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-rw-r--r--audio/filter/af_center.c12
-rw-r--r--audio/filter/af_channels.c34
-rw-r--r--audio/filter/af_delay.c46
-rw-r--r--audio/filter/af_drc.c36
-rw-r--r--audio/filter/af_dummy.c2
-rw-r--r--audio/filter/af_equalizer.c86
-rw-r--r--audio/filter/af_export.c44
-rw-r--r--audio/filter/af_extrastereo.c12
-rw-r--r--audio/filter/af_hrtf.c452
-rw-r--r--audio/filter/af_hrtf.h50
-rw-r--r--audio/filter/af_karaoke.c56
-rw-r--r--audio/filter/af_pan.c20
-rw-r--r--audio/filter/af_sinesuppress.c16
-rw-r--r--audio/filter/af_sub.c30
-rw-r--r--audio/filter/af_surround.c36
-rw-r--r--audio/filter/dsp.h4
-rw-r--r--audio/filter/equalizer.h26
-rw-r--r--audio/filter/filter.c72
-rw-r--r--audio/filter/filter.h2
-rw-r--r--audio/filter/tools.c6
-rw-r--r--audio/filter/window.c6
21 files changed, 524 insertions, 524 deletions
diff --git a/audio/filter/af_center.c b/audio/filter/af_center.c
index 2e626bffb6..106e08412e 100644
--- a/audio/filter/af_center.c
+++ b/audio/filter/af_center.c
@@ -35,7 +35,7 @@
// Data for specific instances of this filter
typedef struct af_center_s
{
- int ch; // Channel number which to insert the filtered data
+ int ch; // Channel number which to insert the filtered data
}af_center_t;
// Initialization and runtime control
@@ -61,12 +61,12 @@ static int control(struct af_instance* af, int cmd, void* arg)
// Filter data through filter
static int filter(struct af_instance* af, struct mp_audio* data, int flags)
{
- struct mp_audio* c = data; // Current working data
+ 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
+ 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
diff --git a/audio/filter/af_channels.c b/audio/filter/af_channels.c
index 6db44ba024..118607b66f 100644
--- a/audio/filter/af_channels.c
+++ b/audio/filter/af_channels.c
@@ -114,7 +114,7 @@ static void copy(struct af_instance *af, void* in, void* out,
}
default:
MP_ERR(af, "Unsupported number of bytes/sample: %i"
- " please report this error on the MPlayer mailing list. \n",bps);
+ " please report this error on the MPlayer mailing list. \n",bps);
}
}
@@ -125,7 +125,7 @@ static int check_routes(struct af_instance *af, int nin, int nout)
int i;
if((s->nr < 1) || (s->nr > AF_NCH)){
MP_ERR(af, "The number of routing pairs must be"
- " between 1 and %i. Current value is %i\n",AF_NCH,s->nr);
+ " between 1 and %i. Current value is %i\n",AF_NCH,s->nr);
return AF_ERROR;
}
@@ -154,22 +154,22 @@ static int control(struct af_instance* af, int cmd, void* arg)
int i;
// Make sure this filter isn't redundant
if(af->data->nch == ((struct mp_audio*)arg)->nch)
- return AF_DETACH;
+ return AF_DETACH;
// If mono: fake stereo
if(((struct mp_audio*)arg)->nch == 1){
- s->nr = MPMIN(af->data->nch,2);
- for(i=0;i<s->nr;i++){
- s->route[i][FR] = 0;
- s->route[i][TO] = i;
- }
+ s->nr = MPMIN(af->data->nch,2);
+ for(i=0;i<s->nr;i++){
+ s->route[i][FR] = 0;
+ s->route[i][TO] = i;
+ }
}
else{
- s->nr = MPMIN(af->data->nch, ((struct mp_audio*)arg)->nch);
- for(i=0;i<s->nr;i++){
- s->route[i][FR] = i;
- s->route[i][TO] = i;
- }
+ s->nr = MPMIN(af->data->nch, ((struct mp_audio*)arg)->nch);
+ for(i=0;i<s->nr;i++){
+ s->route[i][FR] = i;
+ s->route[i][TO] = i;
+ }
}
}
@@ -184,10 +184,10 @@ static int control(struct af_instance* af, int cmd, void* arg)
// Filter data through filter
static int filter(struct af_instance* af, struct mp_audio* data, int flags)
{
- struct mp_audio* c = data; // Current working data
- struct mp_audio* l = af->data; // Local data
+ struct mp_audio* c = data; // Current working data
+ struct mp_audio* l = af->data; // Local data
af_channels_t* s = af->priv;
- int i;
+ int i;
mp_audio_realloc_min(af->data, data->samples);
@@ -197,7 +197,7 @@ static int filter(struct af_instance* af, struct mp_audio* data, int flags)
if(AF_OK == check_routes(af,c->nch,l->nch))
for(i=0;i<s->nr;i++)
copy(af, c->planes[0],l->planes[0],c->nch,s->route[i][FR],
- l->nch,s->route[i][TO],mp_audio_psize(c),c->bps);
+ l->nch,s->route[i][TO],mp_audio_psize(c),c->bps);
// Set output data
c->planes[0] = l->planes[0];
diff --git a/audio/filter/af_delay.c b/audio/filter/af_delay.c
index a8cd79f117..b16cdc1f63 100644
--- a/audio/filter/af_delay.c
+++ b/audio/filter/af_delay.c
@@ -37,10 +37,10 @@
// Data for specific instances of this filter
typedef struct af_delay_s
{
- void* q[AF_NCH]; // Circular queues used for delaying audio signal
- int wi[AF_NCH]; // Write index
- int ri; // Read index
- float d[AF_NCH]; // Delay [ms]
+ void* q[AF_NCH]; // Circular queues used for delaying audio signal
+ int wi[AF_NCH]; // Write index
+ int ri; // Read index
+ float d[AF_NCH]; // Delay [ms]
char *delaystr;
}af_delay_t;
@@ -69,7 +69,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
for(i=0;i<af->data->nch;i++){
s->q[i] = calloc(L,af->data->bps);
if(NULL == s->q[i])
- MP_FATAL(af, "Out of memory\n");
+ MP_FATAL(af, "Out of memory\n");
}
if(AF_OK != af_from_ms(AF_NCH, s->d, s->wi, af->data->rate, 0.0, 1000.0))
@@ -99,12 +99,12 @@ static void uninit(struct af_instance* af)
// Filter data through filter
static int filter(struct af_instance* af, struct mp_audio* data, int flags)
{
- struct mp_audio* c = data; // Current working data
- af_delay_t* s = af->priv; // Setup for this instance
- int nch = c->nch; // Number of channels
- int len = mp_audio_psize(c)/c->bps; // Number of sample in data chunk
- int ri = 0;
- int ch,i;
+ struct mp_audio* c = data; // Current working data
+ af_delay_t* s = af->priv; // Setup for this instance
+ int nch = c->nch; // Number of channels
+ int len = mp_audio_psize(c)/c->bps; // Number of sample in data chunk
+ int ri = 0;
+ int ch,i;
for(ch=0;ch<nch;ch++){
switch(c->bps){
case 1:{
@@ -113,10 +113,10 @@ static int filter(struct af_instance* af, struct mp_audio* data, int flags)
