Mercurial > libavcodec.hg
changeset 11653:28638e0d6e63 libavcodec
WMAVoice postfilter.
| author | rbultje |
|---|---|
| date | Wed, 21 Apr 2010 18:01:34 +0000 |
| parents | 8b6f3d3b55cb |
| children | fe105c6d28c7 |
| files | wmavoice.c wmavoice_data.h |
| diffstat | 2 files changed, 706 insertions(+), 7 deletions(-) [+] |
line wrap: on
line diff
--- a/wmavoice.c Wed Apr 21 17:57:48 2010 +0000 +++ b/wmavoice.c Wed Apr 21 18:01:34 2010 +0000 @@ -36,9 +36,13 @@ #include "acelp_filters.h" #include "lsp.h" #include "libavutil/lzo.h" +#include "avfft.h" +#include "fft.h" #define MAX_BLOCKS 8 ///< maximum number of blocks per frame #define MAX_LSPS 16 ///< maximum filter order +#define MAX_LSPS_ALIGN16 16 ///< same as #MAX_LSPS; needs to be multiple + ///< of 16 for ASM input buffer alignment #define MAX_FRAMES 3 ///< maximum number of frames per superframe #define MAX_FRAMESIZE 160 ///< maximum number of samples per frame #define MAX_SIGNAL_HISTORY 416 ///< maximum excitation signal history @@ -140,8 +144,15 @@ int history_nsamples; ///< number of samples in history for signal ///< prediction (through ACB) + /* postfilter specific values */ int do_apf; ///< whether to apply the averaged ///< projection filter (APF) + int denoise_strength; ///< strength of denoising in Wiener filter + ///< [0-11] + int denoise_tilt_corr; ///< Whether to apply tilt correction to the + ///< Wiener filter coefficients (postfilter) + int dc_level; ///< Predicted amount of DC noise, based + ///< on which a DC removal filter is used int lsps; ///< number of LSPs per frame [10 or 16] int lsp_q_mode; ///< defines quantizer defaults [0, 1] @@ -244,6 +255,34 @@ float synth_history[MAX_LSPS]; ///< see #excitation_history /** * @} + * @defgroup post_filter Postfilter values + * Varibales used for postfilter implementation, mostly history for + * smoothing and so on, and context variables for FFT/iFFT. + * @{ + */ + RDFTContext rdft, irdft; ///< contexts for FFT-calculation in the + ///< postfilter (for denoise filter) + DCTContext dct, dst; ///< contexts for phase shift (in Hilbert + ///< transform, part of postfilter) + float sin[511], cos[511]; ///< 8-bit cosine/sine windows over [-pi,pi] + ///< range + float postfilter_agc; ///< gain control memory, used in + ///< #adaptive_gain_control() + float dcf_mem[2]; ///< DC filter history + float zero_exc_pf[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE]; + ///< zero filter output (i.e. excitation) + ///< by postfilter + float denoise_filter_cache[MAX_FRAMESIZE]; + int denoise_filter_cache_size; ///< samples in #denoise_filter_cache + DECLARE_ALIGNED(16, float, tilted_lpcs_pf)[0x80]; + ///< aligned buffer for LPC tilting + DECLARE_ALIGNED(16, float, denoise_coeffs_pf)[0x80]; + ///< aligned buffer for denoise coefficients + DECLARE_ALIGNED(16, float, synth_filter_out_buf)[80 + MAX_LSPS_ALIGN16]; + ///< aligned buffer for postfilter speech + ///< synthesis + /** + * @} */ } WMAVoiceContext; @@ -313,6 +352,28 @@ flags = AV_RL32(ctx->extradata + 18); s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align); s->do_apf = flags & 0x1; + if (s->do_apf) { + ff_rdft_init(&s->rdft, 7, DFT_R2C); + ff_rdft_init(&s->irdft, 7, IDFT_C2R); + ff_dct_init(&s->dct, 6, DCT_I); + ff_dct_init(&s->dst, 6, DST_I); + + ff_sine_window_init(s->cos, 256); + memcpy(&s->sin[255], s->cos, 256 * sizeof(s->cos[0])); + for (n = 0; n < 255; n++) { + s->sin[n] = -s->sin[510 - n]; + s->cos[510 - n] = s->cos[n]; + } + } + s->denoise_strength = (flags >> 2) & 0xF; + if (s->denoise_strength >= 12) { + av_log(ctx, AV_LOG_ERROR, + "Invalid denoise filter strength %d (max=11)\n", + s->denoise_strength); + return -1; + } + s->denoise_tilt_corr = !!