forked from len0rd/rockbox
codecs: Update libspeex from 1.2beta3 to 1.2rc1
This is a relatively minor bump, but it's the first step towards bringing this current. Change-Id: Iab6c9b0c77f0ba705280434ea74b513364719499
This commit is contained in:
Solomon Peachy
parent
8ef20383b1
commit
547b6a570d
21 changed files with 1406 additions and 1001 deletions
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@ -1,6 +1,6 @@
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/* Copyright (C) 2003 Epic Games (written by Jean-Marc Valin)
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Copyright (C) 2004-2006 Epic Games
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Copyright (C) 2004-2006 Epic Games
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File: preprocess.c
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Preprocessor with denoising based on the algorithm by Ephraim and Malah
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@ -34,24 +34,24 @@
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/*
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Recommended papers:
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Y. Ephraim and D. Malah, "Speech enhancement using minimum mean-square error
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short-time spectral amplitude estimator". IEEE Transactions on Acoustics,
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short-time spectral amplitude estimator". IEEE Transactions on Acoustics,
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Speech and Signal Processing, vol. ASSP-32, no. 6, pp. 1109-1121, 1984.
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Y. Ephraim and D. Malah, "Speech enhancement using minimum mean-square error
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log-spectral amplitude estimator". IEEE Transactions on Acoustics, Speech and
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log-spectral amplitude estimator". IEEE Transactions on Acoustics, Speech and
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Signal Processing, vol. ASSP-33, no. 2, pp. 443-445, 1985.
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I. Cohen and B. Berdugo, "Speech enhancement for non-stationary noise environments".
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Signal Processing, vol. 81, no. 2, pp. 2403-2418, 2001.
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Stefan Gustafsson, Rainer Martin, Peter Jax, and Peter Vary. "A psychoacoustic
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approach to combined acoustic echo cancellation and noise reduction". IEEE
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Stefan Gustafsson, Rainer Martin, Peter Jax, and Peter Vary. "A psychoacoustic
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approach to combined acoustic echo cancellation and noise reduction". IEEE
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Transactions on Speech and Audio Processing, 2002.
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J.-M. Valin, J. Rouat, and F. Michaud, "Microphone array post-filter for separation
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of simultaneous non-stationary sources". In Proceedings IEEE International
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of simultaneous non-stationary sources". In Proceedings IEEE International
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Conference on Acoustics, Speech, and Signal Processing, 2004.
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*/
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@ -75,7 +75,7 @@
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#define LOUDNESS_EXP 5.f
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#define AMP_SCALE .001f
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#define AMP_SCALE_1 1000.f
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#define NB_BANDS 24
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#define SPEECH_PROB_START_DEFAULT QCONST16(0.35f,15)
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@ -117,7 +117,7 @@ static inline spx_word16_t DIV32_16_Q8(spx_word32_t a, spx_word32_t b)
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a = SHL32(a,8);
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return PDIV32_16(a,b);
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}
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}
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static inline spx_word16_t DIV32_16_Q15(spx_word32_t a, spx_word32_t b)
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{
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@ -185,7 +185,7 @@ struct SpeexPreprocessState_ {
