That means it does not have to be set on every update of StreamsConfig.
BUG=webrtc:9586
Change-Id: I6a348160e209042857c4475323466e2aa92adef8
Reviewed-on: https://webrtc-review.googlesource.com/c/116690
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26184}
This fixes a bug where we sometimes extract an Opus CNG packet and the packet after, even though there was big timestamp gap between the packets, which causes expansion during the next GetAudio calls.
Change-Id: I2409ac08df58afc496f74b91981657b7206e8bb1
Bug: webrtc:10167
Reviewed-on: https://webrtc-review.googlesource.com/c/115419
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26179}
That is, the payload packetization overhead (eg. vp8 payload header),
not the RTP headers, extensions, etc.
The encoder and pacer both look at payload rate, but are currently not
aware of the bytes that are added in between them.
Bug: webrtc:10155
Change-Id: I4cdb04849d762360374d47a496983c8c6df191d2
Reviewed-on: https://webrtc-review.googlesource.com/c/115410
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26163}
While this class is deprecated, it's needed as a stop-gap solution.
Other methods to configure the max probe rate all effect the current
estimate and/or trigger new probes to be sent, and we need a way to
configure the max without affecting other behavior.
Bug: webrtc:10070
Change-Id: I2b0ba2fef42d0bab6e5ea7f7c921681557802b4b
Reviewed-on: https://webrtc-review.googlesource.com/c/114880
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26162}
This is a reland of 0cc42d47389c039c57e47d7ec0c76b97e2da2b0b
Original change's description:
> Reland "Default to dlopening the PipeWire."
>
> This is a reland of a099877d8946eb942046ca1295cc142e4fa7ea6f
>
> Original change's description:
> > Reland "Default to dlopening the PipeWire."
> >
> > This is a reland of a13be019017449c57f48203d0fb778f34f7553a7
> >
> > Original change's description:
> > > Default to dlopening the PipeWire.
> > >
> > > Reuse the existing infra from Chromium to do that. Additionally the
> > > target_gen_dir needs to the added to the include directories, otherwise
> > > the Chromium build will fail as it won't find the generated stubs. Also the
> > > pw_properties_new() was replaced with pw_properties_new_string() as it doesn't
> > > require a variadic parameter because the //tools/generate_stubs/generate_stubs.py
> > > doesn't work with them correctly. With all these changes in place the PipeWire
> > > support is enabled when compiling on Linux.
> > >
> > > Bug: chromium:682122
> > > Change-Id: I3bbc5efaecd9a08e20cbcf998b2cb534224eae7d
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/111081
> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > > Reviewed-by: Brave Yao <braveyao@webrtc.org>
> > > Commit-Queue: Tomáš Popela <tomas.popela@gmail.com>
> > > Cr-Commit-Position: refs/heads/master@{#25720}
> >
> > Bug: chromium:682122
> > Change-Id: I3cca3d4d961dc7a088346c8fd3c970d3dfde3b79
> > Reviewed-on: https://webrtc-review.googlesource.com/c/113040
> > Reviewed-by: Weiyong Yao <braveyao@chromium.org>
> > Reviewed-by: Brave Yao <braveyao@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> > Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25981}
>
> Bug: chromium:682122
> Change-Id: Ief26c93069f946f981340664a267fcb412229285
> Reviewed-on: https://webrtc-review.googlesource.com/c/114163
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Brave Yao <braveyao@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26004}
Bug: chromium:682122
Change-Id: I0a4ea7b39be5970f26df6dbc3e437dd63cdb8708
Reviewed-on: https://webrtc-review.googlesource.com/c/116280
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26154}
This CL removes MediaOptmization and folds some of its functionality
into VideoStreamEncoder.
The FPS tracking is now handled by a RateStatistics instance. Frame
dropping is still handled by FrameDropper. Both of these now live
directly in VideoStreamEncoder.
There is no intended change in behavior from this CL, but due to a new
way of measuring frame rate, some minor perf changes can be expected.
A small change in behavior is that OnBitrateUpdated is now called
directly rather than on the next frame. Since both encoding frame and
setting rate allocations happen on the encoder worker thread, there's
really no reason to cache bitrates and wait until the next frame.
