AnotherFoxGuy/webrtc-audio-processing
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Bump to WebRTC M120 release

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Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone.
We're continuing to carry iSAC even though it's gone upstream, but maybe
we'll want to drop that soon.
This commit is contained in:
Arun Raghavan
2023-12-12 10:42:58 -05:00
parent 9a202fb8c2
commit c6abf6cd3f
479 changed files with 20900 additions and 11996 deletions
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11
webrtc/modules/audio_processing/agc2/limiter.h
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@ -11,27 +11,26 @@
#ifndef MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_
#define MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
#include "modules/audio_processing/agc2/fixed_digital_level_estimator.h"
#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
#include "modules/audio_processing/include/audio_frame_view.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
class ApmDataDumper;
class Limiter {
public:
Limiter(size_t sample_rate_hz,
Limiter(int sample_rate_hz,
ApmDataDumper* apm_data_dumper,
std::string histogram_name_prefix);
absl::string_view histogram_name_prefix);
Limiter(const Limiter& limiter) = delete;
Limiter& operator=(const Limiter& limiter) = delete;
~Limiter();
// Applies limiter and hard-clipping to |signal|.
// Applies limiter and hard-clipping to `signal`.
void Process(AudioFrameView<float> signal);
InterpolatedGainCurve::Stats GetGainCurveStats() const;
@ -40,7 +39,7 @@ class Limiter {
// * below kMaximalNumberOfSamplesPerChannel*1000/kFrameDurationMs
// so that samples_per_channel fit in the
// per_sample_scaling_factors_ array.
void SetSampleRate(size_t sample_rate_hz);
void SetSampleRate(int sample_rate_hz);
// Resets the internal state.
void Reset();