VideoStreamEncoder creates VideoEncoders. To pass an Environment to VideoEncoder, it should be available in the VideoStreamEncoder.
Bug: webrtc:15860
Change-Id: Id89ac024ce61fdd9673bb66f03f94f243fc0c7f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341840
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41861}
A call to GetScalabilityMode was added for logging purpose and causes an expectation failure for tests using 4 temporal layers.
Plan is to remove the old GetScalabilityMode and keep only the one that returns an optional.
Change-Id: I0e37a496bb621d9754d6572ef5838b58193aa183
Bug: b/327381318
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341520
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#41838}
This is a reland of commit 050ffefd854f8a57071992238723259e9ae0d85a
Original change's description:
> Extends WebRTC logs for software encoder fallback
>
> This CL extends logging related to HW->SW fallbacks on the encoder
> side in WebRTC. The goal is to make it easier to track down the
> different steps taken when setting up the video encoder and why/when
> HW encoding fails.
>
> Current logs are added on several lines which makes regexp searching
> difficult. This CL adds all related information on one line instead.
>
> Three new search tags are also added VSE (VideoStreamEncoder), VESFW
> (VideoEncoderSoftwareFallbackWrapper) and SEA (SimulcastEncoderAdapter). The idea is to allow searching for the tags to see correlated logs.
>
> It has been verified that these added logs also show up in WebRTC
> logs in Meet.
>
> Logs from the GPU process are not included due to the sandboxed
> nature which makes it much more complex to add to the native
> WebRTC log. I think that these simple logs will provide value as is.
>
> Example: https://gist.github.com/henrik-and/41946f7f0b10774241bd14d7687f770b
>
> Bug: b/322132132
> Change-Id: Iec58c9741a9dd6bab3236a88e9a6e45440f5d980
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339260
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41733}
NOTRY=true
Bug: b/322132132
Change-Id: I25dd34b9ba59ea8502e47b4c89cd111430636e08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339680
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41736}
This reverts commit 050ffefd854f8a57071992238723259e9ae0d85a.
Reason for revert: Breaks downstream project.
Original change's description:
> Extends WebRTC logs for software encoder fallback
>
> This CL extends logging related to HW->SW fallbacks on the encoder
> side in WebRTC. The goal is to make it easier to track down the
> different steps taken when setting up the video encoder and why/when
> HW encoding fails.
>
> Current logs are added on several lines which makes regexp searching
> difficult. This CL adds all related information on one line instead.
>
> Three new search tags are also added VSE (VideoStreamEncoder), VESFW
> (VideoEncoderSoftwareFallbackWrapper) and SEA (SimulcastEncoderAdapter). The idea is to allow searching for the tags to see correlated logs.
>
> It has been verified that these added logs also show up in WebRTC
> logs in Meet.
>
> Logs from the GPU process are not included due to the sandboxed
> nature which makes it much more complex to add to the native
> WebRTC log. I think that these simple logs will provide value as is.
>
> Example: https://gist.github.com/henrik-and/41946f7f0b10774241bd14d7687f770b
>
> Bug: b/322132132
> Change-Id: Iec58c9741a9dd6bab3236a88e9a6e45440f5d980
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339260
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41733}
Bug: b/322132132
Change-Id: I24d0a4e71a43ac192485f1af208563a51d919865
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339661
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41735}
This CL extends logging related to HW->SW fallbacks on the encoder
side in WebRTC. The goal is to make it easier to track down the
different steps taken when setting up the video encoder and why/when
HW encoding fails.
Current logs are added on several lines which makes regexp searching
difficult. This CL adds all related information on one line instead.
Three new search tags are also added VSE (VideoStreamEncoder), VESFW
(VideoEncoderSoftwareFallbackWrapper) and SEA (SimulcastEncoderAdapter). The idea is to allow searching for the tags to see correlated logs.
It has been verified that these added logs also show up in WebRTC
logs in Meet.
Logs from the GPU process are not included due to the sandboxed
nature which makes it much more complex to add to the native
WebRTC log. I think that these simple logs will provide value as is.
Example: https://gist.github.com/henrik-and/41946f7f0b10774241bd14d7687f770b
Bug: b/322132132
Change-Id: Iec58c9741a9dd6bab3236a88e9a6e45440f5d980
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339260
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41733}
Instead embed functionality of the rtc::TaskQueue into destructors and describe the potential race.
Bug: webrtc:14169
Change-Id: I01b570b530986a0d07798893057201493a8bef5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335141
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41592}
This change ensures that the FCA now is informed about a new max fps
when VideoStreamEncoder::OnVideoSourceRestrictionsUpdated is called.
