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-rw-r--r--audio/filter/af_hrtf.c670
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diff --git a/audio/filter/af_hrtf.c b/audio/filter/af_hrtf.c
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index 3c8a89c665..0000000000
--- a/audio/filter/af_hrtf.c
+++ /dev/null
@@ -1,670 +0,0 @@
-/*
- * Experimental audio filter that mixes 5.1 and 5.1 with matrix
- * encoded rear channels into headphone signal using FIR filtering
- * with HRTF.
- *
- * Author: ylai
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv. If not, see <http://www.gnu.org/licenses/>.
- */
-
-//#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <inttypes.h>
-
-#include <math.h>
-#include <libavutil/common.h>
-
-#include "af.h"
-#include "dsp.h"
-
-/* HRTF filter coefficients and adjustable parameters */
-#include "af_hrtf.h"
-
-typedef struct af_hrtf_s {
- /* Lengths */
- int dlbuflen, hrflen, basslen;
- /* L, C, R, Ls, Rs channels */
- float *lf, *rf, *lr, *rr, *cf, *cr;
- const float *cf_ir, *af_ir, *of_ir, *ar_ir, *or_ir, *cr_ir;
- int cf_o, af_o, of_o, ar_o, or_o, cr_o;
- /* Bass */
- float *ba_l, *ba_r;
- float *ba_ir;
- /* Whether to matrix decode the rear center channel */
- int matrix_mode;
- /* How to decode the input:
- 0 = 5/5+1 channels
- 1 = 2 channels
- 2 = matrix encoded 2 channels */
- int decode_mode;
- /* Full wave rectified (FWR) amplitudes and gain used to steer the
- active matrix decoding of front channels (variable names
- lpr/lmr means Lt + Rt, Lt - Rt) */
- float l_fwr, r_fwr, lpr_fwr, lmr_fwr;
- float adapt_l_gain, adapt_r_gain, adapt_lpr_gain, adapt_lmr_gain;
- /* Matrix input decoding require special FWR buffer, since the
- decoding is done in place. */
- float *fwrbuf_l, *fwrbuf_r, *fwrbuf_lr, *fwrbuf_rr;
- /* Rear channel delay buffer for matrix decoding */
- float *rear_dlbuf;
- /* Full wave rectified amplitude and gain used to steer the active
- matrix decoding of center rear channel */
- float lr_fwr, rr_fwr, lrprr_fwr, lrmrr_fwr;
- float adapt_lr_gain, adapt_rr_gain;
- float adapt_lrprr_gain, adapt_lrmrr_gain;
- /* Cyclic position on the ring buffer */
- int cyc_pos;
- int print_flag;
- int mode;
-} af_hrtf_t;
-
-/* Convolution on a ring buffer
- * nx: length of the ring buffer
- * nk: length of the convolution kernel
- * sx: ring buffer
- * sk: convolution kernel
- * offset: offset on the ring buffer, can be
- */
-static float conv(const int nx, const int nk, const float *sx, const float *sk,
- const int offset)
-{
- /* k = reminder of offset / nx */
- int k = offset >= 0 ? offset % nx : nx + (offset % nx);
-
- if(nk + k <= nx)
- return af_filter_fir(nk, sx + k, sk);
- else
- return af_filter_fir(nk + k - nx, sx, sk + nx - k) +
- af_filter_fir(nx - k, sx + k, sk);
-}
-