int wi = s->wi[ch];
ri = s->ri;
for(i=ch;i<len;i+=nch){
- q[wi] = a[i];
- a[i] = q[ri];
- UPDATEQI(wi);
- UPDATEQI(ri);
+ q[wi] = a[i];
+ a[i] = q[ri];
+ UPDATEQI(wi);
+ UPDATEQI(ri);
}
s->wi[ch] = wi;
break;
@@ -127,10 +127,10 @@ static int filter(struct af_instance* af, struct mp_audio* data, int flags)
int wi = s->wi[ch];
ri = s->ri;
for(i=ch;i<len;i+=nch){
- q[wi] = a[i];
- a[i] = q[ri];
- UPDATEQI(wi);
- UPDATEQI(ri);
+ q[wi] = a[i];
+ a[i] = q[ri];
+ UPDATEQI(wi);
+ UPDATEQI(ri);
}
s->wi[ch] = wi;
break;
@@ -141,10 +141,10 @@ static int filter(struct af_instance* af, struct mp_audio* data, int flags)
int wi = s->wi[ch];
ri = s->ri;
for(i=ch;i<len;i+=nch){
- q[wi] = a[i];
- a[i] = q[ri];
- UPDATEQI(wi);
- UPDATEQI(ri);
+ q[wi] = a[i];
+ a[i] = q[ri];
+ UPDATEQI(wi);
+ UPDATEQI(ri);
}
s->wi[ch] = wi;
break;
diff --git a/audio/filter/af_drc.c b/audio/filter/af_drc.c
index fad34ba500..3bcd368d4d 100644
--- a/audio/filter/af_drc.c
+++ b/audio/filter/af_drc.c
@@ -70,8 +70,8 @@ typedef struct af_volume_s
// method 2
int idx;
struct {
- float avg; // average level of the sample
- int len; // sample size (weight)
+ float avg; // average level of the sample
+ int len; // sample size (weight)
} mem[NSAMPLES];
// "Ideal" level
float mid_s16;
@@ -100,8 +100,8 @@ static int control(struct af_instance* af, int cmd, void* arg)
static void method1_int16(af_drc_t *s, struct mp_audio *c)
{
register int i = 0;
- int16_t *data = (int16_t*)c->planes[0]; // Audio data
- int len = c->samples*c->nch; // Number of samples
+ int16_t *data = (int16_t*)c->planes[0]; // Audio data
+ int len = c->samples*c->nch; // Number of samples
float curavg = 0.0, newavg, neededmul;
int tmp;
@@ -142,8 +142,8 @@ static void method1_int16(af_drc_t *s, struct mp_audio *c)
static void method1_float(af_drc_t *s, struct mp_audio *c)
{
register int i = 0;
- float *data = (float*)c->planes[0]; // Audio data
- int len = c->samples*c->nch; // Number of samples
+ float *data = (float*)c->planes[0]; // Audio data
+ int len = c->samples*c->nch; // Number of samples
float curavg = 0.0, newavg, neededmul, tmp;
for (i = 0; i < len; i++)
@@ -179,8 +179,8 @@ static void method1_float(af_drc_t *s, struct mp_audio *c)
static void method2_int16(af_drc_t *s, struct mp_audio *c)
{
register int i = 0;
- int16_t *data = (int16_t*)c->planes[0]; // Audio data
- int len = c->samples*c->nch; // Number of samples
+ int16_t *data = (int16_t*)c->planes[0]; // Audio data
+ int len = c->samples*c->nch; // Number of samples
float curavg = 0.0, newavg, avg = 0.0;
int tmp, totallen = 0;
@@ -204,8 +204,8 @@ static void method2_int16(af_drc_t *s, struct mp_audio *c)
avg /= (float)totallen;
if (avg >= SIL_S16)
{
- s->mul = s->mid_s16 / avg;
- s->mul = MPCLAMP(s->mul, MUL_MIN, MUL_MAX);
+ s->mul = s->mid_s16 / avg;
+ s->mul = MPCLAMP(s->mul, MUL_MIN, MUL_MAX);
}
}
@@ -229,8 +229,8 @@ static void method2_int16(af_drc_t *s, struct mp_audio *c)
static void method2_float(af_drc_t *s, struct mp_audio *c)
{
register int i = 0;
- float *data = (float*)c->planes[0]; // Audio data
- int len = c->samples*c->nch; // Number of samples
+ float *data = (float*)c->planes[0]; // Audio data
+ int len = c->samples*c->nch; // Number of samples
float curavg = 0.0, newavg, avg = 0.0, tmp;
int totallen = 0;
@@ -254,8 +254,8 @@ static void method2_float(af_drc_t *s, struct mp_audio *c)
avg /= (float)totallen;
if (avg >= SIL_FLOAT)
{
- s->mul = s->mid_float / avg;
- s->mul = MPCLAMP(s->mul, MUL_MIN, MUL_MAX);
+ s->mul = s->mid_float / avg;
+ s->mul = MPCLAMP(s->mul, MUL_MIN, MUL_MAX);
}
}
@@ -279,16 +279,16 @@ static int filter(struct af_instance* af, struct mp_audio* data, int flags)
if(af->data->format == (AF_FORMAT_S16))
{
if (s->method == 2)
- method2_int16(s, data);
+ method2_int16(s, data);
else
- method1_int16(s, data);
+ method1_int16(s, data);
}
else if(af->data->format == (AF_FORMAT_FLOAT))
{
if (s->method == 2)
- method2_float(s, data);
+ method2_float(s, data);
else
- method1_float(s, data);
+ method1_float(s, data);
}
return 0;
}
diff --git a/audio/filter/af_dummy.c b/audio/filter/af_dummy.c
index c13c32b968..d920480e01 100644
--- a/audio/filter/af_dummy.c
+++ b/audio/filter/af_dummy.c
@@ -34,7 +34,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
case AF_CONTROL_REINIT: ;
*af->data = *(struct mp_audio*)arg;
MP_VERBOSE(af, "Was reinitialized: %iHz/%ich/%s\n",
- af->data->rate,af->data->nch,af_fmt_to_str(af->data->format));
+ af->data->rate,af->data->nch,af_fmt_to_str(af->data->format));
return AF_OK;
}
return AF_UNKNOWN;
diff --git a/audio/filter/af_equalizer.c b/audio/filter/af_equalizer.c
index 4f5a29706e..83fa80f2b3 100644
--- a/audio/filter/af_equalizer.c
+++ b/audio/filter/af_equalizer.c
@@ -32,41 +32,41 @@
#include "common/common.h"
#include "af.h"
-#define L 2 // Storage for filter taps
-#define KM 10 // Max number of bands
+#define L 2 // Storage for filter taps
+#define KM 10 // Max number of bands
#define Q 1.2247449 /* Q value for band-pass filters 1.2247=(3/2)^(1/2)
- gives 4dB suppression @ Fc*2 and Fc/2 */
+ gives 4dB suppression @ Fc*2 and Fc/2 */
/* Center frequencies for band-pass filters
The different frequency bands are:
- nr. center frequency
- 0 31.25 Hz
- 1 62.50 Hz
- 2 125.0 Hz
- 3 250.0 Hz
- 4 500.0 Hz
- 5 1.000 kHz
- 6 2.000 kHz
- 7 4.000 kHz
- 8 8.000 kHz
- 9 16.00 kHz
+ nr. center frequency
+ 0 31.25 Hz
+ 1 62.50 Hz
+ 2 125.0 Hz
+ 3 250.0 Hz
+ 4 500.0 Hz
+ 5 1.000 kHz
+ 6 2.000 kHz
+ 7 4.000 kHz
+ 8 8.000 kHz
+ 9 16.00 kHz
*/
-#define CF {31.25,62.5,125,250,500,1000,2000,4000,8000,16000}
+#define CF {31.25,62.5,125,250,500,1000,2000,4000,8000,16000}
// Maximum and minimum gain for the bands
-#define G_MAX +12.0
-#define G_MIN -12.0
