mirror of
https://github.com/mpv-player/mpv
synced 2024-12-20 22:02:59 +00:00
7deec05ea0
Change the audio filters to use a double instead of rationals for the ratio of output to input size. The rationals could overflow when calculating the overall ratio of a filter chain and gave no real advantage compared to doubles. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24916 b3059339-0415-0410-9bf9-f77b7e298cf2
665 lines
20 KiB
C
665 lines
20 KiB
C
/* Experimental audio filter that mixes 5.1 and 5.1 with matrix
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encoded rear channels into headphone signal using FIR filtering
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with HRTF.
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*/
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//#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <inttypes.h>
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#include <math.h>
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#include "af.h"
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#include "dsp.h"
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/* HRTF filter coefficients and adjustable parameters */
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#include "af_hrtf.h"
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typedef struct af_hrtf_s {
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/* Lengths */
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int dlbuflen, hrflen, basslen;
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/* L, C, R, Ls, Rs channels */
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float *lf, *rf, *lr, *rr, *cf, *cr;
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float *cf_ir, *af_ir, *of_ir, *ar_ir, *or_ir, *cr_ir;
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int cf_o, af_o, of_o, ar_o, or_o, cr_o;
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/* Bass */
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float *ba_l, *ba_r;
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float *ba_ir;
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/* Whether to matrix decode the rear center channel */
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int matrix_mode;
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/* How to decode the input:
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0 = 5/5+1 channels
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1 = 2 channels
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2 = matrix encoded 2 channels */
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int decode_mode;
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/* Full wave rectified (FWR) amplitudes and gain used to steer the
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active matrix decoding of front channels (variable names
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lpr/lmr means Lt + Rt, Lt - Rt) */
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float l_fwr, r_fwr, lpr_fwr, lmr_fwr;
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float adapt_l_gain, adapt_r_gain, adapt_lpr_gain, adapt_lmr_gain;
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/* Matrix input decoding require special FWR buffer, since the
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decoding is done in place. */
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float *fwrbuf_l, *fwrbuf_r, *fwrbuf_lr, *fwrbuf_rr;
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/* Rear channel delay buffer for matrix decoding */
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float *rear_dlbuf;
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/* Full wave rectified amplitude and gain used to steer the active
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matrix decoding of center rear channel */
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float lr_fwr, rr_fwr, lrprr_fwr, lrmrr_fwr;
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float adapt_lr_gain, adapt_rr_gain;
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float adapt_lrprr_gain, adapt_lrmrr_gain;
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/* Cyclic position on the ring buffer */
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int cyc_pos;
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int print_flag;
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} af_hrtf_t;
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/* Convolution on a ring buffer
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* nx: length of the ring buffer
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* nk: length of the convolution kernel
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* sx: ring buffer
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* sk: convolution kernel
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* offset: offset on the ring buffer, can be
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*/
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static float conv(const int nx, const int nk, float *sx, float *sk,
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const int offset)
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{
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/* k = reminder of offset / nx */
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int k = offset >= 0 ? offset % nx : nx + (offset % nx);
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if(nk + k <= nx)
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return af_filter_fir(nk, sx + k, sk);
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else
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return af_filter_fir(nk + k - nx, sx, sk + nx - k) +
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af_filter_fir(nx - k, sx + k, sk);
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}
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/* Detect when the impulse response starts (significantly) */
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static int pulse_detect(float *sx)
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{
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/* nmax must be the reference impulse response length (128) minus
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s->hrflen */
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const int nmax = 128 - HRTFFILTLEN;
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const float thresh = IRTHRESH;
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int i;
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for(i = 0; i < nmax; i++)
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if(fabs(sx[i]) > thresh)
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return i;
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return 0;
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}
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/* Fuzzy matrix coefficient transfer function to "lock" the matrix on
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a effectively passive mode if the gain is approximately 1 */
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static inline float passive_lock(float x)
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{
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const float x1 = x - 1;
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const float ax1s = fabs(x - 1) * (1.0 / MATAGCLOCK);
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return x1 - x1 / (1 + ax1s * ax1s) + 1;
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}
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/* Unified active matrix decoder for 2 channel matrix encoded surround
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sources */
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static inline void matrix_decode(short *in, const int k, const int il,
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const int ir, const int decode_rear,
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const int dlbuflen,
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float l_fwr, float r_fwr,
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float lpr_fwr, float lmr_fwr,
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float *adapt_l_gain, float *adapt_r_gain,
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float *adapt_lpr_gain, float *adapt_lmr_gain,
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float *lf, float *rf, float *lr,
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float *rr, float *cf)
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{
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const int kr = (k + MATREARDELAY) % dlbuflen;
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float l_gain = (l_fwr + r_fwr) /
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(1 + l_fwr + l_fwr);
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float r_gain = (l_fwr + r_fwr) /
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(1 + r_fwr + r_fwr);
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/* The 2nd axis has strong gain fluctuations, and therefore require
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limits. The factor corresponds to the 1 / amplification of (Lt
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- Rt) when (Lt, Rt) is strongly correlated. (e.g. during
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dialogues). It should be bigger than -12 dB to prevent
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distortion. */
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float lmr_lim_fwr = lmr_fwr > M9_03DB * lpr_fwr ?