(flags & 0x40); + s->dc_level = (flags >> 7) & 0xF; s->lsp_q_mode = !!(flags & 0x2000); s->lsp_def_mode = !!(flags & 0x4000); lsp16_flag = flags & 0x1000; @@ -370,6 +431,366 @@ } /** + * @defgroup postfilter Postfilter functions + * Postfilter functions (gain control, wiener denoise filter, DC filter, + * kalman smoothening, plus surrounding code to wrap it) + * @{ + */ +/** + * Adaptive gain control (as used in postfilter). + * + * Identical to #ff_adaptive_gain_control() in acelp_vectors.c, except + * that the energy here is calculated using sum(abs(...)), whereas the + * other codecs (e.g. AMR-NB, SIPRO) use sqrt(dotproduct(...)). + * + * @param out output buffer for filtered samples + * @param in input buffer containing the samples as they are after the + * postfilter steps so far + * @param speech_synth input buffer containing speech synth before postfilter + * @param size input buffer size + * @param alpha exponential filter factor + * @param gain_mem pointer to filter memory (single float) + */ +static void adaptive_gain_control(float *out, const float *in, + const float *speech_synth, + int size, float alpha, float *gain_mem) +{ + int i; + float speech_energy = 0.0, postfilter_energy = 0.0, gain_scale_factor; + float mem = *gain_mem; + + for (i = 0; i < size; i++) { + speech_energy += fabsf(speech_synth[i]); + postfilter_energy += fabsf(in[i]); + } + gain_scale_factor = (1.0 - alpha) * speech_energy / postfilter_energy; + + for (i = 0; i < size; i++) { + mem = alpha * mem + gain_scale_factor; + out[i] = in[i] * mem; + } + + *gain_mem = mem; +} + +/** + * Kalman smoothing function. + * + * This function looks back pitch +/- 3 samples back into history to find + * the best fitting curve (that one giving the optimal gain of the two + * signals, i.e. the highest dot product between the two), and then + * uses that signal history to smoothen the output of the speech synthesis + * filter. + * + * @param s WMA Voice decoding context + * @param pitch pitch of the speech signal + * @param in input speech signal + * @param out output pointer for smoothened signal + * @param size input/output buffer size + * + * @returns -1 if no smoothening took place, e.g. because no optimal + * fit could be found, or 0 on success. + */ +static int kalman_smoothen(WMAVoiceContext *s, int pitch, + const float *in, float *out, int size) +{ + int n; + float optimal_gain = 0, dot; + const float *ptr = &in[-FFMAX(s->min_pitch_val, pitch - 3)], + *end = &in[-FFMIN(s->max_pitch_val, pitch + 3)], + *best_hist_ptr; + + /* find best fitting point in history */ + do { + dot = ff_dot_productf(in, ptr, size); + if (dot > optimal_gain) { + optimal_gain = dot; + best_hist_ptr = ptr; + } + } while (--ptr >= end); + + if (optimal_gain <= 0) + return -1; + dot = ff_dot_productf(best_hist_ptr, best_hist_ptr, size); + if (dot <= 0) // would be 1.0 + return -1; + + if (optimal_gain <= dot) { + dot = dot / (dot + 0.6 * optimal_gain); // 0.625-1.000 + } else + dot = 0.625; + + /* actual smoothing */ + for (n = 0; n < size; n++) + out[n] = best_hist_ptr[n] + dot * (in[n] - best_hist_ptr[n]); + + return 0; +} + +/** + * Get the tilt factor of a formant filter from its transfer function + * @see #tilt_factor() in amrnbdec.c, which does essentially the same, + * but somehow (??) it does a speech synthesis filter in the + * middle, which is missing here + * + * @param lpcs LPC coefficients + * @param n_lpcs Size of LPC buffer + * @returns the tilt factor + */ +static float tilt_factor(const float *lpcs, int n_lpcs) +{ + float rh0, rh1; + + rh0 = 1.0 + ff_dot_productf(lpcs, lpcs, n_lpcs); + rh1 = lpcs[0] + ff_dot_productf(lpcs, &lpcs[1], n_lpcs - 1); + + return rh1 / rh0; +} + +/** + * Derive denoise filter coefficients (in real domain) from the LPCs. + */ +static void calc_input_response(WMAVoiceContext *s, float *lpcs, + int fcb_type, float *coeffs, int remainder) +{ + float last_coeff, min = 15.0, max = -15.0; + float irange, angle_mul, gain_mul, range, sq; + int n, idx; + + /* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */ + ff_rdft_calc(&s->rdft, lpcs); +#define log_range(var, assign) do { \ + float tmp = log10f(assign); var = tmp; \ + max = FFMAX(max, tmp); min = FFMIN(min, tmp); \ + } while (0) + log_range(last_coeff, lpcs[1] * lpcs[1]); + for (n = 1; n < 64; n++) + log_range(lpcs[n], lpcs[n * 2] * lpcs[n * 2] + + lpcs[n * 2 + 1] * lpcs[n * 2 + 1]); + log_range(lpcs[0], lpcs[0] * lpcs[0]); +#undef log_range + range = max - min; + lpcs[64] = last_coeff; + + /* Now, use this spectrum to pick out these frequencies with higher + * (relative) power/energy (which we then take to be "not noise"), + * and set up a table (still in lpc[]) of (relative) gains per frequency. + * These frequencies will be maintained, while others ("noise") will be + * decreased in the filter output. */ + irange = 64.0 / range; // so irange*(max-value) is in the range [0, 63] + gain_mul = range * (fcb_type == FCB_TYPE_HARDCODED ? (5.0 / 13.0) : + (5.0 / 14.7)); + angle_mul = gain_mul * (8.0 * M_LN10 / M_PI); + for (n = 0; n <= 64; n++) { + float pow; + + idx = FFMAX(0, lrint((max - lpcs[n]) * irange) - 1); + pow = wmavoice_denoise_power_table[s->denoise_strength][idx]; + lpcs[n] = angle_mul * pow; + + /* 70.57 =~ 1/log10(1.0331663) */ + idx = (pow * gain_mul - 0.0295) * 70.570526123; + if (idx > 127) { // fallback if index falls outside table range + coeffs[n] = wmavoice_energy_table[127] * + powf(1.0331663, idx - 127); + } else + coeffs[n] = wmavoice_energy_table[FFMAX(0, idx)]; + } + + /* calculate the Hilbert transform of the gains, which we do (since this + * is a sinus input) by doing a phase shift (in theory, H(sin())=cos()). + * Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the + * "moment" of the LPCs in this filter. */ + ff_dct_calc(&s->dct, lpcs); + ff_dct_calc(&s->dst, lpcs); + + /* Split out the coefficient indexes into phase/magnitude pairs */ + idx = 255 + av_clip(lpcs[64], -255, 255); + coeffs[0] = coeffs[0] * s->cos[idx]; + idx = 255 + av_clip(lpcs[64] - 2 * lpcs[63], -255, 255); + last_coeff = coeffs[64] * s->cos[idx]; + for (n = 63;; n--) { + idx = 255 + av_clip(-lpcs[64] - 2 * lpcs[n - 1], -255, 255); + coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx]; + coeffs[n * 2] = coeffs[n] * s->cos[idx]; + + if (!--n) break; + + idx = 255 + av_clip( lpcs[64] - 2 * lpcs[n - 1], -255, 255); + coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx]; + coeffs[n * 2] = coeffs[n] * s->cos[idx]; + } + coeffs[1] = last_coeff; + + /* move into real domain */ + ff_rdft_calc(&s->irdft, coeffs); + + /* tilt correction and normalize scale */ + memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder)); + if (s->denoise_tilt_corr) { + float tilt_mem = 0; + + coeffs[remainder - 1] = 0; + ff_tilt_compensation(&tilt_mem, + -1.8 * tilt_factor(coeffs, remainder - 1), + coeffs, remainder); + } + sq = (1.0 / 64.0) * sqrtf(1 / ff_dot_productf(coeffs, coeffs, remainder)); + for (n = 0; n < remainder; n++) + coeffs[n] *= sq; +} + +/** + * This function applies a Wiener filter on the (noisy) speech signal as + * a means to denoise it. + * + * - take RDFT of LPCs to get the power spectrum of the noise + speech; + * - using