Commit graph

305 commits

Author SHA1 Message Date
philipel
e6542f2112 Removed unused include from encoded_image.h.
Bug: webrtc:9378
Change-Id: Ie26ab4d30d62ec109a8be638661789399821c162
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179525
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31758}
2020-07-17 14:14:03 +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
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
Ilya Nikolaevskiy
2899b3bc3d Allow HVGA Vp9 SVC to have 2 spatial layers and remove excessive rounding
Bug: webrtc:11652
Change-Id: I8bfa91c3115d6ebb17beefbb2a5e51efbbd599e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177000
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31502}
2020-06-11 13:50:16 +00:00
Mirko Bonadei
2dcf348011 Use absl_deps in order to preapre to the Abseil component build release.
Bug: webrtc:1046390
Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-08 12:59:40 +00:00
Markus Handell
f70fbc8411 Introduces rtc_base/synchronization/mutex.h.
This change introduces a new non-reentrant mutex to WebRTC. It
enables eventual migration to Abseil's mutex.

The mutex types supportable by webrtc::Mutex are

- absl::Mutex
- CriticalSection (Windows only)
- pthread_mutex (POSIX only)

In addition to introducing the mutexes, the CL also changes
PacketBuffer to use the new mutex instead of rtc::CriticalSection.

The method of yielding from critical_section.cc was given a
mini-cleanup and YieldCurrentThread() was added to
rtc_base/synchronization/yield.h/cc.

Additionally, google_benchmark benchmarks for the mutexes were added
(test courtesy of danilchap@), and some results from a pthread/Abseil
shootout were added showing Abseil has the advantage in higher
contention.

Bug: webrtc:11567, webrtc:11634
Change-Id: Iaec324ccb32ec3851bf6db3fd290f5ea5dee4c81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176230
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31443}
2020-06-04 09:55:12 +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
Danil Chapovalov
40f1fe9cff Add unittests to validate scalability structures without encoder
Bug: webrtc:10342
Change-Id: I66407e635502b7c87f8d4ab49c95f5c1326da4a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176412
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31423}
2020-06-03 12:59:25 +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
Mirko Bonadei
9ca7365a8c Deprecate webrtc::NackModule.
This CL moves webrtc::NackModule to a deprecated folder and annotates
the type with RTC_DEPRECATED.

Since the header should not be used outside of WebRTC, this CL doesn't
created a forward header.

Bug: webrtc:11611
Change-Id: I4d5899d473d78b8c7f4a6a018e2805648244b5f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176360
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31394}
2020-05-30 16:34:44 +00:00
Tommi
63673fe2cc Remove locks and dependency on ProcessThread+Module from NackModule2.
Change-Id: I39975e7812d7722fd231ac57e261fd6add9de000
Bug: webrtc:11594
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175341
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31367}
2020-05-27 14:20:34 +00:00
Mirko Bonadei
621c33653f Remove //modules/video_coding:nack_module from API.
Bug: None
Change-Id: I8e6cc61ae8406993909d0ab97896ccbaa89349c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176082
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31349}
2020-05-26 06:48:06 +00:00
Tommi
d3807da009 Fork NackModule and RtpVideoStreamReceiver
Bug: webrtc:11595
Change-Id: I4d14c0bf9c32e09d1624099a256f2778afebd4df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175901
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31337}
2020-05-22 17:07:16 +00:00
Danil Chapovalov
61bc0d1ed3 Introduce ChainDiffCalculator
to convert flags which chains a video frame part of into chain_diffs

Bug: webrtc:10342
Change-Id: I6fb899eae934078223b101c9f85e2ac101980d4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175108
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31306}
2020-05-18 14:22:44 +00:00
Henrik Boström
012aa375b1 Asynchronous QualityScaler: Callback-based CheckQpTask.
This CL breaks up the CheckQp() operation into several steps managed
by the inner helper class CheckQpTask, making responding to high or
low QP an asynchronous operation. Why? Reconfiguring the stream in
response to QP overuse will in the future be handled on a separate
task queue. See Call-Level Adaptation Processing for more details:
https://docs.google.com/document/d/1ZyC26yOCknrrcYa839ZWLxD6o6Gig5A3lVTh4E41074/edit?usp=sharing

