Only allow the application of min input volume if the applied input
volume is above zero. To implement this, add a member variable to
store the applied input volume. Rename the related setter to reflect
its new functionality.
Bug: webrtc:7494
Change-Id: Ia70d5cb4dfd972aad9ef2663a81884f3e5cb0758
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287680
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38878}
Add an override to TurnPort for SendBindingErrorResponse to check
if a matching connection object exists before continuing. This is
needed to match with the check in `TurnPort::DispatchPacket` whereby
we forward calls to `Port` when no matching connections exist.
Bug: chromium:1395625
Change-Id: Idf1f898c2a93de6bc2832268db1cadd52cae23a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287223
Reviewed-by: Sameer Vijaykar <samvi@google.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38871}
At the same time, proper names of some parameters are refactored in SimulcastEncoderAdapter.
Bug: None
Change-Id: Ia036e3f362d1394e90aa26b79953c1ffe75e2fe0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284961
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Chunbo Hua <chunbo.hua@intel.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38870}
Follow-up of https://webrtc-review.googlesource.com/c/src/+/287221,
instead of asking GTest for the test suite and the test name, let's
propagate the test case passed by the user of PCLF.
Bug: b/237982523, webrtc:14757
Change-Id: Ia2a6ed4781f8c53c25b0006b8c7483e08ecead26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287126
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38867}
increasing the line coverage to 100%.
Note that code coverage is still a very basic metric and
the tests still do not enforce the length of the attributes
nor their token-char-ness defined in
https://www.rfc-editor.org/rfc/rfc8830.html#section-2
BUG=webrtc:14745
Change-Id: If5de5e1f8c32f7dae029916c6cd0a4f2b094f672
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286900
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38864}
Pass the correct number of channels needed by the AGC2 input volume
controller. This change doesn't affect the adaptive digital
controller which reads the number of channel from the passed audio
buffer instance for each processed frame.
Note that the `AdaptiveDigitalGainController::Initialize()` impl
was removed in [1], but that CL didn't remove the declaration (done
in this CL).
[1] https://webrtc-review.googlesource.com/c/src/+/287222/5/modules/audio_processing/agc2/adaptive_digital_gain_controller.cc#105
Bug: webrtc:7494
Change-Id: I07369ab4025a251b25c716cf618e4222fdb60fc8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287320
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38863}
When running Gtest-based tests, we use the WebRtcUnitTestDelegate
to call Gtest from the iOS application.
However, if tests take too long, iOS terminates that application.
The test execution code (ios/build/bots/scripts) retries up to three times,
but that functionality doesn't seem to work lately.
Because of this, module_unittests are failing at a fairly high rate.
Instead, use GoogleTestRunner to let XCTest run Gtest-based tests.
This is enabled with the --enable-run-ios-unittests-with-xctest flag,
which is passed when using the --xctest flag in ios/build/bots/scripts/run.py.
Existing XCTest-based tests (eg. sdk_unittest) are not affected
as the --xcode-parallelization flag takes precedence over --xctest.
Bug: None
Change-Id: Ib7f8a6d24f6b25444a47e3a83c0edbe96318be46
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287180
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#38859}
This metadata key is temporary, as explained in bugs.webrtc.org/14757,
this information will be at some point directly accessible via the
webrtc.test_metrics.Metric.test_case field.
Bug: b/237982523, webrtc:14757
Change-Id: Ie77875a33db5961f8a5572bd1b7066ad8ba17291
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287221
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38858}
Add AGC2 digital adaptive config parameters in the field trial
"WebRTC-Audio-InputVolumeControllerExperiment". Rename it as
"WebRTC-Audio-GainController2" to reflect that the override now adjusts
the parameters for both input volume controller and adaptive digital
controller.
Bug: webrtc:7494
Change-Id: Ifbc1b8be76cf23b0b6b74b22b5167a45972cab38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286880
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38855}
Rename MonoInputVolumeController member input_volume_ to reflect its
use to store the most recent input volume recommendation.
Rename the remaining variables named as manager in the unit tests.
Bug: webrtc:7494
Change-Id: I31ffdc131c98061ef2b36f98b685c5182b3c6861
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287123
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38854}
The `WebRTC.Audio.AgcSetLevel` name is misleading and the histogram
is logged for each channel - but the input volume is one for all the
channels.
Changes:
- `WebRTC.Audio.Apm.RecommendedInputVolume.OnChangeToMatchTarget`
is the new name
- Now available not only in `AgcManagerDirect` (AGC1), but also in
`InputVolumeController` (AGC2)
- Logged once and not for each channel
- Also add the following AGC implementation agnostic histograms
- `WebRTC.Audio.Apm.AppliedInputVolume.OnChange`
- `WebRTC.Audio.Apm.RecommendedInputVolume.OnChange`
- Fix `SpeechSamplesReader::Feed()` in the unit tests, which did
not set the applied input volume and apply the recommended one
The histogram definitions are updated in crrev.com/c/4087426.
Bug: webrtc:7494
Change-Id: I03c5dfb08165805215ca2c4bb6509b16de8d68da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287081
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38852}
This will enable loss based bwe v2 by default. The default params were used in Chrome experiment and got positive result. Remove some tests in goog_cc, which are for loss based bwe v1.
Bug: webrtc:12707
Change-Id: Ice126a128f6e8cea8b861f879d09e390ee69e521
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285740
Commit-Queue: Diep Bui <diepbp@google.com>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38851}
These histograms have expired and have no owner.
