Commit graph

371 commits

Author SHA1 Message Date
Sergey Silkin
be71a1ee08 Replace VP9 screen sharing.
- Remove referencing control from encoder wrapper. Use fixed temporal
prediction structure.
- Remove flexible mode from encoder wrapper. It only worked with
referencing control which this CL removes.
- Remove external framerate/bitrate controller. Keep codec's internal
frame dropping enabled at screen sharing.
- Use GetSvcConfig() to configure layering.

Bug: webrtc:9261
Change-Id: I355baa6aab7b98ac5028b3851d1f8ccc82a308e0
Reviewed-on: https://webrtc-review.googlesource.com/76801
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23311}
2018-05-18 15:11:46 +00:00
Stefan Holmer
812ceafb5a Ensure render time is zero when playout delay is zero so that minimal latency in the render pipeline is ensured.
Bug: webrtc:9135
Change-Id: Id9ae8ec59536808ba8923c73dd46abfe3fa6fe79
Reviewed-on: https://webrtc-review.googlesource.com/75600
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23309}
2018-05-18 14:47:26 +00:00
Niels Möller
0a8f43580f Move VideoEncoderConfig from call/ to api/.
Bug: webrtc:8830
Change-Id: I42abd45bff9a70fe00733424b34874925c523dc8
Reviewed-on: https://webrtc-review.googlesource.com/77683
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23303}
2018-05-18 12:58:16 +00:00
Anders Carlsson
7eb8e9fd7b Add RegisterExternalDecoder in VideoCodingModule.
In preparation for landing https://webrtc-review.googlesource.com/c/src/+/72441
a downstream project that uses the VideoCodingModule needs to be able to
inject a decoder object created from the outside, just like how encoders
are possible to inject.

Bug: webrtc:7925
Change-Id: Ibaeffda55f84410436d79f75730e7352e298b9f0
Reviewed-on: https://webrtc-review.googlesource.com/77160
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23297}
2018-05-18 09:43:26 +00:00
Niels Möller
c948fe62fd Delete unneeded includes of call/video_config.h.
Bug: webrtc:8830
Change-Id: I6114b47e5524a6d2450108388236478b1ceafb67
Reviewed-on: https://webrtc-review.googlesource.com/77425
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23295}
2018-05-18 09:00:56 +00:00
Niels Möller
4c8811b255 Delete some obsolete forward declarations
Bug: None
Change-Id: I3a9b59bf3dd63c206854ab949cf2d606046182c9
Reviewed-on: https://webrtc-review.googlesource.com/77427
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23292}
2018-05-18 07:29:25 +00:00
philipel
09133af36f Check number of nalus in packet before checking nalu types.
Bug: chromium:840536
Change-Id: Ia4dcf322ad6290691fd01b58fb02cd868714c92e
Reviewed-on: https://webrtc-review.googlesource.com/77121
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23283}
2018-05-17 12:52:11 +00:00
philipel
d5addcaf1e Add philipel to modules/video_coding/OWNERS.
Bug: None
Change-Id: I982ac84a58aff27a6f8f9cf50005ac732f540785
Reviewed-on: https://webrtc-review.googlesource.com/77362
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23274}
2018-05-17 10:14:31 +00:00
Kári Tristan Helgason
8eeda499f7 Enable more VideoCodecTests on iOS.
Bug: webrtc:4755
Change-Id: I403834dbe04cc3899847eb10e5595a24e6001507
Reviewed-on: https://webrtc-review.googlesource.com/76602
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23273}
2018-05-17 09:13:31 +00:00
Rasmus Brandt
8dd4db49e2 Create/destroying codecs on task queue + switch to TaskQueueForTest.
After https://webrtc-review.googlesource.com/c/src/+/70740, we are
creating/destroying the codecs on a task queue in the VideoStreamEncoder. This
CL updates the VideoCodecTest to do the same.

Also, this CL switches from manually Wait()'ing on the task queue to using
TaskQueueForTest::SendTask.

