Start introducing ArrayView to AudioFrame and code that flows down
from there. In this first step:
* Add `data_view()` that returns a read-only ArrayView for the
audio buffer. When AudioFrame is not initialized however, data_view()
will return a nullptr whereas the current data() method never returns
nullptr.
* Add `mutable_data()` that requires two arguments for properly setting
the samples per channel and number of channels that's required for
accurately reserving the returned mutable ArrayView.
A notable behavior change is that if the requested number of channels
is larger than supported or the calculated buffer size is too large,
the function will trigger a check.
* Add TODOs for following work.
Bug: chromium:335805780
Change-Id: I2937de800422589ebe6a3840b3caadf3d9ff8b00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347982
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42202}
This hard-codes the behavior to mode 3 with a threshold of 0.5 like was
already done by FetchPreEchoConfiguration.
Bug: webrtc:14205
Change-Id: I48d47a77c9df0001460788b504524203417f9647
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/345483
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42015}
Replace num_proc_channels() with num_output_channels() in
GainController2. The number of channels is only used in
InputVolumeController.
Bug: webrtc:7494
Change-Id: I6b3f66980a518401fefab304e18c9910eee28d4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338922
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41717}
rtc::TaskQueue is a wrapper of TaskQueueBase providing no extra functionality in this case
Bug: webrtc:14169
Change-Id: I5eb27a5dbb16f6097a9c71c2633c807808e50c05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333800
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41501}
Reasons:
- the code is no longer used in Chrome
- it is conceptually weird for WebRTC to have JSON parsing in its API
- there are concerns around the reliability of the underlying JSON library
Additionally, this CL removes the rtc_json "poisonous" attribute: the scheme is incompatible and redundant with testonly.
Bug: webrtc:1493351
Change-Id: I0b621b0e3f183df7315919d9c89242fbe387928f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325062
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41014}
rtc::strcpyn second param should be the size of the destination buffer,
not the size of the source string. The result is that the final character
(usually a trailing directory path separator) is lost during the copy.
This has been masked because FormFileName helpfully adds a trailing path
separator if one is missing.
BUG=webrtc:15441
Change-Id: I992e69cad86a7e8bc2057ec629063f34c75fe75f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317502
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40736}
This CL addresses the review comments for
https://webrtc-review.googlesource.com/c/src/+/261221
in the downstream cherry-pick: https://crrev.com/c/4660950.
* Always use size_t{} for casting.
* Remove unneeded size_t casts.
* Avoid using __x as it is reserved for the compiler.
Bug: b:217226507
Change-Id: I13c57cb69d7db066ac9a6dbd15b7f6de54abb613
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311360
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Li-Yu Yu <aaronyu@google.com>
Cr-Commit-Position: refs/heads/main@{#40395}
Several files refer to symbols declared in headers not explicitly
included. This compiles now because libc++ tranitively includes these
headers via other libc++ headers; however, these transitive includes are
not guaranteed to exist and in Chrome, will no longer exist once libc++
is compiled with modules.
Bug: chromium:543704
Change-Id: I638bb02df3d050a48345248e80aebd2dd60956c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295762
Auto-Submit: Alan Zhao <ayzhao@google.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39448}
The code has been running in Chrome since 2020 and ChromeOS since 2022 (https://crrev.com/c/3452884) without issues.
Bug: webrtc:11803
Change-Id: I0c572d362b1f52b4591c7790e11a87c1a1ad1a79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293342
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39316}
- Lowering the energy threshold for updating the accumulated error.
- Not using the pre-echo estimate in the initial frames when the matched filters have been recently initialized.
- Slight speed up for the increases in the accumulated error.
- Not periodically resetting the accumulated error.
Bug: webrtc:14205
Change-Id: Ic337332e263b27d7a3aba0ab4b371517780f9c90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291320
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39175}
Based on offline testing; needed to allow input volume adaptations
more frequently. Note that if the estimated speech level falls in
the target range, the recommended input volume won't change and
hence the new lower threshold won't necessarily increase the
number of adjustments.
Bug: webrtc:7494
Change-Id: Iabb501c188da238ea7b7137175bcfe09239c90a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291110
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39161}
When the `WebRTC-Audio-GainController2` field trial is used, the
initial APM configuration is adjusted depending on its original
values and the field trial parameters.
This CL fixes two cases when the code crashes:
1. when, in the initial APM config, AGC1 is enabled, AGC2 is
disabled and TS is enabled
2. when the initial APM sample rate is different from the
capture one and the VAD APM sub-module is not re-initialized
This CL also improves the unit tests coverage and it has been
tested offline to check that the VAD sub-module is created only
when expected and that AGC2 uses its internal VAD when expected.
The tests ran on a few Wav files with different sample rates and
one AEC dump and on 16 different APM and field trial
configurations.
Bug: chromium:1407341, b/265112132
Change-Id: I7cc267ea81cb02be92c1f37f273b7ae93b6e4634
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290988
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39118}
- Test behavior with no input volume controller
- Test behavior with startup volume higher than the minimum
input volume
Bug: webrtc:7494
Change-Id: I36d48e2bd277b8a71eb6fbb0272c26c7176b3d5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286380
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38932}
Make sure that the input volume controller implementations exhibit
the adaptive behavior regardless of the sample rate and the number
of channels. The newly added tests check that:
- a downward adjustment takes place with clipping input
- an upward adjustment takes place with a too low speech level
- a downward adjustment takes place with a too high speech level
Bug: webrtc:14761
Change-Id: I1795e74c5f219e15107e928ebaca2bfa75214526
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287301
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38930}
In order to experiment with AGC2 and TS at the same time, 3 field
trials are removed and merged into `WebRTC-Audio-GainController2`,
which is existing.
New parameters for the `WebRTC-Audio-GainController2` field trial:
- `switch_to_agc2`: true by default; when true, the gain control
switches to AGC2 (both for the input volume and for the adaptive
digital gain);
- `min_input_volume`: minimum input volume enforced by the input
volume controller when the applied input volume is not zero;
- `disallow_transient_suppressor_usage`: when true, TS is never
created.
Removed field trials:
- `WebRTC-Audio-Agc2-MinInputVolume`: now a parameter of
`WebRTC-Audio-GainController2`;
- `WebRTC-ApmTransientSuppressorKillSwitch`: now a parameter of
`WebRTC-Audio-GainController2`;
- `WebRTC-Audio-TransientSuppressorVadMode`: automatically inferred
from `WebRTC-Audio-GainController2`.
Bug: webrtc:7494
Change-Id: I452798c0862d71f9adae6d163fe841df05ca44d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287861
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38890}
In InputVolumeController, rename AnalyzePreProcess() and Process() to
reflect their use and replace the use of the getter
recommended_input_volume() with an optional return value from the
latter one. The added return value carries the recommended input
volume if the call sequence follows the API contract. Make the member
applied_input_volume_ optional. Restrict the use of the getter
recommended_input_volume() for test use. Add a method
capture_output_used() for test use.
In GainController2, store the output of InputVolumeController::Process()
in a new member variable that's updated in Analyze() and Process(). Use
a trivial getter to read the value in APM.
Bug: webrtc:7494
Change-Id: Ifcfb466c4f558be560eb6d2f45410d04adb7e2ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287862
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38889}
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}
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}
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}
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}
