ffmpeg/libavcodec/s302m.c

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/*
* SMPTE 302M decoder
* Copyright (c) 2008 Laurent Aimar <fenrir@videolan.org>
* Copyright (c) 2009 Baptiste Coudurier <baptiste.coudurier@gmail.com>
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/intreadwrite.h"
#include "libavutil/log.h"
#include "avcodec.h"
#include "internal.h"
#include "mathops.h"
#define AES3_HEADER_LEN 4
static int s302m_parse_frame_header(AVCodecContext *avctx, const uint8_t *buf,
int buf_size)
{
uint32_t h;
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int frame_size, channels, bits;
if (buf_size <= AES3_HEADER_LEN) {
av_log(avctx, AV_LOG_ERROR, "frame is too short\n");
return AVERROR_INVALIDDATA;
}
/*
* AES3 header :
* size: 16
* number channels 2
* channel_id 8
* bits per samples 2
* alignments 4
*/
h = AV_RB32(buf);
frame_size = (h >> 16) & 0xffff;
channels = ((h >> 14) & 0x0003) * 2 + 2;
bits = ((h >> 4) & 0x0003) * 4 + 16;
if (AES3_HEADER_LEN + frame_size != buf_size || bits > 24) {
av_log(avctx, AV_LOG_ERROR, "frame has invalid header\n");
return AVERROR_INVALIDDATA;
}
/* Set output properties */
avctx->bits_per_coded_sample = bits;
if (bits > 16)
avctx->sample_fmt = AV_SAMPLE_FMT_S32;
else
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avctx->channels = channels;
avctx->sample_rate = 48000;
avctx->bit_rate = 48000 * avctx->channels * (avctx->bits_per_coded_sample + 4) +
32 * (48000 / (buf_size * 8 /
(avctx->channels *
(avctx->bits_per_coded_sample + 4))));
return frame_size;
}
static int s302m_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
AVFrame *frame = data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int block_size, ret;
int frame_size = s302m_parse_frame_header(avctx, buf, buf_size);
if (frame_size < 0)
return frame_size;
buf_size -= AES3_HEADER_LEN;
buf += AES3_HEADER_LEN;
/* get output buffer */
block_size = (avctx->bits_per_coded_sample + 4) / 4;
frame->nb_samples = 2 * (buf_size / block_size) / avctx->channels;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
buf_size = (frame->nb_samples * avctx->channels / 2) * block_size;
if (avctx->bits_per_coded_sample == 24) {
uint32_t *o = (uint32_t *)frame->data[0];
for (; buf_size > 6; buf_size -= 7) {
*o++ = (ff_reverse[buf[2]] << 24) |
(ff_reverse[buf[1]] << 16) |
(ff_reverse[buf[0]] << 8);
*o++ = (ff_reverse[buf[6] & 0xf0] << 28) |
(ff_reverse[buf[5]] << 20) |
(ff_reverse[buf[4]] << 12) |
(ff_reverse[buf[3] & 0x0f] << 4);
buf += 7;
}
} else if (avctx->bits_per_coded_sample == 20) {
uint32_t *o = (uint32_t *)frame->data[0];
for (; buf_size > 5; buf_size -= 6) {
*o++ = (ff_reverse[buf[2] & 0xf0] << 28) |
(ff_reverse[buf[1]] << 20) |
(ff_reverse[buf[0]] << 12);
*o++ = (ff_reverse[buf[5] & 0xf0] << 28) |
(ff_reverse[buf[4]] << 20) |
(ff_reverse[buf[3]] << 12);
buf += 6;
}
} else {
uint16_t *o = (uint16_t *)frame->data[0];
for (; buf_size > 4; buf_size -= 5) {
*o++ = (ff_reverse[buf[1]] << 8) |
ff_reverse[buf[0]];
*o++ = (ff_reverse[buf[4] & 0xf0] << 12) |
(ff_reverse[buf[3]] << 4) |
(ff_reverse[buf[2]] >> 4);
buf += 5;
}
}
*got_frame_ptr = 1;
return avpkt->size;
}
AVCodec ff_s302m_decoder = {
.name = "s302m",
.long_name = NULL_IF_CONFIG_SMALL("SMPTE 302M"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_S302M,
.decode = s302m_decode_frame,
.capabilities = AV_CODEC_CAP_DR1,
};