Commit graph

342 commits

Author SHA1 Message Date
Sergey Silkin
691b447c53 Fix returns from IsSameSettings and IsSameRate in codec tests
Swap true/false.

Bug: webrtc:14852
Change-Id: Id82c0180d33bfc4e5237f4800c3e89fe8d17693f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302381
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39917}
2023-04-21 11:29:32 +00:00
Jared Siskin
c018bae807 Format /modules
git ls-files | grep -e  "\(\.h\|\.cc\)$" | grep -e  "^modules/" | xargs clang-format -i ; git cl format
after landing: add to .git-blame-ignore-revs

Bug: webrtc:15082
Change-Id: I2c3cd28740062794f8c10e39d8406aadb9e9a35a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301620
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Jared Siskin <jtsiskin@meta.com>
Cr-Commit-Position: refs/heads/main@{#39901}
2023-04-20 02:02:45 +00:00
Sergey Silkin
26d1b26277 Log metrics even if test failed
Set of codecs for testing is hardcoded to AV1, VP8, VP9, H264, H265. Some codecs may not be available due to lack of support on the platform or due to some issue in our code which would be a regression. Reporting zero metrics for failed tests would allow the perf tool to detect such a regression.

This also enables codec tests by default. The tests should not run on bots since video_codec_perf_tests binary is not included in any test suits yet.

Bug: webrtc:14852
Change-Id: I967160069055036f93e595d328c4d5f1ca483be9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300868
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39840}
2023-04-13 08:49:37 +00:00
Sergey Silkin
b4a45546b7 Dedicated build target for video codec perf tests
Bug: webrtc:14852
Change-Id: Ib56ef931b58058a7d09b97b7013ba39ee1767629
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300740
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39823}
2023-04-12 11:24:48 +00:00
Sergey Silkin
ebb5383fd8 Dump codec input
Add functionality for dumping encoder and decoder input to file in video codec test.

Bug: b/261160916, webrtc:14852
Change-Id: I49a84a886d87903c601cf5c35bd723b6393c2a75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298051
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39626}
2023-03-21 16:54:19 +00:00
Sergey Silkin
aa17f2f0a9 Add Initialize() to Encoder/Decoder API in video codec tester
Initialization of Android HW codecs takes hundreds milliseconds. Exclude this time from frame processing time of first frame by initializing codecs before starting encoding/decoding.

Bug: b/261160916, webrtc:14852
Change-Id: I9ec84c6b12c1d9821b59965cf521170224066563
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298304
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39613}
2023-03-21 08:04:48 +00:00
Sergey Silkin
12669513d2 Use internal codec factories directly
BuiltinVideoEncoderFactory, which was used before, has been started to use SEA since https://webrtc-review.googlesource.com/c/src/+/297740. SEA requires factory lifetime to be ~same as created codec lifetime. Codec test doesn't guarantee this currently.

Bug: b/261160916, webrtc:14852
Change-Id: I75ef99f1c9fe0d7823f31fd07c05a3ca52f7212d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298201
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39600}
2023-03-20 10:33:40 +00:00
Sergey Silkin
0af2bc639a Add H265 to VideoCodecMimeType
This enables testing HW H265 codecs on devices where the support is available.

Bug: b/261160916, webrtc:14852
Change-Id: I32d102fcf483ea4ba46d6f5161342dbb584e7cc9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298040
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39591}
2023-03-17 15:28:11 +00:00
Sergey Silkin
82e8a7fdca Fix frame rate scaling in video codec tests
Swap numerator and denominator values.

