view resample.c @ 12329:6644e439130d libavcodec

Calculate an exact frame size before writing. Now the buffer size requirements can be known exactly, so larger frame sizes can be safely encoded without buffer overwrite.

/*
 * samplerate conversion for both audio and video
 * Copyright (c) 2000 Fabrice Bellard
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg 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 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg 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 FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

/**
 * @file
 * samplerate conversion for both audio and video
 */

#include "avcodec.h"
#include "audioconvert.h"
#include "opt.h"

struct AVResampleContext;

static const char *context_to_name(void *ptr)
{
    return "audioresample";
}

static const AVOption options[] = {{NULL}};
static const AVClass audioresample_context_class = { "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT };

struct ReSampleContext {
    struct AVResampleContext *resample_context;
    short *temp[2];
    int temp_len;
    float ratio;
    /* channel convert */
    int input_channels, output_channels, filter_channels;
    AVAudioConvert *convert_ctx[2];
    enum SampleFormat sample_fmt[2]; ///< input and output sample format
    unsigned sample_size[2];         ///< size of one sample in sample_fmt
    short *buffer[2];                ///< buffers used for conversion to S16
    unsigned buffer_size[2];         ///< sizes of allocated buffers
};

/* 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 *av_audio_resample_init(int output_channels, int input_channels,
                                        int output_rate, int input_rate,
                                        enum SampleFormat sample_fmt_out,
                                        enum SampleFormat sample_fmt_in,
                                        int filter_length, int log2_phase_count,
                                        int linear, double cutoff)
{
    ReSampleContext *s;

    if ( input_channels > 2)
      {
        av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.\n");
        return NULL;
      }

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

    s->sample_fmt [0] = sample_fmt_in;
    s->sample_fmt [1] = sample_fmt_out;
    s->sample_size[0] = av_get_bits_per_sample_format(s->sample_fmt[0])>>3;
    s->sample_size[1] = av_get_bits_per_sample_format(s->sample_fmt[1])>>3;

    if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
        if (!(s->convert_ctx[0] = av_audio_convert_alloc(SAMPLE_FMT_S16, 1,
                                                         s->sample_fmt[0], 1, NULL, 0))) {
            av_log(s, AV_LOG_ERROR,
                   "Cannot convert %s sample format to s16 sample format\n",
                   avcodec_get_sample_fmt_name(s->sample_fmt[0]));
            av_free(s);
            return NULL;
        }
    }

    if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
        if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
                                                         SAMPLE_FMT_S16, 1, NULL, 0))) {
            av_log(s, AV_LOG_ERROR,
                   "Cannot convert s16 sample format to %s sample format\n",
                   avcodec_get_sample_fmt_name(s->sample_fmt[1]));
            av_audio_convert_free(s->convert_ctx[0]);
            av_free(s);
            return NULL;
        }
    }

/*
 * AC-3 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;

#define TAPS 16
    s->resample_context= av_resample_init(output_rate, input_rate,
                         filter_length, log2_phase_count, linear, cutoff);

    *(const AVClass**)s->resample_context = &audioresample_context_class;

    return s;
}

#if LIBAVCODEC_VERSION_MAJOR < 53
ReSampleContext *audio_resample_init(int output_channels, int input_channels,
                                     int output_rate, int input_rate)
{
    return av_audio_resample_init(output_channels, input_channels,
                                  output_rate, input_rate,
                                  SAMPLE_FMT_S16, SAMPLE_FMT_S16,
                                  TAPS, 10, 0, 0.8);
}
#endif

/* 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];
    short *output_bak = NULL;
    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;
    }

    if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
        int istride[1] = { s->sample_size[0] };
        int ostride[1] = { 2 };
        const void *ibuf[1] = { input };
        void       *obuf[1];
        unsigned input_size = nb_samples*s->input_channels*2;

        if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
            av_free(s->buffer[0]);
            s->buffer_size[0] = input_size;
            s->buffer[0] = av_malloc(s->buffer_size[0]);
            if (!s->buffer[0]) {
                av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
                return 0;
            }
        }

        obuf[0] = s->buffer[0];

        if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
                             ibuf, istride, nb_samples*s->input_channels) < 0) {
            av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format conversion failed\n");
            return 0;
        }

        input  = s->buffer[0];
    }

    lenout= 4*nb_samples * s->ratio + 16;

    if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
        output_bak = output;

        if (!s->buffer_size[1] || s->buffer_size[1] < lenout) {
            av_free(s->buffer[1]);
            s->buffer_size[1] = lenout;
            s->buffer[1] = av_malloc(s->buffer_size[1]);
            if (!s->buffer[1]) {
                av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
                return 0;
            }
        }

        output = s->buffer[1];
    }

    /* XXX: move those malloc to resample init code */
    for(i=0; i<s->filter_channels; i++){
        bufin[i]= 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 */
    bufout[0]= av_malloc( lenout * sizeof(short) );
    bufout[1]= 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);
    }

    if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
        int istride[1] = { 2 };
        int ostride[1] = { s->sample_size[1] };
        const void *ibuf[1] = { output };
        void       *obuf[1] = { output_bak };

        if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
                             ibuf, istride, nb_samples1*s->output_channels) < 0) {
            av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format convertion failed\n");
            return 0;
        }
    }

    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_freep(&s->buffer[0]);
    av_freep(&s->buffer[1]);
    av_audio_convert_free(s->convert_ctx[0]);
    av_audio_convert_free(s->convert_ctx[1]);
    av_free(s);
}