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lmr_fwr : M9_03DB * lpr_fwr;
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float lpr_gain = (lpr_fwr + lmr_lim_fwr) /
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(1 + lpr_fwr + lpr_fwr);
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float lmr_gain = (lpr_fwr + lmr_lim_fwr) /
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(1 + lmr_lim_fwr + lmr_lim_fwr);
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float lmr_unlim_gain = (lpr_fwr + lmr_fwr) /
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(1 + lmr_fwr + lmr_fwr);
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float lpr, lmr;
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float l_agc, r_agc, lpr_agc, lmr_agc;
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float f, d_gain, c_gain, c_agc_cfk;
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#if 0
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static int counter = 0;
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static FILE *fp_out;
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if(counter == 0)
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fp_out = fopen("af_hrtf.log", "w");
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if(counter % 240 == 0)
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fprintf(fp_out, "%g %g %g %g %g ", counter * (1.0 / 48000),
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l_gain, r_gain, lpr_gain, lmr_gain);
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#endif
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/*** AXIS NO. 1: (Lt, Rt) -> (C, Ls, Rs) ***/
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/* AGC adaption */
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d_gain = (fabs(l_gain - *adapt_l_gain) +
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fabs(r_gain - *adapt_r_gain)) * 0.5;
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f = d_gain * (1.0 / MATAGCTRIG);
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f = MATAGCDECAY - MATAGCDECAY / (1 + f * f);
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*adapt_l_gain = (1 - f) * *adapt_l_gain + f * l_gain;
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*adapt_r_gain = (1 - f) * *adapt_r_gain + f * r_gain;
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/* Matrix */
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l_agc = in[il] * passive_lock(*adapt_l_gain);
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r_agc = in[ir] * passive_lock(*adapt_r_gain);
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cf[k] = (l_agc + r_agc) * M_SQRT1_2;
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if(decode_rear) {
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lr[kr] = rr[kr] = (l_agc - r_agc) * M_SQRT1_2;
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/* Stereo rear channel is steered with the same AGC steering as
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the decoding matrix. Note this requires a fast updating AGC
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at the order of 20 ms (which is the case here). */
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lr[kr] *= (l_fwr + l_fwr) /
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(1 + l_fwr + r_fwr);
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rr[kr] *= (r_fwr + r_fwr) /
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(1 + l_fwr + r_fwr);
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}
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/*** AXIS NO. 2: (Lt + Rt, Lt - Rt) -> (L, R) ***/
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lpr = (in[il] + in[ir]) * M_SQRT1_2;
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lmr = (in[il] - in[ir]) * M_SQRT1_2;
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/* AGC adaption */
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d_gain = fabs(lmr_unlim_gain - *adapt_lmr_gain);
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f = d_gain * (1.0 / MATAGCTRIG);
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f = MATAGCDECAY - MATAGCDECAY / (1 + f * f);
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*adapt_lpr_gain = (1 - f) * *adapt_lpr_gain + f * lpr_gain;
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*adapt_lmr_gain = (1 - f) * *adapt_lmr_gain + f * lmr_gain;
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/* Matrix */
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lpr_agc = lpr * passive_lock(*adapt_lpr_gain);
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lmr_agc = lmr * passive_lock(*adapt_lmr_gain);
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lf[k] = (lpr_agc + lmr_agc) * M_SQRT1_2;
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rf[k] = (lpr_agc - lmr_agc) * M_SQRT1_2;
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/*** CENTER FRONT CANCELLATION ***/
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/* A heuristic approach exploits that Lt + Rt gain contains the
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information about Lt, Rt correlation. This effectively reshapes
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the front and rear "cones" to concentrate Lt + Rt to C and
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introduce Lt - Rt in L, R. */
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/* 0.67677 is the emprical lower bound for lpr_gain. */
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c_gain = 8 * (*adapt_lpr_gain - 0.67677);
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c_gain = c_gain > 0 ? c_gain : 0;
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/* c_gain should not be too high, not even reaching full
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cancellation (~ 0.50 - 0.55 at current AGC implementation), or
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the center will s0und too narrow. */
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c_gain = MATCOMPGAIN / (1 + c_gain * c_gain);
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c_agc_cfk = c_gain * cf[k];
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lf[k] -= c_agc_cfk;
