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author | anders <anders@b3059339-0415-0410-9bf9-f77b7e298cf2> | 2002-10-01 06:45:08 +0000 |
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committer | anders <anders@b3059339-0415-0410-9bf9-f77b7e298cf2> | 2002-10-01 06:45:08 +0000 |
commit | 1f6c494641c4ca99eec7a9f47818526c72789439 (patch) | |
tree | 17bf220c1c1ac1144dfa8212f09444e51c248a26 /libaf/af_resample.c | |
parent | c0091278d8f15580eecebfc8c454fba250cf4b94 (diff) | |
download | mpv-1f6c494641c4ca99eec7a9f47818526c72789439.tar.bz2 mpv-1f6c494641c4ca99eec7a9f47818526c72789439.tar.xz |
Adding new audio output filter layer libaf
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@7569 b3059339-0415-0410-9bf9-f77b7e298cf2
Diffstat (limited to 'libaf/af_resample.c')
-rw-r--r-- | libaf/af_resample.c | 340 |
1 files changed, 340 insertions, 0 deletions
diff --git a/libaf/af_resample.c b/libaf/af_resample.c new file mode 100644 index 0000000000..ea9919ecdf --- /dev/null +++ b/libaf/af_resample.c @@ -0,0 +1,340 @@ +/*============================================================================= +// +// This software has been released under the terms of the GNU Public +// license. See http://www.gnu.org/copyleft/gpl.html for details. +// +// Copyright 2002 Anders Johansson ajh@atri.curtin.edu.au +// +//============================================================================= +*/ + +/* This audio filter changes the sample rate. */ + +#define PLUGIN + +#include <stdio.h> +#include <stdlib.h> +#include <unistd.h> +#include <inttypes.h> + +#include "../config.h" +#include "../mp_msg.h" +#include "../libao2/afmt.h" + +#include "af.h" +#include "dsp.h" + +/* Below definition selects the length of each poly phase component. + Valid definitions are L8 and L16, where the number denotes the + length of the filter. This definition affects the computational + complexity (see play()), the performance (see filter.h) and the + memory usage. The filterlenght is choosen to 8 if the machine is + slow and to 16 if the machine is fast and has MMX. +*/ + +#if defined(HAVE_SSE) && !defined(HAVE_3DNOW) // This machine is slow + +#define L 8 // Filter length +// Unrolled loop to speed up execution +#define FIR(x,w,y){ \ + int16_t a = (w[0]*x[0]+w[1]*x[1]+w[2]*x[2]+w[3]*x[3]) >> 16; \ + int16_t b = (w[4]*x[4]+w[5]*x[5]+w[6]*x[6]+w[7]*x[7]) >> 16; \ + (y[0]) = a+b; \ +} + +#else /* Fast machine */ + +#define L 16 +// Unrolled loop to speed up execution +#define FIR(x,w,y){ \ + int16_t a = (w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3] ) >> 16; \ + int16_t b = (w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7] ) >> 16; \ + int16_t c = (w[8] *x[8] +w[9] *x[9] +w[10]*x[10]+w[11]*x[11]) >> 16; \ + int16_t d = (w[12]*x[12]+w[13]*x[13]+w[14]*x[14]+w[15]*x[15]) >> 16; \ + y[0] = (a+b+c+d) >> 1; \ +} + +#endif /* Fast machine */ + +// Macro to add data to circular que +#define ADDQUE(xi,xq,in)\ + xq[xi]=xq[xi+L]=(*in);\ + xi=(--xi)&(L-1); + + + +// local data +typedef struct af_resample_s +{ + int16_t* w; // Current filter weights + int16_t** xq; // Circular buffers + int16_t xi; // Index for circular buffers + int16_t wi; // Index for w + uint16_t i; // Number of new samples to put in x queue + uint16_t dn; // Down sampling factor + uint16_t up; // Up sampling factor +} af_resample_t; + +// Euclids algorithm for calculating Greatest Common Divisor GCD(a,b) +inline int gcd(register int a, register int b) +{ + register int r = min(a,b); + a=max(a,b); + b=r; + + r=a%b; + while(r!=0){ + a=b; + b=r; + r=a%b; + } + return b; +} + +static int upsample(af_data_t* c,af_data_t* l, af_resample_t* s) +{ + uint16_t ci = l->nch; // Index for channels + uint16_t len = 0; // Number of input samples + uint16_t nch = l->nch; // Number of channels + uint16_t inc = s->up/s->dn; + uint16_t level = s->up%s->dn; + uint16_t up = s->up; + uint16_t dn = s->dn; + + register int16_t* w = s->w; + register uint16_t wi = 0; + register uint16_t xi = 0; + + // Index current channel + while(ci--){ + // Temporary pointers + register int16_t* x = s->xq[ci]; + register int16_t* in = ((int16_t*)c->audio)+ci; + register int16_t* out = ((int16_t*)l->audio)+ci; + int16_t* end = in+c->len/2; // Block loop end + wi = s->wi; xi = s->xi; + + while(in < end){ + register uint16_t i = inc; + if(wi<level) i++; + + ADDQUE(xi,x,in); + in+=nch; + while(i--){ + // Run the FIR filter + FIR((&x[xi]),(&w[wi*L]),out); + len++; out+=nch; + // Update wi to point at the correct polyphase component + wi=(wi+dn)%up; + } + } + } + // Save values that needs to be kept for next time + s->wi = wi; + s->xi = xi; + return len; +} + +static int downsample(af_data_t* c,af_data_t* l, af_resample_t* s) +{ + uint16_t ci = l->nch; // Index for channels + uint16_t len = 0; // Number of output samples + uint16_t nch = l->nch; // Number of channels + uint16_t inc = s->dn/s->up; + uint16_t level = s->dn%s->up; + uint16_t up = s->up; + uint16_t dn = s->dn; + + register uint16_t i = 0; + register uint16_t wi = 0; + register uint16_t xi = 0; + + // Index current channel + while(ci--){ + // Temporary pointers + register int16_t* x = s->xq[ci]; + register int16_t* in = ((int16_t*)c->audio)+ci; + register int16_t* out = ((int16_t*)l->audio)+ci; + register int16_t* end = in+c->len/2; // Block loop end + i = s->i; wi = s->wi; xi = s->xi; + + while(in < end){ + + ADDQUE(xi,x,in); + in+=nch; + if(!