Instead of "bool AdaptDown()" when high QP is reported,
synchronously returning true or false depending on the result of
adaptation, this CL introduces
  void QualityScalerQpUsageHandlerInterface::OnReportQpUsageHigh(
      rtc::scoped_refptr<QualityScalerQpUsageHandlerCallback>);
Where
  QualityScalerQpUsageHandlerCallback::OnQpUsageHandled(
      bool clear_qp_samples);
Instructs the QualityScaler whether to clear samples before
checking QP the next time or to increase the frequency of checking
(corresponding to AdaptDown's return value prior to this CL).

QualityScaler no longer using AdaptationObserverInterface, this class
is renamed and moved to overuse_frame_detector.h.

The dependency between CheckQpTasks is made explicit with
CheckQpTask::Result and variables like observed_enough_frames_,
adapt_called_ and adapt_failed_ are moved there and given more
descriptive names.

Bug: webrtc:11521
Change-Id: I7faf795aeee5ded18ce75eb1617f88226e337228
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173760
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31140}
2020-04-28 09:00:15 +00:00
Mirko Bonadei
6415dcad7a Remove WebRTC-ExperimentalScreenshareSettings.
This field trial is unused.

Bug: webrtc:11503
Change-Id: Id79b0dc64fed3559b9b63ebcf539e5536ddad589
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173339
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31090}
2020-04-16 18:15:08 +00:00
Evan Shrubsole
ce0a11d5f9 Unify AdaptationReason and AdaptReason enums.
Moves the unified AdaptationReason to the api/ folder.

Bug: webrtc:11392
Change-Id: I28782e82ef6cc3ca3b061f65b0bbdc3766df1f9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172583
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31084}
2020-04-16 13:33:49 +00:00
Danil Chapovalov
a4c4425748 Restore setting encoder speed for AV1 encoder wrapper
Also add simple unittests for the wrapper.

Bug: webrtc:11404
Change-Id: I41d185da9bce392297d1982194c059bddb7881ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171481
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30867}
2020-03-24 12:34:27 +00:00
Rasmus Brandt
2e6bd28381 libvpx-vp8: Add settings struct to constructor.
Migrate the injectable Vp8FrameBufferControllerFactory
into a settings struct, allowing for straight-forward
future extensions.

Bug: webrtc:11436
Change-Id: I53e555eb6ef88cf5b10ee8a43abd6ef9c930d100
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170635
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30839}
2020-03-20 11:35:46 +00:00
Niels Möller
d3da6b05c1 Move EventWrapper class to target video_coding_legacy.
And remove some unneeded logic for WEBRTC_EVENT_INFINITE.

Bug: webrtc:3380
Change-Id: Ibf632493edc6ced1609bd9ced44c2020fe9878cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169846
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30700}
2020-03-06 08:39:35 +00:00
Guido Urdaneta
c8958e5a4f Add RTC_EXPORT to VCMEncodedFrame
This is needed to be able to use webrtc::video_coding::EncodedFrame
is unit tests in Chromium.

TBR=tommi@webrtc.org

Bug: webrtc:11380
Change-Id: Idb3b0ab667a548f5a968e02a8efd91f02585c3f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169451
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30651}
2020-02-28 16:59:10 +00:00
Danil Chapovalov
e209fe6c68 Do not propagate generic descriptor on receiving frame
It was used only for the frame decryptor.
Decryptor needs only raw representation that it can recreate
in a way compatible with the new version of the descriptor.

This relands commit abf73de8ea.
with adjustments.

Change-Id: I935977179bef31d8e1023964b967658e9a7db92d
Bug: webrtc:10342
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168489
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30532}
2020-02-17 14:52:03 +00:00
Mirta Dvornicic
6799d732d5 Delete DefaultVideoBitrateAllocator.
It was removed from tests in https://webrtc-review.googlesource.com/c/src/+/123540.

If simulcast is not used, SimulcastRateAllocator returns the
same allocation as DefaultVideoBitrateAllocator.