this power spectrum, calculate (for each frequency) the Wiener + * filter gain, which depends on the frequency power and desired level + * of noise subtraction (when set too high, this leads to artifacts) + * We can do this symmetrically over the X-axis (so 0-4kHz is the inverse + * of 4-8kHz); + * - by doing a phase shift, calculate the Hilbert transform of this array + * of per-frequency filter-gains to get the filtering coefficients; + * - smoothen/normalize/de-tilt these filter coefficients as desired; + * - take RDFT of noisy sound, apply the coefficients and take its IRDFT + * to get the denoised speech signal; + * - the leftover (i.e. output of the IRDFT on denoised speech data beyond + * the frame boundary) are saved and applied to subsequent frames by an + * overlap-add method (otherwise you get clicking-artifacts). + * + * @param s WMA Voice decoding context + * @param s fcb_type Frame (codebook) type + * @param synth_pf input: the noisy speech signal, output: denoised speech + * data; should be 16-byte aligned (for ASM purposes) + * @param size size of the speech data + * @param lpcs LPCs used to synthesize this frame's speech data + */ +static void wiener_denoise(WMAVoiceContext *s, int fcb_type, + float *synth_pf, int size, + const float *lpcs) +{ + int remainder, lim, n; + + if (fcb_type != FCB_TYPE_SILENCE) { + float *tilted_lpcs = s->tilted_lpcs_pf, + *coeffs = s->denoise_coeffs_pf, tilt_mem = 0; + + tilted_lpcs[0] = 1.0; + memcpy(&tilted_lpcs[1], lpcs, sizeof(lpcs[0]) * s->lsps); + memset(&tilted_lpcs[s->lsps + 1], 0, + sizeof(tilted_lpcs[0]) * (128 - s->lsps - 1)); + ff_tilt_compensation(&tilt_mem, 0.7 * tilt_factor(lpcs, s->lsps), + tilted_lpcs, s->lsps + 2); + + /* The IRDFT output (127 samples for 7-bit filter) beyond the frame + * size is applied to the next frame. All input beyond this is zero, + * and thus all output beyond this will go towards zero, hence we can + * limit to min(size-1, 127-size) as a performance consideration. */ + remainder = FFMIN(127 - size, size - 1); + calc_input_response(s, tilted_lpcs, fcb_type, coeffs, remainder); + + /* apply coefficients (in frequency spectrum domain), i.e. complex + * number multiplication */ + memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size)); + ff_rdft_calc(&s->rdft, synth_pf); + ff_rdft_calc(&s->rdft, coeffs); + synth_pf[0] *= coeffs[0]; + synth_pf[1] *= coeffs[1]; + for (n = 1; n < 128; n++) { + float v1 = synth_pf[n * 2], v2 = synth_pf[n * 2 + 1]; + synth_pf[n * 2] = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1]; + synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1]; + } + ff_rdft_calc(&s->irdft, synth_pf); + } + + /* merge filter output with the history of previous runs */ + if (s->denoise_filter_cache_size) { + lim = FFMIN(s->denoise_filter_cache_size, size); + for (n = 0; n < lim; n++) + synth_pf[n] += s->denoise_filter_cache[n]; + s->denoise_filter_cache_size -= lim; + memmove(s->denoise_filter_cache, &s->denoise_filter_cache[size], + sizeof(s->denoise_filter_cache[0]) * s->denoise_filter_cache_size); + } + + /* move remainder of filter output into a cache for future runs */ + if (fcb_type != FCB_TYPE_SILENCE) { + lim = FFMIN(remainder, s->denoise_filter_cache_size); + for (n = 0; n < lim; n++) + s->denoise_filter_cache[n] += synth_pf[size + n]; + if (lim < remainder) { + memcpy(&s->denoise_filter_cache[lim], &synth_pf[size + lim], + sizeof(s->denoise_filter_cache[0]) * (remainder - lim)); + s->denoise_filter_cache_size = remainder; + } + } +} + +/** + * Averaging projection filter, the postfilter used in WMAVoice. + * + * This uses the following steps: + * - A