Bug: None
Change-Id: Ia0398b24e32e9cc5361ba5ee4c08441116def18e
Reviewed-on: https://webrtc-review.googlesource.com/76800
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23257}
2018-05-16 08:15:23 +00:00
Niels Möller
59130a11b8 Delete deprecated version of VideoCodecInitializer::SetupCodec.
A followup to https://webrtc-review.googlesource.com/71380 and
https://webrtc-review.googlesource.com/69986, deleting the
nack_enabled flag.

Bug: webrtc:8830
Change-Id: Ie53f7a1e131be5347936ed44cee7167295026d57
Reviewed-on: https://webrtc-review.googlesource.com/76760
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23256}
2018-05-16 08:03:24 +00:00
Rasmus Brandt
8694f29b30 Rename VideoProcessorIntegrationTest -> VideoCodecTest.
This CL simply renames the test cases that were not renamed in
prior CLs.

Bug: None
Change-Id: If616eb823e1453bc92ba037722b77a219d54409c
Reviewed-on: https://webrtc-review.googlesource.com/76780
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23240}
2018-05-15 11:28:21 +00:00
Rasmus Brandt
a7e4844298 Move H.264 codec test activation directly behind build flag.
Bug: None
Change-Id: I5dd366aba16b440a8e99826e116db0d6b68bf844
Reviewed-on: https://webrtc-review.googlesource.com/76621
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23231}
2018-05-15 08:00:00 +00:00
Sergey Silkin
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
“Michael
b54500ec32 VP9 SVC minimum bit rate thresholds too low for 720p
Changed minimum bit rate threshold formula to raise the minimum bit rate
at which 720p video is presented in VP9 SVC to ensure that the video
quality for VP9 SVC is the same or better than VP8 SIM.  The minimum bit
rate threshold values for lower resolutions remain largely unchanged.
Also changed maximum bit rate threshold formula to lower the maximum
bit rate for low resolutions (e.g., 180p) in order to ensure higher
frame rates when downlink bit rates are very low (e.g., < 100 kbps).

Bug: webrtc:9242
Change-Id: I8f9c76c9188b98f3fd40a608551b576b0c3b8f34
Reviewed-on: https://webrtc-review.googlesource.com/75244
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23218}
2018-05-14 15:48:29 +00:00
Anders Carlsson
b330688ef7 Fix build errors when rtc_use_builtin_sw_codecs is set to false.
The previous effort of building WebRTC without SW codecs stopped when
libjingle_peerconnection was possible to build. In order to make the
group("default") target build, this basically updates a bunch of
tests to explicitly depend on the built-in software video codecs.

Bug: webrtc:7925
Change-Id: I2715414770c197fca01cb8dbde173a21f4434500
Reviewed-on: https://webrtc-review.googlesource.com/70503
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23216}
2018-05-14 13:24:29 +00:00
Niels Möller
c6ce9c5938 New file api/video/BUILD.gn
Build targets involving files under api/video/ are moved into this
file, from api/BUILD.gn. In addition, drop "_api" part of target
names, and move the header file api/videosinkinterface.h to
api/video/video_sink_interface.h.

Bug: webrtc:9253
Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155
Reviewed-on: https://webrtc-review.googlesource.com/75500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-14 06:57:38 +00:00
Sergey Silkin
377ef24a8f Remove extra reference from GOF.
This removes second reference for frame 3 in GOF predefined for 3
temporal layers since encoder never use that reference.

Bug: webrtc:9245
Change-Id: I6fbdbe7d3c753dda7fbcfcbd05f3530f70f80728
Reviewed-on: https://webrtc-review.googlesource.com/74705
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@google.com>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23193}
2018-05-09 18:18:43 +00:00
Sami Kalliomäki
ee98be7811 Fix handling non-tightly packed ByteBuffers in HardwareVideoDecoder.
Before this CL, there would be an out-of-bounds write in the ByteBuffer
copying when a decoded frame had height != sliceHeight.

