summaryrefslogtreecommitdiffstats
path: root/audio
diff options
context:
space:
mode:
Diffstat (limited to 'audio')
-rw-r--r--audio/audio.c7
-rw-r--r--audio/audio.h1
-rw-r--r--audio/filter/af.c23
-rw-r--r--audio/filter/af_center.c104
-rw-r--r--audio/filter/af_export.c237
-rw-r--r--audio/filter/af_extrastereo.c132
-rw-r--r--audio/filter/af_hrtf.c670
-rw-r--r--audio/filter/af_hrtf.h510
-rw-r--r--audio/filter/af_karaoke.c86
-rw-r--r--audio/filter/af_ladspa.c851
-rw-r--r--audio/filter/af_sinesuppress.c117
-rw-r--r--audio/filter/af_sub.c148
-rw-r--r--audio/filter/af_surround.c246
-rw-r--r--audio/filter/af_sweep.c92
-rw-r--r--audio/filter/dsp.h31
-rw-r--r--audio/filter/filter.c359
-rw-r--r--audio/filter/filter.h74
-rw-r--r--audio/filter/window.c212
-rw-r--r--audio/filter/window.h42
19 files changed, 0 insertions, 3942 deletions
diff --git a/audio/audio.c b/audio/audio.c
index f84d6054bc..4b12992879 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -58,13 +58,6 @@ void mp_audio_set_num_channels(struct mp_audio *mpa, int num_channels)
update_redundant_info(mpa);
}
-// Use old MPlayer/ALSA channel layout.
-void mp_audio_set_channels_old(struct mp_audio *mpa, int num_channels)
-{
- mp_chmap_from_channels_alsa(&mpa->channels, num_channels);
- update_redundant_info(mpa);
-}
-
void mp_audio_set_channels(struct mp_audio *mpa, const struct mp_chmap *chmap)
{
mpa->channels = *chmap;
diff --git a/audio/audio.h b/audio/audio.h
index a0ecb2d7bf..bf5358274a 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -46,7 +46,6 @@ struct mp_audio {
void mp_audio_set_format(struct mp_audio *mpa, int format);
void mp_audio_set_num_channels(struct mp_audio *mpa, int num_channels);
-void mp_audio_set_channels_old(struct mp_audio *mpa, int num_channels);
void mp_audio_set_channels(struct mp_audio *mpa, const struct mp_chmap *chmap);
void mp_audio_copy_config(struct mp_audio *dst, const struct mp_audio *src);
bool mp_audio_config_equals(const struct mp_audio *a, const struct mp_audio *b);
diff --git a/audio/filter/af.c b/audio/filter/af.c
index 6a5b1f42a5..b877ba7661 100644
--- a/audio/filter/af.c
+++ b/audio/filter/af.c
@@ -34,23 +34,12 @@
extern const struct af_info af_info_delay;
extern const struct af_info af_info_channels;
extern const struct af_info af_info_format;
-extern const struct af_info af_info_force;
extern const struct af_info af_info_volume;
extern const struct af_info af_info_equalizer;
extern const struct af_info af_info_pan;
-extern const struct af_info af_info_surround;
-extern const struct af_info af_info_sub;
-extern const struct af_info af_info_export;
extern const struct af_info af_info_drc;
-extern const struct af_info af_info_extrastereo;
extern const struct af_info af_info_lavcac3enc;
extern const struct af_info af_info_lavrresample;
-extern const struct af_info af_info_sweep;
-extern const struct af_info af_info_hrtf;
-extern const struct af_info af_info_ladspa;
-extern const struct af_info af_info_center;
-extern const struct af_info af_info_sinesuppress;
-extern const struct af_info af_info_karaoke;
extern const struct af_info af_info_scaletempo;
extern const struct af_info af_info_bs2b;
extern const struct af_info af_info_lavfi;
@@ -63,24 +52,12 @@ static const struct af_info *const filter_list[] = {
&af_info_volume,
&af_info_equalizer,
&af_info_pan,
- &af_info_surround,
- &af_info_sub,
- &af_info_export,
&af_info_drc,
- &af_info_extrastereo,
&af_info_lavcac3enc,
&af_info_lavrresample,
- &af_info_sweep,
- &af_info_hrtf,
-#if HAVE_LADSPA
- &af_info_ladspa,
-#endif
#if HAVE_RUBBERBAND
&af_info_rubberband,
#endif
- &af_info_center,
- &af_info_sinesuppress,
- &af_info_karaoke,
&af_info_scaletempo,
#if HAVE_LIBBS2B
&af_info_bs2b,
diff --git a/audio/filter/af_center.c b/audio/filter/af_center.c
deleted file mode 100644
index 69e54e81c6..0000000000
--- a/audio/filter/af_center.c
+++ /dev/null
@@ -1,104 +0,0 @@
-/*
- * This filter adds a center channel to the audio stream by
- * averaging the left and right channel.