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rf[k] -= c_agc_cfk;
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cf[k] += c_agc_cfk + c_agc_cfk;
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#if 0
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if(counter % 240 == 0)
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fprintf(fp_out, "%g %g %g %g %g\n",
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*adapt_l_gain, *adapt_r_gain,
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*adapt_lpr_gain, *adapt_lmr_gain,
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c_gain);
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counter++;
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#endif
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}
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static inline void update_ch(af_hrtf_t *s, short *in, const int k)
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{
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const int fwr_pos = (k + FWRDURATION) % s->dlbuflen;
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/* Update the full wave rectified total amplitude */
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/* Input matrix decoder */
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if(s->decode_mode == HRTF_MIX_MATRIX2CH) {
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s->l_fwr += abs(in[0]) - fabs(s->fwrbuf_l[fwr_pos]);
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s->r_fwr += abs(in[1]) - fabs(s->fwrbuf_r[fwr_pos]);
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s->lpr_fwr += abs(in[0] + in[1]) -
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fabs(s->fwrbuf_l[fwr_pos] + s->fwrbuf_r[fwr_pos]);
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s->lmr_fwr += abs(in[0] - in[1]) -
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fabs(s->fwrbuf_l[fwr_pos] - s->fwrbuf_r[fwr_pos]);
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}
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/* Rear matrix decoder */
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if(s->matrix_mode) {
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s->lr_fwr += abs(in[2]) - fabs(s->fwrbuf_lr[fwr_pos]);
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s->rr_fwr += abs(in[3]) - fabs(s->fwrbuf_rr[fwr_pos]);
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s->lrprr_fwr += abs(in[2] + in[3]) -
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fabs(s->fwrbuf_lr[fwr_pos] + s->fwrbuf_rr[fwr_pos]);
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s->lrmrr_fwr += abs(in[2] - in[3]) -
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fabs(s->fwrbuf_lr[fwr_pos] - s->fwrbuf_rr[fwr_pos]);
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}
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switch (s->decode_mode) {
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case HRTF_MIX_51:
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/* 5/5+1 channel sources */
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s->lf[k] = in[0];
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s->cf[k] = in[4];
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s->rf[k] = in[1];
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s->fwrbuf_lr[k] = s->lr[k] = in[2];
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s->fwrbuf_rr[k] = s->rr[k] = in[3];
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break;
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case HRTF_MIX_MATRIX2CH:
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/* Matrix encoded 2 channel sources */
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s->fwrbuf_l[k] = in[0];
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s->fwrbuf_r[k] = in[1];
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matrix_decode(in, k, 0, 1, 1, s->dlbuflen,
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s->l_fwr, s->r_fwr,
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s->lpr_fwr, s->lmr_fwr,
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&(s->adapt_l_gain), &(s->adapt_r_gain),
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&(s->adapt_lpr_gain), &(s->adapt_lmr_gain),
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s->lf, s->rf, s->lr, s->rr, s->cf);
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break;
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case HRTF_MIX_STEREO:
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/* Stereo sources */
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s->lf[k] = in[0];
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s->rf[k] = in[1];
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s->cf[k] = s->lr[k] = s->rr[k] = 0;
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break;
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}
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/* We need to update the bass compensation delay line, too. */
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s->ba_l[k] = in[0] + in[4] + in[2];
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s->ba_r[k] = in[4] + in[1] + in[3];
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}
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/* Initialization and runtime control */
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static int control(struct af_instance_s *af, int cmd, void* arg)
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{
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af_hrtf_t *s = af->setup;
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int test_output_res;
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char mode;
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switch(cmd) {
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case AF_CONTROL_REINIT:
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af->data->rate = ((af_data_t*)arg)->rate;
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if(af->data->rate != 48000) {
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// automatic samplerate adjustment in the filter chain
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// is not yet supported.
