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Updating to SVN Speex again. Missing voice detection.

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git-svn-id: https://mumble.svn.sourceforge.net/svnroot/mumble/trunk@442 05730e5d-ab1b-0410-a4ac-84af385074fa
This commit is contained in:
Thorvald Natvig 2007-05-23 13:57:43 +00:00
parent a1ba43376d
commit 99a73c9e28
3 changed files with 90 additions and 108 deletions
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6
AudioInput.cpp
View File

@ -287,7 +287,6 @@ void AudioInput::encodeAudioFrame() {
max=abs(psSource[i]);
dPeakSignal=20.0*log10((max * 1.0L) / 32768.0L);
CloneSpeexPreprocessState *pps=reinterpret_cast<CloneSpeexPreprocessState *>(sppPreprocess);
// The default is a bit short, increase it
if (! iIsSpeech) {
@ -297,14 +296,15 @@ void AudioInput::encodeAudioFrame() {
} else {
iHoldFrames = 0;
}
/*
CloneSpeexPreprocessState *pps=reinterpret_cast<CloneSpeexPreprocessState *>(sppPreprocess);
if (pps->loudness2 < g.s.iMinLoudness)
pps->loudness2 = g.s.iMinLoudness;
if (isnan(pps->loudness2)) {
qWarning("AudioInput: loudness is nan");
bResetProcessor = true;
}
*/
if (g.s.atTransmit == Settings::Continous)
iIsSpeech = 1;
else if (g.s.atTransmit == Settings::PushToTalk)

123
AudioStats.cpp
View File

@ -46,45 +46,109 @@ struct CloneSpeexEchoState {
int M;
int cancel_count;
int adapted;
int saturated;
int screwed_up;
spx_int32_t sampling_rate;
spx_word16_t spec_average;
spx_word16_t beta0;
spx_word16_t beta_max;
spx_word32_t sum_adapt;
spx_word16_t *e;
spx_word16_t *x;
spx_word16_t *X;
spx_word16_t *d;
spx_word16_t *y;
spx_word16_t leak_estimate;
spx_word16_t *e; /* scratch */
spx_word16_t *x; /* Far-end input buffer (2N) */
spx_word16_t *X; /* Far-end buffer (M+1 frames) in frequency domain */
spx_word16_t *input; /* scratch */
spx_word16_t *y; /* scratch */
spx_word16_t *last_y;
spx_word32_t *Yps;
spx_word16_t *Y;
spx_word16_t *Y; /* scratch */
spx_word16_t *E;
spx_word32_t *PHI;
spx_word32_t *W;
spx_word32_t *power;
spx_float_t *power_1;
spx_word16_t *wtmp;
#ifdef FIXED_POINT
spx_word16_t *wtmp2;
#endif
spx_word32_t *Rf;
spx_word32_t *Yf;
spx_word32_t *Xf;
spx_word32_t *PHI; /* scratch */
spx_word32_t *W; /* (Background) filter weights */
spx_word16_t *foreground; /* Foreground filter weights */
spx_word32_t Davg1; /* 1st recursive average of the residual power difference */
spx_word32_t Davg2; /* 2nd recursive average of the residual power difference */
spx_float_t Dvar1; /* Estimated variance of 1st estimator */
spx_float_t Dvar2; /* Estimated variance of 2nd estimator */
spx_word32_t *power; /* Power of the far-end signal */
spx_float_t *power_1;/* Inverse power of far-end */
spx_word16_t *wtmp; /* scratch */
spx_word32_t *Rf; /* scratch */
spx_word32_t *Yf; /* scratch */
spx_word32_t *Xf; /* scratch */
spx_word32_t *Eh;
spx_word32_t *Yh;
spx_float_t Pey;
spx_float_t Pyy;
spx_float_t Pey;
spx_float_t Pyy;
spx_word16_t *window;
spx_word16_t *prop;
void *fft_table;
spx_word16_t memX, memD, memE;
spx_word16_t preemph;
spx_word16_t notch_radius;
spx_mem_t notch_mem[2];
};
/* NOTE: If you only use speex_echo_cancel() and want to save some memory, remove this */
spx_int16_t *play_buf;
int play_buf_pos;
/** Speex pre-processor state. */
struct CloneSpeexPreprocessState {
/* Basic info */
int frame_size; /**< Number of samples processed each time */
int ps_size; /**< Number of points in the power spectrum */
int sampling_rate; /**< Sampling rate of the input/output */
int nbands;
void *bank;
int denoise_enabled;
int vad_enabled;
int dereverb_enabled;
spx_word16_t reverb_decay;
spx_word16_t reverb_level;
spx_word16_t speech_prob_start;
spx_word16_t speech_prob_continue;
int noise_suppress;
int echo_suppress;
int echo_suppress_active;
CloneSpeexEchoState *echo_state;
spx_word16_t *frame; /**< Processing frame (2*ps_size) */
spx_word16_t *ft; /**< Processing frame in freq domain (2*ps_size) */
spx_word32_t *ps; /**< Current power spectrum */
spx_word16_t *gain2; /**< Adjusted gains */
spx_word16_t *gain_floor; /**< Minimum gain allowed */
spx_word16_t *window; /**< Analysis/Synthesis window */
spx_word32_t *noise; /**< Noise estimate */
spx_word32_t *reverb_estimate; /**< Estimate of reverb energy */
spx_word32_t *old_ps; /**< Power spectrum for last frame */
spx_word16_t *gain; /**< Ephraim Malah gain */
spx_word16_t *prior; /**< A-priori SNR */
spx_word16_t *post; /**< A-posteriori SNR */
spx_word32_t *S; /**< Smoothed power spectrum */
spx_word32_t *Smin; /**< See Cohen paper */
spx_word32_t *Stmp; /**< See Cohen paper */
int *update_prob; /**< Propability of speech presence for noise update */
spx_word16_t *zeta; /**< Smoothed a priori SNR */
spx_word32_t *echo_noise;
spx_word32_t *residual_echo;
spx_word16_t *inbuf; /**< Input buffer (overlapped analysis) */
spx_word16_t *outbuf; /**< Output buffer (for overlap and add) */
int agc_enabled;
float agc_level;
float loudness_accum;
float *loudness_weight; /**< Perceptual loudness curve */
float loudness; /**< Loudness estimate */
float agc_gain; /**< Current AGC gain */
int nb_loudness_adapt; /**< Number of frames used for loudness adaptation so far */
float max_gain; /**< Maximum gain allowed */
float max_increase_step; /**< Maximum increase in gain from one frame to another */
float max_decrease_step; /**< Maximum decrease in gain from one frame to another */
float prev_loudness; /**< Loudness of previous frame */
float init_max; /**< Current gain limit during initialisation */
int nb_adapt; /**< Number of frames used for adaptation so far */
int was_speech;
int min_count; /**< Number of frames processed so far */
void *fft_lookup; /**< Lookup table for the FFT */
};
AudioEchoWidget::AudioEchoWidget(QWidget *p) : QGLWidget(p) {
@ -450,14 +514,6 @@ void AudioStats::on_Tick_timeout() {
txt.sprintf("%06.2f dB",ai->dPeakSignal);
qlSignalLevel->setText(txt);
double level = st->loudness2 / 200.0;
txt.sprintf("%03.0f%%", level);
qlMicVolume->setText(txt);
#ifdef SPEEX_ANCIENT_PP
txt.sprintf("%06.3f",st->Zlast);
#else
float Zframe = 0;
int freq_start = static_cast<int>(300.0f*2.f*st->ps_size/st->sampling_rate);
int freq_end = static_cast<int>(2000.0f*2.f*st->ps_size/st->sampling_rate);
@ -465,14 +521,9 @@ void AudioStats::on_Tick_timeout() {
Zframe += st->zeta[i];
Zframe /= (freq_end-freq_start);
txt.sprintf("%06.3f",Zframe);
#endif
qlMicSNR->setText(txt);
#ifdef SPEEX_ANCIENT_PP
txt.sprintf("%03.0f%%",st->speech_prob * 100.0);
#else
txt.sprintf("%03.0f%%",nTalking ? 100.0 : 0.0);
#endif
qlSpeechProb->setText(txt);
txt.sprintf("%04.1f kbit/s",ai->iBitrate / 1000.0);