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int sampling_rate; /**< Sampling rate of the input/output */
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int nbands;
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FilterBank *bank;
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/* Parameters */
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int denoise_enabled;
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int vad_enabled;
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@ -198,7 +198,9 @@ struct SpeexPreprocessState_ {
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int echo_suppress;
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int echo_suppress_active;
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SpeexEchoState *echo_state;
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spx_word16_t speech_prob; /**< Probability last frame was speech */
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/* DSP-related arrays */
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spx_word16_t *frame; /**< Processing frame (2*ps_size) */
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spx_word16_t *ft; /**< Processing frame in freq domain (2*ps_size) */
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@ -234,7 +236,6 @@ struct SpeexPreprocessState_ {
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float *loudness_weight; /**< Perceptual loudness curve */
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float loudness; /**< Loudness estimate */
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float agc_gain; /**< Current AGC gain */
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int nb_loudness_adapt; /**< Number of frames used for loudness adaptation so far */
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float max_gain; /**< Maximum gain allowed */
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float max_increase_step; /**< Maximum increase in gain from one frame to another */
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float max_decrease_step; /**< Maximum decrease in gain from one frame to another */
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@ -259,7 +260,7 @@ static void conj_window(spx_word16_t *w, int len)
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spx_word16_t tmp;
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#ifdef FIXED_POINT
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spx_word16_t x = DIV32_16(MULT16_16(32767,i),len);
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#else
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#else
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spx_word16_t x = DIV32_16(MULT16_16(QCONST16(4.f,13),i),len);
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#endif
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int inv=0;
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@ -284,10 +285,10 @@ static void conj_window(spx_word16_t *w, int len)
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}
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}
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#ifdef FIXED_POINT
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/* This function approximates the gain function
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y = gamma(1.25)^2 * M(-.25;1;-x) / sqrt(x)
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/* This function approximates the gain function
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y = gamma(1.25)^2 * M(-.25;1;-x) / sqrt(x)
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which multiplied by xi/(1+xi) is the optimal gain
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in the loudness domain ( sqrt[amplitude] )
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Input in Q11 format, output in Q15
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@ -320,7 +321,7 @@ static inline spx_word16_t qcurve(spx_word16_t x)
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static void compute_gain_floor(int noise_suppress, int effective_echo_suppress, spx_word32_t *noise, spx_word32_t *echo, spx_word16_t *gain_floor, int len)
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{
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int i;
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if (noise_suppress > effective_echo_suppress)
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{
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spx_word16_t noise_gain, gain_ratio;
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@ -346,8 +347,8 @@ static void compute_gain_floor(int noise_suppress, int effective_echo_suppress,
