This reverts commit 8b715657fba050d9fd817911f94a8e13b23dfdec.
Reason for revert: A couple of days has passed and we should have enough data points to be able to detect changes.
Original change's description:
> Revert "Enable experiment WebRTC-SendPacketsOnWorkerThread in pc_full_stack_test"
>
> This reverts commit 1b3f531da404c200da09f229799e827250347b60.
>
> Reason for revert: Simulated network changes has been reverted.
> In order to see the effect of this experiment, there should not be other larger changes affecting the metrics of a few runs.
> https://webrtc.googlesource.com/src/+/baf5c9fabd3eba46a2b7747df00b1124a8f5def8
>
> Original change's description:
> > Enable experiment WebRTC-SendPacketsOnWorkerThread in pc_full_stack_test
> >
> > This is a follow up to https://webrtc-review.googlesource.com/c/src/+/278980 to actually enable the experiment in some tests.
> >
> > Bug: webrtc:14502
> > Change-Id: I166f984bcb94527adc6ebb9169b66abf0f105d76
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279140
> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> > Commit-Queue: Per Kjellander <perkj@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#38407}
>
> Bug: webrtc:14502
> Change-Id: I6e5a607a284186895d1ecd622fdf28f5c1ffd187
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279600
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38417}
Bug: webrtc:14502
Change-Id: I6a179e963e54d266ddbf84ce3287c6b61256833f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279901
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38501}
This ensures that if for some reason, the frame buffer becomes
undecodable while waiting to decode a frame, the decoding is halted.
This also guards against receiving an empty temporal unit from the frame
buffer, even though this should never happen when the frame buffer has a
decodable temporal unit.
Bug: chromium:1378253, chromium:1361623
Change-Id: I8c4c897bf474d5cbda5f0f357781bf1dc0701fe4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280701
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38494}
The scalability mode should now be supported for all test configurations.
Bug: none
Change-Id: I79aeb56b35d62265c94edefdbcb10c6835bc2750
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280200
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38482}
This metric was always supposed to be the spec's answer to
googBucketDelay, and is defined as "The total number of seconds that
packets have spent buffered locally before being transmitted onto the
network." But our implementation measured the time between capture and
send, including encode time. This is incorrect and yields a much larger
value than expected.
This CL updated the metric to do what the spec says. Implementation-wise
we measure the time between pushing and popping each packet from the
queue (in modules/pacing/prioritized_packet_queue.cc).
The spec says to increment the delay counter at the same time as we
increment the packet counter in order for the app to be able to do
"delta totalPacketSendDelay / delta packetSent". For this reason,
`total_packet_delay` is added to RtpPacketCounter. (Previously, the
two counters were incremented on different threads and observers.)
Running Google Meet on a good network, I could observe a 2-3 ms average
send delay per packet with this implementation compared to 20-30 ms
with the old implementation. See b/137014977#comment170 for comparison
with googBucketDelay which is a little bit different by design -
totalPacketSendDelay is clearly better than googBucketDelay.
Since none of this depend on the media kind, we can wire up this metric
for audio as well in a follow-up:
https://webrtc-review.googlesource.com/c/src/+/280523
Bug: webrtc:14593
Change-Id: If8fcd82fee74030d0923ee5df2c2aea2264600d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280443
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38480}
defined in
https://w3c.github.io/webrtc-encoded-transform/#rtcrtpsender-extension
Note: this does not implement the "rid(s)" parameter which will be done in a future CL.
VP8 still synchronizes keyframes on all layers even when asked for ones on individual layers while H264 (when implemented as three different encoders in SimulcastEncoderAdapter) can actually utilize this.
This does not change the behavior when receiving a RTCP PLI for a particular layer.
BUG=chromium:1354101
Change-Id: Ic8b14d155242e32c9aeafa55fe6652f346ac76b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274169
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38472}
This reverts commit 1b3f531da404c200da09f229799e827250347b60.
Reason for revert: Simulated network changes has been reverted.
In order to see the effect of this experiment, there should not be other larger changes affecting the metrics of a few runs.
https://webrtc.googlesource.com/src/+/baf5c9fabd3eba46a2b7747df00b1124a8f5def8
Original change's description:
> Enable experiment WebRTC-SendPacketsOnWorkerThread in pc_full_stack_test
>
> This is a follow up to https://webrtc-review.googlesource.com/c/src/+/278980 to actually enable the experiment in some tests.