69
mumble_pch.h
View File

@ -27,75 +27,6 @@
#include <boost/shared_ptr.hpp>
#include <boost/weak_ptr.hpp>
#ifndef JITTER_BUFFER_OK
#define SPEEX_ANCIENT
#endif
#define SPEEX_ANCIENT_PP
#ifdef SPEEX_ANCIENT_PP
typedef SpeexPreprocessState CloneSpeexPreprocessState;
#else
typedef struct {
/* Basic info */
int frame_size; /**< Number of samples processed each time */
int ps_size; /**< Number of points in the power spectrum */
int sampling_rate; /**< Sampling rate of the input/output */
int nbands;
void *bank;
/* Parameters */
int denoise_enabled;
int agc_enabled;
float agc_level;
int vad_enabled;
int dereverb_enabled;
float reverb_decay;
float reverb_level;
float speech_prob_start;
float speech_prob_continue;
int noise_suppress;
int echo_suppress;
int echo_suppress_active;
/* DSP-related arrays */
float *frame; /**< Processing frame (2*ps_size) */
float *ft; /**< Processing frame in freq domain (2*ps_size) */
float *ps; /**< Current power spectrum */
float *gain2; /**< Adjusted gains */
float *gain_floor; /**< Minimum gain allowed */
float *window; /**< Analysis/Synthesis window */
float *noise; /**< Noise estimate */
float *reverb_estimate; /**< Estimate of reverb energy */
float *old_ps; /**< Power spectrum for last frame */
float *gain; /**< Ephraim Malah gain */
float *prior; /**< A-priori SNR */
float *post; /**< A-posteriori SNR */
float *S; /**< Smoothed power spectrum */
float *Smin; /**< See Cohen paper */
float *Stmp; /**< See Cohen paper */
float *update_prob; /**< Propability of speech presence for noise update */
float *zeta; /**< Smoothed a priori SNR */
float *loudness_weight; /**< Perceptual loudness curve */
float *echo_noise;
/* Misc */
float *inbuf; /**< Input buffer (overlapped analysis) */
float *outbuf; /**< Output buffer (for overlap and add) */
int was_speech;
float loudness; /**< loudness estimate */
float loudness2; /**< loudness estimate */
int nb_adapt; /**< Number of frames used for adaptation so far */
int nb_loudness_adapt; /**< Number of frames used for loudness adaptation so far */
int min_count; /**< Number of frames processed so far */
void *fft_lookup; /**< Lookup table for the FFT */
} CloneSpeexPreprocessState;
#endif
#ifndef WIN32
typedef WId HWND;
#endif