-/* Detect when the impulse response starts (significantly) */
-static int pulse_detect(const float *sx)
-{
- /* nmax must be the reference impulse response length (128) minus
- s->hrflen */
- const int nmax = 128 - HRTFFILTLEN;
- const float thresh = IRTHRESH;
- int i;
-
- for(i = 0; i < nmax; i++)
- if(fabs(sx[i]) > thresh)
- return i;
- return 0;
-}
-
-/* Fuzzy matrix coefficient transfer function to "lock" the matrix on
- a effectively passive mode if the gain is approximately 1 */
-static inline float passive_lock(float x)
-{
- const float x1 = x - 1;
- const float ax1s = fabs(x - 1) * (1.0 / MATAGCLOCK);
-
- return x1 - x1 / (1 + ax1s * ax1s) + 1;
-}
-
-/* Unified active matrix decoder for 2 channel matrix encoded surround
- sources */
-static inline void matrix_decode(short *in, const int k, const int il,
- const int ir, const int decode_rear,
- const int dlbuflen,
- float l_fwr, float r_fwr,
- float lpr_fwr, float lmr_fwr,
- float *adapt_l_gain, float *adapt_r_gain,
- float *adapt_lpr_gain, float *adapt_lmr_gain,
- float *lf, float *rf, float *lr,
- float *rr, float *cf)
-{
- const int kr = (k + MATREARDELAY) % dlbuflen;
- float l_gain = (l_fwr + r_fwr) /
- (1 + l_fwr + l_fwr);
- float r_gain = (l_fwr + r_fwr) /
- (1 + r_fwr + r_fwr);
- /* The 2nd axis has strong gain fluctuations, and therefore require
- limits. The factor corresponds to the 1 / amplification of (Lt
- - Rt) when (Lt, Rt) is strongly correlated. (e.g. during
- dialogues). It should be bigger than -12 dB to prevent
- distortion. */
- float lmr_lim_fwr = lmr_fwr > M9_03DB * lpr_fwr ?
- lmr_fwr : M9_03DB * lpr_fwr;
- float lpr_gain = (lpr_fwr + lmr_lim_fwr) /
- (1 + lpr_fwr + lpr_fwr);
- float lmr_gain = (lpr_fwr + lmr_lim_fwr) /
- (1 + lmr_lim_fwr + lmr_lim_fwr);
- float lmr_unlim_gain = (lpr_fwr + lmr_fwr) /
- (1 + lmr_fwr + lmr_fwr);
- float lpr, lmr;
- float l_agc, r_agc, lpr_agc, lmr_agc;
- float f, d_gain, c_gain, c_agc_cfk;
-
-#if 0
- static int counter = 0;
- static FILE *fp_out;
-
- if(counter == 0)
- fp_out = fopen("af_hrtf.log", "w");
- if(counter % 240 == 0)
- fprintf(fp_out, "%g %g %g %g %g ", counter * (1.0 / 48000),
- l_gain, r_gain, lpr_gain, lmr_gain);
-#endif
-
- /*** AXIS NO. 1: (Lt, Rt) -> (C, Ls, Rs) ***/
- /* AGC adaption */
- d_gain = (fabs(l_gain - *adapt_l_gain) +
- fabs(r_gain - *adapt_r_gain)) * 0.5;
- f = d_gain * (1.0 / MATAGCTRIG);
- f = MATAGCDECAY - MATAGCDECAY / (1 + f * f);
- *adapt_l_gain = (1 - f) * *adapt_l_gain + f * l_gain;
- *adapt_r_gain = (1 - f) * *adapt_r_gain + f * r_gain;
- /* Matrix */
- l_agc = in[il] * passive_lock(*adapt_l_gain);
- r_agc = in[ir] * passive_lock(*adapt_r_gain);
- cf[k] = (l_agc + r_agc) * M_SQRT1_2;
- if(decode_rear) {
- lr[kr] = rr[kr] = (l_agc - r_agc) * M_SQRT1_2;
- /* Stereo rear channel is steered with the same AGC steering as
