MaybeInitializeCapture may overwrite the render configuration of a concurrent render reinitialization, leading to a second render reinitialization on the next render processing call.
See bug description for details.
Tested: Verified bitexactness offline (single-threaded) on a large number of aecdumps.
Bug: webrtc:14495
Change-Id: I9b70b454ce1c27859c3414c9c9ec89b7bbe35559
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277380
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38241}
Main changes:
- `AgcManagerDirectTestHelper::FirstProcess()` replaced by
`CallAgcSequence()`, which is API contract compliant
- `ExpectCheckVolumeAndReset()`, `SetVolumeAndProcess()` and
`ExpectInitialize() `removed
- TODOs added for the next batch of improvements
- `AgcManagerDirectTestHelper::mock_agc` now using `NiceMock`
- `AgcManagerDirect::(AnalyzePre)Process()` now receives a
const ref
- `AnalyzePreProcess(const float* const*,size_t )` removed
Bug: webrtc:7494
Change-Id: Ie5bbaa590586dd806b30494fb00ca9c742c241e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273490
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38056}
The current design of the modified getter is error-prone since the
returned value changes meaning based on when (which point in the code)
the getter is called - namely, before `ProcessStream()` is called the
getter returns the stream analog level, after it returns the
recommended level.
Plus, the new implementation, which essentially returns a local
member, removes the risks that the non-trivial implementation
is computationally expensive.
Bug: webrtc:7494, b/241923537
Change-Id: I6714444df27bcc055ae693974ecd1f77f79e6ec0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271580
Reviewed-by: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38055}
The `recommended_stream_analog_level()` getter is used to retrieve
both the applied and the recommended input volume. This behavior is
error-prone since the caller must know what is returned based on
the point in the code (namely, before/after the AGC has changed
the last applied input volume into a recommended level).
This CL is a first step to make clarity on which input volume is
handled in different parts of APM. Next in the pipeline: make
`recommended_stream_analog_level()` a trivial getter that always
returns the recommended level.
Main changes:
- When `recommended_stream_analog_level()` is called but
`set_stream_analog_level()` is not called, APM logs an error
and returns a fall-back volume (which should not be applied
since, when `set_stream_analog_level()` is not called, no
external input volume is expected to be present
- When APM is used without calling the `*_stream_analog_level()`
methods (e.g., when the caller does not provide any input volume),
the recorded AEC dumps won't store `Stream::applied_input_level`
Other changes:
- Removed `AudioProcessingImpl::capture_::prev_analog_mic_level`
- Removed redundant code in `GainController2` around detecting
input volume changes (already done by APM)
- Adapted the `audioproc_f` and `unpack_aecdump` tools
- Data dumps clean-up: the applied and the recommended input
volumes are now recorded in an AGC implementation agnostic way
Bug: webrtc:7494, b/241923537
Change-Id: I3cb4a731fd9f3dc19bf6ac679b7ed8c969ea283b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271544
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38054}
Switching to an AGC implementation agnostic solution for the input
volume emulation functionality offered by the
`capture_levels_adjuster` sub-module.
This CL also fixes a (silent) bug due to which, when the input
volume is emulated via the capture adjuster sub-module, AGC2
reads an incorrect value for the applied input volume.
Tested: audioproc_f with `--analog_mic_gain_emulation 1` used
to verify bit-exactness for one Wav file and one AEC dump for
which the input volume varies.
Bug: webrtc:7494, b/241923537
Change-Id: Ide3085f9a5dfd85888ad812ebd56faa175fb2ba7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273902
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38053}
`EchoControllerObservesAnalogAgc1EchoPathGainChange` is incorrect
since it does not call `set_stream_analog_level()`,
`ProcessCapture()` and `recommended_stream_analog_level()`
according to the contract.
`EchoControllerObservesNoDigitalAgc2EchoPathGainChange` is
useless since AGC2 doesn't have any analog controller at the
moment and the test is not written to explictly trigger digital
gain adaptations.
Bug: webrtc:7494, b/241923537
Change-Id: I56203c736448ec060077b00b57e98cd4c29fa737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271541
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38052}
Making it clear that the field is used to store the applied input
volume and not the recommended input volume.
Bug: webrtc:7494, b/241923537
Change-Id: Ib91bc1a12348f63e3a4ba6e068ed02e40786a87b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271342
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38051}
Fixing a bug due to which the analog controller could not be disabled.
AudioProcessing::Config::GainController1::AnalogGainController::enabled
was ignored and therefore `recommended_stream_analog_level_locked()` in
APM was returning the level recommended by `AgcManagerDirect`.
When the analog controller is disabled, `stream_analog_level()` now
returns the last value set via `set_stream_analog_level()`.
However, the analog controller code is still running and, in particular,
the existing metrics are reported as if the controller were enabled.
This choice was made to reduce the risks of adding bugs in the digital
compression gain selection part, which is tied to the analog
controller. The metric drawback will be solved in a follow-up CL.
