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ffmpeg / libavcodec / resample.c @ 9c89585a

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/*
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 * Sample rate convertion for both audio and video
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 * Copyright (c) 2000 Fabrice Bellard.
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 *
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 * This library is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2 of the License, or (at your option) any later version.
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 *
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 * This library is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with this library; if not, write to the Free Software
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 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
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 */
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#include "avcodec.h"
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typedef struct {
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    /* fractional resampling */
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    UINT32 incr; /* fractional increment */
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    UINT32 frac;
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    int last_sample;
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    /* integer down sample */
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    int iratio;  /* integer divison ratio */
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    int icount, isum;
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    int inv;
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} ReSampleChannelContext;
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struct ReSampleContext {
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    ReSampleChannelContext channel_ctx[2];
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    float ratio;
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    /* channel convert */
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    int input_channels, output_channels, filter_channels;
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};
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#define FRAC_BITS 16
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#define FRAC (1 << FRAC_BITS)
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static void init_mono_resample(ReSampleChannelContext *s, float ratio)
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{
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    ratio = 1.0 / ratio;
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    s->iratio = (int)floor(ratio);
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    if (s->iratio == 0)
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        s->iratio = 1;
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    s->incr = (int)((ratio / s->iratio) * FRAC);
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    s->frac = FRAC;
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    s->last_sample = 0;
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    s->icount = s->iratio;
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    s->isum = 0;
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    s->inv = (FRAC / s->iratio);
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}
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/* fractional audio resampling */
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static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
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{
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    unsigned int frac, incr;
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    int l0, l1;
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    short *q, *p, *pend;
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    l0 = s->last_sample;
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    incr = s->incr;
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    frac = s->frac;
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    p = input;
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    pend = input + nb_samples;
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    q = output;
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    l1 = *p++;
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    for(;;) {
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        /* interpolate */
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        *q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS;
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        frac = frac + s->incr;
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        while (frac >= FRAC) {
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            frac -= FRAC;
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            if (p >= pend)
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                goto the_end;
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            l0 = l1;
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            l1 = *p++;
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        }
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    }
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 the_end:
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    s->last_sample = l1;
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    s->frac = frac;
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    return q - output;
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}
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static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
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{
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    short *q, *p, *pend;
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    int c, sum;
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    p = input;
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    pend = input + nb_samples;
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    q = output;
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    c = s->icount;
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    sum = s->isum;
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    for(;;) {
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        sum += *p++;
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        if (--c == 0) {
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            *q++ = (sum * s->inv) >> FRAC_BITS;
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            c = s->iratio;
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            sum = 0;
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        }
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        if (p >= pend)
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            break;
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    }
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    s->isum = sum;
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    s->icount = c;
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    return q - output;
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}
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/* n1: number of samples */
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static void stereo_to_mono(short *output, short *input, int n1)
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{
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    short *p, *q;
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    int n = n1;
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    p = input;
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    q = output;
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    while (n >= 4) {
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        q[0] = (p[0] + p[1]) >> 1;
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        q[1] = (p[2] + p[3]) >> 1;
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        q[2] = (p[4] + p[5]) >> 1;
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        q[3] = (p[6] + p[7]) >> 1;
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        q += 4;
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        p += 8;
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        n -= 4;
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    }
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    while (n > 0) {
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        q[0] = (p[0] + p[1]) >> 1;
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        q++;
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        p += 2;
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        n--;
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    }
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}
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/* n1: number of samples */
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static void mono_to_stereo(short *output, short *input, int n1)
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{
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    short *p, *q;
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    int n = n1;
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    int v;
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    p = input;
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    q = output;
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    while (n >= 4) {
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        v = p[0]; q[0] = v; q[1] = v;
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        v = p[1]; q[2] = v; q[3] = v;
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        v = p[2]; q[4] = v; q[5] = v;
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        v = p[3]; q[6] = v; q[7] = v;
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        q += 8;
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        p += 4;
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        n -= 4;
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    }
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    while (n > 0) {
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        v = p[0]; q[0] = v; q[1] = v;
