This change adds a few UMAs to inform on the usage of frame
rate constraints related to screenshare sessions.
go/rtc-0hz-present
Bug: chromium:1255737
Change-Id: Icdd011a8e7df837416d603beeb0866d9eb1918e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235368
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35238}
This change
- adds new type VideoTrackSourceConstraints expressing min/max FPS
constraints.
- adds new method VideoTrackSourceInterface::ProcessConstraints.
- adds new method VideoSinkInterface<>::OnConstraintsChanged.
- updates AdaptedVideoTrackSource and VideoBroadcaster to forward
the constraints to sinks.
- adds several unit tests for the added functionality.
- and finally, implements OnConstraintsChanged in VideoStreamEncoder.
Chromium will be updated in coming CLs to supply constraints set
through the MediaStream module.
go/rtc-0hz-present
Bug: chromium:1255737
No-Try: true
Change-Id: Iffef239217269c332a1aaa902ddeae2440929e22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235040
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35197}
Encoder info will be modified at runtime. In fact, we should reduce the
number of 'full' ReconfigureEncoder(). If only need subset of it at
runtime, consider handle it in VideoStreamEncoder::EncodeVideoFrame().
Consider two cases:
Re-configure scalers when encoder info changed. Consider two cases:
1. When the status of the scaler changes from enabled to disabled, if we
don't do this CL, scaler will adapt up/down to trigger an unnecessary
full ReconfigureEncoder() when the scaler should be banned.
2. When the status of the scaler changes from disabled to enabled, if we
don't do this CL, scaler will not work until some code trigger
ReconfigureEncoder(). In extreme cases, the scaler doesn't even work for
a long time when we expect that the scaler should work.
This CL aims to make scalers work properly when encoder info changed.
BUG: None
Change-Id: Iec17730b5fac5e642c0fb2d9b11c5b7434f0a220
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233384
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35175}
Use config from FakeEncoder in some tests.
Bug: none
Change-Id: I1d7e01f604f8aabb5d6815bb519ef2532d024d76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233243
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35138}
First CL to try to understand the extent of the cleanup needed in
order to remove -Wno-shadow and follow Chromium on enabling this
diagnostic.
Bug: webrtc:13219
Change-Id: Ie699762da50fe3dbc08b1fd92220962d4b7da86b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35134}
The |slice_qp_detla| reported by the hardware is not credible, which
causing the quality scaler cannot work properly,the resolution cannot
be adjusted correctly.
To fix this issue, this CL implements a bandwidth scaler which is used
for adjust resolution, this scaler will be used when QP based quality
scaler is not working due to untrusted QP reported by HW AVC encoder.
Bug: webrtc:12942
Change-Id: I2fc5f07a5400ec7e5ead2c2c502faee84d7f2a76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228860
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35120}
min_pacing:8ms, to avoid the situation where bursts of frames are sent
to the decoder at once due to network jitter. The bursts of frames
caused the queues further down in the processing to be full and
therefore drop all frames.
max_decode_queue_size:8, in the event that too many frames have piled
up, do as before and send all frames to the decoder to avoid building
up any latency.
These setting only affect the low-latency video pipeline that is enabled
by setting the playout RTP header extension to min=0ms, max>0ms.
Bug: chromium:1138888
Change-Id: I8154bf3efe7450b770da8387f8fb6b23f6be26bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233220
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35119}
Currently the implementation of FrameTransformers uses distinct,
incompatible types for recevied vs about-to-be-sent frames. This adds a
flag in the interface so we can at least check that we are being given
the correct type. crbug.com/1250638 tracks removing the need for this.
Chrome will be updated after this to check the direction flag and provide
a javascript error if the wrong type of frame is written into the
encoded insertable streams writable stream, rather than crashing.
Bug: chromium:1247260
Change-Id: I9cbb66962ea0718ed47c5e5dba19a8ff9635b0b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232301
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <toprice@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35100}
maxBitrate is set to a minimum of kEncoderMinBitrateKbps in VideoCodecInitializer::SetupCodec and cannot be zero at this point.
Bug: none
Change-Id: I4e062b054d99fabc1a9650260db03dd45b033c3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230142
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35094}
Call SetMaxBitrate when encoder is configured instead of in OnMaybeEncodeFrame (which is called after the initial frame dropping ->
max bitrate is not set for dropped frames).
Added support for single active stream configuration.
Bug: none
Change-Id: I33ff96e7feed70b9ea3c9b3da89f117859108347
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231681
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34973}
The experiment WebRTC-PreStreamDecoders (aka Lazy decoder creation) has
investigated the benefit of only creating a subset of all decoders
during negotiation and the remaining decoders on demand.
