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/*
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
//
// Specifies core class for intelligbility enhancement.
//
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_
#include <complex>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/lapped_transform.h"
#include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h"
struct WebRtcVadInst;
typedef struct WebRtcVadInst VadInst;
namespace webrtc {
// Speech intelligibility enhancement module. Reads render and capture
// audio streams and modifies the render stream with a set of gains per
// frequency bin to enhance speech against the noise background.
// Note: assumes speech and noise streams are already separated.
class IntelligibilityEnhancer {
public:
// Construct a new instance with the given filter bank resolution,
// sampling rate, number of channels and analysis rates.
// |analysis_rate| sets the number of input blocks (containing speech!)
// to elapse before a new gain computation is made. |variance_rate| specifies
// the number of gain recomputations after which the variances are reset.
// |cv_*| are parameters for the VarianceArray constructor for the
// clear speech stream.
// TODO(bercic): the |cv_*|, |*_rate| and |gain_limit| parameters should
// probably go away once fine tuning is done. They override the internal
// constants in the class (kGainChangeLimit, kAnalyzeRate, kVarianceRate).
IntelligibilityEnhancer(int erb_resolution,
int sample_rate_hz,
int channels,
int cv_type,
float cv_alpha,
int cv_win,
int analysis_rate,
int variance_rate,
float gain_limit);
~IntelligibilityEnhancer();
// Reads and processes chunk of noise stream in time domain.
void ProcessCaptureAudio(float* const* audio);
// Reads chunk of speech in time domain and updates with modified signal.
void ProcessRenderAudio(float* const* audio);
private:
enum AudioSource {
kRenderStream = 0, // Clear speech stream.
kCaptureStream, // Noise stream.
};
// Provides access point to the frequency domain.
class TransformCallback : public LappedTransform::Callback {
public:
TransformCallback(IntelligibilityEnhancer* parent, AudioSource source);
// All in frequency domain, receives input |in_block|, applies
// intelligibility enhancement, and writes result to |out_block|.
virtual void ProcessAudioBlock(const std::complex<float>* const* in_block,
int in_channels,
int frames,
int out_channels,
std::complex<float>* const* out_block);
private:
IntelligibilityEnhancer* parent_;
AudioSource source_;
};
friend class TransformCallback;
// Sends streams to ProcessClearBlock or ProcessNoiseBlock based on source.
void DispatchAudio(AudioSource source,
const std::complex<float>* in_block,
std::complex<float>* out_block);
// Updates variance computation and analysis with |in_block_|,
// and writes modified speech to |out_block|.
void ProcessClearBlock(const std::complex<float>* in_block,
std::complex<float>* out_block);
// Computes and sets modified gains.
void AnalyzeClearBlock(float power_target);
// Updates variance calculation for noise input with |in_block|.
void ProcessNoiseBlock(const std::complex<float>* in_block,
std::complex<float>* out_block);
// Returns number of ERB filters.
static int GetBankSize(int sample_rate, int erb_resolution);
// Initializes ERB filterbank.
void CreateErbBank();
// Analytically solves quadratic for optimal gains given |lambda|.
// Negative gains are set to 0. Stores the results in |sols|.
void SolveForGainsGivenLambda(float lambda, int start_freq, float* sols);
// Computes variance across ERB filters from freq variance |var|.
// Stores in |result|.
void FilterVariance(const float* var, float* result);
// Returns dot product of vectors specified by size |length| arrays |a|,|b|.
static float DotProduct(const float* a, const float* b, int length);
static const int kErbResolution;
static const int kWindowSizeMs;
static const int kChunkSizeMs;
static const int kAnalyzeRate; // Default for |analysis_rate_|.
static const int kVarianceRate; // Default for |variance_rate_|.
static const float kClipFreq;
static const float kConfigRho; // Default production and interpretation SNR.
static const float kKbdAlpha;
static const float kGainChangeLimit;
const int freqs_; // Num frequencies in frequency domain.
const int window_size_; // Window size in samples; also the block size.
const int chunk_length_; // Chunk size in samples.
const int bank_size_; // Num ERB filters.
const int sample_rate_hz_;
const int erb_resolution_;
const int channels_; // Num channels.
const int analysis_rate_; // Num blocks before gains recalculated.
const int variance_rate_; // Num recalculations before history is cleared.
intelligibility::VarianceArray clear_variance_;
intelligibility::VarianceArray noise_variance_;
rtc::scoped_ptr<float[]> filtered_clear_var_;
rtc::scoped_ptr<float[]> filtered_noise_var_;
float** filter_bank_; // TODO(ekmeyerson): Switch to using ChannelBuffer.
rtc::scoped_ptr<float[]> center_freqs_;
int start_freq_;
rtc::scoped_ptr<float[]> rho_; // Production and interpretation SNR.
// for each ERB band.
rtc::scoped_ptr<float[]> gains_eq_; // Pre-filter modified gains.
intelligibility::GainApplier gain_applier_;
// Destination buffer used to reassemble blocked chunks before overwriting
// the original input array with modifications.
// TODO(ekmeyerson): Switch to using ChannelBuffer.
float** temp_out_buffer_;
rtc::scoped_ptr<float* []> input_audio_;
rtc::scoped_ptr<float[]> kbd_window_;
TransformCallback render_callback_;
TransformCallback capture_callback_;
rtc::scoped_ptr<LappedTransform> render_mangler_;
rtc::scoped_ptr<LappedTransform> capture_mangler_;
int block_count_;
int analysis_step_;
// TODO(bercic): Quick stopgap measure for voice detection in the clear
// and noise streams.
// Note: VAD currently does not affect anything in IntelligibilityEnhancer.
VadInst* vad_high_;
VadInst* vad_low_;
rtc::scoped_ptr<int16_t[]> vad_tmp_buffer_;
bool has_voice_low_; // Whether voice detected in speech stream.
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_