3633 Commits

Author SHA1 Message Date
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
Erik Språng
1b3f4f9b45 Allow RtpPacketHistory encapsulator function to abort retransmit
Bug: webrtc:10633
Change-Id: I162b2c2f778e8e4c6f31307028db0c352ded2276
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142230
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28312}
2019-06-18 17:59:16 +00:00
Per Kjellander
b762b5b794 Fix potential signed overflow in IntervalBudget::set_target_rate_kbps
This methods multiplies an int with 500. This cl ensure that does not overflow.

BUG=none

Change-Id: I2c4c4c169baf3bbc2cac021e87e396c605cd1815
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141860
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28310}
2019-06-18 15:42:54 +00:00
Erik Språng
58ee187554 Add support within PacedSender and pacer queue for owning rtp packets.
This CL builds on https://webrtc-review.googlesource.com/c/src/+/142165
It adds the parts within the paced sender that uses those send methods.
A follow-up will add the pre-pacer RTP sender parts. That CL will also
add proper integration testing. Here, I mostly add coverage for the new
send methods. When the old code-path is removed, all tests need to be
converted to exclusively use the owned path.

Bug: webrtc:10633
Change-Id: I870d9a2285f07a7b7b0ef6758aa310808f210f28
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142179
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28308}
2019-06-18 15:02:19 +00:00
Sergey Silkin
0c0c9693b6 Add/rewrite H264 VUI video signal type description.
The rewriter updates video signal parameters in VUI such that they
match to given webrtc::ColorSpace.

Bug: webrtc:10723
Change-Id: I8d0593e3cb727bfee7eb00e3f9ff0b41b93b78bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140881
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28306}
2019-06-18 11:41:43 +00:00
Sebastian Jansson
449888ef99 Cleanup of resources from removed remote bitrate estimate test framework.
Bug: webrtc:9883
Change-Id: Id18133a021b3a064b00f0f99b5f30ebb92e89067
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140945
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28305}
2019-06-18 10:22:01 +00:00
Sebastian Jansson
0f557fe1ba Removes unused dependency on RTP/RTCP from loss based controller.
This is part of a series of CLs removing RTP dependencies from GoogCC
implementation.

Bug: webrtc:10749
Change-Id: I73e9402136cc16902d177234d63369938f191e5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142223
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28302}
2019-06-18 07:06:52 +00:00
Erik Språng
9c771c2089 Add TrySendPacket() method to RTP modules.
This method will be called when PacedSender is using the new code path
that directly owns the packets to be sent.

It can be seen as combining a few features of the old code path:
* It checks if this is the correct RTP module and then sends, without
  the need for PacketRouter to poll multiple methods for SSRC etc first.
* It partly corresponds to TimeToSendPacket(), but RTX encapsulation
  now happens pre-pacer and FEC does not need to have a packet history,
  so most of that method is not used.
* It implements most of PrepareAndSendPacket(), such as updating header
  extensions, reporting stats and of course forwards to Transport. It
  now also handles the history a bit differently, since media packets
  will only be stored for potential retransmission post-pacer.

Bug: webrtc:10633
Change-Id: Ie97952eeef6e56e462e115d67f7c7929f36c1817
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142165
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28298}
2019-06-17 15:16:00 +00:00
Sebastian Jansson
e93454a383 Removes AddAndRemoveOld from SendTimeHistory
Bug: webrtc:9883
Change-Id: Id9ae4b2485ba5ff201e77771774cfc83ccbda0d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142220
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28295}
2019-06-17 08:46:54 +00:00
Sebastian Jansson
3d61ab12e6 Adds send time to ReceivedPacket struct.
Bug: webrtc:10742
Change-Id: I7e83d5ec2e23d1db38d02a0c883466ecdcd387c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141874
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28290}
2019-06-14 15:01:36 +00:00
philipel
5e953d7968 Insert startcodes for H264 single NALU packets.
Bug: none
Change-Id: I68ee9b84d75b67044d28a47fe740c79a2cd33517
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142169
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28287}
2019-06-14 13:45:21 +00:00
Chen Xing
12d64deb6c Remove sequence_number from RtpPacketInfo.
This change removes sequence_number from RtpPacketInfo since it's currently not used.