Additional changes:
- log `WebRTC.Audio.GainController.Analog.Enabled` when
AGC1 is created or when its config changes
- first step to replace "analog level" with "input volume"
Bug: webrtc:7909, b/180019868
Change-Id: I28ce9556dd98f3dd9ad546799406c55478730435
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270663
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38044}
Instead of showing individual byte differences, this CL detects
differences in the expected and actual byte streams of the evaluated
AEC dump and, if detected, parses the `audioproc::Event` proto lite
messages and calls EXPECT_EQ() for a subset of individual (sub-)fields.
Note that messages are parsed only if the byte streams of each message
pair do not match, so with no failures the test runs at no extra cost.
Plus, the the added funcionality can only be enabled for local
debugging by flipping the `kDumpWhenExpectMessageEqFails` flag - a
code change cannot land if the flag is set to true.
Note that `MessageDifferencer` (see [1]) could not be used because
it is not implemented for `MessageLite` protos.
[1] https://developers.google.com/protocol-buffers/docs/reference/cpp/google.protobuf.util.message_differencer
Bug: b/241923537
Change-Id: I8e0eda3b1ecfe06945b6dad5ee8871f8200d76d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270922
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37765}
Sample rates not divisible by 100, in particular 11025 Hz and 22050 Hz, have long been used with APM in Chrome, but the support has never been stated explicitly.
This CL makes minor modifications to the APM API to clarify how rates are handled when 10 ms is not an integer number of samples. Unit tests are also extended to cover this case better.
This does not update all references to 10 ms and implicit floor(sample_rate/100) computations, but it does at least take us closer to a correct API.
Note that not all code needs to support these sample rates. For example, audio processing submodules only need to operate on the native APM rates 16000, 32000, 48000.
Bug: chromium:1332484
Change-Id: I1dad15468f6ccb9c0d4d09c5819fe87f8388d5b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268769
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37682}
This is the first step of migrating
AudioProcessing::CreateAndAttachAecDump() from using std::string to
absl::string_view.
Bug: webrtc:13579
Change-Id: I8fc373e7ac55fd8e96bb0b01d1a30e28883ac9a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269400
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37631}
It is now easier to fully test `AgcManagerDirect` with different values
for the used field trials. In particular, this CL adds tests for the
field trial named `WebRTC-Audio-2ndAgcMinMicLevelExperiment`.
1. `UnmutingRaisesTooLowVolume` and `MicVolumeIsLimited`
The expectations for the lowest input volume are not hard-coded anymore
since the parametrized tests use different values for the enforced
minimum.
2. `RecoveryAfterManualLevelChangeBelowMin`
The recovery behavior after manual input volume change depends on
whether the minimum input volume is overridden. When that's the case,
the minimum volume is applied immediately after the manual adjustment.
Hence, the existing test is left and a parametrized version of it has been added to test the "instant recovery" behavior. The latter test is
skipped when the minimum input volume is not overridden since that case
is covered by the existing test.
Bug: chromium:1275566
Change-Id: Ib0d4427b32b88f33138d4062b365916a3c47a406
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268900
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37577}
Stop using TEST_F; that will make it easier to switch to parametric
tests that are needed to correctly test `AgcManagerDirect`.
"Avoid fixtures where reasonable."
Source: https://abseil.io/tips/122
Bug: chromium:1275566
Change-Id: I2d73a0913eb2349144f63bd17ab4d6efa245e472
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268766
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37556}
This reverts commit d0a6fd239c.
Reason for revert: reland the bug fix
Original change's description:
> Revert "`AgcManagerDirect`: stop enforcing min mic level override with 0 level"
>
> This reverts commit e76daab8b3.
>
> Reason for revert: revert required to revert the parent CL
>
> Original change's description:
> > `AgcManagerDirect`: stop enforcing min mic level override with 0 level
> >
> > https://webrtc-review.googlesource.com/c/src/+/250141 introduced a bug
> > due to which the min mic level override is always enforced, if specified
> > even if the user manually adjusts the mic level to zero.
> >
> > This CL fixes that bug, the changes run behind a kill switch.
> >
> > TESTED=Test video call on Chromium on Mac; input volume not adjusted after zeroing it from the system preferences UI
> >
> > Bug: chromium:1275566
> > Change-Id: I18ce2e5970d3002b301f51f84544583c64982d57
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267844
> > Reviewed-by: Hanna Silen <silen@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37460}
>
> Bug: chromium:1275566
> Change-Id: I6d22d8f3fafdc7da3814827b9b69146a506595db
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268468
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37515}
Bug: chromium:1275566
Change-Id: I7198587dec2a153270e8beb714e9dacccdaae806
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268544
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37530}
This reverts commit c9cad23274.