Bug: webrtc:10164
Change-Id: I3d3e1aefe2fcc2bf853cd63c75e008b86eff9241
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168496
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30509}
2020-02-12 21:29:09 +00:00
Danil Chapovalov
02d71fb882 Populate generic descriptor based on GenericFrameInfo when available.
Bug: webrtc:10342
Change-Id: Iff769d2604fd79784bcb09874d2803793d20bde5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167000
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30505}
2020-02-12 10:55:41 +00:00
Danil Chapovalov
02b17a5507 Add helper to calculate frame dependencies based on encoder buffer usage
Bug: webrtc:10342
Change-Id: I1d856d060c2defcd10310f0d8639ce8a9554fff3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168194
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30458}
2020-02-05 16:19:10 +00:00
Niels Möller
2fca97168b Delete header file mock_vcm_callbacks.h
Move definitions of mock classes to the only user, the unit tests for
the deprecated class vcm::VideoReceiver.

Bug: webrtc:7408
Change-Id: I05e38ed8ebbe615bb2db0b631ec914773fb0a520
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168182
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30451}
2020-02-04 14:20:46 +00:00
philipel
190539717b Remove unused NextFrame function from FrameBuffer.
Also updated FrameBuffer unittests to use the GlobalSimulatedTimeController.

Bug: webrtc:7408, webrtc:9378
Change-Id: I8ade27492f66cdd8950b38f5f4a268714dbc35fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164536
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30422}
2020-01-30 12:54:08 +00:00
Danil Chapovalov
64f1f3f04e Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED
Bug: None
Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 15:20:35 +00:00
Qingsi Wang
1c1b99e30f Revert "Delete RtpPayloadParams::SetDependenciesVp8Deprecated as unused"
This reverts commit dc7fe40f49.

Reason for revert: speculative revert for breaking downstream projects

Original change's description:
> Delete RtpPayloadParams::SetDependenciesVp8Deprecated as unused
> 
> Bug: webrtc:10242
> Change-Id: Iddad086d8ce3652bd9f0fb12788d5c73b5ebda76
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161945
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Elad Alon <eladalon@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30159}

TBR=danilchap@webrtc.org,eladalon@webrtc.org,nisse@webrtc.org,philipel@webrtc.org

Change-Id: Ie7f875291610a7b676539a5ccc4bac9a08011f42
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10242
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165240
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30173}
2020-01-07 19:16:48 +00:00
Danil Chapovalov
dc7fe40f49 Delete RtpPayloadParams::SetDependenciesVp8Deprecated as unused
Bug: webrtc:10242
Change-Id: Iddad086d8ce3652bd9f0fb12788d5c73b5ebda76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161945
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30159}
2020-01-07 09:13:29 +00:00
Rasmus Brandt
5cad55b240 Signal requested resolution alignment requirements from sinks to sources.
Bug: webrtc:11218
Change-Id: I593b0515ea389bece472234a3c4082ccc5321ea5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162400
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30113}
2019-12-19 10:39:04 +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
Danil Chapovalov
0682ca9a83 Use AV1 packetizer/depacketizer for AV1 bitstreams
Bug: webrtc:11042
Change-Id: Ibf45a99d8016dccbe109d946ac967efa927312e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161011
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29953}
2019-11-28 18:01:10 +00:00
Artem Titov
5831ddad65 Introduce IVF file reader
Bug: webrtc:10138
Change-Id: I97d332942f4e645527330159efefb1cb1d8034a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160008
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29844}
2019-11-20 13:20:56 +00:00
Danil Chapovalov
aa3f5da8dc Fork VCMPacket for PacketBuffer into own struct
it is easier to reduce and eliminate it when it is not bound to legacy video code

Bug: webrtc:10979
Change-Id: I517e298501b3358a914a23ddce40fcb3075d672d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159707
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29821}
2019-11-18 15:48:07 +00:00
Rasmus Brandt
2b9317ad76 Stop checking VP8BaseHeavyTl3RateAllocation field trial on every frame.
- Centralize field trial string reading to RateControlSettings
- Cache RateControlSettings at all production code use sites

