Commit graph

347 commits

Author SHA1 Message Date
philipel
31718d7ce2 Reland "Add option to disable quality scaling for AV1."
This reverts commit 83102d3907.

Reason for revert: reland with fix

Original change's description:
> Revert "Add option to disable quality scaling for AV1."
>
> This reverts commit 446dbc66fd.
>
> Reason for revert: downstream break
>
> Original change's description:
> > Add option to disable quality scaling for AV1.
> >
> > The main goal of this change is to disable the quality scaler when multiple spatial layers are used.
> >
> > Bug: b/295129711
> > Change-Id: I25e0b7440a8c2adee3e97720a1e0ee5e0a914334
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319181
> > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#40709}
>
> Bug: b/295129711
> Change-Id: Iaeb13951d1b839bc0426120436035843bb3ee98f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320081
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Owners-Override: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40742}

Bug: b/295129711
Change-Id: Iab4846c2cd6074f50a3ebe9551432d449243b5d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320120
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40743}
2023-09-13 15:19:36 +00:00
Philip Eliasson
83102d3907 Revert "Add option to disable quality scaling for AV1."
This reverts commit 446dbc66fd.

Reason for revert: downstream break

Original change's description:
> Add option to disable quality scaling for AV1.
>
> The main goal of this change is to disable the quality scaler when multiple spatial layers are used.
>
> Bug: b/295129711
> Change-Id: I25e0b7440a8c2adee3e97720a1e0ee5e0a914334
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319181
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40709}

Bug: b/295129711
Change-Id: Iaeb13951d1b839bc0426120436035843bb3ee98f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320081
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Owners-Override: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40742}
2023-09-13 12:21:31 +00:00
philipel
46bbf7ec48 Add ScalabilityModeStringToEnum helper function.
Bug: none
Change-Id: Iea602c88afbfe1f8f8e94b353eda96d62b651bd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319882
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40732}
2023-09-11 17:04:27 +00:00
philipel
446dbc66fd Add option to disable quality scaling for AV1.
The main goal of this change is to disable the quality scaler when multiple spatial layers are used.

Bug: b/295129711
Change-Id: I25e0b7440a8c2adee3e97720a1e0ee5e0a914334
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319181
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40709}
2023-09-06 12:37:22 +00:00
Tony Herre
55b593fb6b Remove EncodedFrame::MissingFrame and start removing Decode() param
Remove EncodedFrame::MissingFrame, as it was always false in actual
in-use code anyway, and remove usages of the Decode missing_frames param
within WebRTC. Uses/overrides in other projects will be cleaned up
shortly, allowing that variant to be removed from the interface.

Bug: webrtc:15444
Change-Id: Id299d82e441a351deff81c0f2812707a985d23d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317802
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#40662}
2023-08-30 10:38:35 +00:00
Qiu Jianlin
44943c8064 Add H265 codec name and profile/tier/level utils.
This adds H265 codec name and profile/tier/level handling needed for
H265 SDP negotiation.

Bug: webrtc:13485
Change-Id: I838b910042ce36f8ae3979c41a73ee46935c57d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315900
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#40661}
2023-08-30 08:49:09 +00:00
Sergey Silkin
8efd93dd76 Encoder type agnostic resolution based fallback
WebRTC-Video-EncoderFallbackSettings/resolution_threshold_px:X sets resolution threshold to switch from primary to fallback encoder. When the field trial is present, VP8-specific resolution based fallback settings, provided by WebRTC-VP8-Forced-Fallback-Encoder-v2, are ignored.

Bug: none
Change-Id: I8f2e28438547f3896c7fc288ed6634720328f3a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314760
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40526}
2023-08-08 14:18:31 +00:00
Hans Wennborg
d1780836f4 Fix incorrect use of scoped enumerations in format strings
Scoped enums do not get automatically promoted to their underlying type,
so these uses have undefined behavior and Clang recently started warning
about it.

