This change reduces the risk of echo due to noise in the headroom
of the linear filter.
Changes:
- Use shorter delay headroom
- Delay headroom is specified in samples (not blocks)
- No hysteresis limit when delay is reduced
Bug: chromium:119942,webrtc:10341
Change-Id: I708e80e26d541dff8ca04b6da2d346a1d59cbfcb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126420
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27126}
This CL adds enough simulated time to recover the built up delay. This
makes the test less sensitive to small timing changes. This prepares
for further changes in Scenario test framework.
Bug: webrtc:10365
Change-Id: Iddbe6a57e31f17f590004e29221f907321cbb3d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127567
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27107}
This reverts commit c4b391a257ebf85448e58e73a96eb267635b6d6a.
Reason for revert: issue fixed
Original change's description:
> Revert "NetEQ RTP Play: Optionally write output audio file"
>
> This reverts commit 6330818ec8159ee476481ba4a89f884fb3653f3f.
>
> Reason for revert: This breaks api/test/neteq_simulator_factory.cc, which unfortunately was not caught by our bots.
>
> Original change's description:
> > NetEQ RTP Play: Optionally write output audio file
> >
> > This CL makes the output audio file optional to more
> > quickly run neteq_rtpplay when no audio output is needed.
> > The CL also includes necessary adaptations because of pre-existing
> > dependencies (e.g., the output audio file name is used to create
> > the plotting script file names).
> >
> > The command line arguments are retro-compatible - i.e., same behavior
> > when specifying the output audio file and the new flag
> > --output_files_base_name is not used.
> >
> > This CL also includes a test script with which the retro-compatibility
> > has been verified.
> >
> > Bug: webrtc:10337
> > Change-Id: Ie3f301b3b2ed0682fb74426d9cf452396f2b112b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126224
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27067}
>
> TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org
>
> Change-Id: I0c63a8ba9566ef567ee398f571f2a511916fa742
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10337
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127293
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27078}
TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org
Change-Id: Ia7061f7c2d69db61638ad612e82cd429eb49d539
Bug: webrtc:10337
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127540
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27106}
This adds a flag to audioproc_f to generate a custom call order
file from an AEC dump. This file can be used to get more realism
when simulating with wav-files.
Bug: webrtc:10393
Change-Id: I245533d18affaab2f6cef53138332d7d83c71822
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126782
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27104}
This is a partial fix for regression introduced in
https://webrtc-review.googlesource.com/c/src/+/125461
Currently, the OveruseFrameDetector::OnTargetFramerateUpdated is called
only then the encoder is reconfigured, with the default maxFramerate.
Changing it from default 5 to 60, or even 30 made the detector too
sensitive and it caused adaptation down due to CPU overuse even on
powerful machines.
Bug: webrtc:10310, chromium:940466
Change-Id: I7b0eabfc8f9b502e293af1a5b02fc5d4ab468c14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127280
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27094}
This reverts commit 6330818ec8159ee476481ba4a89f884fb3653f3f.
Reason for revert: This breaks api/test/neteq_simulator_factory.cc, which unfortunately was not caught by our bots.
Original change's description:
> NetEQ RTP Play: Optionally write output audio file
>
> This CL makes the output audio file optional to more
> quickly run neteq_rtpplay when no audio output is needed.
> The CL also includes necessary adaptations because of pre-existing
> dependencies (e.g., the output audio file name is used to create
> the plotting script file names).
>
> The command line arguments are retro-compatible - i.e., same behavior
> when specifying the output audio file and the new flag
> --output_files_base_name is not used.
>
> This CL also includes a test script with which the retro-compatibility
> has been verified.
>
> Bug: webrtc:10337
> Change-Id: Ie3f301b3b2ed0682fb74426d9cf452396f2b112b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126224
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27067}
TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org
Change-Id: I0c63a8ba9566ef567ee398f571f2a511916fa742
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10337
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127293
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27078}
This CL adds a field trial that lets us control the size of the initial probes, how we grow the following probes and how big and frequent our ALR probes are.
Bug: webrtc:10394
Change-Id: I6c7783dfada9aaf55cd836dd8991bb7b8ca4993b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126880
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27077}
This required a fairly extensive overhaul:
* Removed locks from the implementation.
The filter/pin architecture does use multiple threads,
but the state is controlled on one and synchronization
is done via flags that don't require locking.
Note though that the baseclasses used a lot of locking, it's unclear why,
but perhaps there are things I'm not aware of. The locking was not done
consistently though, which doesn't seem to have been a problem.
* Change the code to not mix AddRef/Release and use of explicit 'delete'.
* Removed implementations of interfaces we don't need/use.
