Commit Graph

116 Commits

Author SHA1 Message Date
dfe8ca0d43 Layering and rate allocation for VP9 screen sharing.
- Two quality layers (same resolution, different bitrate).
- Max bitrate of low layer is limited to 200kbps. The choice of the
limit is driven by VP8 screen sharing which limits max bitrate of low
temporal layer to 200kbps. Using the same value for VP9 guarantees
that there will be no regressions for participants with limited
bandwidth.
- Max bitrate of high layer is limited to 500kbps. According to test
results this value is enough to get up to +5dB higher PSNR than VP8
SS provides on 1.2Mbps (max total bitrate for VP8 SS) link.
- Max total sent bitrate is limited to 700kbps. It is 500kbps lower
than that in VP8 SS (1200kbps).

Bug: webrtc:9261
Change-Id: I7919cc3933064664567c39e380a44cad0c65f1e8
Reviewed-on: https://webrtc-review.googlesource.com/76380
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23226}
2018-05-15 07:06:10 +00:00
bd0954e83f Encode key frame when enabling upper layer.
Request key frame when upper spatial layer is enabled dynamically
and inter-layer prediction is disabled or limited to key pictures.

This is needed to force encoder to produce RTP compatible bitstream
where temporal prediction is limited to the same spatial layer.

Bug: webrtc:9217
Change-Id: I4fc1e3f067689ba7b5c6bd1f5af922a0637f03d7
Reviewed-on: https://webrtc-review.googlesource.com/73580
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23102}
2018-05-03 13:22:01 +00:00
6a8f30e5a3 Add control for inter-layer prediction mode.
This allows to control inter-layer prediction at encoding VP9 SVC.
There are three options:
1. Disabled.
2. Enabled for all pictures.
3. Enabled for key pictures, disabled for others.

Inter-layer prediction is enabled for all pictures by default.

Bug: none
Change-Id: I49fe43d8744c92bec349d815100ba158519f0664
Reviewed-on: https://webrtc-review.googlesource.com/71500
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23049}
2018-04-27 06:20:15 +00:00
bc0f0d3ded Rename end_of_superframe to end_of_picture.
For consistency with the VP9 RTP spec which uses term "picture" for set
of frames which belong to the same time instance.

Bug: none
Change-Id: I30e92d5debb008feb58f770b63fe10c2e0029267
Reviewed-on: https://webrtc-review.googlesource.com/72180
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23040}
2018-04-26 15:47:17 +00:00
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
2a1f183e99 Set marker bit on last encoded spatial layer.
In order to handle per-layer frame dropping both VP9 encoder wrapper
and RTP packetizer were modified.

- Encoder wrapper buffers last encoded frame and passes it to
packetizer after frame of next layer is encoded or encoding of
superframe is finished.
- Encoder wrapper sets end_of_superframe flag on last encoded frame of
superframe before passing it to packetizer.
- If end_of_superframe is True then packetizer sets marker bit on last
packet of frame.

Bug: webrtc:9066
Change-Id: I1d45319fbe6bc63d01721ea67bfb7440d4c29275
Reviewed-on: https://webrtc-review.googlesource.com/65540
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22722}
2018-04-04 10:40:19 +00:00
86684960b3 Adding layering configurator and rate allocator for VP9 SVC.
The configurator decides number of spatial layers, their resolution
and bitrate thresholds based on given input resolution and maximum
number of spatial layers.

The allocator distributes available bitrate across spatial and
temporal layers. If there is not enough bitrate to provide acceptable
quality for all spatial layers allocator disables enhancement layers
one by one until the condition is met or number of layers is reduced
to one.

VP9 SVC related unit tests have been updated. Input resolution and
bitrate in these tests have been increased to the level enough to
provide desirable number of spatial layers.

Bug: webrtc:8518
Change-Id: I9df790920227c7f7dd4d42a50a856c22f0f4389b
Reviewed-on: https://webrtc-review.googlesource.com/60340
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22672}
2018-03-29 10:16:47 +00:00
1d2b627438 Use frame generator in video codec unit tests.
There is no need to use real video as input for encoder in unit tests.
Using generator simplifies testing on mobile devices (no need to upload
files to device).

Bug: none
Change-Id: Ic48609cc6f8eecf90d9956edfdd33135be949398
Reviewed-on: https://webrtc-review.googlesource.com/64526
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22648}
2018-03-28 13:07:16 +00:00
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
13e743419d Merge SimulcastStream and SpatialLayer structures.
Both simulcast stream and spatial layer can be described with the same
set of parameters. There is no need in two separate definitions.

1. Original definition of SpatialLayer is removed.
2. SimulcastStream is renamed to SpatialLayer.
3. SimulcastStream is equated to SpatialLayer using typedef.

Bug: webrtc:8518
Change-Id: I90761952b032a1b71fc4bba11f74a6daaf58880a
Reviewed-on: https://webrtc-review.googlesource.com/57102
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22272}
2018-03-02 16:30:19 +00:00
98a867ccd2 Rename VideoCodecTest to VideoCodecUnitTest.
The VideoCodecTest class is a fixture base class for the
libvpx-VP8, libvpx-VP9, and OpenH264 unit tests. It is unrelated
to the VideoProcessor tests, which we colloquially refer to as
the "codec test".

This rename is thus to reduce this confusion. It should have no
functional impact.

Bug: webrtc:8448
Change-Id: If73443bda5df0f29a71ce6ce069ac128795ff0ad
Reviewed-on: https://webrtc-review.googlesource.com/47160
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21867}
2018-02-02 10:27:33 +00:00
46a2765c56 Reland "Update internal SW codecs to return unique_ptrs"
This reverts commit 34c8e6bce8af0c31f2b0b31d691a6a931fa3cb7b.

Reason for revert: Fix Android compilation

Original change's description:
> Revert "Update internal SW codecs to return unique_ptrs"
>
> This reverts commit 4fe6adc06a8524ac25f85260bfe392eb31dae6b4.
>
> 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
34c8e6bce8 Revert "Update internal SW codecs to return unique_ptrs"
This reverts commit 4fe6adc06a8524ac25f85260bfe392eb31dae6b4.

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
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
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
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