Bug: webrtc:9194
Change-Id: Ibb80e5555e8f00d9e1fd4cb8a73f5e4ccd5a0b81
Tested: 640x360 loopback with eglContext == null in AppRTCMobile on Pixel.
Reviewed-on: https://webrtc-review.googlesource.com/74120
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23184}
2018-05-09 09:15:46 +00:00
Niels Möller
8df3a388a3 Deprecate RTPFragmentationHeader argument to VideoDecoder::Decode
Intend to delete in a later cl.

Bug: webrtc:6471
Change-Id: Icf0fcd40e0d3287dc59b684fae6552b40b47204a
Reviewed-on: https://webrtc-review.googlesource.com/39511
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23162}
2018-05-08 08:09:35 +00:00
philipel
156436056e VP9 RTP frame reference finder cleanup.
Bug: chromium:838402
Change-Id: Ib6117db5aeb83fc03f72759be2e42d0a4ceb94b6
Reviewed-on: https://webrtc-review.googlesource.com/60260
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23139}
2018-05-07 11:29:44 +00:00
“Michael
e8492fee6b Mitigate RTP pic ID errors in VP9 SVC non-flexible mode
The code changes in this CL configure VP9 SVC to drop a superframe when
the spatial base layer is dropped and to not drop upper spatial layers
when the spatial base layer is not dropped. The changes are effective in
non-flexible mode when codec_.mode == kRealtimeVideo and
number of spatial layers > 1.

Bug: none
Change-Id: I27481b78f733cfc6c007d1ad9f45d69263853149
Reviewed-on: https://webrtc-review.googlesource.com/74261
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23127}
2018-05-04 16:25:54 +00:00
Kári Tristan Helgason
9d96e92316 Rewrite videoprocessor integrationtest to use public fixture.
This CL creates a test fixture for the videoprocessor integration tests
and exposes it as part of the public API. It also rewrites the current
versions of the tests to build on this new paradigm. The motivation for
this is to easily allow projects that build on top of webrtc to add
integration-level tests for their own custom codec implementations in a
way that does not link them too tightly to the internal implementations
of said tests.

Bug: None
Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2
Reviewed-on: https://webrtc-review.googlesource.com/72481
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23118}
2018-05-04 12:02:44 +00:00
Karl Wiberg
7ba22b8eea Break out the part of the iSAC codec that's used for Voice Activity Detection
The audio processing code is using parts of the iSAC codec to do voice
activity detection (VAD), but it's undesirable for it to pull in the
entire iSAC codec as a dependency. So this CL factors out the parts of
iSAC that's needed for VAD to a separate build target.

Bug: webrtc:8396
Change-Id: I884e25d8fd0bc815fca664352b0573b4b173880e
Reviewed-on: https://webrtc-review.googlesource.com/69640
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23110}
2018-05-04 08:53:34 +00:00
Sergey Silkin
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
philipel
a157e08093 VP9 temporal index bounds check.
Bug: chromium:838672
Change-Id: Ia531327858b6e40cb7fa03ca1b98c120ba4e1389
Reviewed-on: https://webrtc-review.googlesource.com/73701
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23099}
2018-05-03 12:00:41 +00:00
Jonas Olsson
3e18c82820 Reland "Reland "Remove our stream << overloads from non-test build targets.""
This is a reland of d7ee72041f

Original change's description:
> Reland "Remove our stream << overloads from non-test build targets."
>
> This is a reland of c841d18d25
>
> Original change's description:
> > Remove our stream << overloads from non-test build targets.
> >
> > Most are removed entirely, but RtcErrorType, RtpTransceiverDirection, IPAddress and
> > SocketAddress are kept behind gtest's #ifdef UNIT_TEST.
> >
> > Bug: webrtc:8982
> > Change-Id: I36db19891e7d25aeacb08b9a08aa2b4004765e70
> > Reviewed-on: https://webrtc-review.googlesource.com/64143
> > Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> > Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22916}
>
>
> Bug: webrtc:8982
> Change-Id: Ibe08c6270e5e693eb661a6ce9e8f074b34ef8123
> Reviewed-on: https://webrtc-review.googlesource.com/71161
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22949}