Bug: b/261160916, webrtc:14852
Change-Id: Id1fa81ac8e13513005a53b7034f1d38bb1602c2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297960
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39581}
2023-03-16 17:13:59 +00:00
Sergey Silkin
a5f32e445c Set frame capture timestamp
Unlike SW encoder wrappers, Android HW encoder wrapper uses frame capture timestamp instead of RTP timestamp: https://source.corp.google.com/piper///depot/google3/third_party/webrtc/files/stable/webrtc/sdk/android/src/jni/video_frame.cc;rcl=514125748;l=309

Bug: b/261160916, webrtc:14852
Change-Id: Ief76abae659f7ba890371901cc9b505526ac4f97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296500
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39495}
2023-03-07 13:56:14 +00:00
Sergey Silkin
1c1382be0f Dump codec output to ivf/y4m
Bug: b/261160916, webrtc:14852
Change-Id: I19de2210aa03b56752db5ce8b6fd94498123d6f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296260
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39490}
2023-03-07 08:33:39 +00:00
Sergey Silkin
9259b5f72c Add rate adaptation tests
Bug: b/261160916, webrtc:14852
Change-Id: I58b3647218c961dcf0305c3902f79adb448b73e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295866
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39489}
2023-03-06 18:33:16 +00:00
Sergey Silkin
fddc9131a5 Aggregate and log video codec metrics
Bug: b/261160916, webrtc:14852
Change-Id: Idcb7e96b12ca38af49b9b1f10d1e23cc7faac92b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293945
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39427}
2023-03-01 08:27:54 +00:00
Palak Agarwal
a09f21b207 Introduce capture_time_identifier in webrtc::EncodedImage
This CL propagates capture_time_identifier introduced in
webrtc::VideoFrame and propagates it to EncodedImage. For use cases
involving EncodedTransforms, this identifier is further propagated to
TransformableVideoSenderFrame.

VideoEncoder::Encode function is overriden by each encoder. Each of
these overriden functions needs to be changed so that they can handle
this new identifier and propagate its value in the created EncodedImage.

Change-Id: I5bea4c5a3fe714f1198e497a4bcb5fd059afe516
Bug: webrtc:14878
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291800
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#39374}
2023-02-22 17:08:53 +00:00
Sergey Silkin
72b99a1128 Test Android HW codecs
Bug: b/261160916, webrtc:14852
Change-Id: Iebeab244a9ca6ae196735016064ccd056b7c888e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293401
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39326}
2023-02-16 14:01:52 +00:00
Henrik Boström
2e540a28c0 Introduce EncodedImage.SimulcastIndex().
As part of go/unblocking-vp9-simulcast (Step 1), EncodedImage is being
upgraded to be able to differentiate between what is a simulcast index
and what is a spatial index.

In order not to break existing code assuming that "if codec != VP9,
SpatialIndex() is the simulcast index", SimulcastIndex() has fallback
logic to return the value of spatial_index_ in the event that
SetSimulcastIndex() has not been called. This allows migrating external
code from (Set)SpatialIndex() to (Set)SimulcastIndex(). During this
intermediate time, codec gates are still necessary in some places of
the code, see TODOs added.

In a follow-up CL, after having fixed dependencies, we'll be able to
remove the fallback logic and rely on SimulcastIndex() and
SpatialIndex() actually being the advertised index and "if codec..."
hacks will be a thing of the past!

Bug: webrtc:14884
Change-Id: I70095c091d0ce2336640451150888a3c3841df80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293343
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39318}
2023-02-15 15:02:57 +00:00
Sergey Silkin
5dd493b3da Do not use PostDelayedTask in video codec tester's pacer
PostDelayedTask doesn't guarantee task execution order. For example,
if you post two tasks, A and B, back-to-back using the same delay
there is no guarantee that A will be executed before B.

Re-implemented pacing using sleep(). Changed pacer to compute task
scheduled time instead of delay. Sleep time is calculated right before
task start. This provides better accuracy by accounting for any delays
that may happen after pacing time is computed and before task queue is
ready to run the task.

It is tricky to implement pacer tests using simulated clocks. The test
use system time which make them flacky on low performance bots. Keep
the test disabled by default.

