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 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}
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}
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 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}
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}
follow-up from https://webrtc-review.googlesource.com/c/src/+/262810
* replace Time::Millis(0) and TimeDelta::Millis(0) with ::Zero()
* drop unnecessary webrtc namespace from some TimeDeltas
* make TimeDelta do the unit conversion for stats
BUG=webrtc:13756
Change-Id: Ic60625ae0fc7959a47a6be9f5051851feaf76373
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265875
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37664}
Also move ScalabilityModeToString to api and add RTC_EXPORT so that
Chromium can use it.
Bug: chromium:986069
Change-Id: I5dbbb6de9b14ca20f3ae0630552dcd44595ad5ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267780
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#37444}
This CL also removes the existing non-standard implementation of the metric.
Bug: webrtc:14147, webrtc:11789
Change-Id: I70fd1c451dfd59380fe5ce959086f37b31697c16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265360
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37441}
increasing precision since summing up rounded values leads to
a rounding error, in particular for small frames which take very
little time to decode.
BUG=webrtc:12526,webrtc:13756
Change-Id: I647c702808856a002c746ed9f115aa9bcaddc1f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262810
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#37249}
This CL accomplishes three things:
1) It enables feeding frame drop indications into the
AdaptedVideoTrackSource for the benefit of downstream projects.
2) Under zero hertz source delivery, a discarded frame ending a
sequence of frames which happened to contain important information
can be seen as a capture freeze. Avoid this by starting requesting
refresh frames after a grace period.
3) It changes the duration until first refresh frame requests on new
streams to three frame periods.
Bug: chromium:1324120, chromium:1336952
Change-Id: I0214852f1a26540588f6c193dd88a65c34ec0d99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265871
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37238}
This cl/ adds a way of setting an EncoderSelector on a specific
RtpSenderInterface. This makes it possible to easily use different
EncoderSelector on different streams within the same or different PeerConnections.
The cl/ is almost identical to the impl. of RtpSenderInterface::SetFrameEncryptor.
Iff a EncoderSelector is set on the RtpSender, it will take precedence
over the VideoEncoderFactory::GetEncoderSelector.
Bug: webrtc:14122
Change-Id: Ief4f7c06df7f1ef4ce3245de304a48e9de0ad587
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264542
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37150}
This is part of the project to delete the class entirely.
The CL also adds an "use_rtx" parameter to the function for listing
video codecs, rather than filtering those away afterwards.
Bug: webrtc:13931
Change-Id: I96b9b18c694a1c0986ccf22face76ef4c704d372
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262666
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36963}
implements a total frame assembly time statistic that measures the
cumulative time between the arrival of the first packet of a frame
(the lowest reception time) and the time all packets of the frame have
been received (i.e. the highest reception time)
This is similar to totalProcessingDelay
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
in particular with respect to only being incremented for frames that are being decoded but does not include the amount of time spent decoding the frame.
This statistic is useful for evaluating mechanisms like NACK and FEC
and gives some insight into the behavior of the pacer sending the
packets.
Note that for frames with just a single packet the assembly time will be zero. In order to calculate an average assembly time an additional frames_assembled_from_multiple_packets counter for frames with more than a single packet is added.
Currently this is a nonstandard stat so will only show up in webrtc-internals and not in getStats. Formally it can be defined as
totalAssemblyTime of type double
Only exists for video. The sum of the time, in seconds, each video frame takes from the time the first RTP packet is received (reception timestamp) and to the time the last RTP packet of a frame is received.
Given the complexities involved, the time of arrival or the reception timestamp is measured as close to the network layer as possible.
This metric is not incremented for frames that are not decoded, i.e., framesDropped, partialFramesLost or frames that fail decoding for other reasons (if any). Only incremented for frames consisting of more than one RTP packet. The average frame assembly time can be calculated by dividing the totalAssemblyTime with framesAssembledFromMultiplePacket.
framesAssembledFromMultiplePacket of type unsigned long
Only exists for video. It represents the total number of frames correctly decoded for this RTP stream that consist of more than one RTP packet.
For such frames the totalAssemblyTime is incremented.
BUG=webrtc:13986
Change-Id: Ie0ae431d72a57a0001c3240daba8eda35955f04e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260920
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36922}
DSCP is controlled by the spec-compliant API
RTCRtpEncodingParameters.networkPriority[1]. It already has a default
value that is the same as when DSCP is disabled.
- If you want non-default DSCP default values, you need to set
networkPriority and shouldn't need to set a non-standard googDscp flag
for it to have an effect.
- If you want the default DSCP value, you wouldn't change
networkPriority and so you don't care if enable_dscp is true... you'll
get the default regardless.
Drive-by: This CL also adds crbug references to other goog flags.
[1] https://w3c.github.io/webrtc-priority/#dom-rtcrtpencodingparameters-networkpriority
Bug: chromium:1315574
Change-Id: I15a0470fa04f55e2534cee0d240eeb03446c2de6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36550}
Motivation: never used.
Follow-up CL of https://webrtc-review.googlesource.com/c/src/+/250680.
Tested on a custom chromium build by making a video call and with
keyboard activity. The expected logs from the APM transient suppressor
sub-module were shown.
Bug: webrtc:11226
Change-Id: I4186994412dd8ba2e71ed8f9dcc9cf8f8e40fbd3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250667
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36300}
This is a safe cleanup change since top-level const applied to
parameters in function declarations (that are not also
definitions) are ignored by the compiler. Hence, such changes do
not change the type of the declared functions and are simply
no-ops.
Bug: webrtc:13610
Change-Id: Ibafb92c45119a6d8bdb6f9109aa8dad6385163a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249086
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35802}
This is a simple check for upcoming changes for media channels to
be able to check if the state on the network thread is consistent.
Bug: webrtc:11992
Change-Id: I8ed2d091ecf3869a66970fc4733aebf209c4ef82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246681
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35706}
Because of this (seemingly simple) change, I had to change the return
type of transport_name from `const std::string&` to `absl::string_view`
to handle the case when there's no transport assigned.
That in turn caused an avalanche of required updates.
Bug: webrtc:12230, webrtc:11993
Change-Id: I16ec6c6a5fc2f5f7c7de572355a3c6ca924bb9d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244084
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35617}
This makes things slightly simpler for the time being as surrounding
code is being refactored. This also removes a PostTask which has the
effect of shrinking the window between the Pending/Complete
notifications slightly since there's no additional async task
for the 'complete' step.
Bug: webrtc:11993
Change-Id: Ia86779b21c6f87301f37d763f89ace722e06e563
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244081
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35609}
Removes a few constructors where similar ones existed.
Removes MediaConfig dependency from MediaChannel and fixes an iwyu.
Bug: none
Change-Id: I9e34a1da0852c3fb21222161fad315e70598db3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242966
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35608}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
The constants are being made private since no new code should use them.
However, the helper functions sill uses "AV1X" internally for backwards
compatibility.
Bug: webrtc:13166
Change-Id: I0a0cd46f31ca70bb7f395c9b1e9cdb202df11f6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236680
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35289}
This field is unused within WebRTC, and doesn't seem to
be essential for any existing customers.
If this works well, it will be deprecated and removed.
Bug: none
Change-Id: I96d7485e4d094abfa6a8c3d1e6855834c13dedd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189680
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35263}