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}
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#else
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/* This function approximates the gain function
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y = gamma(1.25)^2 * M(-.25;1;-x) / sqrt(x)
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/* This function approximates the gain function
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y = gamma(1.25)^2 * M(-.25;1;-x) / sqrt(x)
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which multiplied by xi/(1+xi) is the optimal gain
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in the loudness domain ( sqrt[amplitude] )
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*/
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@ -391,7 +392,7 @@ static void compute_gain_floor(int noise_suppress, int effective_echo_suppress,
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}
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#endif
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SpeexPreprocessState *speex_preprocess_state_init(int frame_size, int sampling_rate)
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EXPORT SpeexPreprocessState *speex_preprocess_state_init(int frame_size, int sampling_rate)
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{
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int i;
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int N, N3, N4, M;
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@ -413,8 +414,8 @@ SpeexPreprocessState *speex_preprocess_state_init(int frame_size, int sampling_r
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break;
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}
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}
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if (st->ps_size < 3*st->frame_size/4)
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st->ps_size = st->ps_size * 3 / 2;
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#else
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@ -424,7 +425,7 @@ SpeexPreprocessState *speex_preprocess_state_init(int frame_size, int sampling_r
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N = st->ps_size;
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N3 = 2*N - st->frame_size;
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N4 = st->frame_size - N3;
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st->sampling_rate = sampling_rate;
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st->denoise_enabled = 1;
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st->vad_enabled = 0;
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@ -439,15 +440,15 @@ SpeexPreprocessState *speex_preprocess_state_init(int frame_size, int sampling_r
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st->speech_prob_continue = SPEECH_PROB_CONTINUE_DEFAULT;
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st->echo_state = NULL;
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st->nbands = NB_BANDS;
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M = st->nbands;
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st->bank = filterbank_new(M, sampling_rate, N, 1);
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st->frame = (spx_word16_t*)speex_alloc(2*N*sizeof(spx_word16_t));
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st->window = (spx_word16_t*)speex_alloc(2*N*sizeof(spx_word16_t));
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st->ft = (spx_word16_t*)speex_alloc(2*N*sizeof(spx_word16_t));
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st->ps = (spx_word32_t*)speex_alloc((N+M)*sizeof(spx_word32_t));
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st->noise = (spx_word32_t*)speex_alloc((N+M)*sizeof(spx_word32_t));
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st->echo_noise = (spx_word32_t*)speex_alloc((N+M)*sizeof(spx_word32_t));
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@ -460,19 +461,19 @@ SpeexPreprocessState *speex_preprocess_state_init(int frame_size, int sampling_r
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st->gain2 = (spx_word16_t*)speex_alloc((N+M)*sizeof(spx_word16_t));
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st->gain_floor = (spx_word16_t*)speex_alloc((N+M)*sizeof(spx_word16_t));
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st->zeta = (spx_word16_t*)speex_alloc((N+M)*sizeof(spx_word16_t));
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st->S = (spx_word32_t*)speex_alloc(N*sizeof(spx_word32_t));
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st->Smin = (spx_word32_t*)speex_alloc(N*sizeof(spx_word32_t));
