mirror of https://github.com/FFmpeg/FFmpeg.git
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/*
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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 "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, id, 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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id = (h >> 6) & 0x00ff; |
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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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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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