- * There are two runtime controls one for setting which channel
- * to insert the center-audio into called AF_CONTROL_SUB_CH.
- *
- * FIXME: implement a high-pass filter for better results.
- *
- * copyright (c) 2005 Alex Beregszaszi
- *
- * 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 "common/common.h"
-#include "af.h"
-
-// Data for specific instances of this filter
-typedef struct af_center_s
-{
- int ch; // Channel number which to insert the filtered data
-}af_center_t;
-
-// Initialization and runtime control
-static int control(struct af_instance* af, int cmd, void* arg)
-{
- af_center_t* s = af->priv;
-
- switch(cmd){
- case AF_CONTROL_REINIT:{
- // Sanity check
- if(!arg) return AF_ERROR;
-
- af->data->rate = ((struct mp_audio*)arg)->rate;
- mp_audio_set_channels_old(af->data, MPMAX(s->ch+1,((struct mp_audio*)arg)->nch));
- mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
-
- return af_test_output(af,(struct mp_audio*)arg);
- }
- }
- return AF_UNKNOWN;
-}
-
-static int filter_frame(struct af_instance* af, struct mp_audio* data)
-{
- if (!data)
- return 0;
- if (af_make_writeable(af, data) < 0) {
- talloc_free(data);
- return -1;
- }
- struct mp_audio* c = data; // Current working data
- af_center_t* s = af->priv; // Setup for this instance
- float* a = c->planes[0]; // Audio data
- int nch = c->nch; // Number of channels
- int len = c->samples*c->nch; // Number of samples in current audio block
- int ch = s->ch; // Channel in which to insert the center audio
- register int i;
-
- // Run filter
- for(i=0;i<len;i+=nch){
- // Average left and right
- a[i+ch] = (a[i]/2) + (a[i+1]/2);
- }
-
- af_add_output_frame(af, data);
- return 0;
-}
-
-// Allocate memory and set function pointers
-static int af_open(struct af_instance* af){
- af->control=control;
- af->filter_frame = filter_frame;
- return AF_OK;
-}
-
-#define OPT_BASE_STRUCT af_center_t
-const struct af_info af_info_center = {
- .info = "Audio filter for adding a center channel",
- .name = "center",
- .flags = AF_FLAGS_NOT_REENTRANT,
- .open = af_open,
- .priv_size = sizeof(af_center_t),
- .options = (const struct m_option[]) {
- OPT_INTRANGE("channel", ch, 0, 0, AF_NCH - 1, OPTDEF_INT(1)),
- {0}
- },
-};
diff --git a/audio/filter/af_export.c b/audio/filter/af_export.c
deleted file mode 100644
index 6020d9d98e..0000000000
--- a/audio/filter/af_export.c
+++ /dev/null
@@ -1,237 +0,0 @@
-/*
- * This audio filter exports the incoming signal to other processes
- * using memory mapping. The memory mapped area contains a header:
- * int nch,
- * int size,
- * unsigned long long counter (updated every time the contents of
- * the area changes),
- * the rest is payload (non-interleaved).
