comparison resample.c @ 2082:3dc9bbe1b152 libavcodec

polyphase kaiser windowed sinc and blackman nuttall windowed sinc audio resample filters

2081:d3015863f745

 * Sample rate convertion for both audio and video.
 */

#include "avcodec.h"

typedef struct {
    /* fractional resampling */
    uint32_t incr; /* fractional increment */
    uint32_t frac;
    int last_sample;
    /* integer down sample */
    int iratio;  /* integer divison ratio */
    int icount, isum;
    int inv;
} ReSampleChannelContext;

struct ReSampleContext {
    ReSampleChannelContext channel_ctx[2];
    float ratio;
    /* channel convert */
    int input_channels, output_channels, filter_channels;
};


#define FRAC_BITS 16
#define FRAC (1 << FRAC_BITS)

static void init_mono_resample(ReSampleChannelContext *s, float ratio)
{
    ratio = 1.0 / ratio;
    s->iratio = (int)floorf(ratio);
    if (s->iratio == 0)
        s->iratio = 1;
    s->incr = (int)((ratio / s->iratio) * FRAC);
    s->frac = FRAC;
    s->last_sample = 0;
    s->icount = s->iratio;
    s->isum = 0;
    s->inv = (FRAC / s->iratio);
}

/* fractional audio resampling */
static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
{
    unsigned int frac, incr;
    int l0, l1;
    short *q, *p, *pend;

    l0 = s->last_sample;
    incr = s->incr;
    frac = s->frac;

    p = input;
    pend = input + nb_samples;
    q = output;

    l1 = *p++;
    for(;;) {
        /* interpolate */
        *q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS;
        frac = frac + s->incr;
        while (frac >= FRAC) {
            frac -= FRAC;
            if (p >= pend)
                goto the_end;
            l0 = l1;
            l1 = *p++;
        }
    }
 the_end:
    s->last_sample = l1;
    s->frac = frac;
    return q - output;
}

static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
{
    short *q, *p, *pend;
    int c, sum;

    p = input;
    pend = input + nb_samples;
    q = output;

    c = s->icount;
    sum = s->isum;

    for(;;) {
        sum += *p++;
        if (--c == 0) {
            *q++ = (sum * s->inv) >> FRAC_BITS;
            c = s->iratio;
            sum = 0;
        }
        if (p >= pend)
            break;
    }
    s->isum = sum;
    s->icount = c;
    return q - output;
}

/* n1: number of samples */
static void stereo_to_mono(short *output, short *input, int n1)
{
    short *p, *q;

      *output++ = 0;           /* right surroud */
      *output++ = 0;           /* low freq */
    }
}

static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
{
    short *buf1;
    short *buftmp;

    buf1= (short*)av_malloc( nb_samples * sizeof(short) );

    /* first downsample by an integer factor with averaging filter */
    if (s->iratio > 1) {
        buftmp = buf1;
        nb_samples = integer_downsample(s, buftmp, input, nb_samples);
    } else {
        buftmp = input;
    }

    /* then do a fractional resampling with linear interpolation */
    if (s->incr != FRAC) {
        nb_samples = fractional_resample(s, output, buftmp, nb_samples);
    } else {
        memcpy(output, buftmp, nb_samples * sizeof(short));
    }
    av_free(buf1);
    return nb_samples;
}

ReSampleContext *audio_resample_init(int output_channels, int input_channels, 
                                      int output_rate, int input_rate)
{
    ReSampleContext *s;
    int i;

 * expand to 6 channels after the resampling.
 */
    if(s->filter_channels>2)
      s->filter_channels = 2;

    for(i=0;i<s->filter_channels;i++) {
        init_mono_resample(&s->channel_ctx[i], s->ratio);
    }
    return s;
}

/* resample audio. 'nb_samples' is the number of input samples */
/* XXX: optimize it ! */
/* XXX: do it with polyphase filters, since the quality here is
   HORRIBLE. Return the number of samples available in output */
int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
{
    int i, nb_samples1;
    short *bufin[2];
    short *bufout[2];

        memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
        return nb_samples;
    }

    /* XXX: move those malloc to resample init code */
    bufin[0]= (short*) av_malloc( nb_samples * sizeof(short) );
    bufin[1]= (short*) av_malloc( nb_samples * sizeof(short) );
    