TBR=deadbeef@webrtc.org,kwiberg@webrtc.org,asapersson@webrtc.org,jonasolsson@webrtc.org,benwright@webrtc.org

Bug: webrtc:8982
Change-Id: I29247d1c28e99af36ef228d8c75b4adecbd7b199
Reviewed-on: https://webrtc-review.googlesource.com/72681
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23092}
2018-05-03 10:41:41 +00:00
Niels Möller
f133856dfa Delete nack_enabled flag in encoder configuration.
This is a followup to cl https://webrtc-review.googlesource.com/71380,
which reworked the way encoder resilience is done, and made the
nack_enabled flag unused.

Bug: webrtc:8830
Change-Id: I3de2508c97bc71e01c8f2232d16cd1f33e57fe4a
Reviewed-on: https://webrtc-review.googlesource.com/69986
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23080}
2018-05-02 16:05:27 +00:00
Kári Tristan Helgason
cad94449dd Remove H264 CHP field trial code.
Bug: webrtc:8317
Change-Id: I2da3cc6578dd8ff6e88052bc33cd38cb92af46dc
Reviewed-on: https://webrtc-review.googlesource.com/73242
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23077}
2018-05-02 13:42:37 +00:00
Sergey Silkin
5613879b7b Fill drops with last decoded frame.
Fill drops with last decoded frame to make them look like freeze at
playback and to keep decoded spatial layers in sync.

Bug: none
Change-Id: I65f7c21100985c22932a1edd441b6c724833c11e
Reviewed-on: https://webrtc-review.googlesource.com/73685
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23076}
2018-05-02 10:46:06 +00:00
Sergey Silkin
3c30c9cb9f Decode base reference frame if current layer was dropped.
If frame of current layer was dropped, pass base frame to decoder if
non_ref_for_inter_layer_pred is set to true.

Bug: none
Change-Id: If7bf5220b74f424106edf74867c9afa8cc2b1ec5
Reviewed-on: https://webrtc-review.googlesource.com/73440
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23074}
2018-05-02 10:17:01 +00:00
Sergey Silkin
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
Sergey Silkin
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
Niels Möller
65fb4049c1 Don't expose resilience mode in VP8 and VP9 configuration.
This deletes the resilienceOn flag in VideoCodecVP8 and VideoCodecVP9.
Instead, the implementations of VP8 and VP9 set resilience mode
internally, based on the configuration of temporal and spatial layers.

The nack_enabled argument to VideoCodecInitializer::SetupCodec becomes
unused with this cl. In a followup, it will be deleted, together with
the corresponding argument to VideoStreamEncoder methods.

An applications which really wants to configure resilience differently
can do that by injecting an EncoderFactory with encoders behaving
as desired.

Bug: webrtc:8830
Change-Id: I9990faf07d3e95c0fb4a56fcc9a56c2005b4a6fa
Reviewed-on: https://webrtc-review.googlesource.com/71380
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23025}
2018-04-25 13:54:33 +00:00
Anders Carlsson
1f433e46db Mark built-in software video codecs as poisonous.
The goal is to make these injectable, and only VP8 and VP9 specific
targets should depend on them.

Bug: webrtc:7925
Change-Id: Ie9239a54d197fe70c93de0582797211fef6997a2
Reviewed-on: https://webrtc-review.googlesource.com/72082
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23021}
2018-04-25 11:34:33 +00:00
Rasmus Brandt
cd7da92012 Add MediaCodec VP tests for uncommon resolutions.
Bug: None
Change-Id: Ibfc35af3635c3b3a50027c4cd828f78e7a438dcd
Reviewed-on: https://webrtc-review.googlesource.com/72342
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23020}
2018-04-25 11:31:13 +00:00
Rasmus Brandt
abc821cd31 Allow multiple calls to ProcessFramesAndMaybeVerify with frame writers enabled.
Bug: None
Change-Id: Ic6e52401ec2db3d0bcaca3605c28763123a4eeb8
Reviewed-on: https://webrtc-review.googlesource.com/72343
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23016}
2018-04-25 09:46:13 +00:00
Åsa Persson
a945aee72e Make quality scaler downscale faster.
Include dropped frames by the encoder in the frame drop percentage.