Bug: webrtc:10668
Change-Id: I9b45c7476457df1d18173f37c421374108678931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141873
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28281}
2019-06-14 11:21:42 +00:00
Sebastian Jansson
5740afa0a4 Removes SimulatedTimeClient
Bug: webrtc:9883
Change-Id: Id6e760b37360e7dafc67ded99e06128be20797d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141417
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28269}
2019-06-13 15:37:10 +00:00
Sebastian Jansson
f3f57700a8 Using full scenario test client for loss based control test.
Bug: webrtc:9883
Change-Id: I7c3b2561ddba846e4cdde05e1067679ada14ad80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141405
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28267}
2019-06-13 14:44:09 +00:00
Danil Chapovalov
4284828887 Remove deprecated version of RtpPacket::SetPadding that used to randomize padding
was deprecated in
https://webrtc-review.googlesource.com/c/src/+/103983

Bug: None
Change-Id: I617b7b5112446deaa9be983978cabdb247638266
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141865
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28266}
2019-06-13 14:38:38 +00:00
Jakob Ivarsson
79890ef91f Remove sync buffer length from FilteredCurrentDelayMs.
The sync buffer length is already added when the buffer level filter is updated.

Bug: webrtc:10736
Change-Id: Icbd411d4fd7b16f31b800142d1b6a8de79365d91
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140950
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28261}
2019-06-13 09:38:22 +00:00
Niels Möller
44bc19b0f8 Delete TestAudioDeviceModule methods using rtc::PlatformFile
Bug: webrtc:6463
Change-Id: I5d1d9e9036b5e745d5b37c971de91b1b38fdd368
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141666
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28255}
2019-06-12 15:28:41 +00:00
Danil Chapovalov
08fa953711 Reland "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory"
This reverts commit fd5166c305068772d00ad7edf50151bba215400b.

Reason for revert: Stop using CreateTestAudioDeviceModule in downstream

Original change's description:
> Revert "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory"
> 
> This reverts commit fc961357a721cd87dcd45ed409c66cb8cda6f4a2.
> 
> Reason for revert: Breaks downstream importer.
> 
> Original change's description:
> > Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory
> > 
> > Bug: webrtc:10284
> > Change-Id: Ic92f6ff31b40c48a3362745a0a81179af0595fe0
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141409
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28227}
> 
> TBR=danilchap@webrtc.org,kwiberg@webrtc.org
> 
> Change-Id: Id6d7571f48771646ddce0f05139a7ea0107759fb
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10284
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141414
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28228}

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

Change-Id: I42bc19793d48350ca45b751d7e1b26124ac7fbb9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141670
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28254}
2019-06-12 14:44:01 +00:00
Erik Språng
f53cfa9ebe Add new RtpPacketPacer interface, with callback.
This CL just adds the new interfaces, follow-ups will add implementation
in various parts of the code, and then do cleanup once usage of old
interface is gone.

Bug: webrtc:10633
Change-Id: Icd916f4220065c0d0e4f3f0bfaaed248f8c70d08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140891
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28252}
2019-06-12 13:21:54 +00:00
Erik Språng
e794243650 Remove PacedSender::PacketSender interface and use PacketRouter directly
This interface makes future refactoring difficult and is now in practice
only implemented by PacketRouter.

Bug: webrtc:10633
Change-Id: I3fcb8940781aa7431119649bde7594592a8c8851
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141669
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28251}
2019-06-12 13:09:04 +00:00
Chen Xing
9c16af7eb7 Add a tracker for RTCRtpContributingSource and RTCRtpSynchronizationSource.
This change adds a new SourceTracker class that can do spec-compliant tracking of RTCRtpContributingSource and RTCRtpSynchronizationSource when frames are delivered to the RTCRtpReceiver's MediaStreamTrack for playout. It will replace the existing spec-incompliant ContributingSources.

Bug: webrtc:10545 webrtc:10668
Change-Id: I961adaba09d6337f2f36b301a4fabcd20de65271
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140948
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28249}
2019-06-12 12:11:55 +00:00
Elad Alon
5cf390386b Allow Vp8FrameBufferController::UpdateConfiguration to reset set of overrides
Bug: webrtc:10737
Change-Id: Ifdf82f140465d114300eda1e3ba1a26c70aaef76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141663
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28247}
2019-06-12 10:12:44 +00:00
Niels Möller
47ae30358e Delete deprecated method VideoCodingModule::SetReceiverRobustnessMode
Bug: webrtc:7408
Change-Id: Ie0eb5b4b1522948bc215cab6e3aaaa745f4b3698
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141401
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28245}
2019-06-12 06:39:38 +00:00
Elad Alon
a27958421c Inform FrameBufferController of encoder capabilities
Bug: webrtc:10720
Change-Id: I958b83e29e8e87252e08419b8e16d554983a1a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140941
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28242}
2019-06-11 19:26:23 +00:00
Erik Språng
eceb537086 Add RtpPacketHistory::SetSendTime()
This method will be used instead of GetPacketAndSetSendTime() when the
new pacer code path is used, where the packet isn't stored in the
history during pacing.