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st->Stmp = (spx_word32_t*)speex_alloc(N*sizeof(spx_word32_t));
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st->update_prob = (int*)speex_alloc(N*sizeof(int));
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st->inbuf = (spx_word16_t*)speex_alloc(N3*sizeof(spx_word16_t));
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st->outbuf = (spx_word16_t*)speex_alloc(N3*sizeof(spx_word16_t));
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conj_window(st->window, 2*N3);
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for (i=2*N3;i<2*st->ps_size;i++)
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st->window[i]=Q15_ONE;
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if (N4>0)
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{
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for (i=N3-1;i>=0;i--)
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@ -514,7 +515,6 @@ SpeexPreprocessState *speex_preprocess_state_init(int frame_size, int sampling_r
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/*st->loudness = pow(AMP_SCALE*st->agc_level,LOUDNESS_EXP);*/
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st->loudness = 1e-15;
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st->agc_gain = 1;
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st->nb_loudness_adapt = 0;
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st->max_gain = 30;
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st->max_increase_step = exp(0.11513f * 12.*st->frame_size / st->sampling_rate);
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st->max_decrease_step = exp(-0.11513f * 40.*st->frame_size / st->sampling_rate);
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@ -530,7 +530,7 @@ SpeexPreprocessState *speex_preprocess_state_init(int frame_size, int sampling_r
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return st;
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}
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void speex_preprocess_state_destroy(SpeexPreprocessState *st)
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EXPORT void speex_preprocess_state_destroy(SpeexPreprocessState *st)
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{
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speex_free(st->frame);
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speex_free(st->ft);
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@ -573,7 +573,7 @@ static void speex_compute_agc(SpeexPreprocessState *st, spx_word16_t Pframe, spx
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float target_gain;
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float loudness=1.f;
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float rate;
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for (i=2;i<N;i++)
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{
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loudness += 2.f*N*st->ps[i]* st->loudness_weight[i];
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@ -583,7 +583,6 @@ static void speex_compute_agc(SpeexPreprocessState *st, spx_word16_t Pframe, spx
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loudness*2 > pow(st->loudness, 1.0/LOUDNESS_EXP))*/
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if (Pframe>.3f)
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{
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st->nb_loudness_adapt++;
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/*rate=2.0f*Pframe*Pframe/(1+st->nb_loudness_adapt);*/
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rate = .03*Pframe*Pframe;
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st->loudness = (1-rate)*st->loudness + (rate)*pow(AMP_SCALE*loudness, LOUDNESS_EXP);
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@ -592,7 +591,7 @@ static void speex_compute_agc(SpeexPreprocessState *st, spx_word16_t Pframe, spx
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st->init_max *= 1.f + .1f*Pframe*Pframe;
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}
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/*printf ("%f %f %f %f\n", Pframe, loudness, pow(st->loudness, 1.0f/LOUDNESS_EXP), st->loudness2);*/
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target_gain = AMP_SCALE*st->agc_level*pow(st->loudness/(1e-4+st->loudness_accum), -1.0f/LOUDNESS_EXP);
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if ((Pframe>.5 && st->nb_adapt > 20) || target_gain < st->agc_gain)
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@ -605,11 +604,11 @@ static void speex_compute_agc(SpeexPreprocessState *st, spx_word16_t Pframe, spx