zero-synthesis filter (generate excitation from synth signal) + * - Kalman smoothing on excitation, based on pitch + * - Re-synthesized smoothened output + * - Iterative Wiener denoise filter + * - Adaptive gain filter + * - DC filter + * + * @param s WMAVoice decoding context + * @param synth Speech synthesis output (before postfilter) + * @param samples Output buffer for filtered samples + * @param size Buffer size of synth & samples + * @param lpcs Generated LPCs used for speech synthesis + * @param fcb_type Frame type (silence, hardcoded, AW-pulses or FCB-pulses) + * @param pitch Pitch of the input signal + */ +static void postfilter(WMAVoiceContext *s, const float *synth, + float *samples, int size, + const float *lpcs, float *zero_exc_pf, + int fcb_type, int pitch) +{ + float synth_filter_in_buf[MAX_FRAMESIZE / 2], + *synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16], + *synth_filter_in = zero_exc_pf; + + assert(size <= MAX_FRAMESIZE / 2); + + /* generate excitation from input signal */ + ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps); + + if (fcb_type >= FCB_TYPE_AW_PULSES && + !kalman_smoothen(s, pitch, zero_exc_pf, synth_filter_in_buf, size)) + synth_filter_in = synth_filter_in_buf; + + /* re-synthesize speech after smoothening, and keep history */ + ff_celp_lp_synthesis_filterf(synth_pf, lpcs, + synth_filter_in, size, s->lsps); + memcpy(&synth_pf[-s->lsps], &synth_pf[size - s->lsps], + sizeof(synth_pf[0]) * s->lsps); + + wiener_denoise(s, fcb_type, synth_pf, size, lpcs); + + adaptive_gain_control(samples, synth_pf, synth, size, 0.99, + &s->postfilter_agc); + + if (s->dc_level > 8) { + /* remove ultra-low frequency DC noise / highpass filter; + * coefficients are identical to those used in SIPR decoding, + * and very closely resemble those used in AMR-NB decoding. */ + ff_acelp_apply_order_2_transfer_function(samples, samples, + (const float[2]) { -1.99997, 1.0 }, + (const float[2]) { -1.9330735188, 0.93589198496 }, + 0.93980580475, s->dcf_mem, size); + } +} +/** + * @} + */ + +/** * Dequantize LSPs * @param lsps output pointer to the array that will hold the LSPs * @param num number of LSPs to be dequantized @@ -980,6 +1401,7 @@ * * @param ctx WMA Voice decoder context * @param gb bit I/O context (s->gb or one for cross-packet superframes) + * @param frame_idx Frame number within superframe [0-2] * @param samples pointer to output sample buffer, has space for at least 160 * samples * @param lsps LSP array @@ -988,7 +1410,7 @@ * @param synth target buffer for synthesized speech data * @return 0 on success, <0 on error. */ -static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, +static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx, float *samples, const double *lsps, const double *prev_lsps, float *excitation, float *synth) @@ -1113,10 +1535,23 @@ /* Averaging projection filter, if applicable. Else, just copy samples * from synthesis buffer */ if (s->do_apf) { - // FIXME this is where APF would take place, currently not implemented - av_log_missing_feature(ctx, "APF", 0); - s->do_apf = 0; - } //else + double i_lsps[MAX_LSPS]; + float lpcs[MAX_LSPS]; + + for (n = 0; n < s->lsps; n++) // LSF -> LSP + i_lsps[n] = cos(0.5 * (prev_lsps[n] + lsps[n])); + ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); + postfilter(s, synth, samples, 80, lpcs, + &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx], + frame_descs[bd_idx].fcb_type, pitch[0]); + + for (n = 0; n < s->lsps; n++) // LSF -> LSP + i_lsps[n] = cos(lsps[n]); + ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); + postfilter(s, &synth[80], &samples[80], 