Bug: chromium:1456289
Change-Id: I9cf4e5a68378930a3bf7d8ac7b0a21eaf0d12670
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309520
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Auto-Submit: Hans Wennborg <hans@chromium.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40321}
2023-06-21 09:16:11 +00:00
Henrik Boström
2fec64484f Fix L1Tx target bitrate bug when the standard API is used.
There are now multiple ways to configure VP9 L1Tx:
- Legacy API: configure legacy SVC and disable encodings, this gets
  interpreted as disabling spatial layers (non-standard API hack).
- Standard API: configure scalability_mode. This can be done either
  with a single encoding or multiple encodings. As long as only one
  encoding is active we get a single L1Tx ssrc, same as legacy API.

Due to a bug, the ApplySpatialLayerBitrateLimits() logic which tweaks
bitrates was only applied in the legacy API code path, not the standard
API code path, despite both code paths configuring L1Tx.

The issue is that IsSimulcastOrMultipleSpatialLayers() was checking if
`number_of_streams == 1`. This is true in legacy code path but not
standard code path. The fix is to look at
`numberOfSimulcastStreams == 1` instead, which is set to the correct
value regardless of code path used.

This CL adds comments documenting the difference between
`number_of_streams` and `numberOfSimulcastStreams` to reduce the risk
of more mistakes like this in the future.

Bug: chromium:1455039, b:279161263
Change-Id: I69789b68cc5d45ef1b3becd310687c8dec8e7c87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308722
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40287}
2023-06-15 12:48:48 +00:00
Florent Castelli
8c4b9ea535 Remove references to AudioCodec and VideoCodec constructors
The preferred method to create codecs is to use the function
cricket::CreateAudioCodec or cricketCreateVideoCodec.
Empty codec objects are deprecated and should be replaced
with alternatives such as methods returning an
absl::optional object instead.

Bug: webrtc:15214
Change-Id: I7fe40f64673cd407830dbbb0e541b85a3aee93aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307521
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40226}
2023-06-05 23:23:40 +00:00
Jared Siskin
bceec84aee Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
half of the remaining folders

git ls-files | grep -e  "\(\.h\|\.cc\)$" | grep -E "^(api|call|common_audio|examples|media|net|p2p|pc)/" | xargs clang-format -i ; git cl format
after landing: add to .git-blame-ignore-revs

Bug: webrtc:15082
Change-Id: I8b2cac13f4587d3ce9b2fccc7362967283f57ea2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302062
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39977}
2023-05-03 11:09:26 +00:00
philipel
7ceb49c7b8 Fuzzy match the SdpVideoFormat in VideoEncoderFactoryTemplate::CreateVideoEncoder.
Bug: webrtc:13573
Change-Id: I1223f5f989d5298d3a6f7f6d06fd752e650adfd9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300342
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39766}
2023-04-05 11:48:26 +00:00
Henrik Boström
8481f6358e Remove IsSinglecastOrAllNonFirstLayersInactive() helper.
As of recent changes, we can simply look at numberOfSimulcastStreams
because in the {active,inactive,inactive} case we get a single
webrtc::VideoStream here[1] which results in numberOfSimulcastStreams
being 1 here[2].

Looking at numberOfSimulcastStreams instead of using a helper is
preferred because it is more descriptive and in the future, when
{inactive,active,inactive} or {inactive,inactive,active} cases of VP9
simulcast is also supported (webrtc:15046) then this gating will work
even when the first layer is not the active one.

[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/config/encoder_stream_factory.cc;l=146;drc=c99753ac8f051e379ae68e281aaef04b0a5ca8f2

[2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/video_codec_initializer.cc;l=77;drc=4baea5b07f2fd309892845cf2d1c0f4ca77862d3

# No need to wait for win chrome bot, everything else green
NOTRY=True

Bug: webrtc:15046
Change-Id: I8aaea2e8cc350bd01fb00cc7fd85032b7fdfe24d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299942
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39759}
2023-04-04 13:59:07 +00:00
philipel
31af34ba8c Update Dav1dDecoderTemplateAdapater to include AV1 profile1 support.
Bug: webrtc:13573
Change-Id: Ibc81b6400eb8e22ce28ae434c83a22013ef96c79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299946
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39757}
2023-04-04 12:40:17 +00:00
Henrik Boström
4baea5b07f Make VP9 simulcast behave like singlecast for single active layer cases.
Various "if streams == 1" cases are updated to "if
IsSinglecastOrAllNonFirstLayersInactive()" in order not to cause subtle
differences between VP9 {active} and VP9 {active,inactive,inactive}.