To react faster at low framerates:
- Use ExpFilter instead of MovingAverage to filter QP values.
- Reduce sampling interval while waiting for minimum number of needed frames (when not in fast rampup mode).

A separate slower ExpFilter is used for upscaling.

Bug: webrtc:9169
Change-Id: If7ff6c3bd4201fda2da67125889838fe96ce7061
Reviewed-on: https://webrtc-review.googlesource.com/70761
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23014}
2018-04-25 09:08:21 +00:00
Sergey Silkin
c5a131a5fb Add non_ref_for_inter_layer_pred to VP9 RTP.
This converts the reserved bit in VP9 RTP payload descriptor into the
flag which indicates whether current frame can be used for prediction
of next spatial layer or not.

VP9 encoder wrapper sets non_ref_for_inter_layer_pred=false for all
frames for now.

Bug: none
Change-Id: I32f68868686475905fb09173cffd2b6e1bedcb7c
Reviewed-on: https://webrtc-review.googlesource.com/71080
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23010}
2018-04-24 19:20:30 +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
Niels Möller
e8a9c45cc1 Delete enum VP8ResilienceMode.
We only support on (formely kResilientStream) and off (formely
kResilienceOff). The third mode, kResilientFrames, was not
implemented.

Bug: None
Change-Id: Ida82f6a33eda9d943ea70bc8ae4e6bddb720b0e8
Reviewed-on: https://webrtc-review.googlesource.com/71481
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22984}
2018-04-23 15:10:26 +00:00
Karl Wiberg
bb23c838f5 GN hack to tag targets as poisonous (and use it with audio codecs)
Only specially taggged targets may transitively depend on poisonous
targets. We first apply it to audio codecs.

This makes it much clearer exactly what parts of the code still have
dependencies on the audio codecs (and we want to eventually get rid of
pretty much all of them).

Bug: webrtc:8396, webrtc:9121
Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43
Reviewed-on: https://webrtc-review.googlesource.com/69520
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22979}
2018-04-23 13:41:47 +00:00
Sergey Silkin
1322dbc81a Fix calculation of target bitrate of VP9 spatial layer.
This fixes misprint in the code which calculates target bitrate of a
VP9 spatial layer where "-" was used instead of "+".

Bug: none
Change-Id: I17d76a84d00e453c055c068968d7b276e9c23f51
Reviewed-on: https://webrtc-review.googlesource.com/71663
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22974}
2018-04-23 11:31:47 +00:00
Åsa Persson
04d5f1d2e5 QualityScaler: rename classes and methods from "QP" to "Qp".
Bug: none
Change-Id: Iea6d69149912a6804e2a54262e89114f10a49394
Reviewed-on: https://webrtc-review.googlesource.com/71482
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22970}
2018-04-23 08:39:16 +00:00
Taylor Brandstetter
bd7392829a Revert "Reland "Remove our stream << overloads from non-test build targets.""
This reverts commit d7ee72041f.

Reason for revert: Broke downstream build which was using SdpAudioFormat operator<<

Original change's description:
> Reland "Remove our stream << overloads from non-test build targets."
> 
> This is a reland of c841d18d25
> 
> Original change's description:
> > Remove our stream << overloads from non-test build targets.
> >
> > Most are removed entirely, but RtcErrorType, RtpTransceiverDirection, IPAddress and
> > SocketAddress are kept behind gtest's #ifdef UNIT_TEST.
> >
> > Bug: webrtc:8982
> > Change-Id: I36db19891e7d25aeacb08b9a08aa2b4004765e70
> > Reviewed-on: https://webrtc-review.googlesource.com/64143
> > Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> > Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22916}
> 
> TBR=deadbeef@webrtc.org,kwiberg@webrtc.org,asapersson@webrtc.org,jonasolsson@webrtc.org,benwright@webrtc.org
> 
> Bug: webrtc:8982
> Change-Id: Ibe08c6270e5e693eb661a6ce9e8f074b34ef8123
> Reviewed-on: https://webrtc-review.googlesource.com/71161
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22949}