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target_gain = st->max_gain;
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if (target_gain > st->init_max)
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target_gain = st->init_max;
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st->agc_gain = target_gain;
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}
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/*fprintf (stderr, "%f %f %f\n", loudness, (float)AMP_SCALE_1*pow(st->loudness, 1.0f/LOUDNESS_EXP), st->agc_gain);*/
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for (i=0;i<2*N;i++)
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ft[i] *= st->agc_gain;
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st->prev_loudness = loudness;
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@ -629,7 +628,7 @@ static void preprocess_analysis(SpeexPreprocessState *st, spx_int16_t *x)
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st->frame[i]=st->inbuf[i];
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for (i=0;i<st->frame_size;i++)
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st->frame[N3+i]=x[i];
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/* Update inbuf */
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for (i=0;i<N3;i++)
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st->inbuf[i]=x[N4+i];
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@ -648,10 +647,10 @@ static void preprocess_analysis(SpeexPreprocessState *st, spx_int16_t *x)
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st->frame[i] = SHL16(st->frame[i], st->frame_shift);
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}
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#endif
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/* Perform FFT */
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spx_fft(st->fft_lookup, st->frame, st->ft);
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/* Power spectrum */
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ps[0]=MULT16_16(st->ft[0],st->ft[0]);
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for (i=1;i<N;i++)
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@ -669,11 +668,11 @@ static void update_noise_prob(SpeexPreprocessState *st)
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int N = st->ps_size;
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for (i=1;i<N-1;i++)
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st->S[i] = MULT16_32_Q15(QCONST16(.8f,15),st->S[i]) + MULT16_32_Q15(QCONST16(.05f,15),st->ps[i-1])
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st->S[i] = MULT16_32_Q15(QCONST16(.8f,15),st->S[i]) + MULT16_32_Q15(QCONST16(.05f,15),st->ps[i-1])
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+ MULT16_32_Q15(QCONST16(.1f,15),st->ps[i]) + MULT16_32_Q15(QCONST16(.05f,15),st->ps[i+1]);
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st->S[0] = MULT16_32_Q15(QCONST16(.8f,15),st->S[0]) + MULT16_32_Q15(QCONST16(.2f,15),st->ps[0]);
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st->S[N-1] = MULT16_32_Q15(QCONST16(.8f,15),st->S[N-1]) + MULT16_32_Q15(QCONST16(.2f,15),st->ps[N-1]);
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if (st->nb_adapt==1)
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{
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for (i=0;i<N;i++)
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@ -700,12 +699,12 @@ static void update_noise_prob(SpeexPreprocessState *st)
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for (i=0;i<N;i++)
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{
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st->Smin[i] = MIN32(st->Smin[i], st->S[i]);
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st->Stmp[i] = MIN32(st->Stmp[i], st->S[i]);
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st->Stmp[i] = MIN32(st->Stmp[i], st->S[i]);
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}
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}
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for (i=0;i<N;i++)
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{
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if (MULT16_32_Q15(QCONST16(.4f,15),st->S[i]) > ADD32(st->Smin[i],EXTEND32(20)))
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if (MULT16_32_Q15(QCONST16(.4f,15),st->S[i]) > st->Smin[i])
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st->update_prob[i] = 1;
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else
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st->update_prob[i] = 0;