+#define G_MAX +12.0
+#define G_MIN -12.0
// Data for specific instances of this filter
typedef struct af_equalizer_s
{
- float a[KM][L]; // A weights
- float b[KM][L]; // B weights
- float wq[AF_NCH][KM][L]; // Circular buffer for W data
- float g[AF_NCH][KM]; // Gain factor for each channel and band
- int K; // Number of used eq bands
- int channels; // Number of channels
+ float a[KM][L]; // A weights
+ float b[KM][L]; // B weights
+ float wq[AF_NCH][KM][L]; // Circular buffer for W data
+ float g[AF_NCH][KM]; // Gain factor for each channel and band
+ int K; // Number of used eq bands
+ int channels; // Number of channels
float gain_factor; // applied at output to avoid clipping
double p[KM];
} af_equalizer_t;
@@ -108,7 +108,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
if(s->K != KM)
MP_INFO(af, "Limiting the number of filters to"
- " %i due to low sample rate.\n",s->K);
+ " %i due to low sample rate.\n",s->K);
// Generate filter taps
for(k=0;k<s->K;k++)
@@ -144,33 +144,33 @@ static int control(struct af_instance* af, int cmd, void* arg)
// Filter data through filter
static int filter(struct af_instance* af, struct mp_audio* data, int flags)
{
- struct mp_audio* c = data; // Current working data
- af_equalizer_t* s = (af_equalizer_t*)af->priv; // Setup
- uint32_t ci = af->data->nch; // Index for channels
- uint32_t nch = af->data->nch; // Number of channels
+ struct mp_audio* c = data; // Current working data
+ af_equalizer_t* s = (af_equalizer_t*)af->priv; // Setup
+ uint32_t ci = af->data->nch; // Index for channels
+ uint32_t nch = af->data->nch; // Number of channels
while(ci--){
- float* g = s->g[ci]; // Gain factor
- float* in = ((float*)c->planes[0])+ci;
- float* out = ((float*)c->planes[0])+ci;
- float* end = in + c->samples*c->nch; // Block loop end
+ float* g = s->g[ci]; // Gain factor
+ float* in = ((float*)c->planes[0])+ci;
+ float* out = ((float*)c->planes[0])+ci;
+ float* end = in + c->samples*c->nch; // Block loop end
while(in < end){
- register int k = 0; // Frequency band index
- register float yt = *in; // Current input sample
+ register int k = 0; // Frequency band index
+ register float yt = *in; // Current input sample
in+=nch;
// Run the filters
for(;k<s->K;k++){
- // Pointer to circular buffer wq
- register float* wq = s->wq[ci][k];
- // Calculate output from AR part of current filter
- register float w=yt*s->b[k][0] + wq[0]*s->a[k][0] + wq[1]*s->a[k][1];
- // Calculate output form MA part of current filter
- yt+=(w + wq[1]*s->b[k][1])*g[k];
- // Update circular buffer
- wq[1] = wq[0];
- wq[0] = w;
+ // Pointer to circular buffer wq
+ register float* wq = s->wq[ci][k];
+ // Calculate output from AR part of current filter
+ register float w=yt*s->b[k][0] + wq[0]*s->a[k][0] + wq[1]*s->a[k][1];
+ // Calculate output form MA part of current filter
+ yt+=(w + wq[1]*s->b[k][1])*g[k];
+ // Update circular buffer
+ wq[1] = wq[0];
+ wq[0] = w;
}
// Calculate output
*out=yt*s->gain_factor;
diff --git a/audio/filter/af_export.c b/audio/filter/af_export.c
index 542b2a1f9f..faa09bf927 100644
--- a/audio/filter/af_export.c
+++ b/audio/filter/af_export.c
@@ -47,7 +47,7 @@
#define DEF_SZ 512 // default buffer size (in samples)
#define SHARED_FILE "mpv-af_export" /* default file name
- (relative to ~/.mpv/ */
+ (relative to ~/.mpv/ */
#define SIZE_HEADER (2 * sizeof(int) + sizeof(unsigned long long))
@@ -55,12 +55,12 @@
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
+ 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;
@@ -109,7 +109,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
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);
+ s->filename);
return AF_ERROR;
}
@@ -127,7 +127,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
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);
+ s->filename, s->mmap_area);
// Initialize header
*((int*)s->mmap_area) = af->data->nch;
@@ -164,27 +164,27 @@ static void uninit( struct af_instance* af )
*/
static int filter( struct af_instance* af, struct mp_audio* data, int flags)
{
- struct mp_audio* c = data; // Current working data
- af_export_t* s = af->priv; // Setup for this instance
- int16_t* a = c->planes[0]; // Incomming 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
+ struct mp_audio* c = data; // Current working data
+ af_export_t* s = af->priv; // Setup for this instance
+ int16_t* a = c->planes[0]; // Incomming 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;
+ 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
+ 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;
+ flag = 1;
+ break;
}
}
s->wi = wi % s->sz;
@@ -196,7 +196,7 @@ static int filter( struct af_instance* af, struct mp_audio* data, int flags)
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));
+ &(s->count), sizeof(s->count));
}
return 0;
diff --git a/audio/filter/af_extrastereo.c b/audio/filter/af_extrastereo.c
index 1e814e8351..05a1ac8e36 100644
--- a/audio/filter/af_extrastereo.c
+++ b/audio/filter/af_extrastereo.c
@@ -51,11 +51,11 @@ static int control(struct af_instance* af, int cmd, void* arg)
mp_audio_set_num_channels(af->data, 2);
if (af->data->format == AF_FORMAT_FLOAT)
{
- af->filter = play_float;
+ af->filter = play_float;
}// else
{
mp_audio_set_format(af->data, AF_FORMAT_S16);
- af->filter = play_s16;
+ af->filter = play_s16;
}
return af_test_output(af,(struct mp_audio*)arg);
@@ -69,8 +69,8 @@ static int play_s16(struct af_instance* af, struct mp_audio* data, int f)
{
af_extrastereo_t *s = af->priv;
register int i = 0;
- int16_t *a = (int16_t*)data->planes[0]; // Audio data
- int len = data->samples*data->nch; // Number of samples
+ 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)
@@ -91,8 +91,8 @@ static int play_float(struct af_instance* af, struct mp_audio* data, int f)
