Commit graph

18 commits

Author SHA1 Message Date
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
e624d078bc Always update libvpx configuration if bitrate has changed.
OnRatesUpdated() is called every time the bitrate estimate, or once per
second. However, since we don't want to reconfigure libvpx too often,
just in case it interferes with the rate controller, so
ScreenshareLayers contains a boolean |bitrate_update_| which indicate
if the configuration should be updated on a call to
UpdateConfiguration().

However, it two rate updates happened between two frames, the first of
which changes the rates and second one does not, |bitrate_update_| will
be reset to false and the encoder won't get the desired config.

This CL makes sure we update the configuration iff the rate has changed
at any time since the last call to UpdateConfiguration().

Bug: webrtc:9012
Change-Id: I62af36cffe20ecb7d3f403b3eb11f23a9692d719
Reviewed-on: https://webrtc-review.googlesource.com/69040
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22826}
2018-04-11 15:27:56 +00:00
Niels Möller
150dcb0a9a Delete logic to set picture id and tl0 pic idx in encoders.
It would be nice to also delete the fields from CodecSpecificInfo,
but those fields are used on the receive side.

Bug: webrtc:8830
Change-Id: I1a3f13ea2c024cbd73b33fd9dd58e531d3576a55
Reviewed-on: https://webrtc-review.googlesource.com/64780
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22625}
2018-03-27 13:11:16 +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
Erik Språng
bb60a3a5fa Refactor VP8 TemporalLayers
This CL moves all temporal layer rate allocation from
DefaultTemporalLayers and ScreenshareLayers into SimulcastRateAllocator.
This means we don't need an extra call-out to the TemporalLayers
interface to get the last allocation, which simplifies the code path a
lot.

It also paves the wave for removing the TemporalLayersFactory interface
(in a separate cl), which will further simplify the ownership model.

Bug: webrtc:9012
Change-Id: I6540b1848efa1a136dce449f13902ad479d5ee37
Reviewed-on: https://webrtc-review.googlesource.com/62420
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22502}
2018-03-19 18:14:21 +00:00
Anders Carlsson
beabdcb498 Break VP8 temporal_layers dependency on libvpx.
This is in preparation for
https://webrtc-review.googlesource.com/c/src/+/36340

With these changes we can avoid some strange #ifdefs in the code
that uses temporal layers.

Bug: webrtc:7925
Change-Id: I472210738ccc9f73812b8863951befeabec56f15
Reviewed-on: https://webrtc-review.googlesource.com/41280
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21759}
2018-01-25 10:37:21 +00:00
Erik Språng
db9e9d5486 Make screenshare_layers frame dropper less aggressive
Try to use frame timestamps first if they look reasonable, otherwise
use realtime clock.
Also, lower limit from 90% of target to 85%.

Bug: webrtc:4172, chromium:802290
Change-Id: Iad489be7c7cf637345be4795e5089936ab9fab07
Reviewed-on: https://webrtc-review.googlesource.com/41041
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21729}
2018-01-22 21:15:29 +00:00
Erik Språng
27a457d172 Smoother frame dropping when screenshare_layers limits fps
This is a reland of https://webrtc-review.googlesource.com/34380

The main problem with that CL was that we used frame timestamps as basis
for frame dropping, but those might not be continuous or even populated
in some circumstances.

Additionally, I found that the bitrate was off since the encoder does
not not take the dropped frames into account, so if we drop every other
frame continiusoly, the bitrate sent will be around half of the target.

Patch set 1 is the original CL, subsequent patch sets cotains fixes.

Bug: webrtc:4172
Change-Id: I8ec8dddcebf4ce44f28dd9055cf9c46bbd68e4a6
Reviewed-on: https://webrtc-review.googlesource.com/39201
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21601}
2018-01-12 14:40:40 +00:00
Erik Språng
afb3fc3558 Revert "Smoother frame dropping when screenshare_layers limits fps"
This reverts commit 28a06b16cc.

Reason for revert: Causes some unexpected perf changes.

Original change's description:
> Smoother frame dropping when screenshare_layers limits fps
> 
> Currently, when input fps is higher than the configured target fps in
> screenshare_layers, we drop frames with the help of a rate tracker using
> a one second sliding window. This is not optimal.
> For instance, given a 5fps limit and a 30fps capturer, the window will
> not be saturated until we have added the first 5 frames. Then we will
> drop all frames until the oldest one drops out, at which point we can
> immediately fill it's place. This results in quick 5 frame bursts every
> second.
> 
> In addition to rate tracker, also set a limit on minimum interval
> required between input frames, based on target frame rate.
> 
> Bug: webrtc:4172
> Change-Id: I49f0abf4d549673cc6b3fafe580b529ea3feaf57
> Reviewed-on: https://webrtc-review.googlesource.com/34380
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21325}

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

Change-Id: I7ca5b0c847310dbb11dce28773082eac946c0ba4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:4172
Reviewed-on: https://webrtc-review.googlesource.com/34780
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21354}
2017-12-19 11:21:11 +00:00
Erik Språng
28a06b16cc Smoother frame dropping when screenshare_layers limits fps
Currently, when input fps is higher than the configured target fps in
screenshare_layers, we drop frames with the help of a rate tracker using
a one second sliding window. This is not optimal.
For instance, given a 5fps limit and a 30fps capturer, the window will
not be saturated until we have added the first 5 frames. Then we will
drop all frames until the oldest one drops out, at which point we can
immediately fill it's place. This results in quick 5 frame bursts every
second.

In addition to rate tracker, also set a limit on minimum interval
required between input frames, based on target frame rate.

Bug: webrtc:4172
Change-Id: I49f0abf4d549673cc6b3fafe580b529ea3feaf57
Reviewed-on: https://webrtc-review.googlesource.com/34380
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21325}
2017-12-18 15:28:39 +00:00
Erik Språng
13044c1b53 Reduce time between sync frames for temporal layers vp8 screensharing.
This is expected to result in a slight loss of overall quality, but
should offset by quicker switching between temporal layers with flaky
connections.

Bug: webrtc:7694
Change-Id: Ib605802bb59f12773652324ac66cdf4774ae6bb6
Reviewed-on: https://webrtc-review.googlesource.com/6881
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20160}
2017-10-05 11:10:38 +00:00
Ilya Nikolaevskiy
586629155c Implement ScreenshareTemporalLayersChecker
Bug: none
Change-Id: Ic95156d0f47d186e2289264aa9a916511a8e4510
Reviewed-on: https://webrtc-review.googlesource.com/4960
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20143}
2017-10-04 14:18:04 +00:00
Ilya Nikolaevskiy
bf35298996 Implement temporal layers checkers for vp8
All frames are checked against hard-coded dependency graph 
using new helper class. It's invoked in RTC_DCHECK(). Only 
DefaultTemporalLayers are fully implemented in this CL, checker 
for ScreenshareLayers is not doing anything for now.

Bug: none
Change-Id: I3ec017176d8c25f7572c8f161e52f2ebfac8220f
Reviewed-on: https://webrtc-review.googlesource.com/3740
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20066}
2017-10-02 09:14:59 +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/codecs/vp8/screenshare_layers.cc (Browse further)