- the decoding matrix. Note this requires a fast updating AGC
- at the order of 20 ms (which is the case here). */
- lr[kr] *= (l_fwr + l_fwr) /
- (1 + l_fwr + r_fwr);
- rr[kr] *= (r_fwr + r_fwr) /
- (1 + l_fwr + r_fwr);
- }
-
- /*** AXIS NO. 2: (Lt + Rt, Lt - Rt) -> (L, R) ***/
- lpr = (in[il] + in[ir]) * M_SQRT1_2;
- lmr = (in[il] - in[ir]) * M_SQRT1_2;
- /* AGC adaption */
- d_gain = fabs(lmr_unlim_gain - *adapt_lmr_gain);
- f = d_gain * (1.0 / MATAGCTRIG);
- f = MATAGCDECAY - MATAGCDECAY / (1 + f * f);
- *adapt_lpr_gain = (1 - f) * *adapt_lpr_gain + f * lpr_gain;
- *adapt_lmr_gain = (1 - f) * *adapt_lmr_gain + f * lmr_gain;
- /* Matrix */
- lpr_agc = lpr * passive_lock(*adapt_lpr_gain);
- lmr_agc = lmr * passive_lock(*adapt_lmr_gain);
- lf[k] = (lpr_agc + lmr_agc) * M_SQRT1_2;
- rf[k] = (lpr_agc - lmr_agc) * M_SQRT1_2;
-
- /*** CENTER FRONT CANCELLATION ***/
- /* A heuristic approach exploits that Lt + Rt gain contains the
- information about Lt, Rt correlation. This effectively reshapes
- the front and rear "cones" to concentrate Lt + Rt to C and
- introduce Lt - Rt in L, R. */
- /* 0.67677 is the empirical lower bound for lpr_gain. */
- c_gain = 8 * (*adapt_lpr_gain - 0.67677);
- c_gain = c_gain > 0 ? c_gain : 0;
- /* c_gain should not be too high, not even reaching full
- cancellation (~ 0.50 - 0.55 at current AGC implementation), or
- the center will s0und too narrow. */
- c_gain = MATCOMPGAIN / (1 + c_gain * c_gain);
- c_agc_cfk = c_gain * cf[k];
- lf[k] -= c_agc_cfk;
- rf[k] -= c_agc_cfk;
- cf[k] += c_agc_cfk + c_agc_cfk;
-#if 0
- if(counter % 240 == 0)
- fprintf(fp_out, "%g %g %g %g %g\n",
- *adapt_l_gain, *adapt_r_gain,
- *adapt_lpr_gain, *adapt_lmr_gain,
- c_gain);
- counter++;
-#endif
-}
-
-static inline void update_ch(af_hrtf_t *s, short *in, const int k)
-{
- const int fwr_pos = (k + FWRDURATION) % s->dlbuflen;
- /* Update the full wave rectified total amplitude */
- /* Input matrix decoder */
- if(s->decode_mode == HRTF_MIX_MATRIX2CH) {
- s->l_fwr += abs(in[0]) - fabs(s->fwrbuf_l[fwr_pos]);
- s->r_fwr += abs(in[1]) - fabs(s->fwrbuf_r[fwr_pos]);
- s->lpr_fwr += abs(in[0] + in[1]) -
- fabs(s->fwrbuf_l[fwr_pos] + s->fwrbuf_r[fwr_pos]);
- s->lmr_fwr += abs(in[0] - in[1]) -
- fabs(s->fwrbuf_l[fwr_pos] - s->fwrbuf_r[fwr_pos]);
- }
- /* Rear matrix decoder */
- if(s->matrix_mode) {
- s->lr_fwr += abs(in[2]) - fabs(s->fwrbuf_lr[fwr_pos]);
- s->rr_fwr += abs(in[3]) - fabs(s->fwrbuf_rr[fwr_pos]);
- s->lrprr_fwr += abs(in[2] + in[3]) -
- fabs(s->fwrbuf_lr[fwr_pos] + s->fwrbuf_rr[fwr_pos]);
- s->lrmrr_fwr += abs(in[2] - in[3]) -
- fabs(s->fwrbuf_lr[fwr_pos] - s->fwrbuf_rr[fwr_pos]);
- }
-
- switch (s->decode_mode) {
- case HRTF_MIX_51:
- /* 5/5+1 channel sources */
- s->lf[k] = in[0];
- s->cf[k] = in[4];
- s->rf[k] = in[1];