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@ -719,12 +718,12 @@ static void update_noise_prob(SpeexPreprocessState *st)
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void speex_echo_get_residual(SpeexEchoState *st, spx_word32_t *Yout, int len);
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int speex_preprocess(SpeexPreprocessState *st, spx_int16_t *x, spx_int32_t *echo)
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EXPORT int speex_preprocess(SpeexPreprocessState *st, spx_int16_t *x, spx_int32_t *echo)
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{
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return speex_preprocess_run(st, x);
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}
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int speex_preprocess_run(SpeexPreprocessState *st, spx_int16_t *x)
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EXPORT int speex_preprocess_run(SpeexPreprocessState *st, spx_int16_t *x)
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{
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int i;
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int M;
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@ -736,12 +735,12 @@ int speex_preprocess_run(SpeexPreprocessState *st, spx_int16_t *x)
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spx_word16_t Pframe;
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spx_word16_t beta, beta_1;
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spx_word16_t effective_echo_suppress;
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st->nb_adapt++;
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if (st->nb_adapt>20000)
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st->nb_adapt = 20000;
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st->min_count++;
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beta = MAX16(QCONST16(.03,15),DIV32_16(Q15_ONE,st->nb_adapt));
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beta_1 = Q15_ONE-beta;
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M = st->nbands;
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@ -775,7 +774,7 @@ int speex_preprocess_run(SpeexPreprocessState *st, spx_int16_t *x)
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st->update_prob[i] = 0;
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}
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*/
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/* Update the noise estimate for the frequencies where it can be */
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for (i=0;i<N;i++)
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{
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@ -793,17 +792,17 @@ int speex_preprocess_run(SpeexPreprocessState *st, spx_int16_t *x)
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for (i=0;i<N+M;i++)
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{
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spx_word16_t gamma;
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/* Total noise estimate including residual echo and reverberation */
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spx_word32_t tot_noise = ADD32(ADD32(ADD32(EXTEND32(1), PSHR32(st->noise[i],NOISE_SHIFT)) , st->echo_noise[i]) , st->reverb_estimate[i]);
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/* A posteriori SNR = ps/noise - 1*/
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st->post[i] = SUB16(DIV32_16_Q8(ps[i],tot_noise), QCONST16(1.f,SNR_SHIFT));
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st->post[i]=MIN16(st->post[i], QCONST16(100.f,SNR_SHIFT));
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/* Computing update gamma = .1 + .9*(old/(old+noise))^2 */
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gamma = QCONST16(.1f,15)+MULT16_16_Q15(QCONST16(.89f,15),SQR16_Q15(DIV32_16_Q15(st->old_ps[i],ADD32(st->old_ps[i],tot_noise))));
|
||||
|
||||
|
||||
/* A priori SNR update = gamma*max(0,post) + (1-gamma)*old/noise */
|
||||
st->prior[i] = EXTRACT16(PSHR32(ADD32(MULT16_16(gamma,MAX16(0,st->post[i])), MULT16_16(Q15_ONE-gamma,DIV32_16_Q8(st->old_ps[i],tot_noise))), 15));
|
||||
st->prior[i]=MIN16(st->prior[i], QCONST16(100.f,SNR_SHIFT));
|
||||
|
|
@ -824,13 +823,13 @@ int speex_preprocess_run(SpeexPreprocessState *st, spx_int16_t *x)
|
|||
for (i=N;i<N+M;i++)
|
||||
Zframe = ADD32(Zframe, EXTEND32(st->zeta[i]));
|
||||
Pframe = QCONST16(.1f,15)+MULT16_16_Q15(QCONST16(.899f,15),qcurve(DIV32_16(Zframe,st->nbands)));
|
||||
|
||||
|
||||
effective_echo_suppress = EXTRACT16(PSHR32(ADD32(MULT16_16(SUB16(Q15_ONE,Pframe), st->echo_suppress), MULT16_16(Pframe, st->echo_suppress_active)),15));
|
||||
|