80, lpcs, + &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx + 80], + frame_descs[bd_idx].fcb_type, pitch[0]); + } else memcpy(samples, synth, 160 * sizeof(synth[0])); /* Cache values for next frame */ @@ -1355,7 +1790,7 @@ stabilize_lsps(lsps[n], s->lsps); } - if ((res = synth_frame(ctx, gb, + if ((res = synth_frame(ctx, gb, n, &samples[n * MAX_FRAMESIZE], lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1], &excitation[s->history_nsamples + n * MAX_FRAMESIZE], @@ -1381,6 +1816,9 @@ s->lsps * sizeof(*synth)); memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE], s->history_nsamples * sizeof(*excitation)); + if (s->do_apf) + memmove(s->zero_exc_pf, &s->zero_exc_pf[MAX_SFRAMESIZE], + s->history_nsamples * sizeof(*s->zero_exc_pf)); return 0; } @@ -1535,11 +1973,26 @@ return size; } +static av_cold int wmavoice_decode_end(AVCodecContext *ctx) +{ + WMAVoiceContext *s = ctx->priv_data; + + if (s->do_apf) { + ff_rdft_end(&s->rdft); + ff_rdft_end(&s->irdft); + ff_dct_end(&s->dct); + ff_dct_end(&s->dst); + } + + return 0; +} + static av_cold void wmavoice_flush(AVCodecContext *ctx) { WMAVoiceContext *s = ctx->priv_data; int n; + s->postfilter_agc = 0; s->sframe_cache_size = 0; s->skip_bits_next = 0; for (n = 0; n < s->lsps; n++) @@ -1550,6 +2003,16 @@ sizeof(*s->synth_history) * MAX_LSPS); memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err)); + + if (s->do_apf) { + memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0, + sizeof(*s->synth_filter_out_buf) * s->lsps); + memset(s->dcf_mem, 0, + sizeof(*s->dcf_mem) * 2); + memset(s->zero_exc_pf, 0, + sizeof(*s->zero_exc_pf) * s->history_nsamples); + memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache)); + } } AVCodec wmavoice_decoder = { @@ -1559,7 +2022,7 @@ sizeof(WMAVoiceContext), wmavoice_decode_init, NULL, - NULL, + wmavoice_decode_end, wmavoice_decode_packet, CODEC_CAP_SUBFRAMES, .flush = wmavoice_flush,
--- a/wmavoice_data.h Wed Apr 21 17:57:48 2010 +0000 +++ b/wmavoice_data.h Wed Apr 21 18:01:34 2010 +0000 @@ -3020,4 +3020,240 @@ 0, -0.0273968070, -0.0392575669, -0.0276240534 }; +/** + * LUT for 1.071575641632 * pow(1.0331663, n - 127) + */ +static const float wmavoice_energy_table[128] = { + 0.0169982178, 0.0175619858, 0.0181444519, 0.0187462362, + 0.0193679795, 0.0200103437, 0.0206740128, 0.0213596933, + 0.0220681153, 0.0228000330, 0.0235562258, 0.0243374986, + 0.0251446834, 0.0259786395, 0.0268402549, 0.0277304468, + 0.0286501631, 0.0296003830, 0.0305821182, 0.0315964139, + 0.0326443501, 0.0337270424, 0.0348456436, 0.0360013446, + 0.0371953760, 0.0384290090, 0.0397035571, 0.0410203772, + 0.0423808713, 0.0437864880, 0.0452387238, 0.0467391249, + 0.0482892887, 0.0498908657, 0.0515455612, 0.0532551367, + 0.0550214125, 0.0568462692, 0.0587316496, 0.0606795611, + 0.0626920777, 0.0647713419, 0.0669195677, 0.0691390421, + 0.0714321284, 0.0738012678, 0.0762489827, 0.0787778794, + 0.0813906502, 0.0840900769, 0.0868790336, 0.0897604897, + 0.0927375130, 0.0958132732, 0.0989910450, 0.1022742117, + 0.1056662688, 0.1091708280, 0.1127916204, 0.1165325012, + 0.1203974531, 0.1243905911, 0.1285161668, 0.1327785725, + 0.1371823465, 0.1417321773, 0.1464329093, 0.1512895470, + 0.1563072616, 0.1614913951, 0.1668474671, 0.1723811803, + 0.1780984262, 0.1840052921, 0.1901080668, 0.1964132480, + 0.2029275487, 0.2096579046, 0.2166114816, 0.2237956830, + 0.2312181577, 0.2388868085, 0.2468098001, 0.2549955679, + 0.2634528274, 0.2721905830, 0.2812181375, 0.2905451026, + 0.3001814086, 0.3101373153, 0.3204234225, 0.3310506819, + 0.3420304081, 0.3533742912, 0.3650944090, 0.3772032397, + 0.3897136755, 0.4026390362, 