Need to depend on them from Chromium.
Bug: webrtc:7925
Change-Id: Iea1bb3b937c602920bfd87f885c87c790ac7bc17
Reviewed-on: https://webrtc-review.googlesource.com/82061
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23580}
This is a reland of efc71e565e
Differs from the original cl by not widening the type of
VideoCodec::width and VideoCodec::height.
Original change's description:
> Move class VideoCodec from common_types.h to its own api header file.
>
> Bug: webrtc:7660
> Change-Id: I91f19bfc2565461328f30081f8383e136419aefb
> Reviewed-on: https://webrtc-review.googlesource.com/79881
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23544}
Bug: webrtc:7660
Change-Id: I7cf74a85a61ea2b831e6f32b3b3e17514ebefec8
Reviewed-on: https://webrtc-review.googlesource.com/82140
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23569}
And update most internal calls to use it.
Bug: webrtc:5740, webrtc:9372
Change-Id: Ib57d4ebfa7b0729af6d22981a792f0fdadf8a13f
Reviewed-on: https://webrtc-review.googlesource.com/81743
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23567}
This reverts commit 16e28d143a.
Reason for revert: Fix has supposedly landed upstream.
Original change's description:
> Disabling VeryLowBitrateVP9 to unblock roll.
>
> This should be re-enabled very soon since the libvpx thinks this
> is fixed upstream and is only waiting for merge.
>
> TBR=marpan@google.com
>
> Bug: webrtc:9292
> Change-Id: Ib78ea1462059c333b7168a52756329dc9a385b54
> Reviewed-on: https://webrtc-review.googlesource.com/81660
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23525}
TBR=phoglund@webrtc.org,marpan@google.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9292
Change-Id: I995953070536e8ee3540e7c30bc11dc1200e0463
Reviewed-on: https://webrtc-review.googlesource.com/82200
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23552}
This should be re-enabled very soon since the libvpx thinks this
is fixed upstream and is only waiting for merge.
TBR=marpan@google.com
Bug: webrtc:9292
Change-Id: Ib78ea1462059c333b7168a52756329dc9a385b54
Reviewed-on: https://webrtc-review.googlesource.com/81660
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23525}
webrtc:8745 is closed and it was about unblocking the Chromium roll,
webrtc:9352 is the new bug to keep track of the removal of ffmpeg
deprecated functions.
Bug: webrtc:9352
Change-Id: I2818dba804f3d611d4df80559a635e7cf1ee5338
No-Try: True
TBR: phoglund@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/80882
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23506}
This is a reland of 3409cfa378
Needed to change RtpVideoStreamReceiver to stop deregistering a payload
type if two payload types refer to the same codec (which now happens,
with the packetization mode 0/1 payload types). It's not clear why this
was being done in the first place.
Original change's description:
> Start supporting H264 packetization mode 0.
>
> The work was already done to support it, but it wasn't being negotiated
> in SDP.
>
> This means we'll now see 8 H264 payload types instead of 4; one for each
> combination of BP/CBP profiles, packetization modes 0/1, and RTX/non-RTX.
> This could be problematic in the future, since we're starting to run
> out of dynamic payload types (using 25 of 32).
>
> Bug: chromium:600254
> Change-Id: Ief2340db77c796f12980445b547b87e939170fae
> Reviewed-on: https://webrtc-review.googlesource.com/77264
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23372}
Bug: chromium:600254
Change-Id: Ice1acc05acd1543d9b46e918de2bba0694d86259
Reviewed-on: https://webrtc-review.googlesource.com/78399
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23494}
This will allow exposing the interface to downstream users that
want to test VP8 simulcast. No functional changes to the tests
themselves are expected.
Bug: webrtc:9281
Change-Id: I4128b8f35a4412c5b330cf55c8dc0e173d4570da
Reviewed-on: https://webrtc-review.googlesource.com/77361
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23469}
Use new frame dropping mode - FULL_SUPERFRAME_DROP - in VP9 encoder and
configure it to drop entire superframe if any layer is overshooting.
Bug: none
Change-Id: Ie22ed5c175e530bcce365d40cba0d10cb608ad4f
Reviewed-on: https://webrtc-review.googlesource.com/79622
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23447}
It should be the responsibility of the fixture user to provide the exact
codecs that should be tested instead. This reduces the coupling between
the test fixture and the codec instantiation.
Bug: webrtc:9317
Change-Id: I60d8f5c4b516ba33e2293d574ba17602c39f992b
Reviewed-on: https://webrtc-review.googlesource.com/79147
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23425}
Experiment flag added to PeerConnectionInterface::RtcConfiguration and
propagated down to VideoStreamEncoder.
Artificial Sdp parameter is added to the sdp format if the flag is set.