||||
|
||||
compute_gain_floor(st->noise_suppress, effective_echo_suppress, st->noise+N, st->echo_noise+N, st->gain_floor+N, M);
|
||||
|
||||
/* Compute Ephraim & Malah gain speech probability of presence for each critical band (Bark scale)
|
||||
Technically this is actually wrong because the EM gaim assumes a slightly different probability
|
||||
|
||||
/* Compute Ephraim & Malah gain speech probability of presence for each critical band (Bark scale)
|
||||
Technically this is actually wrong because the EM gaim assumes a slightly different probability
|
||||
distribution */
|
||||
for (i=N;i<N+M;i++)
|
||||
{
|
||||
|
|
@ -847,7 +846,7 @@ int speex_preprocess_run(SpeexPreprocessState *st, spx_int16_t *x)
|
|||
#ifdef FIXED_POINT
|
||||
spx_word16_t tmp;
|
||||
#endif
|
||||
|
||||
|
||||
prior_ratio = PDIV32_16(SHL32(EXTEND32(st->prior[i]), 15), ADD16(st->prior[i], SHL32(1,SNR_SHIFT)));
|
||||
theta = MULT16_32_P15(prior_ratio, QCONST32(1.f,EXPIN_SHIFT)+SHL32(EXTEND32(st->post[i]),EXPIN_SHIFT-SNR_SHIFT));
|
||||
|
||||
|
|
@ -872,12 +871,12 @@ int speex_preprocess_run(SpeexPreprocessState *st, spx_int16_t *x)
|
|||
/* Convert the EM gains and speech prob to linear frequency */
|
||||
filterbank_compute_psd16(st->bank,st->gain2+N, st->gain2);
|
||||
filterbank_compute_psd16(st->bank,st->gain+N, st->gain);
|
||||
|
||||
|
||||
/* Use 1 for linear gain resolution (best) or 0 for Bark gain resolution (faster) */
|
||||
if (1)
|
||||
{
|
||||
filterbank_compute_psd16(st->bank,st->gain_floor+N, st->gain_floor);
|
||||
|
||||
|
||||
/* Compute gain according to the Ephraim-Malah algorithm -- linear frequency */
|
||||
for (i=0;i<N;i++)
|
||||
{
|
||||
|
|
@ -887,7 +886,7 @@ int speex_preprocess_run(SpeexPreprocessState *st, spx_int16_t *x)
|
|||
spx_word16_t tmp;
|
||||
spx_word16_t p;
|
||||
spx_word16_t g;
|
||||
|
||||
|
||||
/* Wiener filter gain */
|
||||
prior_ratio = PDIV32_16(SHL32(EXTEND32(st->prior[i]), 15), ADD16(st->prior[i], SHL32(1,SNR_SHIFT)));
|
||||
theta = MULT16_32_P15(prior_ratio, QCONST32(1.f,EXPIN_SHIFT)+SHL32(EXTEND32(st->post[i]),EXPIN_SHIFT-SNR_SHIFT));
|
||||
|
|
@ -898,22 +897,22 @@ int speex_preprocess_run(SpeexPreprocessState *st, spx_int16_t *x)
|
|||
g = EXTRACT16(MIN32(Q15_ONE, MULT16_32_Q15(prior_ratio, MM)));
|
||||
/* Interpolated speech probability of presence */
|
||||
p = st->gain2[i];
|
||||
|
||||
|
||||
/* Constrain the gain to be close to the Bark scale gain */
|
||||
if (MULT16_16_Q15(QCONST16(.333f,15),g) > st->gain[i])
|
||||
g = MULT16_16(3,st->gain[i]);
|
||||
st->gain[i] = g;
|
||||
|
||||
|
||||
/* Save old power spectrum */
|
||||
st->old_ps[i] = MULT16_32_P15(QCONST16(.2f,15),st->old_ps[i]) + MULT16_32_P15(MULT16_16_P15(QCONST16(.8f,15),SQR16_Q15(st->gain[i])),ps[i]);
|
||||
|
||||
|
||||
/* Apply gain floor */
|
||||
if (st->gain[i] < st->gain_floor[i])
|
||||
st->gain[i] = st->gain_floor[i];
|
||||
|
||||
/* Exponential decay model for reverberation (unused) */
|
||||
/*st->reverb_estimate[i] = st->reverb_decay*st->reverb_estimate[i] + st->reverb_decay*st->reverb_level*st->gain[i]*st->gain[i]*st->ps[i];*/
|
||||
|
||||
|
||||
/* Take into account speech probability of presence (loudness domain MMSE estimator) */
|
||||
/* gain2 = [p*sqrt(gain)+(1-p)*sqrt(gain _floor) ]^2 */
|
||||
tmp = MULT16_16_P15(p,spx_sqrt(SHL32(EXTEND32(st->gain[i]),15))) + MULT16_16_P15(SUB16(Q15_ONE,p),spx_sqrt(SHL32(EXTEND32(st->gain_floor[i]),15)));
|
||||
|
|
@ -927,20 +926,20 @@ int speex_preprocess_run(SpeexPreprocessState *st, spx_int16_t *x)
|
|||
{
|
||||
spx_word16_t tmp;
|
||||
spx_word16_t p = st->gain2[i];
|
||||
st->gain[i] = MAX16(st->gain[i], st->gain_floor[i]);
|
||||
st->gain[i] = MAX16(st->gain[i], st->gain_floor[i]);
|
||||
tmp = MULT16_16_P15(p,spx_sqrt(SHL32(EXTEND32(st->gain[i]),15))) + MULT16_16_P15(SUB16(Q15_ONE,p),spx_sqrt(SHL32(EXTEND32(st->gain_floor[i]),15)));
|
||||
st->gain2[i]=SQR16_Q15(tmp);
|
||||
}
|
||||
filterbank_compute_psd16(st->bank,st->gain2+N, st->gain2);
|
||||
}
|
||||
|
||||
|
||||
/* If noise suppression is off, don't apply the gain (but then why call this in the first place!) */
|
||||
if (!st->denoise_enabled)
|
||||
{
|
||||
for (i=0;i<N+M;i++)
|
||||
st->gain2[i]=Q15_ONE;
|
||||
}
|
||||
|
||||
|
||||
/* Apply computed gain */
|
||||
for (i=1;i<N;i++)
|
||||
{
|
||||
|
|
@ -949,7 +948,7 @@ int speex_preprocess_run(SpeexPreprocessState *st, spx_int16_t *x)
|
|||
}
|
||||
st->ft[0] = MULT16_16_P15(st->gain2[0],st->ft[0]);
|
||||
st->ft[2*N-1] = MULT16_16_P15(st->gain2[N-1],st->ft[2*N-1]);
|
||||
|
||||
|
||||
/*FIXME: This *will* not work for fixed-point */
|
||||
#ifndef FIXED_POINT
|
||||
if (st->agc_enabled)
|
||||
|
|