0.4159930832, 0.4297900346, + 0.4440445799, 0.4587718956, 0.4739876619, 0.4897080789, + 0.5059498840, 0.5227303696, 0.5400674019, 0.5579794393, + 0.5764855528, 0.5956054456, 0.6153594745, 0.6357686714, + 0.6568547659, 0.6786402082, 0.7011481929, 0.7244026842, + 0.7484284410, 0.7732510432, 0.7988969192, 0.8253933741, + 0.8527686184, 0.8810517982, 0.9102730265, 0.9404634147, + 0.9716551065, 1.0038813113, 1.0371763400, 1.0715756416 +}; + +/** + * LUT for f(x,y) = pow((y + 6.9) / 64, 0.025 * (x + 1)). + */ +static const float wmavoice_denoise_power_table[12][64] = { + { 0.9458379339, 0.9490436287, 0.9518757236, 0.9544130754, + 0.9567118717, 0.9588135761, 0.9607496688, 0.9625446194, + 0.9642178285, 0.9657849396, 0.9672587526, 0.9686498743, + 0.9699671937, 0.9712182343, 0.9724094211, 0.9735462842, + 0.9746336187, 0.9756756090, 0.9766759291, 0.9776378218, + 0.9785641645, 0.9794575217, 0.9803201890, 0.9811542296, + 0.9819615045, 0.9827436985, 0.9835023412, 0.9842388263, + 0.9849544265, 0.9856503078, 0.9863275406, 0.9869871101, + 0.9876299254, 0.9882568267, 0.9888685922, 0.9894659445, + 0.9900495551, 0.9906200497, 0.9911780119, 0.9917239872, + 0.9922584859, 0.9927819864, 0.9932949377, 0.9937977618, + 0.9942908555, 0.9947745929, 0.9952493267, 0.9957153901, + 0.9961730980, 0.9966227482, 0.9970646231, 0.9974989903, + 0.9979261037, 0.9983462046, 0.9987595223, 0.9991662752, + 0.9995666709, 0.9999609077, 1.0003491745, 1.0007316515, + 1.0011085110, 1.0014799178, 1.0018460292, 1.0022069960 }, + { 0.8946093973, 0.9006838092, 0.9060673931, 0.9109043185, + 0.9152976055, 0.9193234737, 0.9230399260, 0.9264921443, + 0.9297160207, 0.9327405496, 0.9355894944, 0.9382825789, + 0.9408363568, 0.9432648587, 0.9455800822, 0.9477923675, + 0.9499106907, 0.9519428941, 0.9538958704, 0.9557757107, + 0.9575878241, 0.9593370368, 0.9610276730, 0.9626636222, + 0.9642483964, 0.9657851769, 0.9672768552, 0.9687260672, + 0.9701352224, 0.9715065293, 0.9728420173, 0.9741435556, + 0.9754128696, 0.9766515555, 0.9778610927, 0.9790428553, + 0.9801981216, 0.9813280829, 0.9824338513, 0.9835164667, + 0.9845769028, 0.9856160726, 0.9866348334, 0.9876339913, + 0.9886143053, 0.9895764906, 0.9905212223, 0.9914491381, + 0.9923608411, 0.9932569022, 0.9941378627, 0.9950042356, + 0.9958565084, 0.9966951442, 0.9975205834, 0.9983332454, + 0.9991335296, 0.9999218170, 1.0006984708, 1.0014638383, + 1.0022182509, 1.0029620257, 1.0036954662, 1.0044188628 }, + { 0.8461555040, 0.8547882305, 0.8624635555, 0.8693789920, + 0.8756760853, 0.8814598273, 0.8868103032, 0.8917900284, + 0.8964487626, 0.9008267754, 0.9049571273, 0.9088673021, + 0.9125804007, 0.9161160306, 0.9194909803, 0.9227197376, + 0.9258148939, 0.9287874629, 0.9316471355, 0.9344024839, + 0.9370611291, 0.9396298766, 0.9421148300, 0.9445214846, + 0.9468548060, 0.9491192967, 0.9513190517, 0.9534578074, + 0.9555389816, 0.9575657096, 0.9595408742, 0.9614671327, + 0.9633469396, 0.9651825670, 0.9669761222, 0.9687295635, + 0.9704447142, 0.9721232742, 0.9737668316, 0.9753768718, + 0.9769547868, 0.9785018824, 0.9800193854, 0.9815084500, + 0.9829701633, 0.9844055505, 0.9858155796, 0.9872011653, + 0.9885631734, 0.9899024236, 0.9912196934, 0.9925157203, + 0.9937912053, 0.9950468143, 0.9962831814, 0.9975009102, + 0.9987005760, 0.9998827277, 1.0010478892, 1.0021965608, + 1.0033292209, 1.0044463270, 1.0055483173, 1.0066356112 }, + { 0.8003259737, 0.8112313241, 0.8209581209, 0.8297466775, + 0.8377697066, 0.8451556492, 0.8520027051, 0.8583876935, + 0.8643718792, 0.8700049328, 0.8753277020, 0.8803741979, + 0.8851730502, 0.8897485937, 0.8941216918, 0.8983103719, + 0.9023303202, 0.9061952736, 