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Diffstat (limited to 'contrib/ffmpeg/libavcodec/dtsdec.c')
-rw-r--r--contrib/ffmpeg/libavcodec/dtsdec.c268
1 files changed, 268 insertions, 0 deletions
diff --git a/contrib/ffmpeg/libavcodec/dtsdec.c b/contrib/ffmpeg/libavcodec/dtsdec.c
new file mode 100644
index 000000000..6763572dd
--- /dev/null
+++ b/contrib/ffmpeg/libavcodec/dtsdec.c
@@ -0,0 +1,268 @@
+/*
+ * dtsdec.c : free DTS Coherent Acoustics stream decoder.
+ * Copyright (C) 2004 Benjamin Zores <ben@geexbox.org>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "avcodec.h"
+#include <dts.h>
+
+#include <stdlib.h>
+#include <string.h>
+
+#define BUFFER_SIZE 18726
+#define HEADER_SIZE 14
+
+#define CONVERT_LEVEL 1
+#define CONVERT_BIAS 0
+
+typedef struct DTSContext {
+ dts_state_t *state;
+ uint8_t buf[BUFFER_SIZE];
+ uint8_t *bufptr;
+ uint8_t *bufpos;
+} DTSContext;
+
+static inline int16_t
+convert(sample_t s)
+{
+ return s * 0x7fff;
+}
+
+static void
+convert2s16_multi(sample_t *f, int16_t *s16, int flags)
+{
+ int i;
+
+ switch(flags & (DTS_CHANNEL_MASK | DTS_LFE)){
+ case DTS_MONO:
+ for(i = 0; i < 256; i++){
+ s16[5*i] = s16[5*i+1] = s16[5*i+2] = s16[5*i+3] = 0;
+ s16[5*i+4] = convert(f[i]);
+ }
+ case DTS_CHANNEL:
+ case DTS_STEREO:
+ case DTS_DOLBY:
+ for(i = 0; i < 256; i++){
+ s16[2*i] = convert(f[i]);
+ s16[2*i+1] = convert(f[i+256]);
+ }
+ case DTS_3F:
+ for(i = 0; i < 256; i++){
+ s16[5*i] = convert(f[i+256]);
+ s16[5*i+1] = convert(f[i+512]);
+ s16[5*i+2] = s16[5*i+3] = 0;
+ s16[5*i+4] = convert(f[i]);
+ }
+ case DTS_2F2R:
+ for(i = 0; i < 256; i++){
+ s16[4*i] = convert(f[i]);
+ s16[4*i+1] = convert(f[i+256]);
+ s16[4*i+2] = convert(f[i+512]);
+ s16[4*i+3] = convert(f[i+768]);
+ }
+ case DTS_3F2R:
+ for(i = 0; i < 256; i++){
+ s16[5*i] = convert(f[i+256]);
+ s16[5*i+1] = convert(f[i+512]);
+ s16[5*i+2] = convert(f[i+768]);
+ s16[5*i+3] = convert(f[i+1024]);
+ s16[5*i+4] = convert(f[i]);
+ }
+ case DTS_MONO | DTS_LFE:
+ for(i = 0; i < 256; i++){
+ s16[6*i] = s16[6*i+1] = s16[6*i+2] = s16[6*i+3] = 0;
+ s16[6*i+4] = convert(f[i]);
+ s16[6*i+5] = convert(f[i+256]);
+ }
+ case DTS_CHANNEL | DTS_LFE:
+ case DTS_STEREO | DTS_LFE:
+ case DTS_DOLBY | DTS_LFE:
+ for(i = 0; i < 256; i++){
+ s16[6*i] = convert(f[i]);
+ s16[6*i+1] = convert(f[i+256]);
+ s16[6*i+2] = s16[6*i+3] = s16[6*i+4] = 0;
+ s16[6*i+5] = convert(f[i+512]);
+ }
+ case DTS_3F | DTS_LFE:
+ for(i = 0; i < 256; i++){
+ s16[6*i] = convert(f[i+256]);
+ s16[6*i+1] = convert(f[i+512]);
+ s16[6*i+2] = s16[6*i+3] = 0;
+ s16[6*i+4] = convert(f[i]);
+ s16[6*i+5] = convert(f[i+768]);
+ }
+ case DTS_2F2R | DTS_LFE:
+ for(i = 0; i < 256; i++){
+ s16[6*i] = convert(f[i]);
+ s16[6*i+1] = convert(f[i+256]);
+ s16[6*i+2] = convert(f[i+512]);
+ s16[6*i+3] = convert(f[i+768]);
+ s16[6*i+4] = 0;
+ s16[6*i+5] = convert(f[i+1024]);
+ }
+ case DTS_3F2R | DTS_LFE:
+ for(i = 0; i < 256; i++){
+ s16[6*i] = convert(f[i+256]);
+ s16[6*i+1] = convert(f[i+512]);