- *
- * Authors: Anders; Gustavo Sverzut Barbieri <gustavo.barbieri@ic.unicamp.br>
- *
- * 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 <unistd.h>
-#include "config.h"
-
-#include <sys/types.h>
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <fcntl.h>
-
-#include "osdep/io.h"
-
-#include "talloc.h"
-#include "af.h"
-#include "options/path.h"
-
-#define DEF_SZ 512 // default buffer size (in samples)
-#define SHARED_FILE "mpv-af_export" /* default file name
- (relative to ~/.mpv/ */
-
-#define SIZE_HEADER (2 * sizeof(int) + sizeof(unsigned long long))
-
-// Data for specific instances of this filter
-typedef struct af_export_s
-{
- unsigned long long count; // Used for sync
- void* buf[AF_NCH]; // Buffers for storing the data before it is exported
- int sz; // Size of buffer in samples
- int wi; // Write index
- int fd; // File descriptor to shared memory area
- char* filename; // File to export data
- uint8_t *mmap_area; // MMap shared area
-} af_export_t;
-
-
-/* Initialization and runtime control
- af audio filter instance
- cmd control command
- arg argument
-*/
-static int control(struct af_instance* af, int cmd, void* arg)
-{
- af_export_t* s = af->priv;
- switch (cmd){
- case AF_CONTROL_REINIT:{
- int i=0;
- int mapsize;
-
- // Free previous buffers
- free(s->buf[0]);
-
- // unmap previous area
- if(s->mmap_area)
- munmap(s->mmap_area, SIZE_HEADER + (af->data->bps*s->sz*af->data->nch));
- // close previous file descriptor
- if(s->fd)
- close(s->fd);
-
- // Accept only int16_t as input format (which sucks)
- mp_audio_copy_config(af->data, (struct mp_audio*)arg);
- mp_audio_set_format(af->data, AF_FORMAT_S16);
-
- // Allocate new buffers (as one continuous block)
- s->buf[0] = calloc(s->sz*af->data->nch, af->data->bps);
- if(NULL == s->buf[0]) {
- MP_FATAL(af, "Out of memory\n");
- return AF_ERROR;
- }
- for(i = 1; i < af->data->nch; i++)
- s->buf[i] = (uint8_t *)s->buf[0] + i*s->sz*af->data->bps;
-
- if (!s->filename) {
- MP_FATAL(af, "No filename set.\n");
- return AF_ERROR;
- }
-
- // Init memory mapping
- s->fd = open(s->filename, O_RDWR | O_CREAT | O_TRUNC | O_CLOEXEC, 0640);
- MP_INFO(af, "Exporting to file: %s\n", s->filename);
- if(s->fd < 0) {
- MP_FATAL(af, "Could not open/create file: %s\n",
- s->filename);
- return AF_ERROR;
- }
-
- // header + buffer
- mapsize = (SIZE_HEADER + (af->data->bps * s->sz * af->data->nch));
-
- // grow file to needed size
- for(i = 0; i < mapsize; i++){
- char null = 0;
- write(s->fd, (void*) &null, 1);
- }
-
- // mmap size
- s->mmap_area = mmap(0, mapsize, PROT_READ|PROT_WRITE,MAP_SHARED, s->fd, 0);
- if(s->mmap_area == NULL)
- MP_FATAL(af, "Could not mmap file %s\n", s->filename);
- MP_INFO(af, "Memory mapped to file: %s (%p)\n",
- s->filename, s->mmap_area);
-
- // Initialize header
- *((int*)s->mmap_area) = af->data->nch;
- *((int*)s->mmap_area + 1) = s->sz * af->data->bps * af->data->nch;
- msync(s->mmap_area, mapsize, MS_ASYNC);
-
- // Use test_output to return FALSE if necessary
- return af_test_output(af, (struct mp_audio*)arg);
- }
- }
- return AF_UNKNOWN;
-}
-
-/* Free allocated memory and clean up other stuff too.
- af audio filter instance
-*/
-static void uninit( struct af_instance* af )
-{
- af_export_t* s = af->priv;
-
- free(s->buf[0]);
-
- // Free mmaped area
- if(s->mmap_area)
- munmap(s->mmap_area, sizeof(af_export_t));
-
- if(s->fd > -1)
- close(s->fd);
-}
-
-/* Filter data through filter
- af audio filter instance
- data audio data
-*/
-static int filter(struct af_instance *af, struct mp_audio *data)
-{
- if (!data)
- return 0;
- struct mp_audio* c = data; // Current working data
- af_export_t* s = af->priv; // Setup for this instance
- int16_t* a = c->planes[0]; // Incoming sound
- int nch = c->nch; // Number of channels
- int len = c->samples*c->nch; // Number of sample in data chunk
- int sz = s->sz; // buffer size (in samples)
- int flag = 0; // Set to 1 if buffer is filled
-
- int ch, i;
-
- // Fill all buffers
- for(ch = 0; ch < nch; ch++){
- int wi = s->wi; // Reset write index
- int16_t* b = s->buf[ch]; // Current buffer
-
- // Copy data to export buffers
- for(i = ch; i < len; i += nch){
- b[wi++] = a[i];