Reason for revert: add back field trial
Original change's description:
> Min mic analog level: override minimum and behavior on Mac
>
> This CL removes the `WebRTC-Audio-AgcMinMicLevelExperiment` field trial
> and always enables the code path behind that flag on Mac. In summary,
> the analog AGC behaves as follows on Mac:
> 1. the minimum level is overridden to 20
> 2. the minimum is applied even when clipping is detected
> 3. when the level is manually adjusted to 0, the minimum level is
> enforced - i.e., 20
>
> Note that the 3rd property had been unintentionally added when the
> changes were added behind the aforementioned field trial. This will
> be fixed in a follow-up CL.
>
> Bug: chromium:1275566
> Change-Id: If184c4455a0780fcd94f55141af34460c152e3c3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266488
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Reviewed-by: Hanna Silen <silen@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37459}
Bug: chromium:1275566
Change-Id: I00a37ad9e16efc49f721558d25af16efd5f3db8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268540
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37521}
This reverts commit e76daab8b3.
Reason for revert: revert required to revert the parent CL
Original change's description:
> `AgcManagerDirect`: stop enforcing min mic level override with 0 level
>
> https://webrtc-review.googlesource.com/c/src/+/250141 introduced a bug
> due to which the min mic level override is always enforced, if specified
> even if the user manually adjusts the mic level to zero.
>
> This CL fixes that bug, the changes run behind a kill switch.
>
> TESTED=Test video call on Chromium on Mac; input volume not adjusted after zeroing it from the system preferences UI
>
> Bug: chromium:1275566
> Change-Id: I18ce2e5970d3002b301f51f84544583c64982d57
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267844
> Reviewed-by: Hanna Silen <silen@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37460}
Bug: chromium:1275566
Change-Id: I6d22d8f3fafdc7da3814827b9b69146a506595db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268468
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37515}
https://webrtc-review.googlesource.com/c/src/+/250141 introduced a bug
due to which the min mic level override is always enforced, if specified
even if the user manually adjusts the mic level to zero.
This CL fixes that bug, the changes run behind a kill switch.
TESTED=Test video call on Chromium on Mac; input volume not adjusted after zeroing it from the system preferences UI
Bug: chromium:1275566
Change-Id: I18ce2e5970d3002b301f51f84544583c64982d57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267844
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37460}
This CL removes the `WebRTC-Audio-AgcMinMicLevelExperiment` field trial
and always enables the code path behind that flag on Mac. In summary,
the analog AGC behaves as follows on Mac:
1. the minimum level is overridden to 20
2. the minimum is applied even when clipping is detected
3. when the level is manually adjusted to 0, the minimum level is
enforced - i.e., 20
Note that the 3rd property had been unintentionally added when the
changes were added behind the aforementioned field trial. This will
be fixed in a follow-up CL.
Bug: chromium:1275566
Change-Id: If184c4455a0780fcd94f55141af34460c152e3c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266488
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37459}
This CL duplicates a few lines of utility code from
//modules/audio_processing:audioproc_test_utils (which contains more
testonly things) and allows the possibility to remove testonly from
the unpack_aecdump tool.
Bug: b/237526033
Change-Id: If2e1dd4cc825429c496091cf8640c67069fb6e6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267701
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37437}
Look for first echo (and not only the strongest one) on the same matched
filter.
This change is bit exact with previous version when `pre_echo` is false.
Author: Jesús de Vicente Peña <devicentepena@webrtc.org>
Bug: webrtc:14205
Change-Id: I6782eaa1d690b0df78d00f6d425a85c951b2ca9d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266321
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37360}
When the `WebRTC-Audio-TransientSuppressorVadMode-RnnVad` field trial
is set, APM now uses (i) its RNN VAD sub-module to compute the voice
probability, (ii) that probability for TS and (iii) a temporally
delayed version of it for AGC2 (the delay introduced by TS is taken
into account).
Bug: webrtc:13663
Change-Id: Ic0f245c3f00d318c19bb01d3dbc2d5176c90f851
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266362
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37291}
Add a VoiceActivityDetectorWrapper submodule in AudioProcessingImpl
and enable injecting speech probability into GainController2.
Bug: webrtc:13663
Change-Id: I05e13b737d085b45ac8ce76660191867c56834c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265166
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37275}
Puts the whole block in contiguous memory and reduce pointer look-up.
The change has been verified to be bit-exact.
Bug: webrtc:14089
Change-Id: I264aaf764bf53a29f23249105f704b2fdbd7e51c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263203
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36983}
The high-band gain is corrected by fixing the computation of the
low-band energy
Bug: webrtc:14108
Change-Id: I5033287de57aedcd91bb71623ca2862519ffb35b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263201
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36972}
This change adds a Block class to reduce the need for std::vector<std::vector<std::vector<float>>>. This make the code
easier to read and less error prone.
It also enables future changes to the underlying data structure of a
block. For instance, the data of all bands and channels could be stored
in a single vector.
The change has been verified to be bit-exact.
Bug: webrtc:14089
Change-Id: Ied9a78124c0bbafe0e912017aef91f7c311de2ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262252
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36968}
To quote rfc6464:
The audio level for digital silence -- for a muted audio source, for
example -- MUST be represented as 127 (-127 dBov), regardless of the
dynamic range of the encoded audio format.