@ -978,7 +977,7 @@ int speex_preprocess_run(SpeexPreprocessState *st, spx_int16_t *x)
|
|||
}
|
||||
}
|
||||
#endif
|
||||
|
||||
|
||||
/* Synthesis window (for WOLA) */
|
||||
for (i=0;i<2*N;i++)
|
||||
st->frame[i] = MULT16_16_Q15(st->frame[i], st->window[i]);
|
||||
|
|
@ -988,15 +987,16 @@ int speex_preprocess_run(SpeexPreprocessState *st, spx_int16_t *x)
|
|||
x[i] = st->outbuf[i] + st->frame[i];
|
||||
for (i=0;i<N4;i++)
|
||||
x[N3+i] = st->frame[N3+i];
|
||||
|
||||
|
||||
/* Update outbuf */
|
||||
for (i=0;i<N3;i++)
|
||||
st->outbuf[i] = st->frame[st->frame_size+i];
|
||||
|
||||
/* FIXME: This VAD is a kludge */
|
||||
st->speech_prob = Pframe;
|
||||
if (st->vad_enabled)
|
||||
{
|
||||
if (Pframe > st->speech_prob_start || (st->was_speech && Pframe > st->speech_prob_continue))
|
||||
if (st->speech_prob > st->speech_prob_start || (st->was_speech && st->speech_prob > st->speech_prob_continue))
|
||||
{
|
||||
st->was_speech=1;
|
||||
return 1;
|
||||
|
|
@ -1010,7 +1010,7 @@ int speex_preprocess_run(SpeexPreprocessState *st, spx_int16_t *x)
|
|||
}
|
||||
}
|
||||
|
||||
void speex_preprocess_estimate_update(SpeexPreprocessState *st, spx_int16_t *x)
|
||||
EXPORT void speex_preprocess_estimate_update(SpeexPreprocessState *st, spx_int16_t *x)
|
||||
{
|
||||
int i;
|
||||
int N = st->ps_size;
|
||||
|
|
@ -1020,11 +1020,11 @@ void speex_preprocess_estimate_update(SpeexPreprocessState *st, spx_int16_t *x)
|
|||
|
||||
M = st->nbands;
|
||||
st->min_count++;
|
||||
|
||||
|
||||
preprocess_analysis(st, x);
|
||||
|
||||
update_noise_prob(st);
|
||||
|
||||
|
||||
for (i=1;i<N-1;i++)
|
||||
{
|
||||
if (!st->update_prob[i] || st->ps[i] < PSHR32(st->noise[i],NOISE_SHIFT))
|
||||
|
|
@ -1045,7 +1045,7 @@ void speex_preprocess_estimate_update(SpeexPreprocessState *st, spx_int16_t *x)
|
|||
}
|
||||
|
||||
|
||||
int speex_preprocess_ctl(SpeexPreprocessState *state, int request, void *ptr)
|
||||
EXPORT int speex_preprocess_ctl(SpeexPreprocessState *state, int request, void *ptr)
|
||||
{
|
||||
int i;
|
||||
SpeexPreprocessState *st;
|
||||
|
|
@ -1103,7 +1103,7 @@ int speex_preprocess_ctl(SpeexPreprocessState *state, int request, void *ptr)
|
|||
case SPEEX_PREPROCESS_GET_VAD:
|
||||
(*(spx_int32_t*)ptr) = st->vad_enabled;
|
||||
break;
|
||||
|
||||
|
||||
case SPEEX_PREPROCESS_SET_DEREVERB:
|
||||
st->dereverb_enabled = (*(spx_int32_t*)ptr);
|
||||
for (i=0;i<st->ps_size;i++)
|
||||
|
|
@ -1121,7 +1121,7 @@ int speex_preprocess_ctl(SpeexPreprocessState *state, int request, void *ptr)
|
|||
/* FIXME: Re-enable when de-reverberation is actually enabled again */
|
||||
/*(*(float*)ptr) = st->reverb_level;*/
|
||||
break;
|
||||
|
||||
|
||||
case SPEEX_PREPROCESS_SET_DEREVERB_DECAY:
|
||||
/* FIXME: Re-enable when de-reverberation is actually enabled again */
|
||||
/*st->reverb_decay = (*(float*)ptr);*/
|
||||
|
|
@ -1169,17 +1169,51 @@ int speex_preprocess_ctl(SpeexPreprocessState *state, int request, void *ptr)
|
|||
st->echo_state = (SpeexEchoState*)ptr;
|
||||
break;
|
||||
case SPEEX_PREPROCESS_GET_ECHO_STATE:
|
||||
ptr = (void*)st->echo_state;
|
||||
(*(SpeexEchoState**)ptr) = (SpeexEchoState*)st->echo_state;
|
||||
break;
|
||||
#ifndef FIXED_POINT
|
||||
case SPEEX_PREPROCESS_GET_AGC_LOUDNESS:
|
||||
(*(spx_int32_t*)ptr) = pow(st->loudness, 1.0/LOUDNESS_EXP);
|
||||
break;
|
||||
case SPEEX_PREPROCESS_GET_AGC_GAIN:
|
||||
(*(spx_int32_t*)ptr) = floor(.5+8.6858*log(st->agc_gain));
|
||||
break;
|
||||
#endif
|
||||
case SPEEX_PREPROCESS_GET_PSD_SIZE:
|
||||
case SPEEX_PREPROCESS_GET_NOISE_PSD_SIZE:
|
||||
(*(spx_int32_t*)ptr) = st->ps_size;
|
||||
break;
|
||||
case SPEEX_PREPROCESS_GET_PSD:
|
||||
for(i=0;i<st->ps_size;i++)
|
||||
((spx_int32_t *)ptr)[i] = (spx_int32_t) st->ps[i];
|
||||
break;
|
||||
case SPEEX_PREPROCESS_GET_NOISE_PSD:
|
||||
for(i=0;i<st->ps_size;i++)
|
||||
((spx_int32_t *)ptr)[i] = (spx_int32_t) PSHR32(st->noise[i], NOISE_SHIFT);
|
||||
break;
|
||||
case SPEEX_PREPROCESS_GET_PROB:
|
||||
(*(spx_int32_t*)ptr) = MULT16_16_Q15(st->speech_prob, 100);
|
||||
break;
|
||||
#ifndef FIXED_POINT
|
||||
case SPEEX_PREPROCESS_SET_AGC_TARGET:
|
||||
st->agc_level = (*(spx_int32_t*)ptr);
|
||||
if (st->agc_level<1)
|
||||
st->agc_level=1;
|
||||
if (st->agc_level>32768)
|
||||
st->agc_level=32768;
|
||||
break;
|
||||
case SPEEX_PREPROCESS_GET_AGC_TARGET:
|
||||
(*(spx_int32_t*)ptr) = st->agc_level;
|
||||
break;
|
||||
#endif
|
||||
|
||||
default:
|
||||
speex_warning_int("Unknown speex_preprocess_ctl request: ", request);
|
||||
return -1;
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
#ifdef FIXED_DEBUG
|
||||
long long spx_mips=0;
|
||||
#endif
|
||||
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue