Commit graph

168 commits

Author SHA1 Message Date
Emil Vardar
97ba6afb37 Propagate the base QP value from frame header to Decoded callback in dav1d decoder.
Current version of the dav1d decoder does not propagate any QP value to the Decoded callback. This CL updates this such that the base QP gets propagated from the frame header.

Bug: None
Change-Id: Ib7624b7e27d2c973f1821df5688cbb444e4847a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359740
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Cr-Commit-Position: refs/heads/main@{#42790}
2024-08-16 12:02:20 +00:00
Sergey Silkin
e9810a8adb Use GetTemporalLayerSum
Bug: b/337757868
Change-Id: Ieff4c22425bab06c12419d64db7a2eef69cc54d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355962
Auto-Submit: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42630}
2024-07-12 15:25:28 +00:00
Sergey Silkin
f7a1506703 Adjust max consecutive drops depending on target frame rate
Current thresholds were tuned to guarantee no buffer overshoot in an extreme scenario (encoding a high complexity video in a low bitrate).

Bug: b/337757868, webrtc:351644568
Change-Id: I832b2564af6f18f06550338cc9b3618f8acdf831
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356580
Reviewed-by: Dan Tan <dwtan@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42620}
2024-07-10 17:16:18 +00:00
Sergey Silkin
7a6053ae62 Rename minimum_qp to min_qp
For better consistency with the rest codebase (it is min_/max_ for all params in video_encoder.h; only qp is for some reason prefixed with minimum_).

Also fixed constant names in libaom AV1 encoder wrapper (moved min from suffix to prefix, minimum -> min_).

Bug: chromium:328598314
Change-Id: I6d8521a3abff3a0595a5241c02ef4746eb4694df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356600
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42604}
2024-07-08 15:37:23 +00:00
Sergey Silkin
b43cd86e64 Increase frame rate precision in libaom AV1 encoder wrapper
Before this change the AV1 encoder wrapper converted target frame rate from double to integer with rounding to the middle. That approach resulted in a bitrate mismatch caused by rounding error. The mismatch was especially high at low frame rates. For example, at target frame rate 1.4fps the bitrate mismatch reached 40%:

out/debug/video_codec_perf_tests --gtest_also_run_disabled_tests --gtest_filter=*EncodeDecode --framerate_fps=1.4 --width=320 --height=180 --bitrate_kbps=32 --num_frames=600
...
RESULT s0t0_bitrate_mismatch_pct: DISABLED_EncodeDecode= {39.171875,0} n%

After the change the mismatch reduced to ~2% in the same scenario:
RESULT s0t0_bitrate_mismatch_pct: DISABLED_EncodeDecode= {-2.178125,0} n%

Bug: b/337757868
Change-Id: Ia51f92b3dfdce103eed1d04cac0e084b69fa8213
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356500
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42601}
2024-07-08 12:00:43 +00:00
Johannes Kron
216cce5f49 Add minimum_qp to VideoEncoder::EncoderInfo
The minimum QP field will be used to signal what the QP value will be
once the encoder reach its target video quality. This will be used
in the generalized QP convergence detection.

Bug: chromium:328598314
Change-Id: I82299cd921e3c091e651218d1e3f337875176567
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355701
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Markus Handell <handellm@google.com>
Cr-Commit-Position: refs/heads/main@{#42559}
2024-06-28 10:48:22 +00:00
Sergey Silkin
6e37ee34d1 Reuse QP limits from the main encoder config
Set layer QP limits equal to QP limits in the main encoder config. This reduces number of nodes to modify if you need to change the settings.

Bug: b/337757868
Change-Id: Id7f6f9d6527903e8e22ff4fad2c974bee6e87cb3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353982
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42466}
2024-06-12 09:45:52 +00:00
Florent Castelli
99c519b3fd Mass removal of absl_deps in all BUILD.gn files
Bug: webrtc:341803749
Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 15:09:46 +00:00
Sergey Silkin
f5e9f11994 Delete WebRTC-LibaomAv1Encoder-DisableFrameDropping
This was a kill-switch for frame dropping in AV1 encoder. The frame dropping was enabled in June 2023. Since we have not heard about about any issues related to the frame dropping, we can remove the field trial.

Bug: webrtc:42225542
Change-Id: I4b2f1d5ff61e4ae3a4a7fc6711bb83f7d522fc6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349921
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42241}
2024-05-07 07:47:32 +00:00
Danil Chapovalov
652bd288b3 Query EncoderInfoSettings through propagated field trials
Instead of from the global field trial string.

Bug: webrtc:42220378
Change-Id: Iddb41429e388792de02f702b4caa35689c57d9e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347720
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42201}
2024-04-30 11:16:31 +00:00
Wan-Teh Chang
ef16abac3e RTC_CHECK frame buffer has expected width & height
The width and height of mapped_buffer must match the d_w and d_h members
of frame_to_encode_, which is passed to aom_codec_encode().

Bug: b:330482827
Change-Id: I85d8c82133768685565f165eafc893c42dc40b12
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/345807
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Wan-Teh Chang <wtc@google.com>
Cr-Commit-Position: refs/heads/main@{#42036}
2024-04-10 23:04:33 +00:00
Danil Chapovalov
604b4db346 Require webrtc::Environment to create AV1 Encoder
Bug: webrtc:15860
Change-Id: Ic9bf907a7112c786ef01f8b3209caf55a272bac3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/345742
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42008}
2024-04-05 14:30:33 +00:00
Per K
0fa90887c5 Deprecate VideoFrame::timestamp() and set_timestamp
Instead, add rtp_timestamp and set_rtp_timestamp.

Bug: webrtc:13756
Change-Id: Ic4266394003e0d49e525d71f4d830f5e518299cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342781
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41894}
2024-03-13 11:08:37 +00:00
Danil Chapovalov
b4913a549f Add factory functions to pass Environment to VideoEncoders
Bug: webrtc:15860
Change-Id: I4a9d2678dcfe5b0f178863242e27600fcc95325d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342480
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41879}
2024-03-12 09:43:14 +00:00
Johannes Kron
17e358096e Add AV1 encoder speed setting for screen share
There's an AV1 encoder speed setting 11 that is supposed to be used
for screen sharing content.

Bug: chromium:328598314
Change-Id: Id97898554a740eb1684d03c782c718c19f4c95e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342201
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41874}
2024-03-08 14:53:54 +00:00
Sergey Silkin
74a4038ead Limit max frame size in DAV1D decoder
Bug: chromium:325284120
Change-Id: Iea0aea0a17bb0b1f73b3c1cbd408b7a6cd2b216e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340180
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41776}
2024-02-21 11:05:44 +00:00
Jeremy Leconte
f19c7caeb5 Fix crash when rolling libaom.
The crash is caused by https://aomedia.googlesource.com/aom.git/+/77cf417565ad2c527d5c351927f11db3764fd93c%5E%21

Example of the test failure:
https://ci.chromium.org/ui/p/webrtc/builders/try/linux_rel/72442/overview

Bug: None
Change-Id: I088bf7e45452cdaa71802802e431119e755eca24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337320
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41659}
2024-02-02 09:45:05 +00:00
Dan Tan
4860148c51 Add WebRTC-LibaomAv1Encoder-MaxConsecFrameDrop parameter to explicitly limit the maximum consecutive frame drop
Bug: webrtc:15821
Change-Id: Ib8be6827ea57e4e54269b94a0fc9ea81945af09f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337020
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Commit-Queue: Dan Tan <dwtan@google.com>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41648}
2024-01-31 18:35:51 +00:00
Mirko Bonadei
a3d2c58e38 Skip LibaomAv1SvcTest.EncodeAndDecodeAllDecodeTargets/S3T3.
This is temporary while AV1 gets fixed.

Bug: webrtc:15715, b/315476578
Change-Id: I4fdadb97788c934b12b4a3a19dfec1f61a95a3a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330640
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@google.com>
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41345}
2023-12-09 12:24:51 +00:00
Sergey Silkin
50e2054c5b Move setting single spatial layer bitrates to GetVp9SvcConfig
Before this change bitrate limits for VP9 single spatial layer case were set in VideoCodecInitializer. Move this logic to GetVp9SvcConfig. This simplifies replication of WebRTC behaviour in codec level tests. The similar AV1 logic sits in SetAv1SvcConfig, not VideoCodecInitializer.