The behavior in webrtc is correct that digital silence is represented
with 127, but it is also possible to get a value of 127 for not quite
digitally silent audio buffer (as in, not strictly 0s).
Bug: webrtc:14029
Change-Id: I7ff8698a7e4d5c0960c667fd1cc961838e269456
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261244
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36793}
* Structs with user-declared constructors are no longer considered
aggregates, so remove the declarations when possible
* Types of both arguments to "==" must match to avoid "ambiguous
function call" warning
* Various types of math involving enums are deprecated, so replace with
constexprs where necessary
* ABSL_CONST_INIT must be used on definition as well as declaration
* volatile memory may no longer be read from and written to by the same
operator, so replace e.g. "n++" with "n = n + 1"
* Replace an outdated check for no_unique_address support with
__has_cpp_attribute
* std::result_of(f(x)) has been removed, replace with
std::invoke_result(f, x)
Bug: chromium:1284275
Change-Id: I77b366ab1da7eb2c1e4c825b2714417c31ee5903
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261221
Auto-Submit: Peter Kasting <pkasting@chromium.org>
Reviewed-by: Tomas Gunnarsson <tommi@google.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36786}
While the target has a restricted visibility, since it was in rtc_base_approved
public deps, a lot of targets were able to bypass the visibility check.
So we remove the visibility restrictions and use the dependency explicitely
everywhere instead.
Bug: webrtc:8603
Change-Id: I94a03fdf7f94c54ab72081a58dd648e2cca73d17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258944
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36566}
Adds two metrics for stereo detection:
- An enum indicating whether the last 10 seconds contained persistent stereo content or not, logged every 10 seconds.
- An enum indicating whether any persistent stereo content at all has been detected, logged at the end of the AEC lifetime.
These metrics allow us to assess:
- What proportion of all audio is treated as stereo.
- What proportion of sessions encounter any significant stereo content. If this is unexpectedly high, the stereo detection code may need fine tuning.
Metrics are only logged for component lifetimes exceeding 5 seconds: This is to avoid brief AEC lifetimes due to internal resets etc within APM.
Corresponding Chrome CL for XML histogram declarations:
https://crrev.com/c/3579317
Bug: chromium:1295710
Change-Id: I93e2bf74588cf4bb2a8922dbfad079bccab01456
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258760
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36537}
During temporary stereo content when the AEC3 uses a mono reference signal, the signal is downmixed by averaging instead of using only the left channel.
Additionally, temporary stereo content is flagged as an echo path change.
Tested: Modified local build: Verified stereo mode entered / left in accordance with hysteresis and timeout thresholds. Verified temporary stereo detected during temporary stereo playout. Made an aecdump and inspected content.
Bug: chromium:1295710
Change-Id: I6bd53e615dfb3ec39bc1c73275b7d6d599ac7c57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258481
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36504}
Even if playout audio is only very briefly stereo, the AEC will enter stereo processing mode. To save CPU and improve AEC performance, this CL adds a hysteresis period before treating playout as stereo.
The feature is enabled by default in the AEC3 config.
Bug: chromium:1295710
Change-Id: I29116ab2e7823e25a02aa3b66a1c619f1d966d9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258479
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36503}
If playout audio is temporarily stereo, the AEC will currently enter stereo processing mode indefinitely. To save CPU and improve AEC performance, this CL adds support for falling back to mono after a period of no stereo.
The feature is enabled by default in the AEC3 config.
Bug: chromium:1295710
Change-Id: I690b5b22f8407f950bf41f3bcaa9ca0138452157
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258421
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36502}
This CL adds a component in the TS implementation to return a delayed
version of the voice probability values observed when `Suppress()` is
called. That is needed in order to temporally align the voice
probability values to the processed audio since TS adds algorithmic
delay.
Bug: webrtc:13663
Change-Id: I5041ace3939d2ce7ba084ae703428e66f1aa06be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255860
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36496}
The values returned by `TransientSuppressor::Initialize()` and by
`TransientSuppressor::Suppress()` are never used.
Bug: webrtc:13663
Change-Id: I20b8afb5a66f49e5ebaf132acf8bcd1c4292a5f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255822
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36492}
More robust API option that allows to fully initialize TS when created.
Bug: webrtc:13663
Change-Id: I42c38612ef772eb6d0bbde49d04ea39332a0e3c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255821
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36490}
Adding a delay unit to be used in the APM Transient Suppressor (TS)
sub-module through which the observerd voice probabilities are
temporally aligned to the audio processed by TS, which introduces
algorithmic delay.
Bug: webrtc:13663
Change-Id: I2136c303914580851c742d8db89478a13b06dacb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255680
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36487}
It is now required to specify which VAD is used to compute the speech
probability passed when `TransientSuppressor::Suppress()` is called.
In this way, it is possible to adapt parameters and/or logic of a
`TransientSuppressor` implementation to the behavior of the used
VAD. This CL also adds a "no VAD" mode option, which ignores the speech
probability argument passed when `Suppress()` and always applies mild
suppression to preserve transparency.