TBR=deadbeef@webrtc.org,kwiberg@webrtc.org,asapersson@webrtc.org,jonasolsson@webrtc.org,benwright@webrtc.org

Change-Id: I3c2b18ec2877d68a522ecbae7a2955c4eecf36df
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8982
Reviewed-on: https://webrtc-review.googlesource.com/71446
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22963}
2018-04-20 15:58:25 +00:00
Jonas Olsson
d7ee72041f Reland "Remove our stream << overloads from non-test build targets."
This is a reland of c841d18d25

Original change's description:
> Remove our stream << overloads from non-test build targets.
>
> Most are removed entirely, but RtcErrorType, RtpTransceiverDirection, IPAddress and
> SocketAddress are kept behind gtest's #ifdef UNIT_TEST.
>
> Bug: webrtc:8982
> Change-Id: I36db19891e7d25aeacb08b9a08aa2b4004765e70
> Reviewed-on: https://webrtc-review.googlesource.com/64143
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22916}

TBR=deadbeef@webrtc.org,kwiberg@webrtc.org,asapersson@webrtc.org,jonasolsson@webrtc.org,benwright@webrtc.org

Bug: webrtc:8982
Change-Id: Ibe08c6270e5e693eb661a6ce9e8f074b34ef8123
Reviewed-on: https://webrtc-review.googlesource.com/71161
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22949}
2018-04-20 09:09:30 +00:00
Åsa Persson
0ad2d8af39 Minor changes to QualityScaler.
- remove duplicated test, DoesNotDownscaleOnNormalQp
- add test, KeepsScaleOnNormalQp
- make member const

Bug: none
Change-Id: I6599e5eb0d59b67b0af55701accea25a80c7c875
Reviewed-on: https://webrtc-review.googlesource.com/70203
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22935}
2018-04-19 10:00:08 +00:00
Jonas Olsson
31ef5f0d1b Revert "Remove our stream << overloads from non-test build targets."
This reverts commit c841d18d25.

Reason for revert: Breaks internal tests

Original change's description:
> Remove our stream << overloads from non-test build targets.
> 
> Most are removed entirely, but RtcErrorType, RtpTransceiverDirection, IPAddress and
> SocketAddress are kept behind gtest's #ifdef UNIT_TEST.
> 
> Bug: webrtc:8982
> Change-Id: I36db19891e7d25aeacb08b9a08aa2b4004765e70
> Reviewed-on: https://webrtc-review.googlesource.com/64143
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22916}

TBR=deadbeef@webrtc.org,kwiberg@webrtc.org,asapersson@webrtc.org,jonasolsson@webrtc.org,benwright@webrtc.org

Change-Id: Ia3a36cdbdb2a9648a2bce23c314e539124dc9e0d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8982
Reviewed-on: https://webrtc-review.googlesource.com/70640
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22920}
2018-04-18 10:51:28 +00:00
Jonas Olsson
c841d18d25 Remove our stream << overloads from non-test build targets.
Most are removed entirely, but RtcErrorType, RtpTransceiverDirection, IPAddress and
SocketAddress are kept behind gtest's #ifdef UNIT_TEST.

Bug: webrtc:8982
Change-Id: I36db19891e7d25aeacb08b9a08aa2b4004765e70
Reviewed-on: https://webrtc-review.googlesource.com/64143
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22916}
2018-04-18 08:57:24 +00:00
Sergey Silkin
bfd54ef5cb Round down when converting layer bitrate from bits to kilobits.
This aligns rounding in videoprocessor with rounding in encoder wrappers.

Bug: none
Change-Id: I8bdab7c02628b433d35d63c4bf4c841ffb1c2d1b
Reviewed-on: https://webrtc-review.googlesource.com/69983
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22880}
2018-04-16 14:00:18 +00:00