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99eb31e263
* qatar/master: (25 commits) Replace custom DEBUG preprocessor trickery by the standard one. vorbis: Remove non-compiling debug statement. vorbis: Remove pointless DEBUG #ifdef around debug output macros. cook: Remove non-compiling debug output. Remove pointless #ifdefs around function declarations in a header. Replace #ifdef + av_log() combinations by av_dlog(). Replace custom debug output functions by av_dlog(). cook: Remove unused debug functions. Remove stray extra arguments from av_dlog() invocations. targa: fix big-endian build v4l2: remove one forgotten use of AVFormatParameters.pix_fmt. vfwcap: add a framerate private option. v4l2: add a framerate private option. libdc1394: add a framerate private option. fbdev: add a framerate private option. bktr: add a framerate private option. oma: check avio_read() return value nutdec: remove unused variable Remove unused variables swscale: allocate larger buffer to handle altivec overreads. ... Conflicts: ffmpeg.c libavcodec/dca.c libavcodec/dirac.c libavcodec/error_resilience.c libavcodec/h264.c libavcodec/mpeg12.c libavcodec/mpeg4videodec.c libavcodec/mpegvideo.c libavcodec/mpegvideo_enc.c libavcodec/pthread.c libavcodec/rv10.c libavcodec/s302m.c libavcodec/shorten.c libavcodec/truemotion2.c libavcodec/utils.c libavdevice/dv1394.c libavdevice/fbdev.c libavdevice/libdc1394.c libavdevice/v4l2.c libavformat/4xm.c libavformat/apetag.c libavformat/asfdec.c libavformat/avidec.c libavformat/mmf.c libavformat/mpeg.c libavformat/mpegenc.c libavformat/mpegts.c libavformat/oggdec.c libavformat/oggparseogm.c libavformat/rl2.c libavformat/rmdec.c libavformat/rpl.c libavformat/rtpdec_latm.c libavformat/sauce.c libavformat/sol.c libswscale/utils.c tests/ref/vsynth1/error tests/ref/vsynth2/error Merged-by: Michael Niedermayer <michaelni@gmx.at>
151 lines
5.0 KiB
C
151 lines
5.0 KiB
C
/*
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* SMPTE 302M decoder
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* Copyright (c) 2008 Laurent Aimar <fenrir@videolan.org>
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* Copyright (c) 2009 Baptiste Coudurier <baptiste.coudurier@gmail.com>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/intreadwrite.h"
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#include "avcodec.h"
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#define AES3_HEADER_LEN 4
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static int s302m_parse_frame_header(AVCodecContext *avctx, const uint8_t *buf,
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int buf_size)
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{
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uint32_t h;
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int frame_size, channels, bits;
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if (buf_size <= AES3_HEADER_LEN) {
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av_log(avctx, AV_LOG_ERROR, "frame is too short\n");
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return AVERROR_INVALIDDATA;
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}
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/*
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* AES3 header :
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* size: 16
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* number channels 2
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* channel_id 8
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* bits per samples 2
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* alignments 4
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*/
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h = AV_RB32(buf);
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frame_size = (h >> 16) & 0xffff;
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channels = ((h >> 14) & 0x0003) * 2 + 2;
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bits = ((h >> 4) & 0x0003) * 4 + 16;
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if (AES3_HEADER_LEN + frame_size != buf_size || bits > 24) {
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av_log(avctx, AV_LOG_ERROR, "frame has invalid header\n");
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return AVERROR_INVALIDDATA;
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}
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/* Set output properties */
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avctx->bits_per_coded_sample = bits;
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if (bits > 16)
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avctx->sample_fmt = SAMPLE_FMT_S32;
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else
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avctx->sample_fmt = SAMPLE_FMT_S16;
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avctx->channels = channels;
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switch(channels) {
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case 2:
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avctx->channel_layout = AV_CH_LAYOUT_STEREO;
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break;
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case 4:
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avctx->channel_layout = AV_CH_LAYOUT_QUAD;
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break;
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case 8:
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avctx->channel_layout = AV_CH_LAYOUT_5POINT1_BACK | AV_CH_LAYOUT_STEREO_DOWNMIX;
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}
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avctx->sample_rate = 48000;
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avctx->bit_rate = 48000 * avctx->channels * (avctx->bits_per_coded_sample + 4) +
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32 * (48000 / (buf_size * 8 /
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(avctx->channels *
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(avctx->bits_per_coded_sample + 4))));
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return frame_size;
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}
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static int s302m_decode_frame(AVCodecContext *avctx, void *data,
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int *data_size, AVPacket *avpkt)
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{
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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int frame_size = s302m_parse_frame_header(avctx, buf, buf_size);
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if (frame_size < 0)
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return frame_size;
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buf_size -= AES3_HEADER_LEN;
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buf += AES3_HEADER_LEN;
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if (*data_size < 4 * buf_size * 8 / (avctx->bits_per_coded_sample + 4))
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return -1;
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if (avctx->bits_per_coded_sample == 24) {
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uint32_t *o = data;
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for (; buf_size > 6; buf_size -= 7) {
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*o++ = (av_reverse[buf[2]] << 24) |
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(av_reverse[buf[1]] << 16) |
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(av_reverse[buf[0]] << 8);
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*o++ = (av_reverse[buf[6] & 0xf0] << 28) |
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(av_reverse[buf[5]] << 20) |
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(av_reverse[buf[4]] << 12) |
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(av_reverse[buf[3] & 0x0f] << 8);
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buf += 7;
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}
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*data_size = (uint8_t*) o - (uint8_t*) data;
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} else if (avctx->bits_per_coded_sample == 20) {
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uint32_t *o = data;
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for (; buf_size > 5; buf_size -= 6) {
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*o++ = (av_reverse[buf[2] & 0xf0] << 28) |
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(av_reverse[buf[1]] << 20) |
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(av_reverse[buf[0]] << 12);
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*o++ = (av_reverse[buf[5] & 0xf0] << 28) |
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(av_reverse[buf[4]] << 20) |
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(av_reverse[buf[3]] << 12);
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buf += 6;
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}
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*data_size = (uint8_t*) o - (uint8_t*) data;
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} else {
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uint16_t *o = data;
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for (; buf_size > 4; buf_size -= 5) {
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*o++ = (av_reverse[buf[1]] << 8) |
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av_reverse[buf[0]];
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*o++ = (av_reverse[buf[4] & 0xf0] << 12) |
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(av_reverse[buf[3]] << 4) |
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av_reverse[buf[2] & 0x0f];
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buf += 5;
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}
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*data_size = (uint8_t*) o - (uint8_t*) data;
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}
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return buf - avpkt->data;
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}
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AVCodec ff_s302m_decoder = {
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.name = "s302m",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = CODEC_ID_S302M,
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.priv_data_size = 0,
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.decode = s302m_decode_frame,
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.long_name = NULL_IF_CONFIG_SMALL("SMPTE 302M"),
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};
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