mirror of https://git.ffmpeg.org/ffmpeg.git
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 Libav.
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*
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* Libav 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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* Libav 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 Libav; 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 "libavutil/log.h"
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#include "avcodec.h"
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#include "internal.h"
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#include "mathops.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 = AV_SAMPLE_FMT_S32;
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else
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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avctx->channels = channels;
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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 *got_frame_ptr, AVPacket *avpkt)
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{
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AVFrame *frame = data;
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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int block_size, ret;
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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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/* get output buffer */
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block_size = (avctx->bits_per_coded_sample + 4) / 4;
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frame->nb_samples = 2 * (buf_size / block_size) / avctx->channels;
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
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av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
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return ret;
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}
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buf_size = (frame->nb_samples * avctx->channels / 2) * block_size;
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if (avctx->bits_per_coded_sample == 24) {
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uint32_t *o = (uint32_t *)frame->data[0];
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for (; buf_size > 6; buf_size -= 7) {
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*o++ = (ff_reverse[buf[2]] << 24) |
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(ff_reverse[buf[1]] << 16) |
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(ff_reverse[buf[0]] << 8);
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*o++ = (ff_reverse[buf[6] & 0xf0] << 28) |
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(ff_reverse[buf[5]] << 20) |
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(ff_reverse[buf[4]] << 12) |
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(ff_reverse[buf[3] & 0x0f] << 4);
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buf += 7;
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}
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} else if (avctx->bits_per_coded_sample == 20) {
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uint32_t *o = (uint32_t *)frame->data[0];
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for (; buf_size > 5; buf_size -= 6) {
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*o++ = (ff_reverse[buf[2] & 0xf0] << 28) |
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(ff_reverse[buf[1]] << 20) |
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(ff_reverse[buf[0]] << 12);
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*o++ = (ff_reverse[buf[5] & 0xf0] << 28) |
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(ff_reverse[buf[4]] << 20) |
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(ff_reverse[buf[3]] << 12);
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buf += 6;
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}
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} else {
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uint16_t *o = (uint16_t *)frame->data[0];
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for (; buf_size > 4; buf_size -= 5) {
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*o++ = (ff_reverse[buf[1]] << 8) |
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ff_reverse[buf[0]];
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*o++ = (ff_reverse[buf[4] & 0xf0] << 12) |
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(ff_reverse[buf[3]] << 4) |
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(ff_reverse[buf[2]] >> 4);
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buf += 5;
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}
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}
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*got_frame_ptr = 1;
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return avpkt->size;
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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 = AV_CODEC_ID_S302M,
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.decode = s302m_decode_frame,
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.capabilities = CODEC_CAP_DR1,
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.long_name = NULL_IF_CONFIG_SMALL("SMPTE 302M"),
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};
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