{
af_extrastereo_t *s = af->priv;
register int i = 0;
- float *a = (float*)data->planes[0]; // Audio data
- int len = data->samples * data->nch; // Number of samples
+ 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)
diff --git a/audio/filter/af_hrtf.c b/audio/filter/af_hrtf.c
index e329b4e558..e8ab5fc72d 100644
--- a/audio/filter/af_hrtf.c
+++ b/audio/filter/af_hrtf.c
@@ -75,23 +75,23 @@ typedef struct af_hrtf_s {
} 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
+ * 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)
+ 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);
+ 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);
+ 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) */
@@ -104,8 +104,8 @@ static int pulse_detect(const float *sx)
int i;
for(i = 0; i < nmax; i++)
- if(fabs(sx[i]) > thresh)
- return i;
+ if(fabs(sx[i]) > thresh)
+ return i;
return 0;
}
@@ -122,14 +122,14 @@ static inline float passive_lock(float x)
/* 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 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) /
@@ -161,13 +161,13 @@ static inline void matrix_decode(short *in, const int k, const int il,
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);
+ 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;
+ 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;
@@ -179,12 +179,12 @@ static inline void matrix_decode(short *in, const int k, const int il,
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). */
+ 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);
+ (1 + l_fwr + r_fwr);
rr[kr] *= (r_fwr + r_fwr) /
- (1 + l_fwr + r_fwr);
+ (1 + l_fwr + r_fwr);
}
/*** AXIS NO. 2: (Lt + Rt, Lt - Rt) -> (L, R) ***/
@@ -221,9 +221,9 @@ static inline void matrix_decode(short *in, const int k, const int il,
#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);
+ *adapt_l_gain, *adapt_r_gain,
+ *adapt_lpr_gain, *adapt_lmr_gain,
+ c_gain);
counter++;
#endif
}
@@ -237,18 +237,18 @@ static inline void update_ch(af_hrtf_t *s, short *in, const int k)
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]);
+ 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]);
+ 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]);
+ 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]);
+ fabs(s->fwrbuf_lr[fwr_pos] - s->fwrbuf_rr[fwr_pos]);
}
switch (s->decode_mode) {
@@ -265,11 +265,11 @@ static inline void update_ch(af_hrtf_t *s, short *in, const int k)
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);
+ 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 */
@@ -292,30 +292,30 @@ static int control(struct af_instance *af, int cmd, void* arg)
switch(cmd) {
case AF_CONTROL_REINIT:
- af->data->rate = ((struct mp_audio*)arg)->rate;
- if(af->data->rate != 48000) {
- // automatic samplerate adjustment in the filter chain
- // is not yet supported.
- MP_ERR(af, "ERROR: Sampling rate is not 48000 Hz (%d)!\n",
- af->data->rate);
- return AF_ERROR;
- }
- 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);
+ af->data->rate = ((struct mp_audio*)arg)->rate;
+ if(af->data->rate != 48000) {
+ // automatic samplerate adjustment in the filter chain
+ // is not yet supported.
+ MP_ERR(af, "ERROR: Sampling rate is not 48000 Hz (%d)!\n",
+ af->data->rate);
+ return AF_ERROR;
+ }
+ 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
+ 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;
+ s->print_flag = 1;
+ return test_output_res;
}
return AF_UNKNOWN;
@@ -324,21 +324,21 @@ static int control(struct af_instance *af, int cmd, void* arg)
/* 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);
+ 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
@@ -364,29 +364,29 @@ static int filter(struct af_instance *af, struct mp_audio *data, int flags)
mp_audio_realloc_min(af->data, data->samples);
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;
- }
+ 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");
+ MP_INFO(af, "Using active matrix to decode rear center "
+ "channel\n");
}
out = af->data->planes[0];
@@ -410,122 +410,122 @@ static int filter(struct af_instance *af, struct mp_audio *data, int flags)
*/
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;
+ 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;
}
/* Set output data */
@@ -546,13 +546,13 @@ static int allocate(af_hrtf_t *s)
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;
+ malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
if ((s->fwrbuf_r =
- malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
+ malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
if ((s->fwrbuf_lr =
- malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
+ malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
if ((s->fwrbuf_rr =
- malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
+ malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
return 0;
}
@@ -595,16 +595,16 @@ static int af_open(struct af_instance* af)
s->print_flag = 1;
if (allocate(s) != 0) {
- MP_ERR(af, "Memory allocation error.\n");
- return AF_ERROR;
+ 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->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->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));
@@ -614,18 +614,18 @@ static int af_open(struct af_instance* af)
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;
+ 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;
+ 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;
+ s->ba_ir[i] *= BASSGAIN;
return AF_OK;
}
diff --git a/audio/filter/af_hrtf.h b/audio/filter/af_hrtf.h
index 887310b57d..a654dc4f98 100644
--- a/audio/filter/af_hrtf.h
+++ b/audio/filter/af_hrtf.h
@@ -24,38 +24,38 @@
#define HRTF_MIX_MATRIX2CH 2
/* Amplitude scaling factors */
