Commit graph

22 commits

Author SHA1 Message Date
Erik Språng
eee39206a2 Don't poll EncoderInfo from encoder twice per frame
Bug: webrtc:9890
Change-Id: Id4c2062a1c0c6be699f2096b4c0b334c98f3c4ba
Reviewed-on: https://webrtc-review.googlesource.com/c/111083
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25672}
2018-11-16 14:11:38 +00:00
philipel
ee49f7087f Remove VideoEncoder::SetChannelParameters.
The SetChannelParameters function was used when WebRTC supported decoding
with errors, which we no longer do.

This cleanup CL is related to the work tracked by 9946.

Bug: webrtc:9946
Change-Id: Id2d5ed23031388f890c42651bfbe5f79eda701e5
Reviewed-on: https://webrtc-review.googlesource.com/c/108861
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25505}
2018-11-05 17:37:07 +00:00
Erik Språng
c84cd950b7 Move MockVideoDecoder to api/test.
Move MockVideoDecoder from
modules/video_coding/include/mock/mock_video_codec_interface.h
to
api/test/mock_video_decoder.h

The mock encoder has already moved:
https://webrtc-review.googlesource.com/c/src/+/105620

Keeping the old header until downstream projects have been updated.

Bug: webrtc:9722
Change-Id: I4bc849173a04813064212f17761876695ca3fed4
Reviewed-on: https://webrtc-review.googlesource.com/c/105900
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25170}
2018-10-15 13:45:27 +00:00
Erik Språng
4529fbcfab Move TemporalLayers to api/video_codecs.
Also renaming it Vp8TemporalLayers to show that it is codec specific.

Bug: webrtc:9012
Change-Id: I18187538b8142cdd7538f1a4ed1bada09d040f1f
Reviewed-on: https://webrtc-review.googlesource.com/c/104643
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25137}
2018-10-12 09:15:21 +00:00
Oleh Prypin
a1d9ca47f9 Revert "Add ability to specify if rate controller of video encoder is trusted."
This reverts commit 3e335d1423.

Reason for revert: breaks downstream project

Original change's description:
> Add ability to specify if rate controller of video encoder is trusted.
>
> If rate controller is trusted, we disable the frame dropper in the
> media optimization module.
>
> Bug: webrtc:9722
> Change-Id: I821f21fd74a400ee9d5aa3f6b42d4e569033acbe
> Reviewed-on: https://webrtc-review.googlesource.com/c/105020
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25107}

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

Change-Id: Ifdb0aae684894854a184ec1e7423a7c62e7ba237
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9722
Reviewed-on: https://webrtc-review.googlesource.com/c/105360
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25117}
2018-10-11 15:37:40 +00:00
Erik Språng
3e335d1423 Add ability to specify if rate controller of video encoder is trusted.
If rate controller is trusted, we disable the frame dropper in the
media optimization module.

Bug: webrtc:9722
Change-Id: I821f21fd74a400ee9d5aa3f6b42d4e569033acbe
Reviewed-on: https://webrtc-review.googlesource.com/c/105020
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25107}
2018-10-11 09:07:34 +00:00
Erik Språng
8abd56cfdf Split TemporalLayers and TemporalLayers checker, clean up header.
This CL is a step towards making the TemporalLayers landable in api/ :
* It splits TemporalLayers from TemporalLayersChecker
* It initially renames temporal_layer.h to vp8_temporal_layers.h and
  moved it into the include/ folder
* It removes the dependency on VideoCodec, which was essentially only
  used to determine if screenshare_layers or default_temporal_layers
  should be used, and the number of temporal temporal layers to use.

Subsequent CLs will make further cleanup before attempting a move to api

Bug: webrtc:9012
Change-Id: I87ea7aac66d39284eaebd86aa9d015aba2eaaaea
Reviewed-on: https://webrtc-review.googlesource.com/94156
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24920}
2018-10-02 07:52:02 +00:00
Niels Möller
2377588c82 Add accessor methods for RTP timestamp of EncodedImage.
Intention is to make the member private, but downstream callers
must be updated to use the accessor methods first.