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        q += 2;
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        p += 1;
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        n--;
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    }
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}
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/* XXX: should use more abstract 'N' channels system */
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static void stereo_split(short *output1, short *output2, short *input, int n)
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{
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    int i;
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    for(i=0;i<n;i++) {
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        *output1++ = *input++;
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        *output2++ = *input++;
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    }
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}
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static void stereo_mux(short *output, short *input1, short *input2, int n)
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{
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    int i;
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    for(i=0;i<n;i++) {
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        *output++ = *input1++;
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        *output++ = *input2++;
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    }
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}
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static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
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{
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    short *buf1;
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    short *buftmp;
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    buf1= (short*)av_malloc( nb_samples * sizeof(short) );
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    /* first downsample by an integer factor with averaging filter */
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    if (s->iratio > 1) {
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        buftmp = buf1;
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        nb_samples = integer_downsample(s, buftmp, input, nb_samples);
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    } else {
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        buftmp = input;
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    }
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    /* then do a fractional resampling with linear interpolation */
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    if (s->incr != FRAC) {
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        nb_samples = fractional_resample(s, output, buftmp, nb_samples);
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    } else {
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        memcpy(output, buftmp, nb_samples * sizeof(short));
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    }
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    av_free(buf1);
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    return nb_samples;
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}
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ReSampleContext *audio_resample_init(int output_channels, int input_channels, 
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                                      int output_rate, int input_rate)
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{
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    ReSampleContext *s;
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    int i;
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    if (output_channels > 2 || input_channels > 2)
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        return NULL;
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    s = av_mallocz(sizeof(ReSampleContext));
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    if (!s)
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        return NULL;
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    s->ratio = (float)output_rate / (float)input_rate;
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    s->input_channels = input_channels;
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    s->output_channels = output_channels;
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    s->filter_channels = s->input_channels;
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    if (s->output_channels < s->filter_channels)
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        s->filter_channels = s->output_channels;
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    for(i=0;i<s->filter_channels;i++) {
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        init_mono_resample(&s->channel_ctx[i], s->ratio);
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    }
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    return s;
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}
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/* resample audio. 'nb_samples' is the number of input samples */
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/* XXX: optimize it ! */
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/* XXX: do it with polyphase filters, since the quality here is
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   HORRIBLE. Return the number of samples available in output */
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int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
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{
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    int i, nb_samples1;
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    short *bufin[2];
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    short *bufout[2];
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    short *buftmp2[2], *buftmp3[2];
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    int lenout;
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    if (s->input_channels == s->output_channels && s->ratio == 1.0) {
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        /* nothing to do */
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        memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
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        return nb_samples;
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    }
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    /* XXX: move those malloc to resample init code */
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    bufin[0]= (short*) av_malloc( nb_samples * sizeof(short) );
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    bufin[1]= (short*) av_malloc( nb_samples * sizeof(short) );
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    /* make some zoom to avoid round pb */
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    lenout= (int)(nb_samples * s->ratio) + 16;
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    bufout[0]= (short*) av_malloc( lenout * sizeof(short) );
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    bufout[1]= (short*) av_malloc( lenout * sizeof(short) );
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    if (s->input_channels == 2 &&
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        s->output_channels == 1) {
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        buftmp2[0] = bufin[0];
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        buftmp3[0] = output;
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        stereo_to_mono(buftmp2[0], input, nb_samples);
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    } else if (s->output_channels == 2 && s->input_channels == 1) {
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        buftmp2[0] = input;
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        buftmp3[0] = bufout[0];
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    } else if (s->output_channels == 2) {
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        buftmp2[0] = bufin[0];
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        buftmp2[1] = bufin[1];
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        buftmp3[0] = bufout[0];
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        buftmp3[1] = bufout[1];
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        stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
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    } else {
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        buftmp2[0] = input;
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        buftmp3[0] = output;
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    }
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    /* resample each channel */
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    nb_samples1 = 0; /* avoid warning */
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    for(i=0;i<s->filter_channels;i++) {
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        nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples);
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    }
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    if (s->output_channels == 2 && s->input_channels == 1) {
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        mono_to_stereo(output, buftmp3[0], nb_samples1);
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    } else if (s->output_channels == 2) {
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        stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
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    }
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    av_free(bufin[0]);
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    av_free(bufin[1]);
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    av_free(bufout[0]);
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    av_free(bufout[1]);
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    return nb_samples1;
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}
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void audio_resample_close(ReSampleContext *s)
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{
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    av_free(s);
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}