mpv/libmpcodecs/ad_pcm.c

/*
 * This file is part of MPlayer.
 *
 * MPlayer is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * MPlayer 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 General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License along
 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
 */

#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <stdbool.h>

#include "talloc.h"
#include "config.h"
#include "ad_internal.h"
#include "libaf/format.h"
#include "libaf/reorder_ch.h"

static const ad_info_t info = {
    "Uncompressed PCM audio decoder",
    "pcm",
    "Nick Kurshev",
    "A'rpi",
    ""
};

struct ad_pcm_context {
    unsigned char *buffer;
    int buffer_pos;
    int buffer_len;
    int buffer_size;
};

LIBAD_EXTERN(pcm)

static int init(sh_audio_t * sh_audio)
{
    WAVEFORMATEX *h = sh_audio->wf;
    if (!h)
        return 0;
    sh_audio->i_bps = h->nAvgBytesPerSec;
    sh_audio->channels = h->nChannels;
    sh_audio->samplerate = h->nSamplesPerSec;
    sh_audio->samplesize = (h->wBitsPerSample + 7) / 8;
    sh_audio->sample_format = AF_FORMAT_S16_LE; // default
    switch (sh_audio->format) { /* hardware formats: */
    case 0x0:
    case 0x1:                  // Microsoft PCM
    case 0xfffe:               // Extended
        switch (sh_audio->samplesize) {
        case 1: sh_audio->sample_format = AF_FORMAT_U8;     break;
        case 2: sh_audio->sample_format = AF_FORMAT_S16_LE; break;
        case 3: sh_audio->sample_format = AF_FORMAT_S24_LE; break;
        case 4: sh_audio->sample_format = AF_FORMAT_S32_LE; break;
        }
        break;
    case 0x3:                  // IEEE float
        sh_audio->sample_format = AF_FORMAT_FLOAT_LE;
        break;
    case 0x6:  sh_audio->sample_format = AF_FORMAT_A_LAW;      break;
    case 0x7:  sh_audio->sample_format = AF_FORMAT_MU_LAW;     break;
    case 0x11: sh_audio->sample_format = AF_FORMAT_IMA_ADPCM;  break;
    case 0x50: sh_audio->sample_format = AF_FORMAT_MPEG2;      break;
/*    case 0x2000: sh_audio->sample_format=AFMT_AC3; */
    case 0x20776172: // 'raw '
        sh_audio->sample_format = AF_FORMAT_S16_BE;
        if (sh_audio->samplesize == 1)
            sh_audio->sample_format = AF_FORMAT_U8;
        break;
    case 0x736F7774: // 'twos'
        sh_audio->sample_format = AF_FORMAT_S16_BE;
        // intended fall-through
    case 0x74776F73: // 'sowt'
        if (sh_audio->samplesize == 1)
            sh_audio->sample_format = AF_FORMAT_S8;
        break;
    case 0x32336c66: // 'fl32', bigendian float32
    case 0x32334C46: // 'FL32', bigendian float32 in aiff
        sh_audio->sample_format = AF_FORMAT_FLOAT_BE;
        sh_audio->samplesize = 4;
        break;
    case 0x666c3332: // '23lf', little endian float32, MPlayer internal fourCC
    case 0x6D63706C: // 'lpcm'
        sh_audio->sample_format = AF_FORMAT_FLOAT_LE;
        sh_audio->samplesize = 4;
        break;
/*    case 0x34366c66: // 'fl64', bigendian float64
       sh_audio->sample_format=AF_FORMAT_FLOAT_BE;
       sh_audio->samplesize=8;
       break;
    case 0x666c3634: // '46lf', little endian float64, MPlayer internal fourCC
       sh_audio->sample_format=AF_FORMAT_FLOAT_LE;
       sh_audio->samplesize=8;
       break;*/
    case 0x34326e69: // 'in24', bigendian int24
        sh_audio->sample_format = AF_FORMAT_S24_BE;
        sh_audio->samplesize = 3;
        break;
    case 0x696e3234: // '42ni', little endian int24, MPlayer internal fourCC
        sh_audio->sample_format = AF_FORMAT_S24_LE;
        sh_audio->samplesize = 3;
        break;
    case 0x32336e69: // 'in32', bigendian int32
        sh_audio->sample_format = AF_FORMAT_S32_BE;
        sh_audio->samplesize = 4;
        break;
    case 0x696e3332: // '23ni', little endian int32, MPlayer internal fourCC
        sh_audio->sample_format = AF_FORMAT_S32_LE;
        sh_audio->samplesize = 4;
        break;
    default:
        if (sh_audio->samplesize != 2)
            sh_audio->sample_format = AF_FORMAT_U8;
    }
    if (!sh_audio->samplesize)  // this would cause MPlayer to hang later
        sh_audio->samplesize = 2;
    sh_audio->context = talloc_zero(NULL, struct ad_pcm_context);
    return 1;
}

