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authorMichael Niedermayer <michaelni@gmx.at>2004-07-14 01:32:14 +0000
committerMichael Niedermayer <michaelni@gmx.at>2004-07-14 01:32:14 +0000
commit23c992532927afa9d3a00677ff40cd071e21dc8f (patch)
tree232b97558b925172d4c6372c10a5c7e156469f27 /libavcodec/dtsdec.c
parenteb507b21c410515b179c0ca85b3db3d83fc296bd (diff)
libdts support by (Benjamin Zores <ben at geexbox dot org>)
Originally committed as revision 3310 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec/dtsdec.c')
-rw-r--r--libavcodec/dtsdec.c349
1 files changed, 349 insertions, 0 deletions
diff --git a/libavcodec/dtsdec.c b/libavcodec/dtsdec.c
new file mode 100644
index 0000000000..d683a08e61
--- /dev/null
+++ b/libavcodec/dtsdec.c
@@ -0,0 +1,349 @@
+/*
+ * dtsdec.c : free DTS Coherent Acoustics stream decoder.
+ * Copyright (C) 2004 Benjamin Zores <ben@geexbox.org>
+ *
+ * This file is part of libavcodec.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifdef HAVE_AV_CONFIG_H
+#undef HAVE_AV_CONFIG_H
+#endif
+
+#include "avcodec.h"
+#include <dts.h>
+#include "dts_internal.h"
+
+#include <stdlib.h>
+#include <string.h>
+#include <malloc.h>
+#include <math.h>
+
+#define INBUF_SIZE 4096
+#define BUFFER_SIZE 4096
+#define HEADER_SIZE 14
+
+#ifdef LIBDTS_FIXED
+#define CONVERT_LEVEL (1 << 26)
+#define CONVERT_BIAS 0
+#else
+#define CONVERT_LEVEL 1
+#define CONVERT_BIAS 384
+#endif
+
+static void
+pre_calc_cosmod (dts_state_t * state)
+{
+ int i, j, k;
+
+ for (j=0,k=0;k<16;k++)
+ for (i=0;i<16;i++)
+ state->cos_mod[j++] = cos((2*i+1)*(2*k+1)*M_PI/64);
+
+ for (k=0;k<16;k++)
+ for (i=0;i<16;i++)
+ state->cos_mod[j++] = cos((i)*(2*k+1)*M_PI/32);
+
+ for (k=0;k<16;k++)
+ state->cos_mod[j++] = 0.25/(2*cos((2*k+1)*M_PI/128));
+
+ for (k=0;k<16;k++)
+ state->cos_mod[j++] = -0.25/(2.0*sin((2*k+1)*M_PI/128));
+}
+
+static inline
+int16_t convert (int32_t i)
+{
+#ifdef LIBDTS_FIXED
+ i >>= 15;
+#else
+ i -= 0x43c00000;
+#endif
+ return (i > 32767) ? 32767 : ((i < -32768) ? -32768 : i);
+}
+
+void
+convert2s16_2 (sample_t * _f, int16_t * s16)
+{
+ int i;
+ int32_t * f = (int32_t *) _f;
+
+ for (i = 0; i < 256; i++)
+ {
+ s16[2*i] = convert (f[i]);
+ s16[2*i+1] = convert (f[i+256]);
+ }
+}
+
+void
+convert2s16_4 (sample_t * _f, int16_t * s16)
+{
+ int i;
+ int32_t * f = (int32_t *) _f;
+
+ 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]);
+ }
+}
+
+void
+convert2s16_5 (sample_t * _f, int16_t * s16)
+{
+ int i;
+ int32_t * f = (int32_t *) _f;
+
+ for (i = 0; i < 256; i++)
+ {
+ s16[5*i] = convert (f[i]);
+ s16[5*i+1] = convert (f[i+256]);
+ s16[5*i+2] = convert (f[i+512]);
+ s16[5*i+3] = convert (f[i+768]);
+ s16[5*i+4] = convert (f[i+1024]);
+ }
+}
+
+static void
+convert2s16_multi (sample_t * _f, int16_t * s16, int flags)
+{
+ int i;
+ int32_t * f = (int32_t *) _f;
+
+ switch (flags)
+ {
+ 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]);
+ }
+ break;
+ case DTS_CHANNEL:
+ case DTS_STEREO:
+ case DTS_DOLBY:
+ convert2s16_2 (_f, s16);
+ break;
+ case DTS_3F:
+ for (i = 0; i < 256; i++)
+ {
+ s16[5*i] = convert (f[i]);
+ s16[5*i+1] = convert (f[i+512]);
+ s16[5*i+2] = s16[5*i+3] = 0;
+ s16[5*i+4] = convert (f[i+256]);
+ }
+ break;
+ case DTS_2F2R:
+ convert2s16_4 (_f, s16);
+ break;
+ case DTS_3F2R:
+ convert2s16_5 (_f, s16);
+ break;
+ 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+256]);
+ s16[6*i+5] = convert (f[i]);