This CL also affects a line that conditionally sets
`simulcastStream[0].active = codec_active` so it seemed fitting to
improve the test coverage of "if all streams are inactive, don't send".

Bug: webrtc:15028
Change-Id: I8872dc8be0f2dfc1d8914bdba5e6433f9ba8cbfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298881
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39656}
2023-03-23 14:49:22 +00:00
Henrik Boström
62dc65b537 Add test that attempting HW is still possible after SW fallback.
Based on previous discussions I would have thought that this test would
fail, but it turns out that it passes. See referenced bug for context.

Bug: webrtc:15021
Change-Id: I845b48f688fb25942e3b770d50cafbf8a0bafe94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298562
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39625}
2023-03-21 15:02:34 +00:00
Henrik Boström
f6eae959bf Delete EncoderSimulcastProxy in favor of SimulcastEncoderAdapter.
Because the adapter has a passthrough mode, it can already handle both
singlecast and simulcast cases, meaning the proxy is no longer providing
value. Let's delete.

Bug: webrtc:15001
Change-Id: I480eaba599448e9b82b8cf7f829dc35ad6ce0434
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297740
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39579}
2023-03-16 13:25:44 +00:00
Philipp Hancke
b660b7a89c Enable multithreaded OpenH264 encoding behind field trial
This uses the field trial introduced is crbug.com/1406331 and
extends the usage to OpenH264. This simplifies experimentation
whether this change improves performance without requiring
multi-slice encoding.

BUG=webrtc:14368

Change-Id: I0031e59059f7113dd5453234869c957d46f311bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294340
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39371}
2023-02-22 14:26:33 +00:00
Henrik Boström
fbd0ddb32e Introduce WebRTC-VideoEncoderSettings/encoder_thread_limit:X.
As requested by a CEF hosted application (https://crbug.com/1406331)
who want to be able to limit the number of threads in a controlled
environment, this CL adds a flag to control the max limit per encoder.

For plumbing-reasons, this is placed in VideoEncoder::Settings but
with a note that this is considered an experimental API with limited
support. For now only LibvpxVp8Encoder uses it and there are no plans
to roll this out.

I have manually confirmed this is working with printf debugging,
--force-fieldtrials=WebRTC-VideoEncoderSettings/encoder_thread_limit:2
and https://jsfiddle.net/henbos/2bd6m7Lt/

Bug: chromium:1406331
Change-Id: Ib02bd83e2071034874843d3aaa0d3b0adc5bbf46
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293960
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39349}
2023-02-20 14:01:32 +00:00
Wan-Teh Chang
bd86684bf3 Make VideoEncoder::GetEncoderInfo() pure virtual
Bug: webrtc:9722
Change-Id: I831a9c460425be86e5da2761769b8eecf231462f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293386
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Wan-Teh Chang <wtc@google.com>
Cr-Commit-Position: refs/heads/main@{#39319}
2023-02-15 17:26:07 +00:00
Wan-Teh Chang
0c4c9be436 video_encoder.h: update kFullFramerate in comment
During code review, kFullFramerate was renamed kMaxFramerateFraction but
the uses of kFullFramerate in a comment were not updated. See
https://webrtc-review.googlesource.com/c/src/+/117900/3..4

Change-Id: I6b3c06b4c5b302e8ba40bde4ba722b94aab191eb
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292801
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Wan-Teh Chang <wtc@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39285}
2023-02-09 15:17:50 +00:00
Åsa Persson
b7f9113b72 Add API for querying codec support.
Implement
- BuiltinVideoEncoderFactory::QueryCodecSupport
- QualityAnalyzingVideoEncoderFactory::QueryCodecSupport
- FakeWebRtcVideoEncoderFactory::QueryCodecSupport

Bug: webrtc:11607
Change-Id: I9a138bbdc809abf5577dd27d84a51d0ed77d62ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290381
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38994}
2023-01-04 10:04:46 +00:00
Harald Alvestrand
794d599741 Split media_channel and its dependencies from the rtc_media_base target
This helps in figuring out which dependencies exist, and gets closer
to obeying the "one target per .cc file" rule.

