annotate resample.c @ 8991:ca768cb2bfb6 libavcodec

Use last decoded SPS as current SPS in order to parse picture timing SEI correctly. This works around an apparent H.264 standard deficiency. Patch by Ivan Schreter, schreter gmx net

/*
 * 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 libavcodec/resample.c
 * 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 };

struct ReSampleContext {
    const AVClass *av_class;
    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);

    s->av_class= &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, 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, 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, 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, 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);
}