The latest restricted frame rate which is provided to the FCA will
only affect the cadence of repeated non-idle (quality has not
converged) frames and the main goal is to ensure that the FCA reduces
its repeat rate in situations where the video source is constrained.
UpdateVideoSourceRestrictions is added to the FrameCadenceAdapter API
and it is called from the VideoStreamEncoder when its source
parameters (resolution and/or frame rate) are restricted.
This modification has no effect on the flow driven by
ProcessOnDelayedCadence (non repeated frames).
Bug: webrtc:15539
Change-Id: I26dee6480e5137f82c5ccf57091b737cad82dbf6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328300
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41308}
The FrameCadenceAdapterInterface::Callback::OnFrame method in
VideoStreamEncoder only changed frame handling on
frames_scheduled_for_processing being 1. This CL changes
the parameter to more explicitly signal queue overload via
boolean parameter.
Bug: None
Change-Id: I1eb46b34fc4d748b7e2f1921642497c939adf197
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327761
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41226}
The change adds dropped frame reporting for previously dropped frame
and also cleans up the colon list of the VSE.
Bug: None
Change-Id: Iad1c084739e5392ded4f100d940b45adf9b561ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327800
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41225}
This reverts commit 8039cdbe48f8c8bb91fa1761f807005a7b497196.
Reason for revert: remove functionality after measurement complete
Original change's description:
> Measure wall clock time of capture and encode processing.
>
> (NOTE: This and dependent CLs will be reverted in a few days after
> data collection from the field is complete.)
>
> This change introduces a new task queue concept, Voucher. They
> are associated with a currently running task tree. Whenever
> tasks are posted, the current voucher is inherited and set as
> current in the new task.
>
> The voucher exists for as long as there are direct and indirect
> tasks running that descend from the task where the voucher was
> created.
>
> Vouchers aggregate application-specific attachments, which perform
> logic unrelated to Voucher progression. This particular change adds
> an attachment that measures time from capture to all encode operations
> complete, and places it into the WebRTC.Video.CaptureToSendTimeMs UMA.
>
> An accompanying Chrome change crrev.com/c/4992282 ensures survival of
> vouchers across certain Mojo IPC.
>
> Bug: chromium:1498378
> Change-Id: I2a27800a4e5504f219d8b9d33c56a48904cf6dde
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325400
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41061}
Bug: chromium:1498378
Change-Id: I9503575fbc52f1946ca26fc3c17b623ea75cd3c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327023
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41135}
This reverts commit 03bc3a0fa67e274efb4518da005f4c5b77c607e9.
Reason for revert: measurement complete
Original change's description:
> VideoStreamEncoder: exclude screencast from capture time measurement.
>
> This CL avoids measurement for screencast encoding work. The reason is
> screencast can cling on to and re-encode old video frames for which
> webrtc::VideoFrame::reference_time() is unchanged.
>
> Bug: chromium:1498378
> Change-Id: I5bf79d29ef7f57ddff2622cbb6c3436480bd16ba
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326103
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Auto-Submit: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41097}
Bug: chromium:1498378
Change-Id: I42c1a86123eb1d6c7ad7c8981769f5560884a2f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327025
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41133}
This CL avoids measurement for screencast encoding work. The reason is
screencast can cling on to and re-encode old video frames for which
webrtc::VideoFrame::reference_time() is unchanged.
Bug: chromium:1498378
Change-Id: I5bf79d29ef7f57ddff2622cbb6c3436480bd16ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326103
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41097}
(NOTE: This and dependent CLs will be reverted in a few days after
data collection from the field is complete.)
This change introduces a new task queue concept, Voucher. They
are associated with a currently running task tree. Whenever
tasks are posted, the current voucher is inherited and set as
current in the new task.
The voucher exists for as long as there are direct and indirect
tasks running that descend from the task where the voucher was
created.
Vouchers aggregate application-specific attachments, which perform
logic unrelated to Voucher progression. This particular change adds
an attachment that measures time from capture to all encode operations
complete, and places it into the WebRTC.Video.CaptureToSendTimeMs UMA.
An accompanying Chrome change crrev.com/c/4992282 ensures survival of
vouchers across certain Mojo IPC.
Bug: chromium:1498378
Change-Id: I2a27800a4e5504f219d8b9d33c56a48904cf6dde
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41061}
Convert most field trials used in PCLF tests.
Change-Id: I26c0c4b1164bb0870aae1a488942cde888cb459d
Bug: webrtc:10335
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322703
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40909}
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}
Expose new function MaybeCreateFrameDumpingEncoderWrapper that
wraps another passed encoder and dumps its encoded frames out
into a unique IVF file into the directory specified by the
"WebRTC-EncoderDataDumpDirectory" field trial. If the passed
encoder is nullptr, or the field trial is not setup, the function
just returns the passed encoder. The directory specified by the
field trial parameter should be delimited by ';'.