Bug: webrtc:14852
Change-Id: Ie7202ec880d0e4b903e7265721eeef9b3920f21a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324286
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40992}
2023-10-23 14:10:21 +00:00
Sergey Silkin
a4b2b95f99 Restrict ARM-specific VP8/VP9/AV1 settings to mobile platforms
ARM-specific settings were intended to be used on mobile ARM devices which may not be powerful enough. But the settings were also applied to ARM-based Macs. This changes restricts ARM-specific settings to Android and iOS platforms.

Bug: none
Change-Id: I68764b4c0679db07399bba5923f4a6be89c5ad80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321861
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Jerome Jiang <jianj@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40884}
2023-10-06 15:10:04 +00:00
Danil Chapovalov
9c58483b5a Rename EncodedImage property Timetamp to RtpTimestamp
To avoid name collision with Timestamp type,
To avoid confusion with capture time represented as Timestamp

Bug: webrtc:9378
Change-Id: I8438a9cf4316e5f81d98c2af9dc9454c21c78e70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40796}
2023-09-24 20:06:48 +00:00
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
19ff1ad237 Reland "Always use AV1 specific bitrate limits when spatial layers are used."
This reverts commit 030c6ff43f.

Reason for revert: reland with fix

Original change's description:
> Revert "Always use AV1 specific bitrate limits when spatial layers are used."
>
> This reverts commit d2d165d47c.
>
> Reason for revert: All the regressions!
>
> Original change's description:
> > Always use AV1 specific bitrate limits when spatial layers are used.
> >
> > Bug: b/295129711
> > Change-Id: I93569027bea34c43e2a3c4de0875e8bbddd5b64e
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319283
> > Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
> > Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#40719}
>
> Bug: b/295129711
> Change-Id: I5776edbaba33e86eb10414062ef2b29510f40b8d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319880
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#40730}

Bug: b/295129711
Change-Id: I5fe84184d3f3780fdc4e9c1d43c4989d333d44a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319881
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40739}
2023-09-12 13:00:19 +00:00
Philip Eliasson
030c6ff43f Revert "Always use AV1 specific bitrate limits when spatial layers are used."
This reverts commit d2d165d47c.

Reason for revert: All the regressions!

Original change's description:
> Always use AV1 specific bitrate limits when spatial layers are used.
>
> Bug: b/295129711
> Change-Id: I93569027bea34c43e2a3c4de0875e8bbddd5b64e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319283
> Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40719}

Bug: b/295129711
Change-Id: I5776edbaba33e86eb10414062ef2b29510f40b8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319880
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#40730}
2023-09-11 11:57:39 +00:00
philipel
d2d165d47c Always use AV1 specific bitrate limits when spatial layers are used.
Bug: b/295129711
Change-Id: I93569027bea34c43e2a3c4de0875e8bbddd5b64e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319283
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40719}
2023-09-08 09:02:11 +00:00
philipel
8fd09016e6 Reduce number of spatial layers depending on input resolution for AV1
Bug: b/295129711
Change-Id: If54562d6e453209da9f358bbdb2909662e4ab873
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319380
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40713}
2023-09-07 10:29:47 +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
Jerome Jiang
3403acb3c6 av1: 8 threads for >720p and tiles config
Use 8 threads for > 720p
Use 4 tile columns and 2 tile rows for 8 threads
Use 2 tile columns and 2 tile rows for 4 threads

For VGA, 2 tile_col x 2 tile_row configuration has
 - ~9% speedup over 4 tile_col x 1 tile_row
 - ~5% speedup over 1 tile_col x 4 tile_row

Bug: None
Change-Id: I3c1ea948437aece7c6734ce25351158cbdf0a15b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307880
Commit-Queue: Jerome Jiang <jianj@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40237}
2023-06-07 15:33:41 +00:00
Sergey Silkin
d615704551 Enable frame dropping in libaom AV1 encoder
Bug: webrtc:15225
Change-Id: Ife16a61d47d7aa2f20548d30c56bf59844de1b26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307500
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40236}
2023-06-07 13:24:02 +00:00
Danil Chapovalov
0c85f733c9 For AV1, disable error resilience on upper temporal layers
Error resilience is no longer required for upper temporal layers.
Disabling error resilience on the upper layers leads to a ~2% PSNR BD-rate gain.