Finally, this CL adds a field trial to choose which VAD is used by
APM for transient suppression. Wiring the RNN VAD to TS will be done
in a follow-up CL.
Bug: webrtc:13663
Change-Id: I21ed49f91875a4ee0f04db97ea87c0dbc3db7f8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250962
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36485}
The features have two safety fallbacks:
- multichannel config has a killswitch WebRTC-Aec3SetupSpecificDefaultConfigDefaultsKillSwitch
- stereo detection has a killswitch WebRTC-Aec3StereoContentDetectionKillSwitch
Both features are enabled by default in the AEC3 config.
Tested: Bitexact on a large number of aecdumps.
Bug: chromium:1295710
Change-Id: I340cdc9140dacd4ca22d0911eb9f732b6cf8b226
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258129
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36482}
Apart from making the construction more straightforward, this change allows recreating the BlockProcessor at runtime. This is used to change parameterization at runtime in an upcoming CL [1].
[1] https://webrtc-review.googlesource.com/c/src/+/258129
Tested: Bitexact on a large number of aecdumps.
Bug: chromium:1295710
Change-Id: I2e0275c5c97044cb4370042633266b193c06b960
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258100
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36473}
Motivation: never used.
Follow-up CL of https://webrtc-review.googlesource.com/c/src/+/250680.
Tested on a custom chromium build by making a video call and with
keyboard activity. The expected logs from the APM transient suppressor
sub-module were shown.
Bug: webrtc:11226
Change-Id: I4186994412dd8ba2e71ed8f9dcc9cf8f8e40fbd3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250667
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36300}
Add a flag to override the key pressed state when simulating APM.
The current behavior changes as follows:
- Wav files simulation: instead of simulating continuous key press
events only if the transient suppressor (TS) sub-module is active,
allow to simulate the events regardless of whether TS is used;
the default key pressed state is used if the command line flag is
unspecified, otherwise it is overridden (either always false or
always true)
- AEC dump simulation: instead of simulating continuous key press
events when `--ts 2` is specified, allow to simulate the events
regardless of whether TS is used; the state recorded in the AEC
dump is used if the command line flag is unspecified, otherwise
it is overridden (either always false or always true)
- The `--ts 2` option (continuous key events) is now equivalent to
`--ts 1`.
Bug: webrtc:13663
Change-Id: I5ebe96283db73ee235ec2b2795d91d4e241a3527
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256003
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36247}
Increase the history size of clipping_predictor_evaluator_. Use one-sample
accuracy in clipping detection for the evaluator.
Bug: webrtc:12774
Change-Id: I8c1bbfe69fe55af73ce14992e49ef7295b3ce926
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241602
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36218}
- missing negation causes the opposite behavior when
`analog_agc_disable_digital_adaptive` is used
- flag replaced with `analog_agc_use_digital_adaptive_controller`
which is less error-prone
Bug: webrtc:7494
Change-Id: If9e0ba4fc9e539c73269faf9096ca782620dac6d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251322
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36113}
Also test the field trial with valid parameter and non-empty suffix.
Bug: webrtc:7494
Change-Id: I3d871b41dd71c951ac56e180b3c09cda4c3627d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251441
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36031}
This enum is no longer needed. Also moving the last piece of code from
common.h to audio_processing_impl.h, allowing to delete common.h.
Bug: chromium:1271981, b/217349489
Change-Id: If115336c36d6d7b5845a903e421c18aebfe434ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251242
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35946}
When the minimum mic level is overridden via the field trial named
WebRTC-Audio-AgcMinMicLevelExperiment, AGC1 can still lower the gain
beyond the minimum value (namely, when clipping is observed).
This CL changes the behavior of the field trial. When specified, the
override always applies and therefore the mic level is guaranteed to
never become lower than what the field trial specifies.
Tested: RTC call in Chromium with and without --force-fieldtrials="
WebRTC-Audio-AgcMinMicLevelExperiment/Enabled-255"
Bug: chromium:1275566
Change-Id: I42ff45add54c11084f5ca6a2b95887c627c3c3aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250141
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35914}
- Switch from ptr+size to rtc::ArrayView
- Remove `AgcManagerDirect::sample_rate_hz_` since it's always 16 kHz
- Stop passing nullptr in agc_manager_direct_unittest.cc when
`AgcManagerDirect::Process()` is called
- Allow to correctly run the tests added in the child CL (see [1])
[1] https://webrtc-review.googlesource.com/c/src/+/250141
Bug: webrtc:7494
Change-Id: I0292d7038d6510ca7c58af32b6003a1e4b121b00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250541
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35910}
The 6-parameter Initialize method is removed. The has_keyboard parameter
in the StreamConfig constructor is removed together with the underlying
member and helper functions.
Bug: chromium:1271981, b/217349489
Change-Id: I7259a114a395f74f735a9c06510c0fc0f0f008e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250221
Reviewed-by: Sam Zackrisson <saza@google.com>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Auto-Submit: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35908}
This CL removes even more top-level const from parameters in function
declarations. This change is safe because top-level const in function
declarations (not function definitions) are ignored by the compiler
and so change is just a no-op cleanup.