-#define M17_0DB 0.1414213562
-#define M9_03DB 0.3535533906
-#define M6_99DB 0.4472135955
-#define M4_77DB 0.5773502692
-#define M3_01DB 0.7071067812
-#define M1_76DB 0.8164965809
+#define M17_0DB 0.1414213562
+#define M9_03DB 0.3535533906
+#define M6_99DB 0.4472135955
+#define M4_77DB 0.5773502692
+#define M3_01DB 0.7071067812
+#define M1_76DB 0.8164965809
-#define DELAYBUFLEN 1024 /* Length of the delay buffer */
-#define HRTFFILTLEN 64 /* HRTF filter length */
-#define IRTHRESH 0.001 /* Impulse response pruning thresh. */
+#define DELAYBUFLEN 1024 /* Length of the delay buffer */
+#define HRTFFILTLEN 64 /* HRTF filter length */
+#define IRTHRESH 0.001 /* Impulse response pruning thresh. */
-#define AMPLNORM M6_99DB /* Overall amplitude renormalization */
+#define AMPLNORM M6_99DB /* Overall amplitude renormalization */
-#define BASSFILTFREQ 180 /* Bass compensation filter cut (Hz) */
-#define BASSFILTLEN 193 /* Bass compensation filter length */
-#define BASSGAIN M_SQRT2 /* Bass compensation gain */
-#define BASSCROSS 0.35 /* Bass cross talk */
+#define BASSFILTFREQ 180 /* Bass compensation filter cut (Hz) */
+#define BASSFILTLEN 193 /* Bass compensation filter length */
+#define BASSGAIN M_SQRT2 /* Bass compensation gain */
+#define BASSCROSS 0.35 /* Bass cross talk */
-#define FWRDURATION 240 /* FWR average duration (samples) */
-#define MATREARDELAY 720 /* Matrix mode rear delay (samples) */
+#define FWRDURATION 240 /* FWR average duration (samples) */
+#define MATREARDELAY 720 /* Matrix mode rear delay (samples) */
-#define MATAGCTRIG 8.0 /* (Fuzzy) AGC trigger */
-#define MATAGCDECAY 1.0 /* AGC baseline decay rate (1/samp.) */
-#define MATAGCLOCK 0.2 /* AGC range (around 1) where the
- matrix behaves passively */
-#define MATCOMPGAIN 0.37 /* Cross talk compensation gain,
- 0.50 - 0.55 is full cancellation. */
+#define MATAGCTRIG 8.0 /* (Fuzzy) AGC trigger */
+#define MATAGCDECAY 1.0 /* AGC baseline decay rate (1/samp.) */
+#define MATAGCLOCK 0.2 /* AGC range (around 1) where the
+ matrix behaves passively */
+#define MATCOMPGAIN 0.37 /* Cross talk compensation gain,
+ 0.50 - 0.55 is full cancellation. */
-#define CFECHODELAY 360 /* Center front echo delay (samples) */
-#define CFECHOAMPL M17_0DB /* Center front echo amplitude */
+#define CFECHODELAY 360 /* Center front echo delay (samples) */
+#define CFECHOAMPL M17_0DB /* Center front echo amplitude */
-#define STEXPAND2 0.07 /* Stereo expansion / 2 */
+#define STEXPAND2 0.07 /* Stereo expansion / 2 */
/* Head related impulse response (HRIR) derived from KEMAR measurement
data by Bill Gardner <billg@media.mit.edu> and Keith Martin
diff --git a/audio/filter/af_karaoke.c b/audio/filter/af_karaoke.c
index bddb218dde..00c4d16dca 100644
--- a/audio/filter/af_karaoke.c
+++ b/audio/filter/af_karaoke.c
@@ -32,50 +32,50 @@
// Initialization and runtime control
static int control(struct af_instance* af, int cmd, void* arg)
{
- switch(cmd){
- case AF_CONTROL_REINIT:
+ switch(cmd){
+ case AF_CONTROL_REINIT:
mp_audio_copy_config(af->data, (struct mp_audio*)arg);
mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
- return af_test_output(af,(struct mp_audio*)arg);
- }
- return AF_UNKNOWN;
+ return af_test_output(af,(struct mp_audio*)arg);
+ }
+ return AF_UNKNOWN;
}
// Filter data through filter
static int play(struct af_instance* af, struct mp_audio* data, int flags)
{
- struct mp_audio* c = data; // Current working data
- float* a = c->planes[0]; // Audio data
- int nch = c->nch; // Number of channels
- int len = c->samples*nch; // Number of samples in current audio block
- register int i;
+ struct mp_audio* c = data; // Current working data
+ float* a = c->planes[0]; // Audio data
+ int nch = c->nch; // Number of channels
+ int len = c->samples*nch; // Number of samples in current audio block
+ register int i;
- /*
- FIXME1 add a low band pass filter to avoid suppressing
- centered bass/drums
- FIXME2 better calculated* attenuation factor
- */
+ /*
+ FIXME1 add a low band pass filter to avoid suppressing
+ centered bass/drums
+ FIXME2 better calculated* attenuation factor
+ */
- for(i=0;i<len;i+=nch)
- {
- a[i] = (a[i] - a[i+1]) * 0.7;
- a[i+1]=a[i];
- }
+ for(i=0;i<len;i+=nch)
+ {
+ a[i] = (a[i] - a[i+1]) * 0.7;
+ a[i+1]=a[i];
+ }
- return 0;
+ return 0;
}
// Allocate memory and set function pointers
static int af_open(struct af_instance* af){
- af->control = control;
- af->filter = play;
- return AF_OK;
+ af->control = control;
+ af->filter = play;
+ return AF_OK;
}
// Description of this filter
struct af_info af_info_karaoke = {
- .info = "Simple karaoke/voice-removal audio filter",
- .name = "karaoke",
- .flags = AF_FLAGS_NOT_REENTRANT,
- .open = af_open,
+ .info = "Simple karaoke/voice-removal audio filter",
+ .name = "karaoke",
+ .flags = AF_FLAGS_NOT_REENTRANT,
+ .open = af_open,
};
diff --git a/audio/filter/af_pan.c b/audio/filter/af_pan.c
index 01b0575ae3..b966deb9b3 100644
--- a/audio/filter/af_pan.c
+++ b/audio/filter/af_pan.c
@@ -32,7 +32,7 @@
typedef struct af_pan_s
{
int nch; // Number of output channels; zero means same as input
- float level[AF_NCH][AF_NCH]; // Gain level for each channel
+ float level[AF_NCH][AF_NCH]; // Gain level for each channel
char *matrixstr;
}af_pan_t;
@@ -110,14 +110,14 @@ static int control(struct af_instance* af, int cmd, void* arg)
// Filter data through filter
static int filter(struct af_instance* af, struct mp_audio* data, int flags)
{
- struct mp_audio* c = data; // Current working data
- struct mp_audio* l = af->data; // Local data
- af_pan_t* s = af->priv; // Setup for this instance
- float* in = c->planes[0]; // Input audio data
- float* out = NULL; // Output audio data
- float* end = in+c->samples*c->nch; // End of loop