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af_msg(AF_MSG_ERROR,
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"[hrtf] ERROR: Sampling rate is not 48000 Hz (%d)!\n",
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af->data->rate);
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return AF_ERROR;
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}
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af->data->nch = ((af_data_t*)arg)->nch;
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if(af->data->nch == 2) {
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/* 2 channel input */
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if(s->decode_mode != HRTF_MIX_MATRIX2CH) {
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/* Default behavior is stereo mixing. */
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s->decode_mode = HRTF_MIX_STEREO;
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}
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}
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else if (af->data->nch < 5)
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af->data->nch = 5;
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af->data->format = AF_FORMAT_S16_NE;
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af->data->bps = 2;
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test_output_res = af_test_output(af, (af_data_t*)arg);
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af->mul = 2.0 / af->data->nch;
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// after testing input set the real output format
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af->data->nch = 2;
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s->print_flag = 1;
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return test_output_res;
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case AF_CONTROL_COMMAND_LINE:
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sscanf((char*)arg, "%c", &mode);
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switch(mode) {
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case 'm':
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/* Use matrix rear decoding. */
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s->matrix_mode = 1;
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break;
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case 's':
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/* Input needs matrix decoding. */
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s->decode_mode = HRTF_MIX_MATRIX2CH;
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break;
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case '0':
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s->matrix_mode = 0;
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break;
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default:
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af_msg(AF_MSG_ERROR,
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"[hrtf] Mode is neither 'm', 's', nor '0' (%c).\n",
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mode);
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return AF_ERROR;
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}
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s->print_flag = 1;
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return AF_OK;
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}
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return AF_UNKNOWN;
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}
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/* Deallocate memory */
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static void uninit(struct af_instance_s *af)
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{
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if(af->setup) {
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af_hrtf_t *s = af->setup;
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if(s->lf)
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free(s->lf);
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if(s->rf)
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free(s->rf);
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if(s->lr)
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free(s->lr);
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if(s->rr)
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free(s->rr);
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if(s->cf)
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free(s->cf);
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if(s->cr)
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free(s->cr);
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if(s->ba_l)
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free(s->ba_l);
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if(s->ba_r)
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free(s->ba_r);
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if(s->ba_ir)
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free(s->ba_ir);
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if(s->fwrbuf_l)
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free(s->fwrbuf_l);
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if(s->fwrbuf_r)
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free(s->fwrbuf_r);
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if(s->fwrbuf_lr)
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free(s->fwrbuf_lr);
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if(s->fwrbuf_rr)
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free(s->fwrbuf_rr);
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free(af->setup);
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}
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if(af->data)
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free(af->data->audio);
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free(af->data);
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}
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/* Filter data through filter
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Two "tricks" are used to compensate the "color" of the KEMAR data:
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1. The KEMAR data is refiltered to ensure that the front L, R channels
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on the same side of the ear are equalized (especially in the high
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frequencies).
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2. A bass compensation is introduced to ensure that 0-200 Hz are not
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damped (without any real 3D acoustical image, however).