Additionally, sdp format is propagated in vp8 simulcast adapters.
Bug: chromium:794608
Change-Id: I2dec54d19ae7bfbd5f2777ec682da5a84194da94
Reviewed-on: https://webrtc-review.googlesource.com/78500
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23412}
In flexible mode, use VP9E_GET_SVC_REF_FRAME_CONFIG to get indices of
reference frame buffers and buffers update by encoded frame.
Set inter_pic_predicted to true only if encoder actually used temporal
prediction.
Bug: webrtc:9244, webrtc:9270
Change-Id: I4e439abeab9e063d50abdcefc59bf58d6596ea6c
Reviewed-on: https://webrtc-review.googlesource.com/74780
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@google.com>
Cr-Commit-Position: refs/heads/master@{#23410}
Thus we don't need to initialize new members added to the structure
in the future.
Bug: None
Change-Id: Id9f5b127c224660f3016973261045b4231a617c1
Reviewed-on: https://webrtc-review.googlesource.com/79080
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23404}
All decoders are injectable, no need to create built-in codecs from
there.
Bug: webrtc:7925
Change-Id: Iabf3d59a8e4d721ad29386acbf138b7e5992ce5e
Reviewed-on: https://webrtc-review.googlesource.com/72441
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23397}
This is needed for downstream users of the impl, as we currently pull
in Chromium specifics in the android_codec_factory_helper. Further,
the downstream users should explicitly supply their own factories
if they do not want to use the internal ones.
Bug: None
Change-Id: Ia7b01a66aadaba3d5accf44e5ca38e1a319e4e34
Reviewed-on: https://webrtc-review.googlesource.com/78420
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23390}
This reverts commit 3409cfa378.
Reason for revert: Broke WebRtcBrowserTest.RunsAudioVideoWebRTCCallInTwoTabsH264 on Windows 7/10 bots
Original change's description:
> Start supporting H264 packetization mode 0.
>
> The work was already done to support it, but it wasn't being negotiated
> in SDP.
>
> This means we'll now see 8 H264 payload types instead of 4; one for each
> combination of BP/CBP profiles, packetization modes 0/1, and RTX/non-RTX.
> This could be problematic in the future, since we're starting to run
> out of dynamic payload types (using 25 of 32).
>
> Bug: chromium:600254
> Change-Id: Ief2340db77c796f12980445b547b87e939170fae
> Reviewed-on: https://webrtc-review.googlesource.com/77264
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23372}
TBR=deadbeef@webrtc.org,magjed@webrtc.org,sprang@webrtc.org
Change-Id: I2f2a2b4ca20ba883764cd5265911e1453d3df66e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:600254
Reviewed-on: https://webrtc-review.googlesource.com/78398
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23374}
The work was already done to support it, but it wasn't being negotiated
in SDP.
This means we'll now see 8 H264 payload types instead of 4; one for each
combination of BP/CBP profiles, packetization modes 0/1, and RTX/non-RTX.
This could be problematic in the future, since we're starting to run
out of dynamic payload types (using 25 of 32).
Bug: chromium:600254
Change-Id: Ief2340db77c796f12980445b547b87e939170fae
Reviewed-on: https://webrtc-review.googlesource.com/77264
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23372}
These files are required when implementing tests based on the test fixture,
and should be exposed as part of the test api.
This CL also removes a usage of stringstream and fixes some chromium-style
lint issues.
Bug: webrtc:8982, webrtc:163
Change-Id: I132aea0da79a79587887f21897236fc9802b7574
Reviewed-on: https://webrtc-review.googlesource.com/74586
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@{#23346}
Follow up to https://webrtc-review.googlesource.com/c/src/+/39511,
which introduced a new Decode method, without the
RTPFragmentationHeader argument, and deprecated the old method.
Bug: webrtc:6471
Change-Id: Icd3c536ebedd4e3c2d57fdb4d6e078d6ff1de5b6
Reviewed-on: https://webrtc-review.googlesource.com/75180
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23339}
This is a kitchen-sink header, some pieces should be moved to
byteorder.h, the rest likely deleted.
Delete most includes of basictypes.h. In leaf headers,
include stddef.h and stdint.h explicitly where needed.
Bug: webrtc:6853
Change-Id: Ibc809936a8f94d418e4eb650da1e89c1b9142073
Reviewed-on: https://webrtc-review.googlesource.com/77721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23333}
- Limit framerate by dropping frames before encoding.
- The max framerate at screen sharing is set to 5fps.
Bug: webrtc:9261
Change-Id: Icfbbecce33fdce2d746291708db0108e0ba10760
Reviewed-on: https://webrtc-review.googlesource.com/76921
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23316}
- 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}
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}