This CL changes the default value to only create one decoder during
negotiation. This frees up hardware resources and reduces the SDP
negotiation time.
Bug: chromium:1202042
Change-Id: I6e2206839162aa857fcc948ccd53d0ff91cbdeaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231643
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34959}
Setting the rtp header extensions on the packet delivery thread
(currently worker, soon to be network), is now possible without
taking the hit of deleting and recreating the receive stream (and
rtp receiver and related state).
Bug: webrtc:11993
Change-Id: I9bbe306844a25d85d79cd216092ead66eaf68960
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223741
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34953}
Instead offer getters for the sync_group and rtp struct. Both are
a part of the config but expose much less of the config, which has
mutable parts.
Bug: none
Change-Id: Icc8007246e9776a5d20f30cda1a2df3fb7252ffc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229980
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34838}
The config struct is big and in order to control access to its state,
some of which isn't always const, we need to limit raw unlocked access
to it from other classes.
Bug: none
Change-Id: I4513c41486e79ef6c5cfd6376122ab338ad94642
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229921
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34835}
After a send stream is stopped, it can still be re-used and implicitly
restarted by activating layers. This change removes marking the flag
we use for async operations as 'not alive' inside Stop() and only doing
so when the send stream is stopped permanently.
The effect this has is that an implicit start via
UpdateActiveSimulcastLayers() will run and correctly update the states.
Before, if a stream had been stopped, the safety flag would prevent
the async operation from running.
Bug: chromium:1241213
Change-Id: Iebdfabba3e1955aafa364760eebd4f66281bcc60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229304
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34809}
Setting different number of temporal layers is supported by SimulcastEncodeAdapter and LibvpxVp8Encoder will fallback to SimulcastEncoderAdapter if InitEncode fails.
Bug: none
Change-Id: I8a09ee1e6c70a0006317957c0802d019a0d28ca2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228642
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34785}
from runtime check in proxy classes that picks decoder (VCMDecoderDataBase)
to a DCHECK in the VideoDecoder::Settings
Bug: None
Change-Id: Ic8c2e971486a3a7eb247f9d03815aba5ca5a7bad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228644
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34761}
Instead of requiring huge VideoCodec struct, pass single member from it
Bug: webrtc:13045
Change-Id: I1a4a48abd6c407cb9a878daafda5c8a85beff39e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228641
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34753}
When implicit start/stop happens via activation/deactivation of layers
occurs, don't change the state of the 'alive' flag since further
activations following full de-activation need to be allowed to happen
when Stop() has not been called.
Bug: chromium:1234779
Change-Id: Ic3cae387990122eaa2f48de096ff9dafa7c34414
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228242
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34698}
Currently encode_frame_rate is updated (ComputeRate called) when a frame is encoded.
If a stream is stopped, encode_frame_rate will have an old value (the framerate at the time of the last encoded frame) instead of zero.
Bug: webrtc:13037
Change-Id: I1a2122df61e3e8187e57155dda71c0173cda4c5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228220
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34695}
When layers are activated/deactivated via UpdateActiveSimulcastLayers,
the flag wasn't being updated. This resulted in calls to Stop() getting
ignored after an implicit start via activating layers.
Bug: chromium:1234779
Change-Id: I4a72e624874526d27d3e97d6903112367c5e77fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227700
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34654}
Update framerate for top spatial layer instead of per timestamp (to ensure all simulcast layers are updated).
Bug: webrtc:13037
Change-Id: I4fa423dee40d74aee22a87855207b885f0536e25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227344
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34634}
Some hardware H.264 encoders does not place average QP delta in
slice_qp_delta field. Adding an optional flag in EncoderInfo to notify
quality scaler about this.
Bug: webrtc:12942
Change-Id: I3ee29c5ae9bd7bb34d26eba7e6bede3798ca44b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226921
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34627}
The alternative new name proposed, NackTracker, is already in
use in audio_coding.
Fixed: webrtc:11594
Change-Id: I6a05fafc05fa7ddb18ea4f64886a135e5ef59f7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226744
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34539}
As part of go/coil update code search links to not point to the
"master" branch.
Bug: chromium:1226942
Change-Id: I0ae9e84ecc660f789a69fe0b226f93bbc39a8a66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226081
Commit-Queue: Tony Herre <toprice@chromium.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34531}
NackModule2 creates repeating tasks, but as there are
many modules (one per receiver) these tasks execute out
of phase with each other, multipliying the amount of wakeups
caused.
Fix this by creating a single wakeup source that serves all
NackModule2 instances in a call.
Bug: webrtc:12989
Change-Id: Ia9c84307eb57349679e42b673474feb2cb43f08e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226464
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34527}