+ s16[6*i+2] = convert(f[i+768]);
+ s16[6*i+3] = convert(f[i+1024]);
+ s16[6*i+4] = convert(f[i]);
+ s16[6*i+5] = convert(f[i+1280]);
+ }
+ }
+}
+
+static int
+channels_multi(int flags)
+{
+ switch(flags & (DTS_CHANNEL_MASK | DTS_LFE)){
+ case DTS_CHANNEL:
+ case DTS_STEREO:
+ case DTS_DOLBY:
+ return 2;
+ case DTS_2F2R:
+ return 4;
+ case DTS_MONO:
+ case DTS_3F:
+ case DTS_3F2R:
+ return 5;
+ case DTS_MONO | DTS_LFE:
+ case DTS_CHANNEL | DTS_LFE:
+ case DTS_STEREO | DTS_LFE:
+ case DTS_DOLBY | DTS_LFE:
+ case DTS_3F | DTS_LFE:
+ case DTS_2F2R | DTS_LFE:
+ case DTS_3F2R | DTS_LFE:
+ return 6;
+ }
+
+ return -1;
+}
+
+static int
+dts_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
+ uint8_t * buff, int buff_size)
+{
+ DTSContext *s = avctx->priv_data;
+ uint8_t *start = buff;
+ uint8_t *end = buff + buff_size;
+ int16_t *out_samples = data;
+ int sample_rate;
+ int frame_length;
+ int flags;
+ int bit_rate;
+ int len;
+ level_t level;
+ sample_t bias;
+ int nblocks;
+ int i;
+
+ *data_size = 0;
+
+ while(1) {
+ int length;
+
+ len = end - start;
+ if(!len)
+ break;
+ if(len > s->bufpos - s->bufptr)
+ len = s->bufpos - s->bufptr;
+ memcpy(s->bufptr, start, len);
+ s->bufptr += len;
+ start += len;
+ if(s->bufptr != s->bufpos)
+ return start - buff;
+ if(s->bufpos != s->buf + HEADER_SIZE)
+ break;
+
+ length = dts_syncinfo(s->state, s->buf, &flags, &sample_rate,
+ &bit_rate, &frame_length);
+ if(!length) {
+ av_log(NULL, AV_LOG_INFO, "skip\n");
+ for(s->bufptr = s->buf; s->bufptr < s->buf + HEADER_SIZE - 1; s->bufptr++)
+ s->bufptr[0] = s->bufptr[1];
+ continue;
+ }
+ s->bufpos = s->buf + length;
+ }
+
+ level = CONVERT_LEVEL;
+ bias = CONVERT_BIAS;
+
+ flags |= DTS_ADJUST_LEVEL;
+ if(dts_frame(s->state, s->buf, &flags, &level, bias)) {
+ av_log(avctx, AV_LOG_ERROR, "dts_frame() failed\n");
+ goto end;
+ }
+
+ avctx->sample_rate = sample_rate;
+ avctx->channels = channels_multi(flags);
+ avctx->bit_rate = bit_rate;
+
+ nblocks = dts_blocks_num(s->state);
+
+ for(i = 0; i < nblocks; i++) {
+ if(dts_block(s->state)) {
+ av_log(avctx, AV_LOG_ERROR, "dts_block() failed\n");
+ goto end;
+ }
+
+ convert2s16_multi(dts_samples(s->state), out_samples, flags);
+
+ out_samples += 256 * avctx->channels;
+ *data_size += 256 * sizeof(int16_t) * avctx->channels;
+ }
+
+end:
+ s->bufptr = s->buf;
+ s->bufpos = s->buf + HEADER_SIZE;
+ return start - buff;
+}
+
+static int
+dts_decode_init(AVCodecContext * avctx)
+{
+ DTSContext *s = avctx->priv_data;
+ s->bufptr = s->buf;
+ s->bufpos = s->buf + HEADER_SIZE;
+ s->state = dts_init(0);
+ if(s->state == NULL)
+ return -1;
+
+ return 0;
+}
+
+static int
+dts_decode_end(AVCodecContext * avctx)
+{
+ DTSContext *s = avctx->priv_data;
+ dts_free(s->state);
+ return 0;
+}
+
+AVCodec dts_decoder = {
+ "dts",
+ CODEC_TYPE_AUDIO,
+ CODEC_ID_DTS,
+ sizeof(DTSContext),
+ dts_decode_init,
+ NULL,
+ dts_decode_end,
+ dts_decode_frame,
+};
generated by cgit v1.2.3 (git 2.25.1) at 2025-04-28 03:45:35 +0000