AnotherFoxGuy/webrtc-audio-processing
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Update to current webrtc library

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This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.
This commit is contained in:
Arun Raghavan
2020-10-12 18:08:02 -04:00
parent b1b02581d3
commit bcec8b0b21
859 changed files with 76187 additions and 49580 deletions
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31
webrtc/modules/audio_processing/agc/agc.h
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@ -8,29 +8,25 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_H_
#ifndef MODULES_AUDIO_PROCESSING_AGC_AGC_H_
#define MODULES_AUDIO_PROCESSING_AGC_AGC_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_processing/vad/voice_activity_detector.h"
#include "webrtc/typedefs.h"
#include <memory>
#include "modules/audio_processing/vad/voice_activity_detector.h"
namespace webrtc {
class AudioFrame;
class Histogram;
class LoudnessHistogram;
class Agc {
public:
Agc();
virtual ~Agc();
// Returns the proportion of samples in the buffer which are at full-scale
// (and presumably clipped).
virtual float AnalyzePreproc(const int16_t* audio, size_t length);
// |audio| must be mono; in a multi-channel stream, provide the first (usually
// left) channel.
virtual int Process(const int16_t* audio, size_t length, int sample_rate_hz);
virtual void Process(const int16_t* audio, size_t length, int sample_rate_hz);
// Retrieves the difference between the target RMS level and the current
// signal RMS level in dB. Returns true if an update is available and false
@ -39,20 +35,17 @@ class Agc {
virtual void Reset();
virtual int set_target_level_dbfs(int level);
virtual int target_level_dbfs() const { return target_level_dbfs_; }
virtual float voice_probability() const {
return vad_.last_voice_probability();
}
virtual int target_level_dbfs() const;
virtual float voice_probability() const;
private:
double target_level_loudness_;
int target_level_dbfs_;
rtc::scoped_ptr<Histogram> histogram_;
rtc::scoped_ptr<Histogram> inactive_histogram_;
std::unique_ptr<LoudnessHistogram> histogram_;
std::unique_ptr<LoudnessHistogram> inactive_histogram_;
VoiceActivityDetector vad_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_H_
#endif // MODULES_AUDIO_PROCESSING_AGC_AGC_H_