- int nchi = c->nch; // Number of input channels
- int ncho = l->nch; // Number of output channels
+ struct mp_audio* c = data; // Current working data
+ struct mp_audio* l = af->data; // Local data
+ af_pan_t* s = af->priv; // Setup for this instance
+ float* in = c->planes[0]; // Input audio data
+ float* out = NULL; // Output audio data
+ float* end = in+c->samples*c->nch; // End of loop
+ int nchi = c->nch; // Number of input channels
+ int ncho = l->nch; // Number of output channels
register int j,k;
mp_audio_realloc_min(af->data, data->samples);
@@ -130,7 +130,7 @@ static int filter(struct af_instance* af, struct mp_audio* data, int flags)
register float x = 0.0;
register float* tin = in;
for(k=0;k<nchi;k++)
- x += tin[k] * s->level[j][k];
+ x += tin[k] * s->level[j][k];
out[j] = x;
}
out+= ncho;
diff --git a/audio/filter/af_sinesuppress.c b/audio/filter/af_sinesuppress.c
index 9719d46828..6b1d5da50d 100644
--- a/audio/filter/af_sinesuppress.c
+++ b/audio/filter/af_sinesuppress.c
@@ -57,14 +57,14 @@ static int control(struct af_instance* af, int cmd, void* arg)
#if 0
if (((struct mp_audio*)arg)->format == AF_FORMAT_FLOAT)
{
- af->data->format = AF_FORMAT_FLOAT;
- af->data->bps = 4;
- af->play = play_float;
+ af->data->format = AF_FORMAT_FLOAT;
+ af->data->bps = 4;
+ af->play = play_float;
}// else
#endif
{
mp_audio_set_format(af->data, AF_FORMAT_S16);
- af->filter = play_s16;
+ af->filter = play_s16;
}
return af_test_output(af,(struct mp_audio*)arg);
@@ -78,8 +78,8 @@ static int play_s16(struct af_instance* af, struct mp_audio* data, int f)
{
af_sinesuppress_t *s = af->priv;
register int i = 0;
- int16_t *a = (int16_t*)data->planes[0]; // Audio data
- int len = data->samples*data->nch; // Number of samples
+ int16_t *a = (int16_t*)data->planes[0]; // Audio data
+ int len = data->samples*data->nch; // Number of samples
for (i = 0; i < len; i++)
{
@@ -109,8 +109,8 @@ static struct mp_audio* play_float(struct af_instance* af, struct mp_audio* data
{
af_sinesuppress_t *s = af->setup;
register int i = 0;
- float *a = (float*)data->audio; // Audio data
- int len = data->len/4; // Number of samples
+ float *a = (float*)data->audio; // Audio data
+ int len = data->len/4; // Number of samples
float avg, l, r;
for (i = 0; i < len; i+=2)
diff --git a/audio/filter/af_sub.c b/audio/filter/af_sub.c
index 7012e1e129..742d2e7413 100644
--- a/audio/filter/af_sub.c
+++ b/audio/filter/af_sub.c
@@ -41,22 +41,22 @@
// Analog domain biquad section
typedef struct{
- float a[3]; // Numerator coefficients
- float b[3]; // Denominator coefficients
+ float a[3]; // Numerator coefficients
+ float b[3]; // Denominator coefficients
} biquad_t;
// S-parameters for designing 4th order Butterworth filter
static biquad_t sp[2] = {{{1.0,0.0,0.0},{1.0,0.765367,1.0}},
- {{1.0,0.0,0.0},{1.0,1.847759,1.0}}};
+ {{1.0,0.0,0.0},{1.0,1.847759,1.0}}};
// Data for specific instances of this filter
typedef struct af_sub_s
{
- float w[2][4]; // Filter taps for low-pass filter
- float q[2][2]; // Circular queues
- float fc; // Cutoff frequency [Hz] for low-pass filter
- float k; // Filter gain;
- int ch; // Channel number which to insert the filtered data
+ float w[2][4]; // Filter taps for low-pass filter
+ float q[2][2]; // Circular queues
+ float fc; // Cutoff frequency [Hz] for low-pass filter
+ float k; // Filter gain;
+ int ch; // Channel number which to insert the filtered data
}af_sub_t;
@@ -92,7 +92,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
float h0 = (q)[0]; \
float h1 = (q)[1]; \
float hn = (in) - h0 * (w)[0] - h1 * (w)[1]; \
- out = hn + h0 * (w)[2] + h1 * (w)[3]; \
+ out = hn + h0 * (w)[2] + h1 * (w)[3]; \
(q)[1] = h0; \
(q)[0] = hn; \
}
@@ -101,12 +101,12 @@ static int control(struct af_instance* af, int cmd, void* arg)
// Filter data through filter
static int filter(struct af_instance* af, struct mp_audio* data, int flags)
{
- struct mp_audio* c = data; // Current working data
- af_sub_t* s = af->priv; // Setup for this instance
- float* a = c->planes[0]; // Audio data
- int len = c->samples*c->nch; // Number of samples in current audio block
- int nch = c->nch; // Number of channels
- int ch = s->ch; // Channel in which to insert the sub audio
+ struct mp_audio* c = data; // Current working data
+ af_sub_t* s = af->priv; // Setup for this instance
+ float* a = c->planes[0]; // Audio data
+ int len = c->samples*c->nch; // Number of samples in current audio block
+ int nch = c->nch; // Number of channels
+ int ch = s->ch; // Channel in which to insert the sub audio
register int i;
// Run filter
diff --git a/audio/filter/af_surround.c b/audio/filter/af_surround.c
index 9cabeedf9d..06a8c4a761 100644
--- a/audio/filter/af_surround.c
+++ b/audio/filter/af_surround.c
@@ -76,13 +76,13 @@ typedef struct af_surround_s
{
float lq[2*L]; // Circular queue for filtering left rear channel
float rq[2*L]; // Circular queue for filtering right rear channel
- float w[L]; // FIR filter coefficients for surround sound 7kHz low-pass
- float* dr; // Delay queue right rear channel
- float* dl; // Delay queue left rear channel
- float d; // Delay time
- int i; // Position in circular buffer
- int wi; // Write index for delay queue
- int ri; // Read index for delay queue
+ float w[L]; // FIR filter coefficients for surround sound 7kHz low-pass
+ float* dr; // Delay queue right rear channel
+ float* dl; // Delay queue left rear channel
+ float d; // Delay time
+ int i; // Position in circular buffer
+ int wi; // Write index for delay queue
+ int ri; // Read index for delay queue
}af_surround_t;
// Initialization and runtime control
@@ -132,10 +132,10 @@ static int control(struct af_instance* af, int cmd, void* arg)
// The beginnings of an active matrix...