--i){ + // Run the FIR filter + FIR((&x[xi]),(&s->w[wi*L]),out); + len++; out+=nch; + + // Update wi to point at the correct polyphase component + wi=(wi+dn)%up; + + // Insert i number of new samples in queue + i = inc; + if(wi<level) i++; + } + } + } + // Save values that needs to be kept for next time + s->wi = wi; + s->xi = xi; + s->i = i; + + return len; +} + +// Initialization and runtime control +static int control(struct af_instance_s* af, int cmd, void* arg) +{ + switch(cmd){ + case AF_CONTROL_REINIT:{ + af_resample_t* s = (af_resample_t*)af->setup; + af_data_t* n = (af_data_t*)arg; // New configureation + int i,d = 0; + int rv = AF_OK; + + // Make sure this filter isn't redundant + if(af->data->rate == n->rate) + return AF_DETACH; + + // Create space for circular bufers (if nesessary) + if(af->data->nch != n->nch){ + // First free the old ones + if(s->xq){ + for(i=1;i<af->data->nch;i++) + if(s->xq[i]) + free(s->xq[i]); + free(s->xq); + } + // ... then create new + s->xq = malloc(n->nch*sizeof(int16_t*)); + for(i=0;i<n->nch;i++) + s->xq[i] = malloc(2*L*sizeof(int16_t)); + s->xi = 0; + } + + // Set parameters + af->data->nch = n->nch; + af->data->format = AFMT_S16_LE; + af->data->bps = 2; + if(af->data->format != n->format || af->data->bps != n->bps) + rv = AF_FALSE; + n->format = AFMT_S16_LE; + n->bps = 2; + + // Calculate up and down sampling factors + d=gcd(af->data->rate,n->rate); + + // Check if the the design needs to be redone + if(s->up != af->data->rate/d || s->dn != n->rate/d){ + float* w; + float* wt; + float fc; + int j; + s->up = af->data->rate/d; + s->dn = n->rate/d; + + // Calculate cuttof frequency for filter + fc = 1/(float)(max(s->up,s->dn)); + // Allocate space for polyphase filter bank and protptype filter + w = malloc(sizeof(float) * s->up *L); + if(NULL != s->w) + free(s->w); + s->w = malloc(L*s->up*sizeof(int16_t)); + + // Design prototype filter type using Kaiser window with beta = 10 + if(NULL == w || NULL == s->w || + -1 == design_fir(s->up*L, w, &fc, LP|KAISER , 10.0)){ + mp_msg(MSGT_AFILTER,MSGL_ERR,"[resample] Unable to design prototype filter.\n"); + return AF_ERROR; + } + // Copy data from prototype to polyphase filter + wt=w; + for(j=0;j<L;j++){//Columns + for(i=0;i<s->up;i++){//Rows + float t=(float)s->up*32767.0*(*wt); + s->w[i*L+j] = (int16_t)((t>=0.0)?(t+0.5):(t-0.5)); + wt++; + } + } + free(w); + mp_msg(MSGT_AFILTER,MSGL_V,"[resample] New filter designed up: %i down: %i\n", s->up, s->dn); + } + + // Set multiplier + af->mul.n = s->up; + af->mul.d = s->dn; + return rv; + } + case AF_CONTROL_RESAMPLE: + // Reinit must be called after this function has been called + + // Sanity check + if(((int*)arg)[0] <= 8000 || ((int*)arg)[0] > 192000){ + mp_msg(MSGT_AFILTER,MSGL_ERR,"[resample] The output sample frequency must be between 8kHz and 192kHz. Current value is %i \n",((int*)arg)[0]); + return AF_ERROR; + } + + af->data->rate=((int*)arg)[0]; + mp_msg(MSGT_AFILTER,MSGL_STATUS,"[resample] Changing sample rate to %iHz\n",af->data->rate); + return AF_OK; + } + return AF_UNKNOWN; +} + +// Deallocate memory +static void uninit(struct af_instance_s* af) +{ + if(af->data) + free(af->data); +} + +// Filter data through filter +static af_data_t* play(struct af_instance_s* af, af_data_t* data) +{ + int len = 0; // Length of output data + af_data_t* c = data; // Current working data + af_data_t* l = af->data; // Local data + af_resample_t* s = (af_resample_t*)af->setup; + + if(AF_OK != RESIZE_LOCAL_BUFFER(af,data)) + return NULL; + + // Run resampling + if(s->up>s->dn) + len = upsample(c,l,s); + else + len = downsample(c,l,s); + + // Set output data + c->audio = l->audio; + c->len = len*2; + c->rate = l->rate; + + return c; +} + +// Allocate memory and set function pointers +static int open(af_instance_t* af){ + af->control=control; + af->uninit=uninit; + af->play=play; + af->mul.n=1; + af->mul.d=1; + af->data=calloc(1,sizeof(af_data_t)); + af->setup=calloc(1,sizeof(af_resample_t)); + if(af->data == NULL || af->setup == NULL) + return AF_ERROR; + return AF_OK; +} + +// Description of this plugin +af_info_t af_info_resample = { + "Sample frequency conversion", + "resample", + "Anders", + "", + open +}; + |