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*/
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static af_data_t* play(struct af_instance_s *af, af_data_t *data)
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{
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af_hrtf_t *s = af->setup;
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short *in = data->audio; // Input audio data
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short *out = NULL; // Output audio data
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short *end = in + data->len / sizeof(short); // Loop end
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float common, left, right, diff, left_b, right_b;
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const int dblen = s->dlbuflen, hlen = s->hrflen, blen = s->basslen;
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if(AF_OK != RESIZE_LOCAL_BUFFER(af, data))
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return NULL;
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if(s->print_flag) {
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s->print_flag = 0;
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switch (s->decode_mode) {
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case HRTF_MIX_51:
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af_msg(AF_MSG_INFO,
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"[hrtf] Using HRTF to mix %s discrete surround into "
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"L, R channels\n", s->matrix_mode ? "5+1" : "5");
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break;
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case HRTF_MIX_STEREO:
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af_msg(AF_MSG_INFO,
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"[hrtf] Using HRTF to mix stereo into "
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"L, R channels\n");
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break;
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case HRTF_MIX_MATRIX2CH:
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af_msg(AF_MSG_INFO,
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"[hrtf] Using active matrix to decode 2 channel "
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"input, HRTF to mix %s matrix surround into "
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"L, R channels\n", "3/2");
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break;
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default:
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af_msg(AF_MSG_WARN,
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"[hrtf] bogus decode_mode: %d\n", s->decode_mode);
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break;
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}
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if(s->matrix_mode)
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af_msg(AF_MSG_INFO,
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"[hrtf] Using active matrix to decode rear center "
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"channel\n");
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}
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out = af->data->audio;
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/* MPlayer's 5 channel layout (notation for the variable):
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*
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* 0: L (LF), 1: R (RF), 2: Ls (LR), 3: Rs (RR), 4: C (CF), matrix
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* encoded: Cs (CR)
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*
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* or: L = left, C = center, R = right, F = front, R = rear