static float steering_matrix[][12] = {
-// LL RL LR RR LS RS
-// LLs RLs LRs RRs LC RC
- {.707, .0, .0, .707, .5, -.5,
- .5878, -.3928, .3928, -.5878, .5, .5},
+// LL RL LR RR LS RS
+// LLs RLs LRs RRs LC RC
+ {.707, .0, .0, .707, .5, -.5,
+ .5878, -.3928, .3928, -.5878, .5, .5},
};
// Experimental moving average dominance
@@ -144,13 +144,13 @@ static float steering_matrix[][12] = {
// Filter data through filter
static int filter(struct af_instance* af, struct mp_audio* data, int flags){
af_surround_t* s = (af_surround_t*)af->priv;
- float* m = steering_matrix[0];
- float* in = data->planes[0]; // Input audio data
- float* out = NULL; // Output audio data
- float* end = in + data->samples * data->nch;
- int i = s->i; // Filter queue index
- int ri = s->ri; // Read index for delay queue
- int wi = s->wi; // Write index for delay queue
+ float* m = steering_matrix[0];
+ float* in = data->planes[0]; // Input audio data
+ float* out = NULL; // Output audio data
+ float* end = in + data->samples * data->nch;
+ int i = s->i; // Filter queue index
+ int ri = s->ri; // Read index for delay queue
+ int wi = s->wi; // Write index for delay queue
mp_audio_realloc_min(af->data, data->samples);
diff --git a/audio/filter/dsp.h b/audio/filter/dsp.h
index 561b86cfe0..164eccf140 100644
--- a/audio/filter/dsp.h
+++ b/audio/filter/dsp.h
@@ -18,8 +18,8 @@
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
-#ifndef MPLAYER_DSP_H
-#define MPLAYER_DSP_H
+#ifndef MPLAYER_DSP_H
+#define MPLAYER_DSP_H
/* Implementation of routines used for DSP */
diff --git a/audio/filter/equalizer.h b/audio/filter/equalizer.h
index 4935401413..6505379276 100644
--- a/audio/filter/equalizer.h
+++ b/audio/filter/equalizer.h
@@ -26,22 +26,22 @@
typedef struct equalizer_s
{
- float gain; // Gain in dB -15 - 15
- int channel; // Channel number 0 - 5
- int band; // Frequency band 0 - 9
+ float gain; // Gain in dB -15 - 15
+ int channel; // Channel number 0 - 5
+ int band; // Frequency band 0 - 9
}equalizer_t;
/* The different frequency bands are:
-nr. center frequency
-0 31.25 Hz
-1 62.50 Hz
-2 125.0 Hz
-3 250.0 Hz
-4 500.0 Hz
-5 1.000 kHz
-6 2.000 kHz
-7 4.000 kHz
-8 8.000 kHz
+nr. center frequency
+0 31.25 Hz
+1 62.50 Hz
+2 125.0 Hz
+3 250.0 Hz
+4 500.0 Hz
+5 1.000 kHz
+6 2.000 kHz
+7 4.000 kHz
+8 8.000 kHz
9 16.00 kHz
*/
diff --git a/audio/filter/filter.c b/audio/filter/filter.c
index b272125fd8..dbbe743a79 100644
--- a/audio/filter/filter.c
+++ b/audio/filter/filter.c
@@ -58,7 +58,7 @@ inline FLOAT_TYPE af_filter_fir(register unsigned int n, const FLOAT_TYPE* w,
0 < fc < 1 where 1 <=> Fs/2
flags window and filter type as defined in filter.h
variables are ored together: i.e. LP|HAMMING will give a
- low pass filter designed using a hamming window
+ low pass filter designed using a hamming window
opt beta constant used only when designing using kaiser windows
returns 0 if OK, -1 if fail
@@ -66,16 +66,16 @@ inline FLOAT_TYPE af_filter_fir(register unsigned int n, const FLOAT_TYPE* w,
int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc,
unsigned int flags, FLOAT_TYPE opt)
{
- unsigned int o = n & 1; // Indicator for odd filter length
- unsigned int end = ((n + 1) >> 1) - o; // Loop end
- unsigned int i; // Loop index
+ unsigned int o = n & 1; // Indicator for odd filter length
+ unsigned int end = ((n + 1) >> 1) - o; // Loop end
+ unsigned int i; // Loop index
- FLOAT_TYPE k1 = 2 * M_PI; // 2*pi*fc1
+ FLOAT_TYPE k1 = 2 * M_PI; // 2*pi*fc1
FLOAT_TYPE k2 = 0.5 * (FLOAT_TYPE)(1 - o);// Constant used if the filter has even length
- FLOAT_TYPE k3; // 2*pi*fc2 Constant used in BP and BS design
- FLOAT_TYPE g = 0.0; // Gain
- FLOAT_TYPE t1,t2,t3; // Temporary variables
- FLOAT_TYPE fc1,fc2; // Cutoff frequencies
+ FLOAT_TYPE k3; // 2*pi*fc2 Constant used in BP and BS design
+ FLOAT_TYPE g = 0.0; // Gain
+ FLOAT_TYPE t1,t2,t3; // Temporary variables
+ FLOAT_TYPE fc1,fc2; // Cutoff frequencies
// Sanity check
if(!w || (n == 0)) return -1;
@@ -113,28 +113,28 @@ int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc,
// where x is zero. To make sure nothing strange happens, we set this
// value separately.
