3 * Copyright (c) 2008 Laurent Aimar <fenrir@videolan.org>
4 * Copyright (c) 2009 Baptiste Coudurier <baptiste.coudurier@gmail.com>
6 * This file is part of FFmpeg.
8 * FFmpeg is free software; you can redistribute it and/or
9 * modify it under the terms of the GNU Lesser General Public
10 * License as published by the Free Software Foundation; either
11 * version 2.1 of the License, or (at your option) any later version.
13 * FFmpeg is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16 * Lesser General Public License for more details.
18 * You should have received a copy of the GNU Lesser General Public
19 * License along with FFmpeg; if not, write to the Free Software
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
23 #include "libavutil/intreadwrite.h"
26 #define AES3_HEADER_LEN 4
28 static int s302m_parse_frame_header(AVCodecContext *avctx, const uint8_t *buf,
32 int frame_size, channels, bits;
34 if (buf_size <= AES3_HEADER_LEN) {
35 av_log(avctx, AV_LOG_ERROR, "frame is too short\n");
36 return AVERROR_INVALIDDATA;
49 frame_size = (h >> 16) & 0xffff;
50 channels = ((h >> 14) & 0x0003) * 2 + 2;
51 bits = ((h >> 4) & 0x0003) * 4 + 16;
53 if (AES3_HEADER_LEN + frame_size != buf_size || bits > 24) {
54 av_log(avctx, AV_LOG_ERROR, "frame has invalid header\n");
55 return AVERROR_INVALIDDATA;
58 /* Set output properties */
59 avctx->bits_per_coded_sample = bits;
61 avctx->sample_fmt = SAMPLE_FMT_S32;
63 avctx->sample_fmt = SAMPLE_FMT_S16;
65 avctx->channels = channels;
68 avctx->channel_layout = AV_CH_LAYOUT_STEREO;
71 avctx->channel_layout = AV_CH_LAYOUT_QUAD;
74 avctx->channel_layout = AV_CH_LAYOUT_5POINT1_BACK | AV_CH_LAYOUT_STEREO_DOWNMIX;
76 avctx->sample_rate = 48000;
77 avctx->bit_rate = 48000 * avctx->channels * (avctx->bits_per_coded_sample + 4) +
78 32 * (48000 / (buf_size * 8 /
80 (avctx->bits_per_coded_sample + 4))));
85 static int s302m_decode_frame(AVCodecContext *avctx, void *data,
86 int *data_size, AVPacket *avpkt)
88 const uint8_t *buf = avpkt->data;
89 int buf_size = avpkt->size;
91 int frame_size = s302m_parse_frame_header(avctx, buf, buf_size);
95 buf_size -= AES3_HEADER_LEN;
96 buf += AES3_HEADER_LEN;
98 if (*data_size < 4 * buf_size * 8 / (avctx->bits_per_coded_sample + 4))
101 if (avctx->bits_per_coded_sample == 24) {
103 for (; buf_size > 6; buf_size -= 7) {
104 *o++ = (av_reverse[buf[2]] << 24) |
105 (av_reverse[buf[1]] << 16) |
106 (av_reverse[buf[0]] << 8);
107 *o++ = (av_reverse[buf[6] & 0xf0] << 28) |
108 (av_reverse[buf[5]] << 20) |
109 (av_reverse[buf[4]] << 12) |
110 (av_reverse[buf[3] & 0x0f] << 4);
113 *data_size = (uint8_t*) o - (uint8_t*) data;
114 } else if (avctx->bits_per_coded_sample == 20) {
116 for (; buf_size > 5; buf_size -= 6) {
117 *o++ = (av_reverse[buf[2] & 0xf0] << 28) |
118 (av_reverse[buf[1]] << 20) |
119 (av_reverse[buf[0]] << 12);
120 *o++ = (av_reverse[buf[5] & 0xf0] << 28) |
121 (av_reverse[buf[4]] << 20) |
122 (av_reverse[buf[3]] << 12);
125 *data_size = (uint8_t*) o - (uint8_t*) data;
128 for (; buf_size > 4; buf_size -= 5) {
129 *o++ = (av_reverse[buf[1]] << 8) |
131 *o++ = (av_reverse[buf[4] & 0xf0] << 12) |
132 (av_reverse[buf[3]] << 4) |
133 (av_reverse[buf[2]] >> 4);
136 *data_size = (uint8_t*) o - (uint8_t*) data;
139 return buf - avpkt->data;
143 AVCodec ff_s302m_decoder = {
145 .type = AVMEDIA_TYPE_AUDIO,
146 .id = CODEC_ID_S302M,
148 .decode = s302m_decode_frame,
149 .long_name = NULL_IF_CONFIG_SMALL("SMPTE 302M"),