+ }
+ break;
+ 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+256]);
+ s16[6*i+1] = convert (f[i+512]);
+ s16[6*i+2] = s16[6*i+3] = s16[6*i+4] = 0;
+ s16[6*i+5] = convert (f[i]);
+ }
+ break;
+ 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+768]);
+ s16[6*i+2] = s16[6*i+3] = 0;
+ s16[6*i+4] = convert (f[i+512]);
+ s16[6*i+5] = convert (f[i]);
+ }
+ break;
+ case DTS_2F2R | 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] = 0;
+ s16[6*i+5] = convert (f[i]);
+ }
+ break;
+ 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+768]);
+ s16[6*i+2] = convert (f[i+1024]);
+ s16[6*i+3] = convert (f[i+1280]);
+ s16[6*i+4] = convert (f[i+512]);
+ s16[6*i+5] = convert (f[i]);
+ }
+ break;
+ }
+}
+
+static int
+channels_multi (int flags)
+{
+ if (flags & DTS_LFE)
+ return 6;
+ else if (flags & 1) /* center channel */
+ return 5;
+ else if ((flags & DTS_CHANNEL_MASK) == DTS_2F2R)
+ return 4;
+ else
+ return 2;
+}
+
+static int
+dts_decode_frame (AVCodecContext *avctx, void *data, int *data_size,
+ uint8_t *buff, int buff_size)
+{
+ uint8_t * start = buff;
+ uint8_t * end = buff + buff_size;
+ *data_size = 0;
+
+ static uint8_t buf[BUFFER_SIZE];
+ static uint8_t * bufptr = buf;
+ static uint8_t * bufpos = buf + HEADER_SIZE;
+
+ static int sample_rate;
+ static int frame_length;
+ static int flags;
+ int bit_rate;
+ int len;
+ dts_state_t *state = avctx->priv_data;
+
+ while (1)
+ {
+ len = end - start;
+ if (!len)
+ break;
+ if (len > bufpos - bufptr)
+ len = bufpos - bufptr;
+ memcpy (bufptr, start, len);
+ bufptr += len;
+ start += len;
+ if (bufptr == bufpos)
+ {
+ if (bufpos == buf + HEADER_SIZE)
+ {
+ int length;
+
+ length = dts_syncinfo (state, buf, &flags, &sample_rate,
+ &bit_rate, &frame_length);
+ if (!length)
+ {
+ av_log (NULL, AV_LOG_INFO, "skip\n");
+ for (bufptr = buf; bufptr < buf + HEADER_SIZE-1; bufptr++)
+ bufptr[0] = bufptr[1];
+ continue;
+ }
+ bufpos = buf + length;
+ }
+ else
+ {
+ level_t level;
+ sample_t bias;
+ int i;
+
+ flags = 2; /* ???????????? */
+ level = CONVERT_LEVEL;
+ bias = CONVERT_BIAS;
+
+ flags |= DTS_ADJUST_LEVEL;
+ if (dts_frame (state, buf, &flags, &level, bias))
+ goto error;
+ for (i = 0; i < dts_blocks_num (state); i++)
+ {
+ if (dts_block (state))
+ goto error;
+ {
+ int chans;
+ chans = channels_multi (flags);
+ convert2s16_multi (dts_samples (state), data,
+ flags & (DTS_CHANNEL_MASK | DTS_LFE));
+
+ data += 256 * sizeof (int16_t) * chans;
+ *data_size += 256 * sizeof (int16_t) * chans;
+ }
+ }
+ bufptr = buf;
+ bufpos = buf + HEADER_SIZE;
+ continue;
+ error:
+ av_log (NULL, AV_LOG_ERROR, "error\n");
+ bufptr = buf;
+ bufpos = buf + HEADER_SIZE;
+ }
+ }
+ }
+
+ return buff_size;
+}
+
+static int
+dts_decode_init (AVCodecContext *avctx)
+{
+ dts_state_t * state;
+ int i;
+
+ state = avctx->priv_data;
+ memset (state, 0, sizeof (dts_state_t));
+
+ state->samples = (sample_t *) memalign (16, 256 * 12 * sizeof (sample_t));
+ if (state->samples == NULL)
+ return 1;
+
+ for (i = 0; i < 256 * 12; i++)
+ state->samples[i] = 0;
+
+ /* Pre-calculate cosine modulation coefficients */
+ pre_calc_cosmod (state);
+ state->downmixed = 1;
+
+ return 0;
+}
+
+static int
+dts_decode_end (AVCodecContext *s)
+{
+ return 0;
+}
+
+AVCodec dts_decoder = {
+ "dts",
+ CODEC_TYPE_AUDIO,
+ CODEC_ID_DTS,
+ sizeof (dts_state_t),
+ dts_decode_init,
+ NULL,
+ dts_decode_end,
+ dts_decode_frame,
+};