An edge case though is that if a new bitrate is set before the first
frame, we must remember that bitrate and then apply it after the video
bitrate allocator has been first created.
In addition to existing unit tests, manual tests have been used to
confirm that frame dropping works as expected with misbehaving encoders.
Bug: webrtc:10164
Change-Id: I7ee9c8d3c4f2bcf23c8c420310b05a4d35d94744
Reviewed-on: https://webrtc-review.googlesource.com/c/115620
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26147}
This updates some tests to use AudioProcesing::Config() and
AudioProcessing::GetStatistics() instead.
Some tests are left with voice_detection() because
a) not all tests make sense to run both APIs in parallel, and
b) we want test coverage of the old VoiceDetection until it is removed.
Bug: webrtc:9947
Change-Id: Ifb21a1e6e931d7ad3c3a4e38f5cc4f146da3c9a3
Reviewed-on: https://webrtc-review.googlesource.com/c/116160
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26134}
When using WebRTC in iOS this Warning is shown for every single call even if stereo is not being used at all.
Change-Id: I0cc71620b9deb0692544101d78c0801968edbb26
Bug: webrtc:10146
Change-Id: I0cc71620b9deb0692544101d78c0801968edbb26
Reviewed-on: https://webrtc-review.googlesource.com/c/85283
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26133}
Fix a bug introduced in (https://webrtc-review.googlesource.com/c/src/+/105102) that causes cwnd pushback only active when there is network condition changes.
Bug: None
Change-Id: I8164d5663304ce2e445db09205f706011ff7d784
Reviewed-on: https://webrtc-review.googlesource.com/c/115945
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26122}
The ANA frame length controller requires the provided frame lengths supported by the encoder to be ordered. A data structural guarantee of such was in an earlier version but was accidentally removed since https://codereview.webrtc.org/2429503002. This CL uses std::set to ensure that again.
Change-Id: Ia197dbf6a34f02506e81c9f49d6cd60e4cdacef4
BUG: webrtc:6303
Reviewed-on: https://webrtc-review.googlesource.com/c/115946
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26119}
This adds a second (!) VoiceDetection instance in APM, activated via webrtc::AudioProcessing::Config and which reports its values in the webrtc::AudioProcessingStats struct.
The alternative is to reuse the existing instance, but that would require adding a proxy interface returned by AudioProcessing::voice_detection() to update the internal config of AudioProcessingImpl when calling voice_detection()->Enable().
Complexity-wise, no reasonable client will enable both interfaces simultaneously, so the footprint is negligible.
Bug: webrtc:9947
Change-Id: I7d8e28b9bf06abab8f9c6822424bdb9d803b987d
Reviewed-on: https://webrtc-review.googlesource.com/c/115243
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26101}
This reverts commit d1208c26b1cdb536fdec942207033711101d5d26.
Reason for revert: This cl causes the crashing issue as in
chromium:916961 at starting desktop capture on Windows.
Original change's description:
> Desktop capturer: Add OnDisplayChanged callback
>
> This adds support for a new DesktopCapturer::Callback method
> OnDisplayChanged that is sent at the start of a desktop capture
> session and whenever the display geometry changes.
>
> This cl adds the basic structure to call this api at the start
> of the capture session. Currently Windows only.
>
> A follow-up cl will add support to call this whenever the display
> geometry changes.
>
> Bug: webrtc:10122, chromium:915411
> Change-Id: Ie7283be5992454180daab1a60f58a3b2efdfed56
> Reviewed-on: https://webrtc-review.googlesource.com/c/114020
> Commit-Queue: Gary Kacmarcik <garykac@chromium.org>
> Reviewed-by: Brave Yao <braveyao@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26053}
TBR=jamiewalch@chromium.org,braveyao@webrtc.org,braveyao@chromium.org,garykac@chromium.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10122, chromium:915411, chromium:916961
Change-Id: Id0471e01bb90bb5accdf58262ae2b130cf343ecd
Reviewed-on: https://webrtc-review.googlesource.com/c/115433
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26095}
Rids can now be sent using rtp_sender.
Hooking up the rid values in the voice and video engine is still WIP.
Bug: webrtc:10074
Change-Id: I245c7ecb23b67fc0ba65caaa5dbb4fcfd60c81bb
Reviewed-on: https://webrtc-review.googlesource.com/c/114505
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26092}
This CL guards against null pointer dereference, as caught by
clang static analyzer [1].
