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/*
* Copyright (c) 2012 Laurent Aimar
*
* 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
*/
#include "libavutil/intreadwrite.h"
#include "avcodec.h"
#include "internal.h"
typedef struct DVAudioContext {
int block_size;
int is_12bit;
int is_pal;
int16_t shuffle[2000];
} DVAudioContext;
static av_cold int decode_init(AVCodecContext *avctx)
{
DVAudioContext *s = avctx->priv_data;
int i;
if (avctx->channels != 2) {
av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
return AVERROR(EINVAL);
}
if (avctx->codec_tag == 0x0215) {
s->block_size = 7200;
s->is_pal = 0;
} else if (avctx->codec_tag == 0x0216) {
s->block_size = 8640;
s->is_pal = 1;
} else {
return AVERROR(EINVAL);
}
s->is_12bit = avctx->bits_per_raw_sample == 12;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
for (i = 0; i < FF_ARRAY_ELEMS(s->shuffle); i++) {
const unsigned a = s->is_pal ? 18 : 15;
const unsigned b = 3 * a;
s->shuffle[i] = 80 * ((21 * (i % 3) + 9 * (i / 3) + ((i / a) % 3)) % b) +
(2 + s->is_12bit) * (i / b) + 8;
}
return 0;
}
static inline int dv_get_audio_sample_count(const uint8_t *buffer, int dsf)
{
int samples = buffer[0] & 0x3f; /* samples in this frame - min samples */
switch ((buffer[3] >> 3) & 0x07) {
case 0:
return samples + (dsf ? 1896 : 1580);
case 1:
return samples + (dsf ? 1742 : 1452);
case 2:
default:
return samples + (dsf ? 1264 : 1053);
}
}
static inline uint16_t dv_audio_12to16(uint16_t sample)
{
uint16_t shift, result;
sample = (sample < 0x800) ? sample : sample | 0xf000;
shift = (sample & 0xf00) >> 8;
if (shift < 0x2 || shift > 0xd) {
result = sample;
} else if (shift < 0x8) {
shift--;
result = (sample - (256 * shift)) << shift;
} else {
shift = 0xe - shift;
result = ((sample + ((256 * shift) + 1)) << shift) - 1;
}
return result;
}
static int decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *pkt)
{
DVAudioContext *s = avctx->priv_data;
AVFrame *frame = data;
const uint8_t *src = pkt->data;
int16_t *dst;
int ret, i;
if (pkt->size != s->block_size)
return AVERROR_INVALIDDATA;
frame->nb_samples = dv_get_audio_sample_count(pkt->data + 244, s->is_pal);
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
dst = (int16_t *)frame->data[0];
for (i = 0; i < frame->nb_samples; i++) {
const uint8_t *v = &src[s->shuffle[i]];
if (s->is_12bit) {
*dst++ = dv_audio_12to16((v[0] << 4) | ((v[2] >> 4) & 0x0f));
*dst++ = dv_audio_12to16((v[1] << 4) | ((v[2] >> 0) & 0x0f));
} else {
*dst++ = AV_RB16(&v[0]);
*dst++ = AV_RB16(&v[s->is_pal ? 4320 : 3600]);
}
}
*got_frame_ptr = 1;
return pkt->size;
}
AVCodec ff_dvaudio_decoder = {
.name = "dvaudio",
.long_name = NULL_IF_CONFIG_SMALL("Ulead DV Audio"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_DVAUDIO,
.init = decode_init,
.decode = decode_frame,
.capabilities = AV_CODEC_CAP_DR1,
.priv_data_size = sizeof(DVAudioContext),
};
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