This information is now readily available. Let's expose it.
In practise we don't pace audio by default and the delay is ~0, however
we can tell that this metric is working as intended by setting
PacingController's pace_audio_ to true via the "WebRTC-Pacer-BlockAudio"
field trial. In this case chrome://webrtc-internals/ plots neats graphs
for audio send delay.
Bug: webrtc:10635
Change-Id: Iecfd93bb84ec61e5d54232769a9e7a500601b199
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280523
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38483}
Also run IWYU on a file picked for testing and check in the result.
Bug: none
Change-Id: Ide36bc59d126064f2bab7af441f72a6e8477c848
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280601
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38481}
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}
ScalabilityMode should be validated against the currently
allowed codecs or the currently used codec.
Bug: webrtc:11607
Change-Id: Id2e6cbfad4f089de450150e1203657ed316e2f29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277403
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38433}
Manually tested with libwebrtc built for Android and a solution running into the same problem as the linked repro in chromium:1348132.
Bug: chromium:1348132
Change-Id: I88260b9ac72c67f1a6ad927e075d1490ac06ce91
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278241
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38335}
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/ 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 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 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}
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}
I am working on enabling AV1 profile-1 support for Chrome Remote
Desktop and I noticed that when our host adds both AV1 profile-0
and AV1 profile-1 codecs to the SDP, the second codec is stripped.
I tracked the problem down to this class as the IsSameCodecSpecific
function was not looking at the format params to determine whether
the additional AV1 codec entries were duplciates of the first.
Bug: chromium:1329660
Change-Id: I6ee0c264657203631a43f74f64e08153dca4f63a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273981
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Joe Downing <joedow@google.com>
Cr-Commit-Position: refs/heads/main@{#38020}
In rare cases, it is possible to queue a call to SendData from the signaling
thread on a channel being closed or already closed in the network thread.
By keeping track of currently open streams, we avoid sending messages
with a stream id of channels that the other side already considers closed
and has already reused for a new channel.
This caused rare messages to be delivered on the wrong data channel if
a message was quickly sent, channel closed and a new one reopened.
Bug: webrtc:14277
Change-Id: If35fed8d12d5d2c18cdc6601085d8b632c37a0ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272624
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37880}
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}
This way we're sure instantiation, configuration and decode calls all
happen on the decoder queue - making thread checking easier in the
actual decoder classes.
Bug: None
Change-Id: Ia98f47009f26b34eb8dad2ee0b4ddcde082d1994
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272022
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37825}
A follow-up change will combine the setters for ulpfec and red payload
types, since they're entwined.
Bug: webrtc:11993
Change-Id: Ifea7fe9f4ebc7ac88a62db6cd6748f4d3c20db4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271482
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37785}
This allows for `config_.rtp.nack.rtp_history_ms` to be modified
without deleting and recreating video receive streams.
Bug: webrtc:11993
Change-Id: I8ba132b22fe0e6de03d1c42fc38a570cbe138817
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269301
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37701}
The line "No audio processing module present [...]" has mislead people several times (see linked bug for one example) and does not add any information that cannot already relatively easily be inferred from platform configuration.
Other removed lines are duplicate (already logged via AudioOptions::ToString()) or never runs (ApplyOptions always returns true + empty #elif).
Bug: b/238780321#comment34
Change-Id: Ie0fbd6675ec963c1180a7f614ec74bba5e850777
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270483
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37697}
By attaching the sink when all is set up, we make it possible for an
application to listen for that event - and only then start producing
frames. Otherwise frames risk being dropped during the setup phase.
Bug: webrtc:14276
Change-Id: I0a906681fc526b0ee88c60a842afb0d68e21de14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270660
Auto-Submit: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37696}
...to allow for turning on/off loss notifications for video receive
streams without tearing down and recreating the whole stream.
Bug: webrtc:11993
Change-Id: Ia961bd343ce816ffe3414f11e3a58bb3c235307c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269252
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37688}
This is a step towards getting rid of reconfiguration via tearing down
and reconstructing receive streams when parts of the configuration
change at runtime.
Bug: webrtc:11993
Change-Id: I337e523f17805b75826ddbd75bd3d0eb6e910bd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269250
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37680}
...for payload type changes and avoid recreating the main video stream.
Bug: webrtc:11993
Change-Id: I03e44ff25d88caeb082a2e44b2e802d3b9392feb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269244
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37666}
This reduces the number of times we recreate video receive streams
and prepares for not having to do that for flexfec streams and avoid
having to recreate a video receive stream when flexfec config changes.
Bug: webrtc:11993
Change-Id: I11134b6a72eb162623ebbc12521d409da8616107
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261941
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37641}
