A recent refactoring introduced a subtle difference in how encoded
frame timing is measured. See
https://webrtc-review.googlesource.com/c/src/+/124122/14/video/video_stream_encoder.cc#b1278
After that change, the encode was considered done after encoding was
done. The old behavior included the time needed to call stats and
the video sink, which might include video quality tests related tasks
and RTP packetization.
In order to preserve the old behavior I'm moving timestamping to after
packetization again.
Note that the timing frame info still has a separate timestamp that
does explicitly measure encode time. This is used by the experimental
new overuse detector, so the effect of this change will be transient
anyhow.
Bug: chromium:941457, webrtc:10164
Change-Id: Ia990a1ceaeaf2c45d5df2a32d4f017cdb08e3c55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127569
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27108}
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}
* Removing unused return values.
* Using TaskQueueForTest to do blocking calls.
* Improving naming.
This prepares for future work to run scenario tests in simulated time.
Bug: webrtc:10365
Change-Id: I2c100e9c20f4b85e85d7b455ea01944f6a14e08f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127561
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27105}
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 reland of 184f6d5d75c198cb7b70b8f9b75e0b5096c6e577.
Incorrect build dependencies in downstream tests have been fixed,
and an initialization bug in this CL has also been fixed.
Original change's description:
> Change clip_name -> clip_path in VideoQualityTestFixture::Params::Video.
>
> This allows external users of this test fixture to specify a custom
> path, rather than just a custom file name.
>
> Bug: webrtc:10349
> Change-Id: I84e886c8bc28583017ce9ed7b9e7ee6a8e95730f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126227
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27033}
TBR: kwiberg@webrtc.org
Bug: webrtc:10349
Change-Id: I0ec9dd26cd96c3db8ac8482893a26e62a1b1eefc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127181
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27102}
Specifying relative path like "~" somehow doesn't work in some cases.
To pass an absolute path some parameter rewriting is required.
Also, as the stream gets recreated several times at the beginning of the
call, append the time to the filename to make them unique.
Bug: none
Change-Id: I1c914f9081adb4ac5c34584d96e542e8a863547b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126700
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27101}
This CL adds a mode to simulate roughly what GoogCC could have been
doing during the recording of an rtc event log by using the logged
events as input to GoogCC and visualizing the resulting target rate.
This is similar to the existing simulated_sendside_bwe mode, but uses
the new NetworkControllerInterface to ensure more reliable GoogCC
simulation.
Bug: None
Change-Id: I57894aa666151efc8405407d928b5257fb9b7d61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123924
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27095}
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}
MediaChannel accepted the RtpPacket buffers through non-const pointer.
This is both unclear and introduces questions regarding if the buffer is
actually copied or not.
This change modifies the method to accept by value to reduce ambiguity.
Usage of the non-const data() method which could potentially copy the
buffer contents is also reduced in favor of cdata() which never copies.
Bug: None
Change-Id: I3b2daef0d31cb6aacceb46c86da3a40ce836242b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127340
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27090}
This simple fuzzer is intended to detect potential issues in the field trial
parsing code. Since these can be set by the browser it is better to have some
fuzzing coverage around this area.
Bug: webrtc:10395
Change-Id: I1b8b859d2107a0bc99cb7520cf0ef96f3d110547
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127121
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27087}
Using simulated time implementation of TaskQueue with TaskQueueForTest.
Bug: webrtc:10365
Change-Id: I9f853133aa9fcd5858a5e2e1e0f47ab1a6a5ca18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127292
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27081}
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}
Implicit bool conversions behave wierdly in a bunch of cases, so let's make it explicit.
Bug: None
Change-Id: I15933e90d57c57218eed9608407aace5a640a6ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127284
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27076}
Introduce a handle for route created with network emulation layer,
that can be used to remove it in future properly.
Bug: webrtc:10138
Change-Id: I9fb847caeee24333bafb328727711af005b09224
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127283
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27074}
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}