if (o){
- w[end] = fc1 * w[end] * 2.0;
- g=w[end];
+ w[end] = fc1 * w[end] * 2.0;
+ g=w[end];
}
// Create filter
for (i=0 ; i<end ; i++){
- t1 = (FLOAT_TYPE)(i+1) - k2;
- w[end-i-1] = w[n-end+i] = w[end-i-1] * sin(k1 * t1)/(M_PI * t1); // Sinc
- g += 2*w[end-i-1]; // Total gain in filter
+ t1 = (FLOAT_TYPE)(i+1) - k2;
+ w[end-i-1] = w[n-end+i] = w[end-i-1] * sin(k1 * t1)/(M_PI * t1); // Sinc
+ g += 2*w[end-i-1]; // Total gain in filter
}
}
else{ // High pass filter
if (!o) // High pass filters must have odd length
- return -1;
+ return -1;
w[end] = 1.0 - (fc1 * w[end] * 2.0);
g= w[end];
// Create filter
for (i=0 ; i<end ; i++){
- t1 = (FLOAT_TYPE)(i+1);
- w[end-i-1] = w[n-end+i] = -1 * w[end-i-1] * sin(k1 * t1)/(M_PI * t1); // Sinc
- g += ((i&1) ? (2*w[end-i-1]) : (-2*w[end-i-1])); // Total gain in filter
+ t1 = (FLOAT_TYPE)(i+1);
+ w[end-i-1] = w[n-end+i] = -1 * w[end-i-1] * sin(k1 * t1)/(M_PI * t1); // Sinc
+ g += ((i&1) ? (2*w[end-i-1]) : (-2*w[end-i-1])); // Total gain in filter
}
}
}
@@ -151,32 +151,32 @@ int af_filter_design_fir(unsigned int n, FLOAT_TYPE* w, const FLOAT_TYPE* fc,
if(flags & BP){ // Band pass
// Calculate center tap
if (o){
- g=w[end]*(fc1+fc2);
- w[end] = (fc2 - fc1) * w[end] * 2.0;
+ g=w[end]*(fc1+fc2);
+ w[end] = (fc2 - fc1) * w[end] * 2.0;
}
// Create filter
for (i=0 ; i<end ; i++){
- t1 = (FLOAT_TYPE)(i+1) - k2;
- t2 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2
- t3 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1
- g += w[end-i-1] * (t3 + t2); // Total gain in filter
- w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3);
+ t1 = (FLOAT_TYPE)(i+1) - k2;
+ t2 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2
+ t3 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1
+ g += w[end-i-1] * (t3 + t2); // Total gain in filter
+ w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3);
}
}
else{ // Band stop
if (!o) // Band stop filters must have odd length
- return -1;
+ return -1;
w[end] = 1.0 - (fc2 - fc1) * w[end] * 2.0;
g= w[end];
// Create filter
for (i=0 ; i<end ; i++){
- t1 = (FLOAT_TYPE)(i+1);
- t2 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1
- t3 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2
- w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3);
- g += 2*w[end-i-1]; // Total gain in filter
+ t1 = (FLOAT_TYPE)(i+1);
+ t2 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1
+ t3 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2
+ w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3);
+ g += 2*w[end-i-1]; // Total gain in filter
}
}
}
@@ -224,12 +224,12 @@ static void af_filter_prewarp(FLOAT_TYPE* a, FLOAT_TYPE fc, FLOAT_TYPE fs)
Arguments:
a - s-domain numerator coefficients
b - s-domain denominator coefficients
- k - filter gain factor. Initially set to 1 and modified by each
+ k - filter gain factor. Initially set to 1 and modified by each
biquad section in such a way, as to make it the
coefficient by which to multiply the overall filter gain
in order to achieve a desired overall filter gain,
specified in initial value of k.
- fs - sampling rate (Hz)
+ fs - sampling rate (Hz)
coef - array of z-domain coefficients to be filled in.
Return: On return, set coef z-domain coefficients and k to the gain
@@ -319,13 +319,13 @@ static void af_filter_bilinear(const FLOAT_TYPE* a, const FLOAT_TYPE* b, FLOAT_T
Arguments:
a - s-domain numerator coefficients, a[1] is always assumed to be 1.0
b - s-domain denominator coefficients
- Q - Q value for the filter
- k - filter gain factor. Initially set to 1 and modified by each
+ Q - Q value for the filter
+ k - filter gain factor. Initially set to 1 and modified by each
biquad section in such a way, as to make it the
coefficient by which to multiply the overall filter gain
in order to achieve a desired overall filter gain,
specified in initial value of k.
- fs - sampling rate (Hz)
+ fs - sampling rate (Hz)
coef - array of z-domain coefficients to be filled in.
Note: Upon return from each call, the k argument will be set to a
diff --git a/audio/filter/filter.h b/audio/filter/filter.h
index aed33352c2..f3dea88838 100644
--- a/audio/filter/filter.h
+++ b/audio/filter/filter.h
@@ -49,7 +49,7 @@
#define WINDOW_MASK 0x0000001F
// Parallel filter design
-#define FWD 0x00000001 // Forward indexing of polyphase filter
+#define FWD 0x00000001 // Forward indexing of polyphase filter
#define REW 0x00000002 // Reverse indexing of polyphase filter
#define ODD 0x00000010 // Make filter HP
diff --git a/audio/filter/tools.c b/audio/filter/tools.c
index 217224150e..231477927f 100644
--- a/audio/filter/tools.c
+++ b/audio/filter/tools.c
@@ -115,9 +115,9 @@ int af_test_output(struct af_instance* af, struct mp_audio* out)
float af_softclip(float a)
{
if (a >= M_PI/2)
- return 1.0;
+ return 1.0;
else if (a <= -M_PI/2)
- return -1.0;
+ return -1.0;
else
- return sin(a);
+ return sin(a);
}
diff --git a/audio/filter/window.c b/audio/filter/window.c
index a970bdcbea..089c629f8a 100644
--- a/audio/filter/window.c
+++ b/audio/filter/window.c
@@ -59,7 +59,7 @@ void af_window_triang(int n, FLOAT_TYPE* w)
FLOAT_TYPE k1 = (FLOAT_TYPE)(n & 1);
FLOAT_TYPE k2 = 1/((FLOAT_TYPE)n + k1);
int end = (n + 1) >> 1;
- int i;
+ int i;
// Calculate window coefficients
for (i=0 ; i<end ; i++)
@@ -77,7 +77,7 @@ void af_window_triang(int n, FLOAT_TYPE* w)
*/
void af_window_hanning(int n, FLOAT_TYPE* w)
{
- int i;
+ int i;
FLOAT_TYPE k = 2*M_PI/((FLOAT_TYPE)(n+1)); // 2*pi/(N+1)
// Calculate window coefficients
@@ -201,7 +201,7 @@ void af_window_kaiser(int n, FLOAT_TYPE* w, FLOAT_TYPE b)
{
FLOAT_TYPE tmp;
FLOAT_TYPE k1 = 1.0/besselizero(b);
- int k2 = 1 - (n & 1);
+ int k2 = 1 - (n & 1);
int end = (n + 1) >> 1;
int i;
generated by cgit v1.2.3 (git 2.39.1) at 2024-10-02 07:21:26 +0000