diff options
Diffstat (limited to 'audio/filter/af_equalizer.c')
| -rw-r--r-- | audio/filter/af_equalizer.c | 86 |
1 files changed, 43 insertions, 43 deletions
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; |