Bug: webrtc:9378
Change-Id: I3495bd8d545b7234fbea10abfd14f082caa420b6
Reviewed-on: https://webrtc-review.googlesource.com/82160
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24352}
2018-08-21 09:15:51 +00:00
Erik Språng
d92288f5ba Add experimental shortened 2-temporal-layer setting
Also adjust to base-layer fraction for the shortened 3-tl pattern to be
60%, just like the 2-tl setting.

This CL removes direct use of the allocation matrix and moves it behind
a static getter.

Bug: webrtc:9477
Change-Id: Ifd7d1edffa0555024fd252834357b926997d13b5
Reviewed-on: https://webrtc-review.googlesource.com/86681
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23834}
2018-07-04 09:25:21 +00:00
Sergio Garcia Murillo
43800f95bf Generalize SimulcastEncoderAdapter, use for H264 & VP8.
* Move SimulcastEncoderAdapter out under modules/video_coding
* Move SimulcastRateAllocator back out to modules/video_coding/utility
* Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility
* Move any VP8 specific code - such as temporal layer bitrate budgeting -
  under codec type dependent conditionals.
* Plumb the simulcast index for H264 in the codec specific and RTP format data structures.

TBR=sprang@webrtc.org,stefan@webrtc.org,titovartem@webrtc.org

Bug: webrtc:5840
Change-Id: I2d3b130622dd7ceec5528f3ab6c46f109e6bafb8
Reviewed-on: https://webrtc-review.googlesource.com/84743
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23715}
2018-06-21 15:57:43 +00:00
Mirko Bonadei
6f440ed5b5 Revert "Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8."
This reverts commit 07efe436c9.

Reason for revert: Breaks downstream project.

cricket::GetSimulcastConfig method signature has been updated.
I think you can get away with a default value for temporal_layers_supported (and then you can remove it after a few days when projects will be updated).


Original change's description:
> Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8.
> 
> * Move SimulcastEncoderAdapter out under modules/video_coding
> * Move SimulcastRateAllocator back out to modules/video_coding/utility
> * Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility
> * Move any VP8 specific code - such as temporal layer bitrate budgeting -
>   under codec type dependent conditionals.
> * Plumb the simulcast index for H264 in the codec specific and RTP format data structures.
> 
> Bug: webrtc:5840
> Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e
> Reviewed-on: https://webrtc-review.googlesource.com/64100
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23705}

TBR=sprang@webrtc.org,stefan@webrtc.org,mflodman@webrtc.org,hta@webrtc.org,sergio.garcia.murillo@gmail.com,titovartem@webrtc.org,agouaillard@gmail.com

Change-Id: Ic9d3b1eeaf195bb5ec2063954421f5e77866d663
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:5840
Reviewed-on: https://webrtc-review.googlesource.com/84760
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23710}
2018-06-21 13:41:14 +00:00
Sergio Garcia Murillo
07efe436c9 Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8.
* Move SimulcastEncoderAdapter out under modules/video_coding
* Move SimulcastRateAllocator back out to modules/video_coding/utility
* Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility
* Move any VP8 specific code - such as temporal layer bitrate budgeting -
  under codec type dependent conditionals.
* Plumb the simulcast index for H264 in the codec specific and RTP format data structures.

Bug: webrtc:5840
Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e
Reviewed-on: https://webrtc-review.googlesource.com/64100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23705}
2018-06-21 12:23:03 +00:00
Yves Gerey
665174fdbb Reformat the WebRTC code base
Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +00:00
Erik Språng
566124a6df Move BitrateAllocation to api/ and rename it VideoBitrateAllocation
Since the webrtc_common build target does not have visibility set, we
cannot easily use BitrateAllocation in other parts of Chromium.
This is currently blocking parts of chromium:794608, and I know of other
usage outside webrtc already, so moving it to api/ should be warranted.

Also, since there's some naming confusion and this class is video
specific rename it VideoBitrateAllocation. This also fits with the
standard interface for producing these: VideoBitrateAllocator.