0.9099173316, 0.9135072091, + 0.9169744409, 0.9203275502, 0.9235741882, 0.9267212496, + 0.9297749699, 0.9327410079, 0.9356245146, 0.9384301933, + 0.9411623497, 0.9438249364, 0.9464215906, 0.9489556668, + 0.9514302661, 0.9538482608, 0.9562123167, 0.9585249126, + 0.9607883576, 0.9630048062, 0.9651762722, 0.9673046403, + 0.9693916775, 0.9714390425, 0.9734482944, 0.9754209007, + 0.9773582446, 0.9792616307, 0.9811322918, 0.9829713934, + 0.9847800389, 0.9865592739, 0.9883100900, 0.9900334289, + 0.9917301853, 0.9934012104, 0.9950473143, 0.9966692689, + 0.9982678100, 0.9998436400, 1.0013974295, 1.0029298194, + 1.0044414224, 1.0059328250, 1.0074045889, 1.0088572520 }, + { 0.7569786654, 0.7698939195, 0.7814501054, 0.7919210783, + 0.8015042240, 0.8103467104, 0.8185613167, 0.8262364557, + 0.8334427763, 0.8402376615, 0.8466683811, 0.8527743561, + 0.8585888194, 0.8641400582, 0.8694523567, 0.8745467247, + 0.8794414652, 0.8841526254, 0.8886943552, 0.8930791981, + 0.8973183276, 0.9014217415, 0.9053984227, 0.9092564737, + 0.9130032283, 0.9166453478, 0.9201889007, 0.9236394320, + 0.9270020224, 0.9302813390, 0.9334816797, 0.9366070112, + 0.9396610028, 0.9426470554, 0.9455683275, 0.9484277579, + 0.9512280860, 0.9539718690, 0.9566614986, 0.9592992147, + 0.9618871182, 0.9644271823, 0.9669212630, 0.9693711079, + 0.9717783651, 0.9741445900, 0.9764712529, 0.9787597445, + 0.9810113822, 0.9832274148, 0.9854090274, 0.9875573457, + 0.9896734398, 0.9917583281, 0.9938129803, 0.9958383209, + 0.9978352315, 0.9998045539, 1.0017470919, 1.0036636145, + 1.0055548568, 1.0074215229, 1.0092642871, 1.0110837959 }, + { 0.7159791370, 0.7306629191, 0.7438433845, 0.7558198318, + 0.7668086064, 0.7769714272, 0.7864325139, 0.7952894548, + 0.8036203840, 0.8114888792, 0.8189474022, 0.8260397728, + 0.8328029877, 0.8392685815, 0.8454636629, 0.8514117142, + 0.8571332177, 0.8626461513, 0.8679663850, 0.8731080020, + 0.8780835596, 0.8829043049, 0.8875803529, 0.8921208349, + 0.8965340237, 0.9008274393, 0.9050079382, 0.9090817905, + 0.9130547454, 0.9169320882, 0.9207186893, 0.9244190474, + 0.9280373261, 0.9315773876, 0.9350428208, 0.9384369673, + 0.9417629433, 0.9450236603, 0.9482218422, 0.9513600421, + 0.9544406555, 0.9574659338, 0.9604379957, 0.9633588374, + 0.9662303420, 0.9690542879, 0.9718323569, 0.9745661408, + 0.9772571477, 0.9799068082, 0.9825164805, 0.9850874551, + 0.9876209597, 0.9901181627, 0.9925801775, 0.9950080658, + 0.9974028405, 0.9997654692, 1.0020968764, 1.0043979464, + 1.0066695255, 1.0089124239, 1.0111274185, 1.0133152537 }, + { 0.6772002277, 0.6934309881, 0.7080464599, 0.7213643301, + 0.7336148970, 0.7449707526, 0.7555647772, 0.7655015856, + 0.7748651015, 0.7837237382, 0.7921340426, 0.8001433220, + 0.8077915768, 0.8151129499, 0.8221368310, 0.8288887107, + 0.8353908496, 0.8416628090, 0.8477218755, 0.8535834053, + 0.8592611049, 0.8647672624, 0.8701129393, 0.8753081305, + 0.8803618988, 0.8852824894, 0.8900774261, 0.8947535945, + 0.8993173131, 0.9037743949, 0.9081302004, 0.9123896841, + 0.9165574352, 0.9206377129, 0.9246344779, 0.9285514202, + 0.9323919830, 0.9361593853, 0.9398566405, 0.9434865742, + 0.9470518396, 0.9505549317, 0.9539981992, 0.9573838564, + 0.9607139933, 0.9639905847, 0.9672154989, 0.9703905051, + 0.9735172803, 0.9765974162, 0.9796324243, 0.9826237418, + 0.9855727362, 0.9884807098, 0.9913489039, 0.9941785028, + 0.9969706369, 0.9997263861, 1.0024467831, 1.0051328157, + 1.0077854297, 1.0104055314, 1.0129939892, 1.0155516364 }, + { 0.6405216642, 0.6580962612, 0.6739722363, 0.6884795488, + 0.7018580813, 0.7142880714, 0.7259086094, 0.7368294324, + 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