- s->fwrbuf_lr[k] = s->lr[k] = in[2];
- s->fwrbuf_rr[k] = s->rr[k] = in[3];
- break;
- case HRTF_MIX_MATRIX2CH:
- /* Matrix encoded 2 channel sources */
- s->fwrbuf_l[k] = in[0];
- s->fwrbuf_r[k] = in[1];
- matrix_decode(in, k, 0, 1, 1, s->dlbuflen,
- s->l_fwr, s->r_fwr,
- s->lpr_fwr, s->lmr_fwr,
- &(s->adapt_l_gain), &(s->adapt_r_gain),
- &(s->adapt_lpr_gain), &(s->adapt_lmr_gain),
- s->lf, s->rf, s->lr, s->rr, s->cf);
- break;
- case HRTF_MIX_STEREO:
- /* Stereo sources */
- s->lf[k] = in[0];
- s->rf[k] = in[1];
- s->cf[k] = s->lr[k] = s->rr[k] = 0;
- break;
- }
-
- /* We need to update the bass compensation delay line, too. */
- // TODO: should this use lf/cf/rf etc. instead?
- s->ba_l[k] = in[0];
- s->ba_r[k] = in[1];
- if (s->decode_mode == HRTF_MIX_51) {
- s->ba_l[k] += in[4] + in[2];
- s->ba_r[k] += in[4] + in[3];
- }
-}
-
-static void clear_coeff(af_hrtf_t *s, float *c)
-{
- memset(c, 0, s->dlbuflen * sizeof(float));
-}
-
-static void reset(af_hrtf_t *s)
-{
- clear_coeff(s, s->lf);
- clear_coeff(s, s->rf);
- clear_coeff(s, s->lr);
- clear_coeff(s, s->rr);
- clear_coeff(s, s->cf);
- clear_coeff(s, s->cr);
- clear_coeff(s, s->ba_l);
- clear_coeff(s, s->ba_r);
- clear_coeff(s, s->fwrbuf_l);
- clear_coeff(s, s->fwrbuf_r);
- clear_coeff(s, s->fwrbuf_lr);
- clear_coeff(s, s->fwrbuf_rr);
-}
-
-/* Initialization and runtime control */
-static int control(struct af_instance *af, int cmd, void* arg)
-{
- af_hrtf_t *s = af->priv;
- int test_output_res;
-
- switch(cmd) {
- case AF_CONTROL_REINIT:
- reset(s);
- af->data->rate = 48000;
- mp_audio_set_channels_old(af->data, ((struct mp_audio*)arg)->nch);
- if(af->data->nch == 2) {
- /* 2 channel input */
- if(s->decode_mode != HRTF_MIX_MATRIX2CH) {
- /* Default behavior is stereo mixing. */
- s->decode_mode = HRTF_MIX_STEREO;
- }
- } else if (af->data->nch < 5) {
- mp_audio_set_channels_old(af->data, 5);
- }
- mp_audio_set_format(af->data, AF_FORMAT_S16);
- test_output_res = af_test_output(af, (struct mp_audio*)arg);
- // after testing input set the real output format
- mp_audio_set_num_channels(af->data, 2);
- s->print_flag = 1;
- return test_output_res;
- case AF_CONTROL_RESET:
- reset(s);
- return AF_OK;
- }
-
- return AF_UNKNOWN;
-}
-
-/* Deallocate memory */
-static void uninit(struct af_instance *af)
-{
- af_hrtf_t *s = af->priv;
-
- free(s->lf);
- free(s->rf);
- free(s->lr);
- free(s->rr);
- free(s->cf);
- free(s->cr);
- free(s->ba_l);
- free(s->ba_r);
- free(s->ba_ir);
- free(s->fwrbuf_l);
- free(s->fwrbuf_r);
- free(s->fwrbuf_lr);
- free(s->fwrbuf_rr);
-}
-
-/* Filter data through filter
-
-Two "tricks" are used to compensate the "color" of the KEMAR data:
-
-1. The KEMAR data is refiltered to ensure that the front L, R channels
-on the same side of the ear are equalized (especially in the high
-frequencies).