Bug: b/261160916, webrtc:14852
Change-Id: I88e1a2001e6d33cf3bb7fe16730ec28abf90acc8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291804
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39302}
2023-02-13 15:07:16 +00:00
Sergey Silkin
c6ff4bc793 Do not transfer ownership of codecs to tester
Passing of ownership of codecs to tester is not strictly needed. We may need to continue using a codec after test. For example, to check codec state or to use the same codec instance in next test.

Bug: b/261160916, webrtc:14852
Change-Id: I179b262116d7de76b8171f0409f943ad6d87433e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291802
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39256}
2023-02-03 14:29:43 +00:00
Sergey Silkin
6c60f72a6b Refactor video codec testing stats
This CL introduces VideoCodecStats and VideoCodecStatsImpl which provide baseline functionalities for storing, slicing and aggregation of encoded and/or decoded video frame statistics. To facilitate metrics logging (not implemented yet), SamplesStatsCounter is used for stream parameters.

VideoCodecStats/VideoCodecStatsImpl will replace existing VideoCodecTestStats/VideoCodecTestStatsImpl.

Bug: b/261160916, webrtc:14852
Change-Id: I0f96ce1ed9be3aee2a702804612524676c9882fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291323
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39248}
2023-02-02 15:56:40 +00:00
Sergey Silkin
2e1a9a4ae0 Add video codec tester.
This tester is an improved version of VideoProcessor and VideoCodecTestFixture and will eventually replace them.

The tester provides better separation between codecs and testing logic. Its knowledge about codecs is limited to frame encode/decode calls and frame ready callbacks. Instantiation and configuration of codecs are the test responsibilities.

Other differences:
- Run encoding and decoding in separate threads
- Run quality analysis in a separate thread
- Reference frame buffering is moved into video source (which re-read frames from the file).
- Make it possible to run decode-only tests

This CL is MVP implementation: it adds only 1 test (video_codec_test.cc, ConstantRate/EncodeDecodeTest) and the test is disabled for now.

Bug: b/261160916
Change-Id: Ida24a2fca1b1496237fa695c812084877c76379f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283525
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38901}
2022-12-15 14:32:53 +00:00
Sergey Silkin
1985b5a927 Refactor YUV frame reader
Purposes of this refactoring:
1. Add functionality for reading a specified frame.
2. Change resolution and frame rate on per-frame basis.

Both features are needed for https://webrtc-review.googlesource.com/c/src/+/283525

Bug: b/261160916
Change-Id: I6d60e62dbc3913c43b5c1b491690f5cb4a8632dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285483
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38829}
2022-12-06 16:23:48 +00:00
Philipp Hancke
279b4b7d4f generateKeyframe: pass frame_types in bypass mode
Passes frame_types to the underlying encoder in bypass mode.
For libvpx this has no effect, for H264 this changes the behavior
to allow generating keyframes on a per-layer basis.

BUG=chromium:1354101

Change-Id: I26fc22d9e2ec4681a57ce591e9eafd0b1ec962b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285083
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38821}
2022-12-06 10:54:59 +00:00
Sergey Silkin
7404f07ad9 Sync target rates
Cache target bit- and framerate in a frame_num -> rates map and fetch
the rates accociated with the current frame when needed. This solves
the issue when wrong target rates may be used due to frames buffering
in encoder.

Bug: b/254447893
Change-Id: I369c8d8e71234c957dc2362b055061d12cec818f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283841
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38673}
2022-11-18 10:22:01 +00:00
philipel
ef005bc924 Unwrap the presentation timestamp before calling aom_codec_encode in LibaomAv1Encoder.
Bug: webrtc:14673
Change-Id: I0358fed5ac0839994482c5fb049c13e442f82c82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283701
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38657}
2022-11-17 08:32:18 +00:00
philipel
0c4563c0c4 Remove the libaom av1 decoder.
Bug: webrtc:14267
Change-Id: I95a416b25fa20d4dea6896e05beb59789621f1fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268305
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@{#38253}
2022-09-30 08:42:25 +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
1262eb5ebc Move EncoderStreamFactory into own file
This cl/ is a NOP refactoring,
moving the EncoderStreamFactory from within webrtc_video_engine.cc
into own file in video/. simulcast.cc is collateral.