Test failures seem unrelated, so using No-Try.

No-Try: true
Bug: webrtc:14775
Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 12:15:22 +00:00
Chunbo Hua
46ad25119c Make requested_resolution_alignment of webrtc::EncoderInfo as uint32_t.
At the same time, proper names of some parameters are refactored in SimulcastEncoderAdapter.

Bug: None
Change-Id: Ia036e3f362d1394e90aa26b79953c1ffe75e2fe0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284961
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Chunbo Hua <chunbo.hua@intel.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38870}
2022-12-12 06:02:59 +00:00
philipel
99c4c73dbf Add FuzzyMatchSdpVideoFormat convenience function for VideoEncoderFactoryTemplate.
Bug: webrtc:13573
Change-Id: I6813f2a2524271be7862b700da4831575ec6e206
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279701
Commit-Queue: Xavier Lepaul‎ <xalep@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38463}
2022-10-25 08:30:25 +00:00
Mirko Bonadei
9d9c2d5795 Make header files self contained.
This CL adds #includes to header files in order to make them
self contained after the preprocessor pass.

Bug: b/251890128
Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38327}
2022-10-08 08:38:36 +00:00
Emil Lundmark
ae5677639c Revise video owners
Bug: None
No-try: True
Change-Id: Ibc8dcb22d0ca81897dc63d39ff13372b0fc7302d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277402
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Fanny Linderborg <linderborg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38255}
2022-09-30 08:44:30 +00:00
Tommi
96c1a9b9e2 Clean up decoders when stopping video receive stream.
This updates VideoReceiveStream2::Stop() to symmetrically tear down
state that's built up in VideoReceiveStream2::Start().

Bug: webrtc:11993, webrtc:14486
Change-Id: I41f4feea5584e5baaeed2143432136f8b9761321
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272537
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38244}
2022-09-29 12:03:13 +00:00
Jonas Oreland
6c2dae21e9 Move VideoEncoderConfig from api/ into video/config
This cl move VideoEncoderConfig from api/ to video/config.

VideoStreamEncoderInterface and VideoStreamEncoderObserver
are moved as collateral.

brandt@ think that the reason these were in api/ in the
first place had to downstream project.

Functionality wise, this is a NOP, but it makes it easier
to modify the encoder (config).

Bug: webrtc:14451
Change-Id: I2610d815aeb186298498e7102cac773ecac8cd36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277002
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38242}
2022-09-29 09:44:43 +00:00
Jonas Oreland
7252348d76 Create EncoderStreamFactory in VideoStreamEncoder
This cl/ changes so that the EncoderStreamFactory is
not created inside WebRtcVideoSendStream (webrtc_video_engine).

The benifit of this is that the VideoStreamEncoder can then
amend the EncoderStreamFactory with state (and types)
w/o exposing it in VideoEncoderConfig.

I.e as an alternative to changes done inside
https://webrtc-review.googlesource.com/c/src/+/276742.

The fake_webrtc_call is modified to (if needed) create
it's own EncoderStreamFactory if needed.

Note: this cl/ will have to be merged with with
https://webrtc-review.googlesource.com/c/src/+/277002.

Bug: webrtc:14451
Change-Id: I3d896b227d39725ba6409622e8d09d14bd45d5fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277160
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38237}
2022-09-28 17:47:52 +00:00
Jonas Oreland
0deda15c96 Reland "RtpEncodingParameters::request_resolution patch 1"
This reverts commit b625101da8.