-
-2. A bass compensation is introduced to ensure that 0-200 Hz are not
-damped (without any real 3D acoustical image, however).
-*/
-static int filter(struct af_instance *af, struct mp_audio *data)
-{
- af_hrtf_t *s = af->priv;
-
- if (!data)
- return 0;
- struct mp_audio *outframe =
- mp_audio_pool_get(af->out_pool, &af->fmt_out, data->samples);
- if (!outframe) {
- talloc_free(data);
- return -1;
- }
- mp_audio_copy_attributes(outframe, data);
-
- short *in = data->planes[0]; // Input audio data
- short *out = outframe->planes[0]; // Output audio data
- short *end = in + data->samples * data->nch; // Loop end
- float common, left, right, diff, left_b, right_b;
- const int dblen = s->dlbuflen, hlen = s->hrflen, blen = s->basslen;
-
- if(s->print_flag) {
- s->print_flag = 0;
- switch (s->decode_mode) {
- case HRTF_MIX_51:
- MP_INFO(af, "Using HRTF to mix %s discrete surround into "
- "L, R channels\n", s->matrix_mode ? "5+1" : "5");
- break;
- case HRTF_MIX_STEREO:
- MP_INFO(af, "Using HRTF to mix stereo into "
- "L, R channels\n");
- break;
- case HRTF_MIX_MATRIX2CH:
- MP_INFO(af, "Using active matrix to decode 2 channel "
- "input, HRTF to mix %s matrix surround into "
- "L, R channels\n", "3/2");
- break;
- default:
- MP_WARN(af, "bogus decode_mode: %d\n", s->decode_mode);
- break;
- }
-
- if(s->matrix_mode)
- MP_INFO(af, "Using active matrix to decode rear center "
- "channel\n");
- }
-
- /* MPlayer's 5 channel layout (notation for the variable):
- *
- * 0: L (LF), 1: R (RF), 2: Ls (LR), 3: Rs (RR), 4: C (CF), matrix
- * encoded: Cs (CR)
- *
- * 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] = av_clip_int16(left + diff);
- out[1] = av_clip_int16(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] = av_clip_int16(left);
- out[1] = av_clip_int16(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;
- }
-
- talloc_free(data);
- af_add_output_frame(af, outframe);
- return 0;
-}
-
-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(struct af_instance* af)
-{
- int i;
- af_hrtf_t *s;
- float fc;
-
- af->control = control;
- af->uninit = uninit;
- af->filter_frame = filter;
-
- s = af->priv;
-
- 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;
-
- switch (s->mode) {
- case 0: /* Use matrix rear decoding. */
- s->matrix_mode = 1;
- break;
- case 1: /* Input needs matrix decoding. */
- s->decode_mode = HRTF_MIX_MATRIX2CH;
- break;
- case 2:
- s->matrix_mode = 0;
- break;
- }
-
- s->print_flag = 1;
-
- if (allocate(s) != 0) {
- MP_ERR(af, "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) {
- MP_ERR(af, "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) {
- MP_ERR(af, "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;
-}
-
-#define OPT_BASE_STRUCT af_hrtf_t
-const struct af_info af_info_hrtf = {
- .info = "HRTF Headphone",
- .name = "hrtf",
- .open = af_open,
- .priv_size = sizeof(af_hrtf_t),
- .options = (const struct m_option[]) {
- OPT_CHOICE("mode", mode, 0, ({"m", 0}, {"s", 1}, {"0", 2})),
- {0}
- },
-};
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-27 15:23:19 +0000