Bug: webrtc:13610
Change-Id: Icf6868c27b1fdb9d9915b3a7020eb34bdcf07a09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249989
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35866}
This is a safe cleanup change since top-level const applied to
parameters in function declarations (that are not also
definitions) are ignored by the compiler. Hence, such changes do
not change the type of the declared functions and are simply
no-ops.
Bug: webrtc:13610
Change-Id: Ibafb92c45119a6d8bdb6f9109aa8dad6385163a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249086
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35802}
Just applied a short sed script. See bug description for
the motiviation for this change.
This is the command that was used to generate the changes:
$ find . -type f \( -iname '*.cc' -o -iname '*.h' \) -print0 | \
xargs -0 sed -i -e 's/(const override)/(const, override)/'
Bug: webrtc:13090
Change-Id: Iec7d280f9d55263a972dbb3bd644ebfcd2eb38cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249088
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35801}
Changing to an index for-loop (instead of using std::max_element & std::distance) tracking even & odd elements separately allows the compiler to produce code with less pipeline stall.
Bug: None
Change-Id: Iaa3e820a3a3b61e2eb276f0dac9106c848db1891
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/240061
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Christian Schuldt <cschuldt@google.com>
Cr-Commit-Position: refs/heads/main@{#35729}
Manually unrolling the multiply-and-accumulate loop of the matched filter allows interleaving of instruction, which gives a significant saving.
Bug: None
Change-Id: Ie7a7d92bd453d81e9dd61812781a7b6d62e1f1f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/240321
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Christian Schuldt <cschuldt@google.com>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35566}
Important: This change does not in any way affect echo cancellation or standardized stats. The user audio experience is unchanged. Only non-standard stats are affected. Echo return loss metrics are unchanged. Residual echo likelihood {recent max} will no longer be computed by default.
Important: The echo detector is no longer enabled by default.
API change, PSA: https://groups.google.com/g/discuss-webrtc/c/mJV5cDysBDI/m/7PTPBjVHCgAJ
This CL removes the default usage of the residual echo detector in APM.
It can now only be used via injection and the helper function webrtc::CreateEchoDetector. See how the function audio_processing_unittest.cc:CreateApm() changed, for an example.
The echo detector implementation is marked poisonous, to avoid accidental dependencies.
Some cleanup is done:
- EchoDetector::PackRenderAudioBuffer is declared in one target but is defined in another target. It is not necessary to keep in the API. It is made an implementation detail, and the echo detector input is documented in the API.
- The internal state of APM is large and difficult to track. Submodule pointers that are set permanently on construction are now appropriately marked const.
Tested:
- existing + new unit tests
- audioproc_f is bitexact on a large number of aecdumps
Bug: webrtc:11539
Change-Id: I00cc2ee112fedb06451a533409311605220064d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239652
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35550}
Reducing pointer following. This will allow the compiler to optimize more efficiently with the "-fno-strict-aliasing" flag.
Bug: None
Change-Id: I8e2d841fa543b28c59eb08c654a2b0515ab39d69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241780
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Christian Schuldt <cschuldt@google.com>
Cr-Commit-Position: refs/heads/main@{#35548}
Unloading states and coefficients to local variables avoids excessive memory access when building with "-fno-strict-aliasing".
Bug: None
Change-Id: I90bf81ae794c21e9e41500c5040387cf67ebdd38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/240320
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Christian Schuldt <cschuldt@google.com>
Cr-Commit-Position: refs/heads/main@{#35518}
Changing to an index for-loop (instead of a range for-loop) allows the compiler (clang for x86 at least) to unroll it x2.
Bug: None
Change-Id: I9b9612a8513a06e8aa3b12ae39f6911217da55fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239741
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Christian Schuldt <cschuldt@google.com>
Cr-Commit-Position: refs/heads/main@{#35478}
Changing to an index for-loop (instead of using std::transform) allows the compiler (clang for x86 at least) to use 3 different registers in the loop rather than just 1, resulting in less pipeline stall (I'd assume). Interestingly, the compiler unrolls the loop(s) completely in both cases.
Bug: None
Change-Id: I586773bc525e91bb6eb6638d5399928482306b9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239364
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Christian Schuldt <cschuldt@google.com>
Cr-Commit-Position: refs/heads/main@{#35453}
Reducing pointer following. This will allow the compiler to optimize more efficiently with the "-fno-strict-aliasing" flag.
Bug: None
Change-Id: Ic126bd2d53969a7e9d15e1c1081d5278e27a816c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238664
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Christian Schuldt <cschuldt@google.com>
Cr-Commit-Position: refs/heads/main@{#35414}
Reducing pointer following. This will allow the compiler to optimize more efficiently with the "-fno-strict-aliasing" flag.