It also removes a useless field initialization, which happened
to trigger a false positive from said analyser.
[1] https://chromium.googlesource.com/chromium/src/+/HEAD/docs/clang_static_analyzer.md
Bug: webrtc:8793
Bug: webrtc:9855
Change-Id: Ia0fee24395eb2df16b526bbdffa5da6275b0909a
Reviewed-on: https://webrtc-review.googlesource.com/c/115044
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#26091}
The new controller behaves mostly like before, but increases the target
rate on timer update rather than when feedback is received. This makes
the behavior easier to predict. It also uses a duration parameter to
track the increase, removing the meed for the minimum rate increase
constants that exists in the previous solution.
Bug: webrtc:9718
Change-Id: Iae31a9ba2d6474a8236f8eb72f86ff434f1d1fc6
Reviewed-on: https://webrtc-review.googlesource.com/c/114681
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26088}
Just ignoring single_packet_reduction_len is wrong, because if the
fragment is put in a single packet it might still be the first or the
last packet in the whole sequence.
Bug: none
Change-Id: I4a2fbebe1d49cbef9298bb32d9cecaa617e4dfc3
Reviewed-on: https://webrtc-review.googlesource.com/c/115403
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26084}
The BBR controller can still be injected, but the trials
will no longer work. This reduces the binary size.
Bug: webrtc:8415
Change-Id: I2c32c414d08ef0cc16bfd72651535a755cde9916
Reviewed-on: https://webrtc-review.googlesource.com/c/114120
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26077}
Set spatial index of assembled VP9 picture equal to spatial index of
its top spatial layer frame.
Bug: webrtc:10151
Change-Id: Iae40505864b14b01cc6787f8da99a9e3fe283956
Reviewed-on: https://webrtc-review.googlesource.com/c/115280
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26075}
This adds support for a new DesktopCapturer::Callback method
OnDisplayChanged that is sent at the start of a desktop capture
session and whenever the display geometry changes.
This cl adds the basic structure to call this api at the start
of the capture session. Currently Windows only.
A follow-up cl will add support to call this whenever the display
geometry changes.
Bug: webrtc:10122, chromium:915411
Change-Id: Ie7283be5992454180daab1a60f58a3b2efdfed56
Reviewed-on: https://webrtc-review.googlesource.com/c/114020
Commit-Queue: Gary Kacmarcik <garykac@chromium.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26053}
This will be used to calculate a lower bound for the round trip time in
a later CL.
Bug: webrtc:9718
Change-Id: I0a1d22045961fe6bd343d1d6ce9b36490b036bb1
Reviewed-on: https://webrtc-review.googlesource.com/c/114680
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26050}
This replaces the current usage of AudioProcessing::level_estimator()
in that test.
The unit tests that specifically test the level_estimator API are left
in place, until the level_estimator API itself is removed.
Bug: webrtc:9947
Change-Id: I73301c1478d2c9763bb49598a692142229102876
Reviewed-on: https://webrtc-review.googlesource.com/c/114550
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26049}
If we're in ALR, the acked rate is going to be significantly lower than
the current estimate for the link capacity. If we need to back off in
this situation (usually caused by latency spikes), this CL makes us back
off relative to current estimate if. We then immediately send a new
probe just in case the network did actually change.
All of this is behind experiment flags for now.
Bug: webrtc:10144
Change-Id: I062a259c36417eea2211d44592ef7fc979aa22b7
Reviewed-on: https://webrtc-review.googlesource.com/c/113880
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26045}
Previous attempt: https://codereview.webrtc.org/1882733006/. There
might be some benefit of having dummy encoder/decoder available in
video_loopback.
Bug: webrtc:5791
Change-Id: Iec316296754178c92b18dd3cf92f67ce6aed9439
Reviewed-on: https://webrtc-review.googlesource.com/c/112596
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26043}
For somewhat similar funtionality, GoogCcNetworkController can
be used via GoogCcNetworkControllerFactory.
Bug: webrtc:9586
Change-Id: I298050184513f50c1b9ef5c21b8c9b7a6ca46fd5
Reviewed-on: https://webrtc-review.googlesource.com/c/114543
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26040}