    /* make some zoom to avoid round pb */
    lenout= (int)(nb_samples * s->ratio) + 16;
    bufout[0]= (short*) av_malloc( lenout * sizeof(short) );
    bufout[1]= (short*) av_malloc( lenout * sizeof(short) );

    if (s->input_channels == 2 &&
        s->output_channels == 1) {
        buftmp2[0] = bufin[0];
        buftmp3[0] = output;
        stereo_to_mono(buftmp2[0], input, nb_samples);
    } else if (s->output_channels >= 2 && s->input_channels == 1) {
        buftmp2[0] = input;
        buftmp3[0] = bufout[0];
    } else if (s->output_channels >= 2) {
        buftmp2[0] = bufin[0];
        buftmp2[1] = bufin[1];
        buftmp3[0] = bufout[0];
        buftmp3[1] = bufout[1];
        stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
    } else {
        buftmp2[0] = input;
        buftmp3[0] = output;
    }

    /* resample each channel */
    nb_samples1 = 0; /* avoid warning */
    for(i=0;i<s->filter_channels;i++) {
        nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples);
    }

    if (s->output_channels == 2 && s->input_channels == 1) {
        mono_to_stereo(output, buftmp3[0], nb_samples1);
    } else if (s->output_channels == 2) {

    return nb_samples1;
}

void audio_resample_close(ReSampleContext *s)
{
    av_free(s);
}

2082:3dc9bbe1b152

 * Sample rate convertion for both audio and video.
 */

#include "avcodec.h"

struct AVResampleContext;

struct ReSampleContext {
    struct AVResampleContext *resample_context;
    short *temp[2];
    int temp_len;
    float ratio;
    /* channel convert */
    int input_channels, output_channels, filter_channels;
};

/* n1: number of samples */
static void stereo_to_mono(short *output, short *input, int n1)
{
    short *p, *q;

      *output++ = 0;           /* right surroud */
      *output++ = 0;           /* low freq */
    }
}

ReSampleContext *audio_resample_init(int output_channels, int input_channels, 
                                      int output_rate, int input_rate)
{
    ReSampleContext *s;
    int i;

 * expand to 6 channels after the resampling.
 */
    if(s->filter_channels>2)
      s->filter_channels = 2;

    s->resample_context= av_resample_init(output_rate, input_rate);
    
    return s;
}

/* resample audio. 'nb_samples' is the number of input samples */
/* XXX: optimize it ! */
int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
{
    int i, nb_samples1;
    short *bufin[2];
    short *bufout[2];

        memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
        return nb_samples;
    }

    /* XXX: move those malloc to resample init code */
    for(i=0; i<s->filter_channels; i++){
        bufin[i]= (short*) av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
        memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
        buftmp2[i] = bufin[i] + s->temp_len;
    }
    
    /* make some zoom to avoid round pb */
    lenout= (int)(nb_samples * s->ratio) + 16;
    bufout[0]= (short*) av_malloc( lenout * sizeof(short) );
    bufout[1]= (short*) av_malloc( lenout * sizeof(short) );

    if (s->input_channels == 2 &&
        s->output_channels == 1) {
        buftmp3[0] = output;
        stereo_to_mono(buftmp2[0], input, nb_samples);
    } else if (s->output_channels >= 2 && s->input_channels == 1) {
        buftmp3[0] = bufout[0];
        memcpy(buftmp2[0], input, nb_samples*sizeof(short));
    } else if (s->output_channels >= 2) {
        buftmp3[0] = bufout[0];
        buftmp3[1] = bufout[1];
        stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
    } else {
        buftmp3[0] = output;
        memcpy(buftmp2[0], input, nb_samples*sizeof(short));
    }

    nb_samples += s->temp_len;

    /* resample each channel */
    nb_samples1 = 0; /* avoid warning */
    for(i=0;i<s->filter_channels;i++) {
        int consumed;
        int is_last= i+1 == s->filter_channels;

        nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last);
        s->temp_len= nb_samples - consumed;
        s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short));
        memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short));
    }

    if (s->output_channels == 2 && s->input_channels == 1) {
        mono_to_stereo(output, buftmp3[0], nb_samples1);
    } else if (s->output_channels == 2) {

    return nb_samples1;
}

void audio_resample_close(ReSampleContext *s)
{
    av_resample_close(s->resample_context);
    av_freep(&s->temp[0]);
    av_freep(&s->temp[1]);
    av_free(s);
}