mirror of
https://github.com/nyanmisaka/ffmpeg-rockchip.git
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134 lines
3.8 KiB
C
134 lines
3.8 KiB
C
/*
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* Copyright (c) 2012 Laurent Aimar
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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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#include "internal.h"
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#include "dvaudio.h"
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typedef struct DVAudioContext {
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int block_size;
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int is_12bit;
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int is_pal;
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int16_t shuffle[2000];
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} DVAudioContext;
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static av_cold int decode_init(AVCodecContext *avctx)
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{
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DVAudioContext *s = avctx->priv_data;
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int i;
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if (avctx->channels != 2) {
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av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
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return AVERROR(EINVAL);
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}
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if (avctx->codec_tag == 0x0215) {
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s->block_size = 7200;
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} else if (avctx->codec_tag == 0x0216) {
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s->block_size = 8640;
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} else if (avctx->block_align == 7200 ||
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avctx->block_align == 8640) {
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s->block_size = avctx->block_align;
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} else {
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return AVERROR(EINVAL);
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}
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s->is_pal = s->block_size == 8640;
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s->is_12bit = avctx->bits_per_coded_sample == 12;
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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avctx->channel_layout = AV_CH_LAYOUT_STEREO;
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for (i = 0; i < FF_ARRAY_ELEMS(s->shuffle); i++) {
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const unsigned a = s->is_pal ? 18 : 15;
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const unsigned b = 3 * a;
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s->shuffle[i] = 80 * ((21 * (i % 3) + 9 * (i / 3) + ((i / a) % 3)) % b) +
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(2 + s->is_12bit) * (i / b) + 8;
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}
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return 0;
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}
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static inline uint16_t dv_audio_12to16(uint16_t sample)
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{
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uint16_t shift, result;
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sample = (sample < 0x800) ? sample : sample | 0xf000;
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shift = (sample & 0xf00) >> 8;
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if (shift < 0x2 || shift > 0xd) {
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result = sample;
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} else if (shift < 0x8) {
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shift--;
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result = (sample - (256 * shift)) << shift;
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} else {
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shift = 0xe - shift;
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result = ((sample + ((256 * shift) + 1)) << shift) - 1;
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}
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return result;
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}
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static int decode_frame(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *pkt)
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{
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DVAudioContext *s = avctx->priv_data;
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AVFrame *frame = data;
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const uint8_t *src = pkt->data;
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int16_t *dst;
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int ret, i;
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if (pkt->size < s->block_size)
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return AVERROR_INVALIDDATA;
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frame->nb_samples = dv_get_audio_sample_count(pkt->data + 244, s->is_pal);
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
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return ret;
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dst = (int16_t *)frame->data[0];
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for (i = 0; i < frame->nb_samples; i++) {
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const uint8_t *v = &src[s->shuffle[i]];
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if (s->is_12bit) {
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*dst++ = dv_audio_12to16((v[0] << 4) | ((v[2] >> 4) & 0x0f));
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*dst++ = dv_audio_12to16((v[1] << 4) | ((v[2] >> 0) & 0x0f));
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} else {
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*dst++ = AV_RB16(&v[0]);
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*dst++ = AV_RB16(&v[s->is_pal ? 4320 : 3600]);
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}
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}
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*got_frame_ptr = 1;
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return s->block_size;
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}
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AVCodec ff_dvaudio_decoder = {
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.name = "dvaudio",
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.long_name = NULL_IF_CONFIG_SMALL("Ulead DV Audio"),
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_DVAUDIO,
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.init = decode_init,
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.decode = decode_frame,
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.capabilities = AV_CODEC_CAP_DR1,
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.priv_data_size = sizeof(DVAudioContext),
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};
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