PeerStreamer

/*

 * 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

 */

/**

 * @file resample.c

 * 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;

    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 */

    }

}

ReSampleContext *audio_resample_init(int output_channels, int input_channels,

                                      int output_rate, int input_rate)

{

    ReSampleContext *s;

    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;

    s->resample_context= av_resample_init(output_rate, input_rate, 16, 10, 0, 1.0);

    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];

    short *buftmp2[2], *buftmp3[2];

    int lenout;

    if (s->input_channels == s->output_channels && s->ratio == 1.0 && 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 */

    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) {

        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);

    }

    for(i=0; i<s->filter_channels; i++)

        av_free(bufin[i]);

    av_free(bufout[0]);

    av_free(bufout[1]);

    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);

}