Reason for revert: Found problem that was specific how
configuration is handled for VP9. A 1-line change in webrtc_video_engine.cc line 3715.
Thanks Rasmus and great that this was tested!

Original change's description:
> Revert "RtpEncodingParameters::request_resolution patch 1"
>
> This reverts commit ef7359e679.
>
> Reason for revert: Breaks downstream test
>
> Original change's description:
> > RtpEncodingParameters::request_resolution patch 1
> >
> > This patch adds RtpEncodingParameters::request_resolution
> > with documentation and plumming. No behaviour is changed yet.
> >
> > Bug: webrtc:14451
> > Change-Id: I1f4f83a312ee8c293e3d8f02b950751e62048304
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276262
> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> > Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#38172}
>
> Bug: webrtc:14451
> Change-Id: I4b9590e23ec38e9e1c2e51a4600ef96b129439f2
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276541
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Owners-Override: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38176}

Bug: webrtc:14451
Change-Id: Ica9b74180bce22d09bf289126bb5ac137bf9eb70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276543
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38178}
2022-09-23 11:48:19 +00:00
Björn Terelius
b625101da8 Revert "RtpEncodingParameters::request_resolution patch 1"
This reverts commit ef7359e679.

Reason for revert: Breaks downstream test

Original change's description:
> RtpEncodingParameters::request_resolution patch 1
>
> This patch adds RtpEncodingParameters::request_resolution
> with documentation and plumming. No behaviour is changed yet.
>
> Bug: webrtc:14451
> Change-Id: I1f4f83a312ee8c293e3d8f02b950751e62048304
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276262
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38172}

Bug: webrtc:14451
Change-Id: I4b9590e23ec38e9e1c2e51a4600ef96b129439f2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276541
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Owners-Override: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38176}
2022-09-23 08:27:47 +00:00
Jonas Oreland
ef7359e679 RtpEncodingParameters::request_resolution patch 1
This patch adds RtpEncodingParameters::request_resolution
with documentation and plumming. No behaviour is changed yet.

Bug: webrtc:14451
Change-Id: I1f4f83a312ee8c293e3d8f02b950751e62048304
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276262
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38172}
2022-09-22 14:16:20 +00:00
Åsa Persson
178937de8e api/video_codecs: Add scalability mode helper functions.
Bug: webrtc:13960
Change-Id: I8c86e440223929594aec06e8b32b4bbe73546a94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273303
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37994}
2022-09-02 13:40:38 +00:00
Åsa Persson
ecfe8da46b Add support for more scalability modes (1.5:1 resolution ratio).
Added modes:
- S2T1h
- S2T2h
- S2T3h
- S3T1h
- S3T2h
- S3T3h

Bug: webrtc:13960
Change-Id: I618a30c68b0ce1609847ee33a2298fe8fa0720c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273664
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37968}
2022-08-31 11:01:16 +00:00
Florent Castelli
33155d763c svc: Remove references to bogus modes
Those never existed, were likely a copy-paste error in the spec
that we somehow inherited.

Bug: webrtc:11607
Change-Id: Ib4a038f061123e879f1099656273f6392f092213
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273485
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37953}
2022-08-30 14:03:21 +00:00
Åsa Persson
6d0516412e Add support for scalability modes S2T2, S3T1, S3T2.
Bug: webrtc:13960
Change-Id: Icafd3a5a3f8889777d65da5313b24e56a57af4d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273301
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37943}
2022-08-30 09:51:11 +00:00
Niels Möller
a16786b87f Delete nisse@webrtc.org from OWNERS files
Bug: None
Change-Id: I65fd526b236850f6df0de4f9022c77937b82f11a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269213
Auto-Submit: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37636}
2022-07-28 08:47:38 +00:00
philipel
f012bfaf96 Use Video{Encoder, Decoder}FactoryTemplate instead of Internal{Encoder, Decoder}Factory.
Bug: webrtc:13573
Change-Id: Id0e46a9b6053dedae3cbf0e5581768868900630b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269247
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37596}
2022-07-22 12:07:25 +00:00
Niels Möller
f2d090a0de Add temporary method SimulcastStream::SetNumberOfTemporalLayers
Similar to https://webrtc-review.googlesource.com/c/src/+/267843, it
turns out we need the setter method too to transition downstream code.

