Bump to WebRTC M131 release
Ongoing fixes and improvements, transient suppressor is gone. Also, dropping isac because it doesn't seem to be useful, and is just build system deadweight now. Upstream references: Version: 131.0.6778.200 WebRTC: 79aff54b0fa9238ce3518dd9eaf9610cd6f22e82 Chromium: 2a19506ad24af755f2a215a4c61f775393e0db42
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@ -14,6 +14,7 @@
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#include <vector>
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#include "absl/strings/string_view.h"
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#include "api/audio/audio_frame.h"
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#include "modules/audio_processing/agc2/fixed_digital_level_estimator.h"
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#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
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#include "modules/audio_processing/include/audio_frame_view.h"
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@ -23,23 +24,25 @@ class ApmDataDumper;
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class Limiter {
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public:
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Limiter(int sample_rate_hz,
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ApmDataDumper* apm_data_dumper,
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// See `SetSamplesPerChannel()` for valid values for `samples_per_channel`.
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Limiter(ApmDataDumper* apm_data_dumper,
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size_t samples_per_channel,
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absl::string_view histogram_name_prefix);
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Limiter(const Limiter& limiter) = delete;
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Limiter& operator=(const Limiter& limiter) = delete;
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~Limiter();
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// Applies limiter and hard-clipping to `signal`.
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void Process(AudioFrameView<float> signal);
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void Process(DeinterleavedView<float> signal);
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InterpolatedGainCurve::Stats GetGainCurveStats() const;
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// Supported rates must be
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// * supported by FixedDigitalLevelEstimator
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// * below kMaximalNumberOfSamplesPerChannel*1000/kFrameDurationMs
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// so that samples_per_channel fit in the
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// per_sample_scaling_factors_ array.
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void SetSampleRate(int sample_rate_hz);
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// Supported values must be
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// * Supported by FixedDigitalLevelEstimator
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// * Below or equal to kMaximalNumberOfSamplesPerChannel so that samples
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// fit in the per_sample_scaling_factors_ array.
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void SetSamplesPerChannel(size_t samples_per_channel);
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// Resets the internal state.
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void Reset();
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