Reland of https://webrtc-review.googlesource.com/c/src/+/302001

Bug: webrtc:15106
Change-Id: I72ca9d504a7848dda934cbd52669027061742256
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305782
Reviewed-by: Jerome Jiang <jianj@google.com>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@google.com>
Cr-Commit-Position: refs/heads/main@{#40099}
2023-05-22 08:14:08 +00:00
Jeremy Leconte
67f2109544 Revert "For AV1, disable error resilience on upper temporal layers. Error resilience is no longer required for upper temporal layers. Disabling error resilience on the upper layers leads to a ~2% PSNR BD-rate gain."
This reverts commit 2080dacfb7.

Reason for revert: This CL is causing a lot of flakiness on iOS bots
https://ci.chromium.org/p/webrtc/builders/ci/iOS%20Debug%20%28simulator%29

Original change's description:
> For AV1, disable error resilience on upper temporal layers. Error resilience is no longer required for upper temporal layers. Disabling error resilience on the upper layers leads to a ~2% PSNR BD-rate gain.
>
> Bug: webrtc:15106
> Change-Id: Id92d51defbd26c1a77e3c9fe19607e9db4a3e7c1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302001
> Reviewed-by: Marco Paniconi <marpan@webrtc.org>
> Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39900}

Bug: webrtc:15106
Change-Id: I24515280113ed6681c9766026ec24d689035c031
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301983
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Owners-Override: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39903}
2023-04-20 09:24:52 +00:00
Michael Horowitz
2080dacfb7 For AV1, disable error resilience on upper temporal layers. Error resilience is no longer required for upper temporal layers. Disabling error resilience on the upper layers leads to a ~2% PSNR BD-rate gain.
Bug: webrtc:15106
Change-Id: Id92d51defbd26c1a77e3c9fe19607e9db4a3e7c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302001
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39900}
2023-04-20 00:17:45 +00:00
Yu-Chen (Eric) Sun
35f2b89ee4 Fix the issue 15059: wrong libaom initialized target bitrate
Fix Issue 15059: The target bitrate was mistakenly set to be the maximal

bitrate when initializing the libaom encoder.

Bug: webrtc:15059
Change-Id: I38498d4cce7b0a9c26736d9f1096178dd2e1fef6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300004
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39822}
2023-04-12 10:42:58 +00:00
Wan-Teh Chang
ad192a8c5e Remove extraneous opening parenthesis in comment
Bug: None
Change-Id: I8f1939caa43a7eb48dc5a6276520b39429062b30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298000
Auto-Submit: Wan-Teh Chang <wtc@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Wan-Teh Chang <wtc@google.com>
Cr-Commit-Position: refs/heads/main@{#39587}
2023-03-17 14:31:15 +00:00
Wan-Teh Chang
8f29b42670 Validate encoder_settings_.qpMax
libaom uses the quantizer as an index for an array of size 64, so
encoder_settings_.qpMax must be <= 63.

Add a comment to LibaomAv1Encoder::SetSvcParams() to explain why the
method doesn't initialize svc_params.layer_target_bitrate.

Bug: None
Change-Id: I26be80de005752214365abbe8b9b32dc976cee0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293680
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Wan-Teh Chang <wtc@google.com>
Cr-Commit-Position: refs/heads/main@{#39572}
2023-03-16 02:57:44 +00:00
Danil Chapovalov
8eb7aef196 Relax expectation on the libaom rate controller
Bug: b/271819773
Change-Id: I580b6fde352d1f23773fd394b0ee1543724b828f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296323
Reviewed-by: Jerome Jiang <jianj@google.com>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39482}
2023-03-06 13:58:51 +00:00
Michael Horowitz
b27efd487d Add option to configure AV1 EncoderInfo resolution_bitrate_limits.
bug: webrtc:14931
Change-Id: I8ade2a888d29f76a0f690fc3541b45b7304ad4d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294600
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39426}
2023-02-28 20:48:33 +00:00
Erik Språng
ff1cf61cf3 Fix potentially bad rate control with libaom av1 encoder.
This can happen when the encoder uses real presentation timestamps that
originate with the input frames. By using those, the encoder can bypass
webrtc frame dropping logic and may severely over/under-shoot if the
timestamps are very precise. In practice, this seems rather common on
Chrome on Windows.