>
> Bug: webrtc:14502
> Change-Id: I166f984bcb94527adc6ebb9169b66abf0f105d76
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279140
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38407}
Bug: webrtc:14502
Change-Id: I6e5a607a284186895d1ecd622fdf28f5c1ffd187
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279600
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38417}
The parameterized tests run in two versions.
1. Default with separate network and worker thread
2. The network thread is used as a worker thread. All packet sending and receiving is done on that thread.
Bug: webrtc:14502
Change-Id: Iba09295f6c1d7030d726f387f9cab8c2bf7c03d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278980
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38379}
This spills to a few more clasess....
Change-Id: Iea79e3b4ac86b30db6f13da89a47ab7000c5440a
Bug: webrtc:14502
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277803
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38334}
This patch
1) modifies VideoAdapter to use requested_resolution
instead on OnOutputFormatRequest, iff there are no active encoders
that is not using requested_resolution (i.e all "old" encoder(s) are
not active).
2) modifies VideoBroadcaster to not broadcast wants from
encoders that are not active (iff there is an active encoder
using requested_resolution).
3) fixes a bug in encoder_stream_factor in that the
requested_resolution was not propagated to return value
(must have been lost in merge?).
Bug: webrtc:14451
Change-Id: I00e0907f0fe9329141ed169576fa46cdc5384886
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278360
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@{#38323}
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}
Reasons:
1) It is not used by `PeerConnection` (only in tests)
2) We have no plans on using it
3) The code is functionally untouched since many years
Bug: b/249972434
Change-Id: I1d30edd34231f25d86e8495ff71f1786ba2b0a1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277445
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38260}
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 updates VideoReceiveStream2::Stop() to symmetrically tear down
state that's built up in VideoReceiveStream2::Start().
Bug: webrtc:11993, webrtc:14486
Change-Id: I41f4feea5584e5baaeed2143432136f8b9761321
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272537
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38244}
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}
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 experiment has been approved for a full launch. Changing the
default value so that no decoder is created before the stream starts.
All decoders are created lazily on demand when we receive payload
data of the corresponding type.
Bug: chromium:1319864
Change-Id: Ifb412bbe49a7577a45c340496d5b8572ebc1ba44
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277120
Auto-Submit: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38232}
This cl is a preparation for https://webrtc-review.googlesource.com/c/src/+/277002
so that downstream tests can be modified first.
Bug: webrtc:14451
Change-Id: I03b613db695c533bd77ab4bd2f8b35677094443c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277003
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38231}
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}
Creates the EmulatedSFUConfig that will receive the parameters for
controlling the virtual SFU used in the call.
Its current only field is the previous target_spatial_index from
VideoSimulcastConfig.
This allow to filter out the bottom layers for SVC S mode tests
and enable them.
Bug: webrtc:11607
Change-Id: Id4f3a96b3a03b9be7155796c3bafefce01f32b7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274162
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38182}
New ctor added without optional and media specific fields.
Bug: webrtc:10739, b/246753278
Change-Id: I7e15849aced6ed0a7ada725ea171a15ea1e9bc5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275941
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38124}
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}
* Migrate callback to one-time absl::AnyInvocable.
* Clean tests to use MockFunction.
* Use main thread instead of helper function in unittests.
* Fix some spelling mistakes.
Bug: None
Change-Id: I6145f5f5e2748dfa5278898cfdfd762c1840ff8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274170
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38016}
The new frame buffer is already the default. Sync decoding can now be
inferred by the presence of a metronome rather than using the field
trial.
Tests have been updated to use the DecodeSynchronizer rather than the
field trial.
Bug: webrtc:14003
Change-Id: I33b457feaf4eac1500f3bf828680e445ae4d93cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274163
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38011}
The frame cadence adapter would previously crash on encountering
unconfigured layer updates. Fix this to silently ignore such updates.
Fixed: webrtc:14417
Change-Id: I524aa196f6aed10ffb8cd4ddcb4b053c71dfabba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273325
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37980}
This moves the ownership away from VideoReceiveStream2 and closer to
VCMDecoderDataBase. That facilitates unregistration (upcoming change)
without recreating receive streams.
Bug: none
Change-Id: I812175134730a0ffbf7077fd149c8489481c73d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272481
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37866}