|
|
*
|
|
* Filter notation:
|
|
*
|
|
* CF
|
|
* OF AF
|
|
* Ear->
|
|
* OR AR
|
|
* CR
|
|
*
|
|
* or: C = center, A = same side, O = opposite, F = front, R = rear
|
|
*/
|
|
|
|
while(in < end) {
|
|
const int k = s->cyc_pos;
|
|
|
|
update_ch(s, in, k);
|
|
|
|
/* Simulate a 7.5 ms -20 dB echo of the center channel in the
|
|
front channels (like reflection from a room wall) - a kind of
|
|
psycho-acoustically "cheating" to focus the center front
|
|
channel, which is normally hard to be perceived as front */
|
|
s->lf[k] += CFECHOAMPL * s->cf[(k + CFECHODELAY) % s->dlbuflen];
|
|
s->rf[k] += CFECHOAMPL * s->cf[(k + CFECHODELAY) % s->dlbuflen];
|
|
|
|
switch (s->decode_mode) {
|
|
case HRTF_MIX_51:
|
|
case HRTF_MIX_MATRIX2CH:
|
|
/* Mixer filter matrix */
|
|
common = conv(dblen, hlen, s->cf, s->cf_ir, k + s->cf_o);
|
|
if(s->matrix_mode) {
|
|
/* In matrix decoding mode, the rear channel gain must be
|
|
renormalized, as there is an additional channel. */
|
|
matrix_decode(in, k, 2, 3, 0, s->dlbuflen,
|
|
s->lr_fwr, s->rr_fwr,
|
|
s->lrprr_fwr, s->lrmrr_fwr,
|
|
&(s->adapt_lr_gain), &(s->adapt_rr_gain),
|
|
&(s->adapt_lrprr_gain), &(s->adapt_lrmrr_gain),
|
|
s->lr, s->rr, NULL, NULL, s->cr);
|
|
common +=
|
|
conv(dblen, hlen, s->cr, s->cr_ir, k + s->cr_o) *
|
|
M1_76DB;
|
|
left =
|
|
( conv(dblen, hlen, s->lf, s->af_ir, k + s->af_o) +
|
|
conv(dblen, hlen, s->rf, s->of_ir, k + s->of_o) +
|
|
(conv(dblen, hlen, s->lr, s->ar_ir, k + s->ar_o) +
|
|
conv(dblen, hlen, s->rr, s->or_ir, k + s->or_o)) *
|
|
M1_76DB + common);
|
|
right =
|
|
( conv(dblen, hlen, s->rf, s->af_ir, k + s->af_o) +
|
|
conv(dblen, hlen, s->lf, s->of_ir, k + s->of_o) +
|
|
(conv(dblen, hlen, s->rr, s->ar_ir, k + s->ar_o) +
|
|
conv(dblen, hlen, s->lr, s->or_ir, k + s->or_o)) *
|
|
M1_76DB + common);
|
|
} else {
|
|
left =
|
|
( conv(dblen, hlen, s->lf, s->af_ir, k + s->af_o) +
|
|
conv(dblen, hlen, s->rf, s->of_ir, k + s->of_o) +
|
|
conv(dblen, hlen, s->lr, s->ar_ir, k + s->ar_o) +
|
|
conv(dblen, hlen, s->rr, s->or_ir, k + s->or_o) +
|
|
common);
|
|
right =
|
|
( conv(dblen, hlen, s->rf, s->af_ir, k + s->af_o) +
|
|
conv(dblen, hlen, s->lf, s->of_ir, k + s->of_o) +
|
|
conv(dblen, hlen, s->rr, s->ar_ir, k + s->ar_o) +
|
|
conv(dblen, hlen, s->lr, s->or_ir, k + s->or_o) +
|
|
common);
|
|
}
|
|
break;
|
|
case HRTF_MIX_STEREO:
|
|
left =
|
|
( conv(dblen, hlen, s->lf, s->af_ir, k + s->af_o) +
|
|
conv(dblen, hlen, s->rf, s->of_ir, k + s->of_o));
|
|
right =
|
|
( conv(dblen, hlen, s->rf, s->af_ir, k + s->af_o) +
|
|
conv(dblen, hlen, s->lf, s->of_ir, k + s->of_o));
|
|
break;
|
|
default:
|
|
/* make gcc happy */
|
|
left = 0.0;
|
|
right = 0.0;
|
|
break;
|
|
}
|
|
|
|
/* Bass compensation for the lower frequency cut of the HRTF. A
|
|
cross talk of the left and right channel is introduced to
|
|
match the directional characteristics of higher frequencies.
|
|
The bass will not have any real 3D perception, but that is
|
|
OK (note at 180 Hz, the wavelength is about 2 m, and any
|
|
spatial perception is impossible). */
|
|
left_b = conv(dblen, blen, s->ba_l, s->ba_ir, k);
|
|
right_b = conv(dblen, blen, s->ba_r, s->ba_ir, k);
|
|
left += (1 - BASSCROSS) * left_b + BASSCROSS * right_b;
|
|
right += (1 - BASSCROSS) * right_b + BASSCROSS * left_b;
|
|
/* Also mix the LFE channel (if available) */
|
|
if(data->nch >= 6) {
|
|
left += in[5] * M3_01DB;
|
|
right += in[5] * M3_01DB;
|
|
}
|
|
|
|
/* Amplitude renormalization. */
|
|
left *= AMPLNORM;
|
|
right *= AMPLNORM;
|
|
|
|
switch (s->decode_mode) {
|
|
case HRTF_MIX_51:
|
|
case HRTF_MIX_STEREO:
|
|
/* "Cheating": linear stereo expansion to amplify the 3D
|
|
perception. Note: Too much will destroy the acoustic space
|
|
and may even result in headaches. */
|
|