- if(wi >= sz){ // Don't write outside the end of the buffer
- flag = 1;
- break;
- }
- }
- s->wi = wi % s->sz;
- }
-
- // Export buffer to mmaped area
- if(flag){
- // update buffer in mapped area
- memcpy(s->mmap_area + SIZE_HEADER, s->buf[0], sz * c->bps * nch);
- s->count++; // increment counter (to sync)
- memcpy(s->mmap_area + SIZE_HEADER - sizeof(s->count),
- &(s->count), sizeof(s->count));
- }
-
- af_add_output_frame(af, data);
- return 0;
-}
-
-/* Allocate memory and set function pointers
- af audio filter instance
- returns AF_OK or AF_ERROR
-*/
-static int af_open( struct af_instance* af )
-{
- af->control = control;
- af->uninit = uninit;
- af->filter_frame = filter;
- af_export_t *priv = af->priv;
-
- if (!priv->filename || !priv->filename[0]) {
- MP_FATAL(af, "no export filename given");
- return AF_ERROR;
- }
-
- return AF_OK;
-}
-
-#define OPT_BASE_STRUCT af_export_t
-const struct af_info af_info_export = {
- .info = "Sound export filter",
- .name = "export",
- .open = af_open,
- .priv_size = sizeof(af_export_t),
- .options = (const struct m_option[]) {
- OPT_STRING("filename", filename, 0),
- OPT_INTRANGE("buffersamples", sz, 0, 1, 2048, OPTDEF_INT(DEF_SZ)),
- {0}
- },
-};
diff --git a/audio/filter/af_extrastereo.c b/audio/filter/af_extrastereo.c
deleted file mode 100644
index 49222ebfdc..0000000000
--- a/audio/filter/af_extrastereo.c
+++ /dev/null
@@ -1,132 +0,0 @@
-/*
- * Copyright (C) 2004 Alex Beregszaszi & Pierre Lombard
- *
- * 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 <limits.h>
-
-#include "common/common.h"
-#include "af.h"
-
-// Data for specific instances of this filter
-typedef struct af_extrastereo_s
-{
- float mul;
-}af_extrastereo_t;
-
-// Initialization and runtime control
-static int control(struct af_instance* af, int cmd, void* arg)
-{
- switch(cmd){
- case AF_CONTROL_REINIT:{
- // Sanity check
- if(!arg) return AF_ERROR;
-
- mp_audio_copy_config(af->data, (struct mp_audio*)arg);
- mp_audio_force_interleaved_format(af->data);
- mp_audio_set_num_channels(af->data, 2);
- if (af->data->format != AF_FORMAT_FLOAT)
- mp_audio_set_format(af->data, AF_FORMAT_S16);
-
- return af_test_output(af,(struct mp_audio*)arg);
- }
- }
- return AF_UNKNOWN;
-}
-
-// Filter data through filter
-static void play_s16(af_extrastereo_t *s, struct mp_audio* data)
-{
- register int i = 0;
- int16_t *a = (int16_t*)data->planes[0]; // Audio data
- int len = data->samples*data->nch; // Number of samples
- int avg, l, r;
-
- for (i = 0; i < len; i+=2)
- {
- avg = (a[i] + a[i + 1]) / 2;
-
- l = avg + (int)(s->mul * (a[i] - avg));
- r = avg + (int)(s->mul * (a[i + 1] - avg));
-
- a[i] = MPCLAMP(l, SHRT_MIN, SHRT_MAX);
- a[i + 1] = MPCLAMP(r, SHRT_MIN, SHRT_MAX);
- }
-}
-
-static void play_float(af_extrastereo_t *s, struct mp_audio* data)
-{
- register int i = 0;
- float *a = (float*)data->planes[0]; // Audio data
- int len = data->samples * data->nch; // Number of samples
- float avg, l, r;
-
- for (i = 0; i < len; i+=2)
- {
- avg = (a[i] + a[i + 1]) / 2;
-
- l = avg + (s->mul * (a[i] - avg));
- r = avg + (s->mul * (a[i + 1] - avg));
-
- a[i] = af_softclip(l);
- a[i + 1] = af_softclip(r);
- }
-}
-
-static int filter_frame(struct af_instance *af, struct mp_audio *data)
-{
- if (!data)
- return 0;
- if (af_make_writeable(af, data) < 0) {
- talloc_free(data);
- return -1;
- }
- if (data->format == AF_FORMAT_FLOAT) {
- play_float(af->priv, data);
- } else {
- play_s16(af->priv, data);
- }
- af_add_output_frame(af, data);
- return 0;
-}
-
-// Allocate memory and set function pointers
-static int af_open(struct af_instance* af){
- af->control=control;
- af->filter_frame = filter_frame;
-
- return AF_OK;
-}
-
-#define OPT_BASE_STRUCT af_extrastereo_t
-const struct af_info af_info_extrastereo = {
- .info = "Increase difference between audio channels",
- .name = "extrastereo",
- .flags = AF_FLAGS_NOT_REENTRANT,
- .open = af_open,
- .priv_size = sizeof(af_extrastereo_t),
- .options = (const struct m_option[]) {
- OPT_FLOAT("mul", mul, 0, OPTDEF_FLOAT(2.5)),
- {0}
- },
-};
diff --git a/audio/filter/af_hrtf.c b/audio/filter/af_hrtf.c
deleted file mode 100644
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;
-