Bug: None
Change-Id: I0dbce9cc97fea0bc780982e7ef270b417a8c15bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158664
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29680}
2019-11-04 13:50:59 +00:00
Danil Chapovalov
09860e0bc3 Split out counting unique rtp timestamps from packet_buffer
Bug: None
Change-Id: Ia6fd05f284e8304cf56ab9ddf944fb222a4c9573
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158676
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29656}
2019-10-30 15:27:48 +00:00
Danil Chapovalov
fbec2ec292 Detach H264 sps pps tracker from VCMPacket
Bug: webrtc:10979
Change-Id: I6ec5db570c3957dd068109accad88d2f62304163
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158523
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29639}
2019-10-29 09:52:38 +00:00
Erik Språng
3eae7e4e3c Add exponential backoff of retransmissions for a given packet
Bug: webrtc:8624
Change-Id: I8900c54935bf1da11ac74665426b0d198bd6d814
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/30900
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29625}
2019-10-28 10:06:23 +00:00
Danil Chapovalov
ce1ffcdc06 change PacketBuffer to return it's result rather that use callback
Bug: None
Change-Id: I8cc05dd46e811d6db37af520d2106af21c671def
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157893
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29589}
2019-10-23 16:50:57 +00:00
Mirko Bonadei
3663ed3ad6 Move rtc_base/ignore_wundef.h to its own target.
Since rtc_base/ignore_wundef.h doesn't have any dependency, it is easy to
move it to its own target and allow its dependant to avoid to take a
dependency rtc_base:on rtc_base_approved.

Bug: webrtc:9419
Change-Id: I17f205b0cb2b21cad388b04e60082df9398dffdf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157428
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29548}
2019-10-19 10:50:36 +00:00
Danil Chapovalov
05269ecd0b Rename PacketBuffer tests to follow conventions
Rename file with tests to match code under test.
Rename fixture by moving 'Test' from prefix to suffix

Bug: None
Change-Id: I54c36d3b517bde7cdffa3a7e74528cc464ea7ad7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157301
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29532}
2019-10-18 09:05:06 +00:00
Mirko Bonadei
86d053c2db Use source_sets in component builds and static_library in release builds.
Static libraries don't guarantee that an exported symbol gets linked
into a shared library (and in order to support Chromium's component
build mode, WebRTC needs to be linked as a shared library).

Source sets always pass all the object files to the linker.

On the flip side, source_sets link more object files in release builds
and to avoid this, this CL introduces a the GN template "rtc_library" that
expands to static_library during release builds and to source_set during
component builds.

See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set

Bug: webrtc:9419
Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:17:18 +00:00
Elad Alon
c67a4d63dd Fix WebRTC-Video-MinVideoBitrate for VP9
Make sure the experiment-derived value is used for VP9.

Bug: webrtc:11024
Change-Id: I80b6d388486f2dec793bc8ca872babe6165dcfb2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156562
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29453}
2019-10-11 17:56:51 +00:00
Danil Chapovalov
7c06777ab0 Cleanup includes in modules/include/module_common_types.h
Add missing includes to files that were transactivly depending on removed includes.

Bug: None
Change-Id: Id5923bb8dc3e1d8fbb664e460278ad3e5993be7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155963
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29396}
2019-10-07 16:06:26 +00:00
Niels Möller
834a554962 Include module_common_types.h only where needed
Bug: None
Change-Id: I73d493f8f186b429c7be808f4dfac0398f150931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153891
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29277}
2019-09-24 08:22:38 +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
Johannes Kron
7ddea57e94 Add field-trial parameter to enable tests simulating a slow decoder
This CL adds a field trial parameter WebRTC-SlowDownDecoder that is
used to simulate a slow decoder. The parameter specifies how many
extra ms it takes to decode each video frame. This must only be used
in manual testing.

Bug: None
Change-Id: Iad4079100d67b95c224277aaeaf572e38068717f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151911
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29153}
2019-09-11 11:08:59 +00:00
Niels Möller
2eecfc1f9a Trim dependencies in modules/video_coding/
And move jitter_buffer_common.h to the legacy build target.

Bug: None
Change-Id: I986649f2f0773cadfa7dd9c8b533af7ecf01f3a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152382
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29150}
2019-09-11 09:31:10 +00:00