Bug: webrtc:10633
Change-Id: Ie168125d949cef617ade3868a1858ed1dffe909c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140892
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28240}
2019-06-11 17:49:51 +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
Jonas Olsson
5b2eda4895 Sanity-check field trial string at initialization.
It's easy to make small errors when building field trial strings, and
those errors can cause all sorts of weird problems. This CL checks if
the FT string has an odd number of delimiters, duplicate
names or any trailing chars.

If so we'll log a error message. On debug builds we'll also crash.

Bug: webrtc:10729
Change-Id: Iebf7155d9b117a02d1e9cfe7f64408e11df2aec5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140866
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28234}
2019-06-11 14:11:06 +00:00
Niels Möller
e8e4dc4c8b Change StartAecDump methods to work with FILE* and FileWrapper
Bug: webrtc:6463
Change-Id: Id275975decb9b2876021ced19ee9f279b07bea53
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140283
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28233}
2019-06-11 13:43:36 +00:00
Sebastian Jansson
7742b21839 Removes legacy bitrate controller.
Bug: webrtc:9883
Change-Id: I66af2597059fc1f38c78682f6884361a4d16c4a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141408
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28232}
2019-06-11 13:16:05 +00:00
Sebastian Jansson
28aced5c3c Adds debug logs for loss based controller.
Bug: webrtc:9883
Change-Id: I525d88b1eba22f6198da04b1e18c26ec4a15c42d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141406
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28230}
2019-06-11 13:08:55 +00:00
Philip Eliasson
fd5166c305 Revert "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory"
This reverts commit fc961357a721cd87dcd45ed409c66cb8cda6f4a2.

Reason for revert: Breaks downstream importer.

Original change's description:
> Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory
> 
> Bug: webrtc:10284
> Change-Id: Ic92f6ff31b40c48a3362745a0a81179af0595fe0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141409
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28227}

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

Change-Id: Id6d7571f48771646ddce0f05139a7ea0107759fb
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141414
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28228}
2019-06-11 12:32:23 +00:00
Danil Chapovalov
fc961357a7 Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory
Bug: webrtc:10284
Change-Id: Ic92f6ff31b40c48a3362745a0a81179af0595fe0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141409
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28227}
2019-06-11 12:15:44 +00:00
Sebastian Jansson
7a994334f1 Avoids crash in VCMJitterEstimator due to incorrectly calculated mean FPS.
Bug: webrtc:10728
Change-Id: I9e50ebdcfd9ef7d408cbf00111f29f0c80fd6c75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141402
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28226}
2019-06-11 12:04:52 +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
Sebastian Jansson
4ad51d8b31 Removes SendSideCongestionController.
Bug: webrtc:9586
Change-Id: Id6f3508eb19f277d74c34edfbcaeb8a22320b030
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140286
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28222}
2019-06-11 11:09:24 +00:00
Niels Möller
ab6fc1154f Delete RtpRtcp methods SetKeyFrameRequestMethod and RequestKeyFrame
These are replaced with the methods SendPictureLossIndication and
SendFullIntraRequest, added in cl
https://webrtc-review.googlesource.com/c/src/+/140043.

Also delete the corresponding state variable
RtpRtcpImpl::key_frame_req_method_, the enum KeyFrameRequestMethod,
and the nearby unused enum RtpRtcpPacketType.

Bug: None
Change-Id: I1ac2e4ce6dbe20d1d1cbb3d5b2256ea55b341a57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141403
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28221}
2019-06-11 10:42:04 +00:00
Oleh Prypin
240b893fd2 Reland "Link fewer X11-related libraries"
This is a reland of 829a5dce900b781376ea72ba5d869c5dbe58a82d
(unchanged)
Downstream is fixed in https://chromium-review.googlesource.com/1651786

Original change's description:
> Link fewer X11-related libraries
>
> Use an explicit list and don't add X11 dependency to rtc_base.
>
> Allow skipping code that depends on rarer extensions such as Xdamage, Xfixes.
>
> Bug: None
> Change-Id: Icb8d20a267358f5cd3f1ff2af31a669e0670d2f6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140865
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28204}

Bug: None
Change-Id: I51c60f7713eb49889cebbca2242f33b81b0aef39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141301
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28218}
2019-06-11 07:34:49 +00:00
Marina Ciocea
4606ded90c Revert "Link fewer X11-related libraries"
This reverts commit 829a5dce900b781376ea72ba5d869c5dbe58a82d.