Bug: None
Change-Id: I7cde835161e2d3e85fc7c919556fa9a9e87ef6df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238169
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Christian Schuldt <cschuldt@google.com>
Cr-Commit-Position: refs/heads/main@{#35393}
New limiter tuning to more quickly go back to 0 dB after the limiter
kicks in and the input peak level goes back to normal.
Bug: webrtc:7494
Change-Id: I1050957ca4caf12c4562b899b16c306957dce169
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237701
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35384}
Reducing pointer following. This will allow the compiler to optimize more efficiently with the "-fno-strict-aliasing" flag.
Bug: None
Change-Id: Ib1fd3a1cf3f89471b0ec87404650a6061eec5e2d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237782
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Christian Schuldt <cschuldt@google.com>
Cr-Commit-Position: refs/heads/main@{#35374}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
Implement AnalogGainStatsReporter and add it in AudioProcessingImpl.
This class computes statistics for analog gain updates and
periodically reports them into a histogram.
The added histograms for analog gain update statistics:
- WebRTC.Audio.ApmAnalogGainDecreaseRate
- WebRTC.Audio.ApmAnalogGainIncreaseRate
- WebRTC.Audio.ApmAnalogGainUpdateRate
- WebRTC.Audio.ApmAnalogGainDecreaseAverage
- WebRTC.Audio.ApmAnalogGainIncreaseAverage
- WebRTC.Audio.ApmAnalogGainUpdateAverage
The histograms are defined in
https://chromium-review.googlesource.com/c/chromium/src/+/3207987
Bug: webrtc:12774
Change-Id: I3c58d4bb3eb034a11c3f39ab8edb2bc67c5fd5e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234140
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35301}
Remove analog_level_minimum and analog_level_maximum from
AudioProcessing GainController1 and replace their use with fixed
values 0 and 255, respectively.
Bug: webrtc:12774
Change-Id: Ia4bfe5ed43a65f1587ed67f36bfbb2966b6fdf26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235822
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35297}
The class has also been renamed to better reflect its purpose.
Bug: webrtc:7494
Change-Id: I223a364ab4f8b8a5fef765848bf05675d045cefd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236343
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35277}
This CL includes two changes that break bit-exactness, but that haven't
changed the way AGC2 behaves - the new behavior has been verified with
audioproc_f on a collection of AEC dumps and Wav files (42 recordings
in total).
1) The fixed digital controller can directly be initialized in the
`GainController2` ctor. Before, `SetGainFactor()` was called after the
creation of the object and that caused an initial ramp up lasting one
10 ms frame from -inf to 0 dB. As an effect of the new initialization,
the initial ramp up doesn't happen anymore.
2) In [1] the AGC2 VAD has been moved from the adaptive digital
controller into `GainController2`. In order to not break bit-exactness,
the VAD was placed after the fixed digital controller and before the
adaptive digital one. However, to reduce the chance of incorrect
estimation of the speech probability, the VAD should analyze the
audio before any digital processing is applied inside AGC2.
[1] https://webrtc-review.googlesource.com/c/src/+/234583
Bug: webrtc:7494
Change-Id: I9418229cbe537014fed8271c5550c3ce2bc88e26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235240
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35252}
Bit exactness verified with audioproc_f on a collection of AEC dumps
and Wav files (42 recordings in total).
Bug: webrtc:7494
Change-Id: Id9849c4463791f5a203afe31efc163efb4d4458e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234583
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35248}
Only used in unit tests and a duplication of what `capture_output_rms_`
already does.
This CL also removes `AudioProcessingStats::output_rms_dbfs`, which is
now unused.
Bug: webrtc:5298
Fix: chromium:1261339
Change-Id: I6e583c11d4abb58444c440509a8495a7f5ebc589
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235664
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35246}
Also stop using ApplyConfig() and in [1] fix the build errors when
WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE is defined.
[1] modules/audio_processing/test/audio_processing_builder_for_testing.cc
Bug: webrtc:5298
Change-Id: I50dc5668b952e7ca7fa83c7a5182c013e928c450
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235365
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35228}
Not passing the sample rate to the `VoiceActivityDetectorWrapper` ctor
yet since that would require an unnecessary refactoring of `AdaptiveAgc`
which will soon be removed.
Instead, to ensure correct initialization until the child CL [1] lands,
`VoiceActivityDetectorWrapper::initialized_` is temporarily added.
Bit exactness verified with audioproc_f on a collection of AEC dumps
and Wav files (42 recordings in total).
[1] https://webrtc-review.googlesource.com/c/src/+/234583
Bug: webrtc:7494
Change-Id: I4b4be7b8106ba36c958d91bf263a7b30271a1ee3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234587
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35213}
Internal refactoring of AGC2 to decouple the VAD, its wrapper and the
peak and RMS level measurements.
Bit exactness verified with audioproc_f on a collection of AEC dumps
and Wav files (42 recordings in total).
Bug: webrtc:7494
Change-Id: Ib560f1fcaa601557f4f30e47025c69e91b1b62e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234524
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35208}
When `AudioProcessingImpl::ApplyConfig()` is called, AGC2 is initialized
and then the new config is applied. That is error prone and for example
breaks bit exactness in [1].