Remove to clean up the code and save memory.
Fixed: chromium:1117100
Change-Id: I24a009d8e432109c1d62c4a3a16eff5cd21c8541
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286660
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38850}
This MidCallProbingRampupTriggeredByUpdatedBitrateConstraints blocks https://webrtc-review.googlesource.com/c/src/+/285740 submitting. I was able to complete the test locally, but cannot manage to do so remotely.
Bug: none
Change-Id: I75979af25552b4a31487a26e40857a713299e0eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287022
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Diep Bui <diepbp@google.com>
Cr-Commit-Position: refs/heads/main@{#38848}
This is in the webrtc-stats spec at
https://www.w3.org/TR/webrtc-stats/#dom-rtcoutboundrtpstreamstats-scalabilitymode.
This adds the scalability mode to CodecSpecificInfo which is used to
plumb the modes for each simulcast layer.
TBR=orphis@webrtc.org
Tested: Compiled into Chrome and confirmed the scalability mode set for AV1, VP9, VP8 and H264 software encoders in chrome://webrtc-internals.
Bug: webrtc:14730
Change-Id: I71ceba8f6485a4f4a73e0856031b8d5f16f913f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285085
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38847}
This allows callers to differentiate on whether they need the
channel for sending or receiving purposes.
Note: This CL is incomplete, in that many places cast the pointers
to the concrete subclasses "VideoMediaChannel" and "AudioMediaChannel", which are not split into sending and receiving APIs.
The long term goal is to make two MediaChannel-like class APIs, with distinct implementations, and let the RtpSender and RtpReceiver manage those objects, rather than keeping them in the RtpTransceiver.
Bug: webrtc:13931
Change-Id: I8d56defe2287bd6552b71571cc6a5ec842927fa4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287040
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38844}
The problem occurs when more than one call is made to the method RunToNextGetAudio. Except for the first call to that method, the clock was not properly updated on the first iteration of the inner loop in RunToNextGetAudio.
Pair: lionelk@webrtc.org
Bug: webrtc:14735
Change-Id: If6fb5c2c700b0f715f626fedf95672a56b04ab12
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285942
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38843}
Now that `InputVolumeController` is finalized, it's time to
consolidate AGC2.
Main changes:
- Remove `AdaptiveDigitalGainController`: it's too simple to justify
a dedicated class and some components of it are also used by
`InputVolumeController`
- Remove unwanted temporal dependency: make `InputVolumeController`
adapt the volume based on the current speech level estimation and
not on the estimation from the previous frame
Tested: AGC2 adaptive digital bit-exactness verified
Bug: webrtc:7494
Change-Id: I175c2741cafc52be81794219c996a3824c3bbf5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280560
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38841}
https://www.rfc-editor.org/rfc/rfc8830.html#section-3.2.2
says
Check if a MediaStream with the same WebIDL "id"
attribute already exists. If not, create it.
Ignoring duplicates here satisfies this and brings the behavior
closer to Firefox:
https://github.com/w3c/webrtc-pc/issues/2803
Also make tests use a std::string for the sdp input string.
BUG=webrtc:14745
Change-Id: Iccaabc08d865b779416f6ba4d2dfd5cff04133f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286422
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38840}
Regardless of the APM config, the transient suppressor (TS) submodule
won't be created if the `WebRTC-ApmTransientSuppressorKillSwitch`
field trial, disabled by default, is enabled.
Bug: webrtc:13663
Change-Id: Ic1ef9aa57c728296d671d4ef253630c581a86610
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286382
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38839}
Add a legend when on the python plots generated with neteq_rtpplay.
Bug: None
Change-Id: I4299858bb9e8e59564c824c99272e4fabc610162
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286840
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38838}
removing the temporary limitation to 32 characters
since metrics suggests this is now fixed.
Metrics removal:
https://chromium-review.googlesource.com/c/chromium/src/+/4079261
BUG=webrtc:12517,chromium:1375724
Change-Id: I11bec89463044afa99eeef2b3ecbe108eaa5c954
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286620
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38836}
https://crrev.com/c/2738677 added a variable to
base/allocator/partition_allocator/build_overrides/partition_alloc.gni
and this change prevents chromium roll.
Instead of adding a variable to WebRTC's partition_alloc.gni,
import that file. This will avoid repeating these operations in the
future.
Bug: None
Change-Id: I8ad2e8900d5ca7828cf415ecf7933c8eb1d5160a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286201
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38834}
Fixing errors like this:
Evaluation of CheckChangeOnCommit failed: can only concatenate str (not "list") to str, Traceback (most recent call last):
...
File "/path/to/webrtc/src/infra/specs/PRESUBMIT.py", line 31, in CheckPatchFormatted
results.append(output_api.PresubmitError('Error calling "' + cmd + '"'))
TypeError: can only concatenate str (not "list") to str
Bug: None
Change-Id: Ia0b1c7a80a2752934c02d932a9206114769bcaa1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286547
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38831}
This is required to compile the default target in Android bots.
Bug: webrtc:14743
Change-Id: Ib8248e3d874b45eb59283f9503e07eadcd53bad7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286545
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38830}
Purposes of this refactoring:
1. Add functionality for reading a specified frame.
2. Change resolution and frame rate on per-frame basis.
Both features are needed for https://webrtc-review.googlesource.com/c/src/+/283525
Bug: b/261160916
Change-Id: I6d60e62dbc3913c43b5c1b491690f5cb4a8632dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285483
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38829}