Bug: webrtc:11607
Change-Id: I50df5e9c5d9301717d527089de61fcf783267ee1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268760
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37545}
2022-07-18 12:19:52 +00:00
philipel
98c78cdd20 VideoEncoderFactoryTemplate cleanup.
Bug: webrtc:13573
Change-Id: Id70e64adba6c5d76132dc0edb0c93937e3e894f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268542
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37524}
2022-07-14 16:16:47 +00:00
philipel
f0232f31fb New VideoDecoderFactoryTemplate.
The VideoDecoderFactoryTemplate takes decoder implementations as template arguments, making it possible to easily implement a VideoDecoderFactory only using the implementations required for the particular application. This will replace the BuiltinVideoDecoderFactory.

Bug: webrtc:13573
Change-Id: I0213acd20b69dacf06fc6934851b73bd19b1afc8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268470
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37523}
2022-07-14 15:38:20 +00:00
philipel
02bfcf5132 Compare only SdpVideoFormat::name and SdpVideoFormat::parameters in the VideoEncoderFactoryTemplate.
Since https://webrtc-review.googlesource.com/c/src/+/267780 supported scalability modes are also used to compare for equality between SdpVideoFormats(?).

Bug: webrtc:14267, webrtc:13573
Change-Id: I2f3c2fca93bac6fadd222f776f672c9bd3f1de0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268304
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37510}
2022-07-13 10:54:00 +00:00
Niels Möller
79924579f3 Add temporary accessors for numberOfTemporalLayers
Intended to be used in downstream code when deleting deleting this
attribute.

Bug: webrtc:11607
Change-Id: I39417997a2ec2e72d726da476b5bce88abe267b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267843
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37445}
2022-07-05 13:52:15 +00:00
Byoungchan Lee
a1a7c638ec Let PCF.GetRtpSenderCapabilities return codecs' scalabilityModes.
Also move ScalabilityModeToString to api and add RTC_EXPORT so that
Chromium can use it.

Bug: chromium:986069
Change-Id: I5dbbb6de9b14ca20f3ae0630552dcd44595ad5ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267780
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#37444}
2022-07-05 13:28:33 +00:00
Niels Möller
6939f63ca1 Update old TODO comments
Bug: None
Change-Id: I96850df6cfa19303043108a59ef60d7b686ec747
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267661
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37436}
2022-07-05 09:59:33 +00:00
Niels Möller
865e45d14e Add default values for SimulcastStream members
The default values are zero, for consistency with the memset of VideoCodec. Except for numberOfTemporalLayers; This cl sets
numberOfTemporalLayers to 1 by default. The intention is to be able to
delete exlpicit setting of .numberOfTemporalLayers = 1 in downstream
code, to ease replacing it with a scalability mode.

Bug: webrtc:11607
Change-Id: I9de442f1893d474ea360f9b33364a00627f6c3be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267662
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37430}
2022-07-05 08:37:43 +00:00
Niels Möller
3c24c096ef Add support for scalability modes L2T3 and S2T3
Bug: webrtc:11607
Change-Id: I1d0bd171564d2852f2f6ee2bbee26c7a1c0e1c3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267103
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37389}
2022-07-01 08:17:04 +00:00
Sergio Garcia Murillo
179f40e81a add 422 8 and 10 bit decoding support
Bug: webrtc:14195
Change-Id: I2048d567850ae669d76d9e593752683f3c76499f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266180
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37306}
2022-06-22 15:08:44 +00:00
Niels Möller
9bda4905d2 Add public visibility for scalability_mode target
Bug: webrtc:11607
Change-Id: I93681ef6ff5918bd3bb26b006a9db9cab1805df2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265862
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37223}
2022-06-15 11:26:38 +00:00