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}
It's routed to the old jitter buffer, which is instantiated but
unused.
Bug: webrtc:7408
Change-Id: I8fe2d8a2bfa6c2119945cc294d17982af4f830cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127541
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27100}
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}
This is a reland of 1b871d07532c25d2f27e4db192cb9ce2229b1cee
Original change's description:
> Replacing rtc::Thread with task queue for TestAudioDeviceModule.
>
> This prepares for running it in simulated time.
>
> TBR=henrika@webrtc.org
>
> Bug: webrtc:10465
> Change-Id: I9b29b8af9aeba7f0c8209ca77294a63d8068ff1a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126481
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27083}
TBR=henrika@webrtc.org
Bug: webrtc:10465
Change-Id: Icda8043fb5b1156129bc3b706bf8f190782b0921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127520
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27093}
This prepares from removing the overload in a followup CL.
Bug: webrtc:10365
Change-Id: I80db16e7d37944e3dc7d2799bbf45ef8f439a22c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126860
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27091}
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}
StringToNumber is directly used in parsing the SDP so it should be fuzzed.
Bug: webrtc:10395
Change-Id: I85b520fbefd34d3dba49950c5ff297b482c572b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127123
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27089}
The generic video depacketizer was missed in the initial fuzzing pass.
Bug: webrtc:10395
Change-Id: I166f27fc5897a2eafe38dad8e074834fefcc330e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127041
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27088}
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}
This prepares for running it in simulated time.
TBR=henrika@webrtc.org
Bug: webrtc:10465
Change-Id: I9b29b8af9aeba7f0c8209ca77294a63d8068ff1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126481
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27083}
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}
Also fix minor issues in this class.
Bug: webrtc:10138
Change-Id: Icb3ec7f6296c34da260e701ec51d7b87ce62a4d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127281
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27073}
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 makes the error messages more informative.
Bug: webrtc:10398
Change-Id: I161aa4bfa53f60d06500d42c883f18add4c58531
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127200
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#27071}
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}
The fuzzer times out on too long inputs.
This CL limits tests to 400 000 bytes, ~ 12 seconds of 8 kHz float audio.
Bug: chromium:940209
Change-Id: I86b772f9d8989a8b129d933d25ece3631a6a365f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126780
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27059}
Call DataChannelObserver::OnBufferedAmountChange on each successful send.
Previously, the observer would get notified of buffered amount changes only when
queued send data is consumed. Data gets queued only if it cannot be sent right
away. According to the WebRTC standard[1], bufferedamount should be increased
before each sent and decreased after each successful sent. Update implementation
to be standard compliant.
Design doc: http://doc/1lorHBn-GMn5U0T0RQANxrsW0pXhw8XGZM-xZyVUOW90
[1] https://w3c.github.io/webrtc-pc/#dom-datachannel-bufferedamount
Bug: chromium:878682
Change-Id: Ife009d30c4a18dced9a54cf600a445bb1f02561d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123237
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27057}