Bug: webrtc:14451
Change-Id: Ia69b9241d8cd8a12be6628d887701f2e244c07cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276861
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38224}
2022-09-27 17:29:11 +00:00
Artem Titov
dab4cea30d Migrate VideoCodecTestFixture on new perf metrics logging API
Bug: b/246095034
Change-Id: I312f2643e4c84cdfa3e8fef7078a2decbbfef978
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276629
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38217}
2022-09-27 08:31:12 +00:00
Sergey Silkin
5d80958d5e Name rate adaptation tests
Bug: none
Change-Id: I10f9b2b14b5fa464d5a88d485e8c7ea1be636faf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272680
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37884}
2022-08-24 08:36:39 +00:00
Markus Handell
2cfc1af78a Update rtc::Event::Wait call sites to use TimeDelta.
Bug: webrtc:14366
Change-Id: I949c1d26f030696b18153afef977633c9a5bd4cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272003
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37835}
2022-08-19 10:07:28 +00:00
Danil Chapovalov
e519f38eaa Remove rtc::Location from SendTask test helper
that rtc::Location parameter was used only as extra information for the
RTC_CHECKs directly in the function, thus call stack of the crash should
provide all the information about the caller.

Bug: webrtc:11318
Change-Id: Iec6dd2c5de547f3e1601647a614be7ce57a55734
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270920
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37748}
2022-08-11 12:55:32 +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
philipel
26910ffe22 Make dav1d the default AV1 decoder.
Bug: chromium:1330308, b/234414450
Change-Id: Idc35c9e3612843001f8d7d9361f3769b45350e63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268183
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37483}
2022-07-07 14:04:46 +00:00
Danil Chapovalov
0be8eba07e Migrate pacing and video_coding to absl::AnyInvocable based TaskQueueBase interface
Bug: webrtc:14245
Change-Id: Icfab3e6548055ea72a199a226eca5233b1ead20d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267983
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37467}
2022-07-06 15:46:04 +00:00
Artem Titov
c374d11fac Move to_queued_task.h and pending_task_safety_flag.h into public API
Bug: b/235812579
Change-Id: I9fa3dc4a65044df8b44fff4e9bfeac7233fa381c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266080
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37248}
2022-06-17 09:20:39 +00:00
Niels Möller
c0a9f35248 Define SimulcastStream as an alias for SpatialLayer
Step one in making it a separate type, that will be done as a
followup, after downstream code is updated to use the new name.

Bug: webrtc:11607
Change-Id: I6fa664a0729b1cfd71b7f02b6441880beee0e741
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262806
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36946}
2022-05-20 13:12:21 +00:00
Niels Möller
807328fec7 Move frame drop config to VideoCodec and VideoEncoderConfig.
Intend to delete corresponding codec-specific settings in a followup.

Bug: webrtc:6883
Change-Id: I78ab07729a5aee1055f80d39d8f7289beb6721e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262244
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36882}
2022-05-13 13:40:14 +00:00
Jerome Jiang
0bb7cbc278 Lower av1 test psnr threshold
AV1 Realtime encoder stats changed

Bug: None
Change-Id: I50a8d36c45a775b3c0127476fb32c3d68d288508
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260480
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jerome Jiang <jianj@google.com>
Cr-Commit-Position: refs/heads/main@{#36712}
2022-04-29 16:12:32 +00:00
Niels Möller
79d566b0cf New enum ScalabilityMode.
Used instead of string representation in lower-levels of encoder configuration, to avoid string comparisons (with risk of misspelling) in lots of places.