The new function is wired up in VideoStreamEncoder.
Bug: b/296242528
Change-Id: I6143adf899f78fcc03d4239a86c68dcbab483f1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317200
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40600}
which makes it possible to understand which error occured.
BUG=chromium:1366910
Change-Id: Ided288ea7aa7c6cb283f7d46692c67efb15764d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316863
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40573}
It used input frame resolution before this change which caused unnecessary resolution adaptations when resolution scaling is used.
Found that initial frame dropping was always enabled for AV1 SVC. After fixing DropDueToSize the AV1 SVC tests [1] started to fail ("number of encoded temporal layers is less than expected") on bots. The tests encode 1850x1110 in L3T3 for 5s using the default 300kbps start bitrate. Before the fix the initial frame dropping kicked in and reduced the resolution to a level that let encoder to generate all temporal layers. After the fix the resolution stayed at 1850x1110 and encoder dropped all T1 and T2 layer frames. Mitigated this by increasing test duration from 5 to 10s. This gives enough time for BWE to ramp up and for encoder to generate (stop dropping) all temporal layers.
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/test/svc_e2e_tests.cc;l=460;bpv=1
Bug: chromium:1466809
Change-Id: I16802689e234f8fc16f891f024d5f644985de01c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315142
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40536}
EncoderStreamFactory has two code paths for creating a stream: the
"simulcast path" and the "default path". Only the former cares about
encoding paramter's maxBitrate. The latter assumes that
`encoder_config.max_bitrate_bps` already encompasses the maxBitrate of
the first encoding, but this is not always the case.
As of M113, when scalability mode is specified, {active,inactive} does
not count as simulcast stream but as a default stream represented by
encoding[0].
The problem is that `encoder_config.max_bitrate_bps` only includes
`encodings[0].max_bitrate_bps` when `encodings.size() == 1` which isn't
the case here.
This CL fixes the problem by making the "create default stream" code
path look at the first encoding's maxBitrate and remove existing
assumptions that `encoder_config.max_bitrate_bps` encompasses
`encodings[0].max_bitrate_bps`. This is a step in the right direction
since we're trying to remove all special cases and have encodings map
1:1 with SSRCs, so the "max bps of entire stream" should indeed be a
separate limit than the per-encoding limits and it was confusing that
sometimes it included and sometimes it excluded encoding[0]'s limit.
This issue did not happen in {inactive,active} since that code path
counts as "simulcast stream", so "default stream" is only ever
applicable for index 0.
TESTED=Simulcast Playground, see https://crbug.com/1455962.
Bug: chromium:1455962
Change-Id: I7c44925b780623b5979751e8959e972293648a3d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313282
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@{#40482}
This extension is documented to carry one bit: Screenshare.
It's been used for carrying simulcast layers and experiment IDs.
This CL removes that usage.
Bug: webrtc:15383
Change-Id: I048b283cde59bf1f607d8abdd53ced07a7add6f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312420
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40457}
allowing for better correlation with MaybeEncodeVideoFrame
which also logs the ntp timestamp.
BUG=None
Change-Id: I00fc99e69cd703f6da3f25043361d68b3cb3f3fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311542
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40415}
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}
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}
The goal of the VP9 simulcast project is that when `scalability_mode`
is set, multiple encodings are always interpreted as simulcast, even
if VP9 or AV1 is used. This CL makes this so, but only if the flag
"WebRTC-AllowDisablingLegacyScalability" is "/Enabled/". This allows us
to make "SendingThreeEncodings_VP9_Simulcast" EXPECT VP9 simulcast.
When we are ready to ship we will remove the need to use the field
trial, but before we ship this we'll want to revisit if
SvcRateAllocator can be updated to support simulcast. (Today if we use
SvcRateAllocator when VP9 simulcast is used, all encodings except the
first one get bitrate=0, causing the test to fail because media is not
flowing on all layers.) For now, a TODO is added.
Bug: webrtc:14884
Change-Id: Ie20ae748b0c0405162f3a1b015ab94956ef83dae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297340
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39552}
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}
This is a reland of commit 8ad4924936dea2bd97990b0a951df93f7526f0ff
See diff between latest Patch Set and PS1. Fixes include:
- VideoStreamEncoder's call to bitrate_adjuster_->OnEncodedFrame()
is updated to take stream index (spatial or simulcast index) instead
of only looking at SpatialIndex().
- Migrate test-only helpers to use Spatial/SimulcastIndex correctly.
The fixes are to migrate
some test-only helpers that we had forgot to fix that are used by
external tests.