Bug: chromium:794608
Change-Id: I4c0fae40f9365e860c605a76a4f67ecc9b9cf9fe
Reviewed-on: https://webrtc-review.googlesource.com/70783
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22986}
2018-04-23 15:31:27 +00:00
Erik Språng
82fad3d513 Remove TemporalLayersFactory and associated classes
As the rate allocation has been moved into entirely into
SimulcastRateAllocator, and the listeners are thus no longer needed,
this class doesn't fill any other purpose than to determine if
ScreenshareLayers or TemporalLayers should be created for a given
simulcast stream. This can however be done just from looking at the
VideoCodec instance, so changing this into a static factory method.

Due to dependencies from upstream projects, keep the class name and
field in VideoCodec around for now.

Bug: webrtc:9012
Change-Id: I028fe6b2a19e0d16b35956cc2df01dcf5bfa7979
Reviewed-on: https://webrtc-review.googlesource.com/63264
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22529}
2018-03-21 10:20:48 +00:00
Niels Möller
bf3dbb4a69 Delete payload_type from VCMEncoderDatabase and vcm::VideoSender.
Bug: webrtc:8830
Change-Id: Ie6a874023618a5540e138b34edfcad1ce6e8d391
Reviewed-on: https://webrtc-review.googlesource.com/62102
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22474}
2018-03-16 13:43:01 +00:00
Magnus Jedvert
46a2765c56 Reland "Update internal SW codecs to return unique_ptrs"
This reverts commit 34c8e6bce8.

Reason for revert: Fix Android compilation

Original change's description:
> Revert "Update internal SW codecs to return unique_ptrs"
>
> This reverts commit 4fe6adc06a.
>
> Reason for revert: Breaks android compile.
>
> Original change's description:
> > Update internal SW codecs to return unique_ptrs
> >
> > TBR=stefan@webrtc.org
> >
> > Bug: webrtc:7925
> > Change-Id: I84239b071a2608d928f09b06809090eec5eafb14
> > Reviewed-on: https://webrtc-review.googlesource.com/21165
> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20650}
>
> TBR=magjed@webrtc.org,sprang@webrtc.org,stefan@webrtc.org
>
> Change-Id: If33c3a0ee0dfce63d105558a2897a472f0633306
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:7925
> Reviewed-on: https://webrtc-review.googlesource.com/22540
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20652}

TBR=magjed@webrtc.org,sprang@webrtc.org,stefan@webrtc.org

Change-Id: Ic8551af4687e927c9b605060155abdd5bc6208b2
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/22541
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20655}
2017-11-13 14:23:58 +00:00
Magnus Jedvert
34c8e6bce8 Revert "Update internal SW codecs to return unique_ptrs"
This reverts commit 4fe6adc06a.

Reason for revert: Breaks android compile.

Original change's description:
> Update internal SW codecs to return unique_ptrs
> 
> TBR=stefan@webrtc.org
> 
> Bug: webrtc:7925
> Change-Id: I84239b071a2608d928f09b06809090eec5eafb14
> Reviewed-on: https://webrtc-review.googlesource.com/21165
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20650}

TBR=magjed@webrtc.org,sprang@webrtc.org,stefan@webrtc.org

Change-Id: If33c3a0ee0dfce63d105558a2897a472f0633306
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/22540
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20652}
2017-11-13 13:02:30 +00:00
Magnus Jedvert
4fe6adc06a Update internal SW codecs to return unique_ptrs
TBR=stefan@webrtc.org

Bug: webrtc:7925
Change-Id: I84239b071a2608d928f09b06809090eec5eafb14
Reviewed-on: https://webrtc-review.googlesource.com/21165
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20650}
2017-11-13 12:31:18 +00:00
Niels Möller
a0565999db Delete VCMSendStatisticsCallback and corresponding use of ProcessThread
Bug: webrtc:8422
Change-Id: I5863266a0226d475c4fdd810f2f6f1acdf922df3
Reviewed-on: https://webrtc-review.googlesource.com/14880
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20440}
2017-10-26 08:13:55 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/modules/video_coding/video_sender_unittest.cc (Browse further)