Bug: webrtc:11607
Change-Id: I4d51c2265aac297c29976d2aa601d8ffb33b7326
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259870
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36706}
2022-04-29 12:16:42 +00:00
Niels Möller
c7b690272d Update modules/ to not use implicit conversion from scoped_refptr<T> to T*.
Bug: webrtc:13464
Change-Id: I3906e91906edbf80d558e5c367d6b9429497c021
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259762
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36606}
2022-04-21 14:23:24 +00:00
Björn Terelius
0c68a7aaa7 Use WebRTC's Java VM initialization in tests.
WebRTC should not depend on chromium's //base.

Bug: webrtc:13662
Change-Id: Ie660aa0f2477cc747830bba611aa23ed5e732385
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256364
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36581}
2022-04-20 08:41:48 +00:00
Niels Möller
cc171956f8 Rename scalability mode "NONE" to "L1T1".
Bug: webrtc:11607
Change-Id: I81e8ead4a2cc15de6c21c7ee852e909af38b0567
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258127
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36474}
2022-04-07 07:32:15 +00:00
Jerome Jiang
aaf9d051c7 Lower hd av1 quality threshold
Bug: None
Change-Id: I2b7bfbd8f5a2be13ede11df30272e5b001471453
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255021
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36180}
2022-03-11 20:54:03 +00:00
Jerome Jiang
ddcfe405d3 Change PSNR threshold for av1 test
Bug: None
Change-Id: I47101a6625c2f1704599ea60ad3f2c05370da66e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254101
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jerome Jiang <jianj@google.com>
Cr-Commit-Position: refs/heads/main@{#36151}
2022-03-08 22:14:51 +00:00
“Michael
15ee87fe0e Use VideoCodec complexity to determine AV1 encoder cpu_speed.
Bug: webrtc:13744
Change-Id: Ib6d62dcdf7346d886c0aca09735c7d5c1f3e2455
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252340
Reviewed-by: Erik Språng <sprang@webrtc.org>
Auto-Submit: Michael Horowitz <mhoro@google.com>
Commit-Queue: Michael Horowitz <mhoro@google.com>
Cr-Commit-Position: refs/heads/main@{#36125}
2022-03-03 19:06:17 +00:00
“Michael
3147e29c4e Refactor encoder-complexity param in VideoCodec w/backward compatibility
Move complexity parameter to the main VideoCodec class to enable
additional video codecs to use the parameter without creating a new
codec-specific structure.

Bug: webrtc:13694
Change-Id: Icb7cf640b178875d799f39ade8b5084e3222bb1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251921
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Michael Horowitz <mhoro@google.com>
Cr-Commit-Position: refs/heads/main@{#36040}
2022-02-21 19:40:44 +00:00
Sergey Silkin
454d2309de Add bitrate adaptation tests
Bug: none
Change-Id: I3e2c503efc7a85a3daaa40cd8118c1b02d3b81cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251680
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36013}
2022-02-16 10:25:13 +00:00
Sergey Silkin
0b02d637c0 Calculate max/avg encode/decode latency in codec tests
Bug: none
Change-Id: Ie42461dd06b1764c99308393477921ea25319ab4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251687
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36007}
2022-02-15 18:14:41 +00:00
Sergey Silkin
eb6c6fcf27 Fix delta frame delay calculation
The issue was introduced in https://webrtc-review.googlesource.com/c/src/+/132460.

Bug: webrtc:10412
Change-Id: I92d1bd2be63ea34d150145cec63c282f7aa49ce8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251683
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35989}
2022-02-14 11:15:50 +00:00
Bohan Li
1f06508350 Update av1 video codec test threshold.
Bug: webrtc:13638
Change-Id: If17ae2519451953ac3e6054cb6cdc376594fe3d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250040
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Reviewed-by: Fyodor Kyslov <kyslov@google.com>
Commit-Queue: Bohan Li <bohanli@google.com>
Cr-Commit-Position: refs/heads/main@{#35874}
2022-02-01 20:43:08 +00:00