annotate resample.c @ 1757:3906ddbaffec libavcodec

optimization & bugfix extracted from the 4k line diff between ffmpeg 0.4.7 and http://www.alicestreet.com/ffh263.html the other parts of the diff where 1. spelling fixes (rejected as only a small part of it could be applied automatically) 2. cosmetics (reindention, function reordering, var renaming, ...) with bugs (rejected) 3. rtp related stuff (rejetced as it breaks several codecs) 4. some changes to the intra/inter decission & scene change detection (quality tests needed first)

/*
 * Sample rate convertion for both audio and video
 * Copyright (c) 2000 Fabrice Bellard.
 *
 * This library is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2 of the License, or (at your option) any later version.
 *
 * This library 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
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with this library; if not, write to the Free Software
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 */

/**
 * @file resample.c
 * 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;
    int n = n1;

    p = input;
    q = output;
    while (n >= 4) {
        q[0] = (p[0] + p[1]) >> 1;
        q[1] = (p[2] + p[3]) >> 1;
        q[2] = (p[4] + p[5]) >> 1;
        q[3] = (p[6] + p[7]) >> 1;
        q += 4;
        p += 8;
        n -= 4;
    }
    while (n > 0) {
        q[0] = (p[0] + p[1]) >> 1;
        q++;
        p += 2;
        n--;
    }
}

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

    p = input;
    q = output;
    while (n >= 4) {
        v = p[0]; q[0] = v; q[1] = v;
        v = p[1]; q[2] = v; q[3] = v;
        v = p[2]; q[4] = v; q[5] = v;
        v = p[3]; q[6] = v; q[7] = v;
        q += 8;
        p += 4;
        n -= 4;
    }
    while (n > 0) {
        v = p[0]; q[0] = v; q[1] = v;
        q += 2;
        p += 1;
        n--;
    }
}

/* XXX: should use more abstract 'N' channels system */
static void stereo_split(short *output1, short *output2, short *input, int n)
{
    int i;

    for(i=0;i<n;i++) {
        *output1++ = *input++;
        *output2++ = *input++;
    }
}

static void stereo_mux(short *output, short *input1, short *input2, int n)
{
    int i;

    for(i=0;i<n;i++) {
        *output++ = *input1++;
        *output++ = *input2++;
    }
}

static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
{
    int i;
    short l,r;

    for(i=0;i<n;i++) {
      l=*input1++;
      r=*input2++;
      *output++ = l;           /* left */
      *output++ = (l/2)+(r/2); /* center */
      *output++ = r;           /* right */
      *output++ = 0;           /* left surround */
      *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;
    
    if ( input_channels > 2)
      {
	av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.");
	return NULL;
      }

    s = av_mallocz(sizeof(ReSampleContext));
    if (!s)
      {
	av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.");
	return NULL;
      }

    s->ratio = (float)output_rate / (float)input_rate;
    
    s->input_channels = input_channels;
    s->output_channels = output_channels;
    
    s->filter_channels = s->input_channels;
    if (s->output_channels < s->filter_channels)
        s->filter_channels = s->output_channels;

/*
 * ac3 output is the only case where filter_channels could be greater than 2.
 * input channels can't be greater than 2, so resample the 2 channels and then
 * 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];
    short *buftmp2[2], *buftmp3[2];
    int lenout;

    if (s->input_channels == s->output_channels && s->ratio == 1.0) {
        /* nothing to do */
        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) {
        stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
    } else if (s->output_channels == 6) {
        ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
    }

    av_free(bufin[0]);
    av_free(bufin[1]);

    av_free(bufout[0]);
    av_free(bufout[1]);
    return nb_samples1;
}

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