Original change's description:
> Make SimulcastIndex() and SpatialIndex() distinct (remove fallback).
>
> This CL removes the fallback logic to return the other index when the
> one requested has not been set. This means we can remove the codec gates
> that was previously needed because SpatialIndex() had multiple meanings,
> resolving the TODOs previously added in
> https://webrtc-review.googlesource.com/c/src/+/293343.
>
> We have already migrated all known external dependencies from
> SpatialIndex() to SimulcastIndex() where necessary, unblocking this CL.
>
> PSA: https://groups.google.com/g/discuss-webrtc/c/SDAVg6xJ3gY
>
> Bug: webrtc:14884
> Change-Id: I82787505ab10be151e5f64965b270c45465d63a9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293740
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39343}
Bug: webrtc:14884
Change-Id: Ib966924efca1a040dae881599f0789a7f2ab24a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294284
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39358}
This makes the max bitrate configured in RTP parameter to still be
respected, instead of being overridden by defaults in the case that an
encoder with untrusted QP is reconfigured (e.g. due to some adaptation).
Bug: webrtc:14914
Change-Id: I2d3ff645c069b80ec2e36887e6ce0ecd09a7ecbf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293944
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39357}
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}
Declare two VideoEncoder::EncoderInfo local variables as const. This
makes it clear that they are equal to the return value of
encoder_->GetEncoderInfo().
Bug: None
Change-Id: I08ab34e670e6eb1cb3c67a48b4e9826902d9d9fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293385
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Wan-Teh Chang <wtc@google.com>
Cr-Commit-Position: refs/heads/main@{#39321}
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}
When disabling a spatial layer, reconfiguration of the encoder is not
necessary (bitrate will never be assigned to the inactive layer anway).
This CL however makes sure we reconfigure the encoder when a spatial
layer is activated. Some encoder implementations may encoder the wrong
number of spatial layers if the active layers have not beens set
correctly.
Bug: webrtc:14809, b/261097903
Change-Id: I8d34aaec95eb50a9717c06ea38f25088e5a96429
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290560
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38999}
As the synchronous version only posts a task to recreate the encoder
later, it is not possible to catch errors and state changes that
could appear then.
The asynchronous version of SetParameters() aims to solve this by
providing a callback to wait for the completion of the encoder
reconfiguration, allowing any error to be propagate and subsequent
getParameters() call to have up to date information.
Bug: webrtc:11607
Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38627}
and do the resolution of rids to layers. This has no effect yet
since the simulcast encoder adapter (SimulcastEncoderAdapter::Encode), the VP8 encoder (LibvpxVp8Encoder::Encode) and the OpenH264 encoder (H264EncoderImpl::Encode) all generate a key frame for all layers whenever a key frame is requested on one layer.
BUG=chromium:1354101
Change-Id: I13f5f1bf136839a68942b0f6bf4f2d5890415250
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280945
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38565}
This cl/ adds resource adapation to the requested_resolution
feature. The restrictions that are sent to the video source
are also saved inside video_stream_encoder and used when
determining layer resolution.
Anticipated further patches
4) Let VideoSource do adaption if possible
Bug: webrtc:14451
Change-Id: Ia9b990a6b92b76af7ff6665a562f84585f79c35b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277580
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@{#38306}
This cl/ implements configuring of encode resolution
in the video_stream_encoder (webrtc_video_engine) in
a way that is independent of frame resolution (i.e
not using scale_resolution_down_by).
The cl/ reuses the VideoAdapter as is, and hence
the output resolution will be the same as it is today.
Anticipated further patches
3) Hook up resource adaptation
4) Let VideoSource do adaption if possible
Bug: webrtc:14451
Change-Id: I881b031c5b23be26cacfe138730154f1cb1b66a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276742
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38245}
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}
The Chromium RTCVideoEncoder unfortunately doesn't set if the
result is at target quality, and the definition of the threshold
is buried in libvpx_vp8_encoder.h.
This change
* Updates VideoStreamEncoder to postprocess an incoming EncodedImage
by interpreting the incoming QP information instead.
* Updates the related VideoStreamEncoder test to simulate an encoder
producing images around the QP threshold.
* Updates the steady state VP8 screencast QP threshold to a central
include file.
* Moves this and previously existing EncodedImage post-processing to a
new method AugmentEncodedImage.
Bug: b/245029833
Change-Id: I69ae29ffe501e84f28908f7d9a8cfd066ba82b43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275380
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38091}
BlockingCall doesn't take rtc::Location parameter and thus most of the dependencies on location can be removed
Bug: webrtc:11318
Change-Id: I91a17e342dd9a9e3e2c8f7fbe267474c98a8d0e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274620
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38045}