Changes:
- `GainController2` must be created by passing configuration,
sample rate and number of channels
- `GainController2::ApplyConfig()` removed
Bit exactness verified with audioproc_f on a collection of AEC dumps
and Wav files (42 recordings in total).
[1] https://webrtc-review.googlesource.com/c/src/+/234587.
Bug: webrtc:7494
Change-Id: I251e03603394a4fc8769b9b5c197a157893676a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235060
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35206}
This change improves echo canceller transparency by enabling the use
of a non-capped ERLE when computing the residual echo spectrum for
dominant nearend detection.
Experimentation has shown that the feature improves echo canceller
transparency and user ratings.
Implementation CL:
https://webrtc-review.googlesource.com/c/src/+/221920
Bug: webrtc:12870
Change-Id: I7dc66810e8300cd35321bcd5b9fae9bc3386836d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234841
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35186}
Internal refactoring of AGC2. This CL is needed in preparation for its
child CL to correctly show the upcoming changes in the diff.
Bug: webrtc:7494
Change-Id: If7f837e064243d5ffe09e21fc68f489bb00dfdc5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234527
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35170}
Move the check for analog gain changes so that it can be used
independently of echo_controller. This change is needed to land
https://webrtc-review.googlesource.com/c/src/+/234140.
Bug: webrtc:12774
Change-Id: I9ea127b0a4d374f31493d6f8afcacee40fa9257c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234383
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35159}
This CL does *not* change the behavior of the AGC2 adaptive digital
controller - bitexactness verified with audioproc_f on a collection of
AEC dumps and Wav files (42 recordings in total).
Tested: compiled Chrome with this patch and made an appr.tc test call
Bug: webrtc:7494
Change-Id: Ia8a9f6fbc3a3459b888a2eed87e108f0d39cfe99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233520
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35140}
First CL to try to understand the extent of the cleanup needed in
order to remove -Wno-shadow and follow Chromium on enabling this
diagnostic.
Bug: webrtc:13219
Change-Id: Ie699762da50fe3dbc08b1fd92220962d4b7da86b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35134}
Instead of using two different headroom parameters, namely
`kHeadroomDbfs` and `kSaturationProtectorExtraHeadroomDb`, only use
the former that now also accounts for the deleted one - i.e., it equals
the sum of the two headrooms. In this way, tuning AGC2 will be easier.
This CL does *not* change the behavior of the AGC2 adaptive digital
controller - bitexactness verified with audioproc_f on a collection of
AEC dumps and Wav files (42 recordings in total).
The unit tests changes in agc2/saturation_protector_unittest.cc are
required since `extra_headroom_db` is removed and the changes in
agc2/adaptive_digital_gain_applier_unittest.cc are required because
`AdaptiveDigitalGainApplier` depends on `kHeadroomDbfs` which has been
updated as stated above.
Bug: webrtc:7494
Change-Id: I0a2a710bbede0caa53938090a004d185fdefaeb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232905
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35109}
This CL improves `GainController2::CheckGainAdaptiveDigital`, namely:
- correctly initialize AGC2 with the correct number of channels
- attenuate the input signal in order to avoid that the target gain is
set to zero (which was the case before)
- run AG2 adaptive digital for a longer period to allow time to trigger
the adaptive behavior (namely, from 2s to 10s)
- minor code style improvements
Bug: webrtc:7494
Change-Id: Ib41de088b341bb30460238b83e306a507b2bc5af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233101
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35099}
Add histograms WebRTC.Audio.Agc.ClippingPredictor.Precision and WebRTC.Audio.Agc.ClippingPredictor.Recall. The histograms are defined in https://chromium-review.googlesource.com/c/chromium/src/+/3150271.
Bug: webrtc:12774
Change-Id: I1561ec7a61377c262f636d7aa3a5d5fd60a8839d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231460
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35083}
Thanks to the elimination of `ExperimentalNs`, there is no need anymore
to pass `webrtc::Config` to build APM.
Hence, `AudioProcessingBuilder::Create(const webrtc::Config&)` is also
removed.
Bug: webrtc:5298
Change-Id: I0a3482376a7753434486fe564681f7b9f83939c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232128
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35025}
To focus on the ability to predict clipping, the clipping predictor
evaluator doesn't increment the true positive count anymore when a
prediction is simultaneously observed with a detection.
Note that `WebRTC.Audio.Agc.ClippingPredictor.F1Score` is still used
to log the F1 score - i.e., the histogram hasn't been renamed.
Bug: webrtc:12774
Change-Id: Ia987e568a6df2a3ddba7fa1b5697d6feda22d20c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231233
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34942}
Evaluate the clipping predictor whenever injected but keep using the
predictions only when allowed.
Bug: webrtc:12774
Change-Id: I9e8930a528d1d514d52b821a28b6c8ad0c3aeb5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231137
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34937}