Reason for revert: Suspect of breaking autoroll to chromium: https://crrev.com/c/1651414.

Original change's description:
> Link fewer X11-related libraries
> 
> Use an explicit list and don't add X11 dependency to rtc_base.
> 
> Allow skipping code that depends on rarer extensions such as Xdamage, Xfixes.
> 
> Bug: None
> Change-Id: Icb8d20a267358f5cd3f1ff2af31a669e0670d2f6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140865
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28204}

TBR=zijiehe@chromium.org,mbonadei@webrtc.org,henrika@webrtc.org,oprypin@webrtc.org,guidou@webrtc.org

Change-Id: Ie106de700fc017dafa603d15eaec88c4790dacfb
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141300
Reviewed-by: Marina Ciocea <marinaciocea@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28215}
2019-06-10 19:53:42 +00:00
Niels Möller
646e096e03 Allow null FILE* to AecDumpFactory::Create
Also add Create method with a FileWrapper argument, extracted from cl
https://webrtc-review.googlesource.com/c/src/+/140283

Bug: webrtc:6463
Change-Id: If86f17d5fda964df0b5874bd5a15fcaed899dd4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140946
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28211}
2019-06-10 14:09:19 +00:00
Niels Möller
8169db6df2 Expose target bandwidth fraction for transport feedback for field trial
Bug: None
Change-Id: I5b3db8ff18460ac0bfd5e3db994642a21ba90d9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140944
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28210}
2019-06-10 12:22:34 +00:00
Chen Xing
5d24b16c77 Prepare for splitting the api/video:video_frames build rule.
This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files.

Bug: webrtc:10668
Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 11:50:51 +00:00
Sebastian Jansson
12561e06ce Removes legacy remote bitrate estimator test framework.
It isn't actively maintained to be up to date but depends on internals
increasing maintenance burden. There's now better tools available to
evaluate congestion controller performance that should be used if we
want to start maintaining this functionality again.

Bug: webrtc:9883
Change-Id: I097747e6f31a3d1522ef8dfda84f995e33f3a697
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140887
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28205}
2019-06-10 09:12:25 +00:00
Oleh Prypin
829a5dce90 Link fewer X11-related libraries
Use an explicit list and don't add X11 dependency to rtc_base.

Allow skipping code that depends on rarer extensions such as Xdamage, Xfixes.

Bug: None
Change-Id: Icb8d20a267358f5cd3f1ff2af31a669e0670d2f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140865
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28204}
2019-06-10 08:44:17 +00:00
Pablo Barrera González
7541141b99 Fix assert check in neteq_quality_test
While reading inpùt files until their end, the assert should be
ASSERT_TRUE.

Change-Id: Ib60b68173b58b77d9789c544c7cb647a752a24d1
Bug: webrtc:10690
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140280
Commit-Queue: Pablo Barrera González <barrerap@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28202}
2019-06-10 08:01:26 +00:00
Sebastian Jansson
24cf2606e4 Adds TCP fairness test to receive side congestion controller.
Bug: webrtc:9883
Change-Id: I3697491285e4f70b8f7857198e4e1ccb0097da5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140883
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28196}
2019-06-07 13:49:37 +00:00
Sebastian Jansson
b13ccc5288 Adds TCP fairness test to GoogCC.
Bug: webrtc:9883
Change-Id: Ie78e51edb08f6c22dbf02168b1d3b067b2c0c55e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140293
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28193}
2019-06-07 11:52:03 +00:00
“Michael
d3a4ebe332 Control rtt_mult addition cap via experiment.
Bug: webrtc:10717
Change-Id: I68f7d8216e1a1611e692dd82ba96890cad98c7de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140284
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28191}
2019-06-07 09:43:26 +00:00
Jonas Olsson
944dacec25 Make interval budget use ratio instead of percent
All usages compare the budget usage to ratios, so we can skip a few
multiplications.

Bug: webrtc:10719
Change-Id: I0205d74762043d972c087c152915e4fdd9510057
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140289
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28190}
2019-06-07 08:53:57 +00:00
Mirko Bonadei
71061bcca8 Replace calls to deprecated googletest APIs.
SetUpTestCase/TearDownTestCase -> SetUpTestSuite/TearDownTestSuite.

TBR=kwiberg@webrtc.org

Bug: None
Change-Id: I6d873c62d6b5c9d7100624d00e1c4894d686a9f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140041
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28189}
2019-06-07 06:41:20 +00:00