diff = STEXPAND2 * (left - right);
|
|
out[0] = (int16_t)(left + diff);
|
|
out[1] = (int16_t)(right - diff);
|
|
break;
|
|
case HRTF_MIX_MATRIX2CH:
|
|
/* Do attempt any stereo expansion with matrix encoded
|
|
sources. The L, R channels are already stereo expanded
|
|
by the steering, any further stereo expansion will sound
|
|
very unnatural. */
|
|
out[0] = (int16_t)left;
|
|
out[1] = (int16_t)right;
|
|
break;
|
|
}
|
|
|
|
/* Next sample... */
|
|
in = &in[data->nch];
|
|
out = &out[af->data->nch];
|
|
(s->cyc_pos)--;
|
|
if(s->cyc_pos < 0)
|
|
s->cyc_pos += dblen;
|
|
}
|
|
|
|
/* Set output data */
|
|
data->audio = af->data->audio;
|
|
data->len = data->len / data->nch * 2;
|
|
data->nch = 2;
|
|
|
|
return data;
|
|
}
|
|
|
|
static int allocate(af_hrtf_t *s)
|
|
{
|
|
if ((s->lf = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
|
|
if ((s->rf = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
|
|
if ((s->lr = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
|
|
if ((s->rr = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
|
|
if ((s->cf = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
|
|
if ((s->cr = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
|
|
if ((s->ba_l = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
|
|
if ((s->ba_r = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
|
|
if ((s->fwrbuf_l =
|
|
malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
|
|
if ((s->fwrbuf_r =
|
|
malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
|
|
if ((s->fwrbuf_lr =
|
|
malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
|
|
if ((s->fwrbuf_rr =
|
|
malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
|
|
return 0;
|
|
}
|
|
|
|
/* Allocate memory and set function pointers */
|
|
static int af_open(af_instance_t* af)
|
|
{
|
|
int i;
|
|
af_hrtf_t *s;
|
|
float fc;
|
|
|
|
af->control = control;
|
|
af->uninit = uninit;
|
|
af->play = play;
|
|
af->mul = 1;
|
|
af->data = calloc(1, sizeof(af_data_t));
|
|
af->setup = calloc(1, sizeof(af_hrtf_t));
|
|
if((af->data == NULL) || (af->setup == NULL))
|
|
return AF_ERROR;
|
|
|
|
s = af->setup;
|
|
|
|
s->dlbuflen = DELAYBUFLEN;
|
|
s->hrflen = HRTFFILTLEN;
|
|
s->basslen = BASSFILTLEN;
|
|
|
|
s->cyc_pos = s->dlbuflen - 1;
|
|
/* With a full (two axis) steering matrix decoder, s->matrix_mode
|
|
should not be enabled lightly (it will also steer the Ls, Rs
|
|
channels). */
|
|
s->matrix_mode = 0;
|
|
s->decode_mode = HRTF_MIX_51;
|
|
|
|
s->print_flag = 1;
|
|
|
|
if (allocate(s) != 0) {
|
|
af_msg(AF_MSG_ERROR, "[hrtf] Memory allocation error.\n");
|
|
return AF_ERROR;
|
|
}
|
|
|
|
for(i = 0; i < s->dlbuflen; i++)
|
|
s->lf[i] = s->rf[i] = s->lr[i] = s->rr[i] = s->cf[i] =
|
|
s->cr[i] = 0;
|
|
|
|
s->lr_fwr =
|
|
s->rr_fwr = 0;
|
|
|
|
s->cf_ir = cf_filt + (s->cf_o = pulse_detect(cf_filt));
|
|
s->af_ir = af_filt + (s->af_o = pulse_detect(af_filt));
|
|
s->of_ir = of_filt + (s->of_o = pulse_detect(of_filt));
|
|
s->ar_ir = ar_filt + (s->ar_o = pulse_detect(ar_filt));
|
|
s->or_ir = or_filt + (s->or_o = pulse_detect(or_filt));
|
|
s->cr_ir = cr_filt + (s->cr_o = pulse_detect(cr_filt));
|
|
|
|
if((s->ba_ir = malloc(s->basslen * sizeof(float))) == NULL) {
|
|
af_msg(AF_MSG_ERROR, "[hrtf] Memory allocation error.\n");
|
|
return AF_ERROR;
|
|
}
|
|
fc = 2.0 * BASSFILTFREQ / (float)af->data->rate;
|
|
if(af_filter_design_fir(s->basslen, s->ba_ir, &fc, LP | KAISER, 4 * M_PI) ==
|
|
-1) {
|
|
af_msg(AF_MSG_ERROR, "[hrtf] Unable to design low-pass "
|
|
"filter.\n");
|
|
return AF_ERROR;
|
|
}
|
|
for(i = 0; i < s->basslen; i++)
|
|
s->ba_ir[i] *= BASSGAIN;
|
|
|
|
return AF_OK;
|
|
}
|
|
|
|
/* Description of this filter */
|
|
af_info_t af_info_hrtf = {
|
|
"HRTF Headphone",
|
|
"hrtf",
|
|
"ylai",
|
|
"",
|
|
AF_FLAGS_REENTRANT,
|
|
af_open
|
|
};
|