Bug: aomedia:3391
Change-Id: I2be5eed4fabc86dac8a6c7bfdd068c2dcb5a3743
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294740
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39382}
2023-02-23 18:54:57 +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
Wan-Teh Chang
f6eb9d64b2 Declare kMinimumFrameRate for AV1 codec as double
The kMinimumFrameRate constant is only used in a comparison with
RateControlParameters::framerate_fps, which is of the double type.
Declare kMinimumFrameRate as double to match.

Note: The kMinimumFrameRate constant was added in
https://webrtc-review.googlesource.com/c/src/+/170360.

Bug: webrtc:11404
Change-Id: I11769867d4e52a720219c8a0ade8e8b74d13ca86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293384
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@{#39320}
2023-02-15 17:27:34 +00:00
Wan-Teh Chang
1f39162528 Fill fps allocation for LibaomAv1Encoder correctly
The elements of the fps_allocation vector are fractions of the maximum
frame rate. Each fraction is represented as an 8-bit unsigned integer,
where 0 = 0% and 255 = 100%.

The original code (added in
https://webrtc-review.googlesource.com/c/src/+/201384) sets the elements
of the fps_allocation vector to frame rates rather than frame rate
fractions. Perhaps fps_allocation could be renamed to avoid this kind of
confusion.

modules_unittests --gtest_filter=LibaomAv1EncoderTest.*

Tested: 
Change-Id: Icd050da3b3c2cff31913c3430f7b6b6e9829b9fa
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292784
Commit-Queue: Wan-Teh Chang <wtc@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39286}
2023-02-09 16:00:55 +00:00
Evan Shrubsole
097fc347ec [Unwrap] Prepare SequenceNumberUnwrapper for migrations
This is in prep for the migration of all unwrappers to
SequenceNumberUnwrapper as a standard implementation.

This moves the SeqNumUnwapper to its own header and adds 2 methods to
SeqNumUnwrapper which are defined by other unwrappers:
* PeekUnwrap
* Reset

It also adds two implementations for RtpTimestamps and
RtpSequenceNumbers.

Bug: webrtc:13982
Change-Id: I5baefb2de1db92fe1bb600760bd63b71e9310eb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288742
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39030}
2023-01-09 11:42:20 +00:00
Evan Shrubsole
9b235cd93b Add scalability mode to RTCOutboundRtpStreamStats stats
This is in the webrtc-stats spec at
https://www.w3.org/TR/webrtc-stats/#dom-rtcoutboundrtpstreamstats-scalabilitymode.

This adds the scalability mode to CodecSpecificInfo which is used to
plumb the modes for each simulcast layer.

TBR=orphis@webrtc.org

Tested: Compiled into Chrome and confirmed the scalability mode set for AV1, VP9, VP8 and H264 software encoders in chrome://webrtc-internals.
Bug: webrtc:14730
Change-Id: I71ceba8f6485a4f4a73e0856031b8d5f16f913f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285085
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38847}
2022-12-08 11:46:06 +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
Zhaoliang Ma
dd65499002 Skip one data copy on dav1d decoding
This CL wraps the |Dav1dPicture| data directly for |VideoFrame| using
instead of copy data out to new buffer.

Bug: None
Change-Id: I21ceffb5cac7dda4a44eafbd0ed221974b8d45ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276526
Commit-Queue: Zhaoliang Ma <zhaoliang.ma@intel.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38194}
2022-09-26 08:37:24 +00:00
Joe Downing
59020bd88b Add AV1 profile-1 video decode support to WebRTC
The Chrome Remote Desktop team is looking to support AV1 profile-1
w/ I444 for screen sharing however only I420 is currently supported.

This CL adds I444 support for the Dav1dDecoder, which appears to be
the preferred decoder and adds profile-1 to the
InternalDecoderFactory when the Dav1dDecoder is being used.

I've tested this CL using a CRD host w/ I444 enabled and it seems to
work as expected, though I've only tested on a debug build so I plan
to do some perf testing once this is available in a release build.

Bug: chromium:1329660
Change-Id: I2b8b7b7fd530727456ac5c46e694e7dbad6deff2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273986
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Joe Downing <joedow@google.com>
Cr-Commit-Position: refs/heads/main@{#38022}
2022-09-06 16:31:40 +00:00