc:10191
> Change-Id: I506cc50a90c73a6a4f6a3de36de0999cca72f5ba
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126230
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27035}

TBR=danilchap@webrtc.org,kwiberg@webrtc.org

Change-Id: If98324f88e4b3d18bf2fe33597dfb9711867c243
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10191
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126484
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27041}
2019-03-08 16:14:54 +00:00