* Similarly some methods now return E_NOTIMPL.
* Added some utilities to make use of COM interfaces and concepts, easier.
BUG=webrtc:10374
TBR=mbonadei@webrtc.org
Change-Id: Iaedb1157d37ef5d5c75f727dba3d7de75ce22cd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125086
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Christian Fremerey <chfremer@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27072}
This CL makes the output audio file optional to more
quickly run neteq_rtpplay when no audio output is needed.
The CL also includes necessary adaptations because of pre-existing
dependencies (e.g., the output audio file name is used to create
the plotting script file names).
The command line arguments are retro-compatible - i.e., same behavior
when specifying the output audio file and the new flag
--output_files_base_name is not used.
This CL also includes a test script with which the retro-compatibility
has been verified.
Bug: webrtc:10337
Change-Id: Ie3f301b3b2ed0682fb74426d9cf452396f2b112b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126224
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27067}
This is not used in practice as there's functionality on
other levels that serves the same purpose.
Bug: None
Change-Id: I0488dc42459b07607363eba0f2b06f4c50f7cda4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125520
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27061}
This fixes an issue where the time between freezes dropped in
perf tests. This was triggered by resetting and updating the bitrates
immediately if the min allocatable bitrate changed, causing a drop in
target bitrate. With this CL, the change in min bitrate will not take
effect until we get more data.
Bug: chromium:940349
Change-Id: Ia680a5f1cfe71847ef90669987e7b89b240b9524
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126625
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27054}
This is a reland after changes to the downstream project
VP9 screenshare is not used currently, and with these values according
to local testing with screenshare_loopback, we get performance not worse than current vp8 settings for similar uplink and downlink values.
Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126226
Bug: webrtc:10257
Change-Id: Ib21d7678bd839a3c47457515b0d768c0b979ea40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126524
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27040}
This reverts commit 12abf671fd2028ca249e95bdf0456b15a48136f8.
Reason for revert: Breaks downstream project.
Original change's description:
> Reland "Tune vp9 screenshare bitrate and framerate of spatial layers"
>
> This is a reland without any changes as it seems problems with webrtc-in-chrome importer were flakes or
> caused by some issues within chrome codebase.
>
> Tune vp9 screenshare bitrate and framerate of spatial layers
>
> VP9 screenshare is not used currently, and with these values according
> to local testing with screenshare_loopback, we get performance not worse than current vp8 settings for similar uplink and downlink values.
>
> Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126226
>
> Bug: webrtc:10257
> Change-Id: Ie819d8bbab4f14877daac733d162e5ae7ebf2a8e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126460
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27036}
TBR=ilnik@webrtc.org,jeroendb@webrtc.org,kron@webrtc.org
Change-Id: I9ad9017b054213f931b3b39c641060d35565f17d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10257
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126523
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27037}
This is a reland without any changes as it seems problems with webrtc-in-chrome importer were flakes or
caused by some issues within chrome codebase.
Tune vp9 screenshare bitrate and framerate of spatial layers
VP9 screenshare is not used currently, and with these values according
to local testing with screenshare_loopback, we get performance not worse than current vp8 settings for similar uplink and downlink values.
Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126226
Bug: webrtc:10257
Change-Id: Ie819d8bbab4f14877daac733d162e5ae7ebf2a8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126460
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27036}
This reverts commit aaf3cb3adb618a9f9b14931876b9050201396bee.
Reason for revert: Chrome importer consitently failing after this change
Original change's description:
> Tune vp9 screenshare bitrate and framerate of spatial layers
>
> VP9 screenshare is not used currently, and with these values according
> to local testing with screenshare_loopback, we get performance not worse
> than current vp8 settings for similar uplink and downlink values.
>
> Bug: webrtc:10257
> Change-Id: Icabac04fbd3d616412bbae59291a1fc026d0a504
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126226
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27023}
TBR=ilnik@webrtc.org,kron@webrtc.org
Change-Id: I1ef1eeec8fe87a7662a354ef6362b7d463b2bb4c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10257
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126340
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Commit-Queue: Jeroen de Borst <jeroendb@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27027}
VP9 screenshare is not used currently, and with these values according
to local testing with screenshare_loopback, we get performance not worse
than current vp8 settings for similar uplink and downlink values.
Bug: webrtc:10257
Change-Id: Icabac04fbd3d616412bbae59291a1fc026d0a504
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126226
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27023}
This cl deprecates the FrameType enum, and adds aliases AudioFrameType
and VideoFrameType.
After downstream usage is updated, the enums will be separated
and be moved out of common_types.h.
Bug: webrtc:6883
Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27011}