static int preinit(sh_audio_t *sh)
{
    sh->audio_out_minsize = 2048;
    return 1;
}

static void uninit(sh_audio_t *sh)
{
    talloc_free(sh->context);
}

static int control(sh_audio_t *sh, int cmd, void *arg, ...)
{
    struct ad_pcm_context *ctx = sh->context;
    int skip;
    switch (cmd) {
    case ADCTRL_RESYNC_STREAM:
        ctx->buffer_len = 0;
        return true;
    case ADCTRL_SKIP_FRAME:
        skip = sh->i_bps / 16;
        skip = skip & (~3);
        demux_read_data(sh->ds, NULL, skip);
        return CONTROL_TRUE;
    }
    return CONTROL_UNKNOWN;
}

static int decode_audio(sh_audio_t *sh_audio, unsigned char *buf, int minlen,
                        int maxlen)
{
    int unitsize = sh_audio->channels * sh_audio->samplesize;
    minlen = (minlen + unitsize - 1) / unitsize * unitsize;
    if (minlen > maxlen)
        // if someone needs hundreds of channels adjust audio_out_minsize
        // based on channels in preinit()
        return -1;

    int len = 0;
    struct ad_pcm_context *ctx = sh_audio->context;
    while (len < minlen) {
        if (ctx->buffer_len - ctx->buffer_pos <= 0) {
            double pts;
            unsigned char *ptr;
            int plen = ds_get_packet_pts(sh_audio->ds, &ptr, &pts);
            if (plen < 0)
                break;
            if (ctx->buffer_size < plen) {
                talloc_free(ctx->buffer);
                ctx->buffer = talloc_size(ctx, plen);
                ctx->buffer_size = plen;
            }
            memcpy(ctx->buffer, ptr, plen);
            ctx->buffer_len = plen;
            ctx->buffer_pos = 0;
            if (pts != MP_NOPTS_VALUE) {
                sh_audio->pts = pts;
                sh_audio->pts_bytes = 0;
            }
        }
        int from_stored = ctx->buffer_len - ctx->buffer_pos;
        if (from_stored > minlen - len)
            from_stored = minlen - len;
        memcpy(buf + len, ctx->buffer + ctx->buffer_pos, from_stored);
        ctx->buffer_pos += from_stored;
        sh_audio->pts_bytes += from_stored;
        len += from_stored;
    }
    if (len % unitsize) {
        mp_msg(MSGT_DECAUDIO, MSGL_WARN, "[ad_pcm] discarding partial sample "
               "at end\n");
        len -= len % unitsize;
    }
    if (len == 0)
        len = -1;               // The loop above only exits at error/EOF
    if (len > 0 && sh_audio->channels >= 5) {
        reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
                            AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
                            sh_audio->channels, len / sh_audio->samplesize,
                            sh_audio->samplesize);
    }
    return len;
}