* Copyright (C) 2006 Benjamin Larsson
* Copyright (C) 2007 Konstantin Shishkov
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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 2.1 of the License, or (at your option) any later version.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav 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 FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/common.h"
#include "libavutil/intmath.h"
#include "libavutil/intreadwrite.h"
-#include "libavcore/audioconvert.h"
+#include "libavutil/audioconvert.h"
#include "avcodec.h"
#include "dsputil.h"
#include "fft.h"
#include "dca.h"
#include "synth_filter.h"
#include "dcadsp.h"
+#include "fmtconvert.h"
+
+#if ARCH_ARM
+# include "arm/dca.h"
+#endif
//#define TRACE
-#define DCA_PRIM_CHANNELS_MAX (7)
-#define DCA_SUBBANDS (32)
-#define DCA_ABITS_MAX (32) /* Should be 28 */
-#define DCA_SUBSUBFRAMES_MAX (4)
-#define DCA_SUBFRAMES_MAX (16)
-#define DCA_BLOCKS_MAX (16)
-#define DCA_LFE_MAX (3)
+#define DCA_PRIM_CHANNELS_MAX (7)
+#define DCA_SUBBANDS (32)
+#define DCA_ABITS_MAX (32) /* Should be 28 */
+#define DCA_SUBSUBFRAMES_MAX (4)
+#define DCA_SUBFRAMES_MAX (16)
+#define DCA_BLOCKS_MAX (16)
+#define DCA_LFE_MAX (3)
enum DCAMode {
DCA_MONO = 0,
DCA_EXT_EXSS_XLL = 0x200, ///< lossless extension in ExSS
};
+/* -1 are reserved or unknown */
+static const int dca_ext_audio_descr_mask[] = {
+ DCA_EXT_XCH,
+ -1,
+ DCA_EXT_X96,
+ DCA_EXT_XCH | DCA_EXT_X96,
+ -1,
+ -1,
+ DCA_EXT_XXCH,
+ -1,
+};
+
+/* extensions that reside in core substream */
+#define DCA_CORE_EXTS (DCA_EXT_XCH | DCA_EXT_XXCH | DCA_EXT_X96)
+
/* Tables for mapping dts channel configurations to libavcodec multichannel api.
* Some compromises have been made for special configurations. Most configurations
* are never used so complete accuracy is not needed.
* L = left, R = right, C = center, S = surround, F = front, R = rear, T = total, OV = overhead.
* S -> side, when both rear and back are configured move one of them to the side channel
* OV -> center back
- * All 2 channel configurations -> CH_LAYOUT_STEREO
+ * All 2 channel configurations -> AV_CH_LAYOUT_STEREO
*/
-
-static const int64_t dca_core_channel_layout[] = {
- AV_CH_FRONT_CENTER, ///< 1, A
- AV_CH_LAYOUT_STEREO, ///< 2, A + B (dual mono)
- AV_CH_LAYOUT_STEREO, ///< 2, L + R (stereo)
- AV_CH_LAYOUT_STEREO, ///< 2, (L+R) + (L-R) (sum-difference)
- AV_CH_LAYOUT_STEREO, ///< 2, LT +RT (left and right total)
- AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER, ///< 3, C+L+R
- AV_CH_LAYOUT_STEREO|AV_CH_BACK_CENTER, ///< 3, L+R+S
- AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER|CH_BACK_CENTER, ///< 4, C + L + R+ S
- AV_CH_LAYOUT_STEREO|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT, ///< 4, L + R +SL+ SR
- AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT, ///< 5, C + L + R+ SL+SR
- AV_CH_LAYOUT_STEREO|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT|AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER, ///< 6, CL + CR + L + R + SL + SR
- AV_CH_LAYOUT_STEREO|AV_CH_BACK_LEFT|AV_CH_BACK_RIGHT|AV_CH_FRONT_CENTER|AV_CH_BACK_CENTER, ///< 6, C + L + R+ LR + RR + OV
- AV_CH_FRONT_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER|AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_BACK_CENTER|AV_CH_BACK_LEFT|AV_CH_BACK_RIGHT, ///< 6, CF+ CR+LF+ RF+LR + RR
- AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_FRONT_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER|AV_CH_LAYOUT_STEREO|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT, ///< 7, CL + C + CR + L + R + SL + SR
- AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER|AV_CH_LAYOUT_STEREO|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT|AV_CH_BACK_LEFT|AV_CH_BACK_RIGHT, ///< 8, CL + CR + L + R + SL1 + SL2+ SR1 + SR2
- AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_FRONT_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER|AV_CH_LAYOUT_STEREO|AV_CH_SIDE_LEFT|AV_CH_BACK_CENTER|AV_CH_SIDE_RIGHT, ///< 8, CL + C+ CR + L + R + SL + S+ SR
+static const uint64_t dca_core_channel_layout[] = {
+ AV_CH_FRONT_CENTER, ///< 1, A
+ AV_CH_LAYOUT_STEREO, ///< 2, A + B (dual mono)
+ AV_CH_LAYOUT_STEREO, ///< 2, L + R (stereo)
+ AV_CH_LAYOUT_STEREO, ///< 2, (L + R) + (L - R) (sum-difference)
+ AV_CH_LAYOUT_STEREO, ///< 2, LT + RT (left and right total)
+ AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER, ///< 3, C + L + R
+ AV_CH_LAYOUT_STEREO | AV_CH_BACK_CENTER, ///< 3, L + R + S
+ AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER, ///< 4, C + L + R + S
+ AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT, ///< 4, L + R + SL + SR
+
+ AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_SIDE_LEFT |
+ AV_CH_SIDE_RIGHT, ///< 5, C + L + R + SL + SR
+
+ AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |
+ AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER, ///< 6, CL + CR + L + R + SL + SR
+
+ AV_CH_LAYOUT_STEREO | AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT |
+ AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER, ///< 6, C + L + R + LR + RR + OV
+
+ AV_CH_FRONT_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |
+ AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_BACK_CENTER |
+ AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT, ///< 6, CF + CR + LF + RF + LR + RR
+
+ AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER |
+ AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |
+ AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT, ///< 7, CL + C + CR + L + R + SL + SR
+
+ AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER |
+ AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT |
+ AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT, ///< 8, CL + CR + L + R + SL1 + SL2 + SR1 + SR2
+
+ AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER |
+ AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO |
+ AV_CH_SIDE_LEFT | AV_CH_BACK_CENTER | AV_CH_SIDE_RIGHT, ///< 8, CL + C + CR + L + R + SL + S + SR
};
static const int8_t dca_lfe_index[] = {
- 1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3
+ 1, 2, 2, 2, 2, 3, 2, 3, 2, 3, 2, 3, 1, 3, 2, 3
};
static const int8_t dca_channel_reorder_lfe[][9] = {
{ 3, 2, 4, 0, 1, 5, 8, 7, 6},
};
-#define DCA_DOLBY 101 /* FIXME */
+#define DCA_DOLBY 101 /* FIXME */
-#define DCA_CHANNEL_BITS 6
-#define DCA_CHANNEL_MASK 0x3F
+#define DCA_CHANNEL_BITS 6
+#define DCA_CHANNEL_MASK 0x3F
-#define DCA_LFE 0x80
+#define DCA_LFE 0x80
-#define HEADER_SIZE 14
+#define HEADER_SIZE 14
-#define DCA_MAX_FRAME_SIZE 16384
-#define DCA_MAX_EXSS_HEADER_SIZE 4096
+#define DCA_MAX_FRAME_SIZE 16384
+#define DCA_MAX_EXSS_HEADER_SIZE 4096
-#define DCA_BUFFER_PADDING_SIZE 1024
+#define DCA_BUFFER_PADDING_SIZE 1024
/** Bit allocation */
typedef struct {
static BitAlloc dca_scalefactor; ///< scalefactor VLCs
static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs
-static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba, int idx)
+static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba,
+ int idx)
{
- return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + ba->offset;
+ return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) +
+ ba->offset;
}
typedef struct {
AVCodecContext *avctx;
+ AVFrame frame;
/* Frame header */
int frame_type; ///< type of the current frame
int samples_deficit; ///< deficit sample count
float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment
/* Primary audio coding side information */
- int subsubframes[DCA_SUBFRAMES_MAX]; ///< number of subsubframes
- int partial_samples[DCA_SUBFRAMES_MAX]; ///< partial subsubframe samples count
+ int subsubframes[DCA_SUBFRAMES_MAX]; ///< number of subsubframes
+ int partial_samples[DCA_SUBFRAMES_MAX]; ///< partial subsubframe samples count
int prediction_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction mode (ADPCM used or not)
int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction VQ coefs
int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< bit allocation index
int lfe_scale_factor;
/* Subband samples history (for ADPCM) */
- float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
- DECLARE_ALIGNED(16, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512];
- DECLARE_ALIGNED(16, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32];
+ DECLARE_ALIGNED(16, float, subband_samples_hist)[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
+ DECLARE_ALIGNED(32, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512];
+ DECLARE_ALIGNED(32, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32];
int hist_index[DCA_PRIM_CHANNELS_MAX];
- DECLARE_ALIGNED(16, float, raXin)[32];
+ DECLARE_ALIGNED(32, float, raXin)[32];
int output; ///< type of output
- float add_bias; ///< output bias
float scale_bias; ///< output scale
- DECLARE_ALIGNED(16, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
- DECLARE_ALIGNED(16, float, samples)[(DCA_PRIM_CHANNELS_MAX+1)*256];
- const float *samples_chanptr[DCA_PRIM_CHANNELS_MAX+1];
+ DECLARE_ALIGNED(32, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
+ DECLARE_ALIGNED(32, float, samples)[(DCA_PRIM_CHANNELS_MAX + 1) * 256];
+ const float *samples_chanptr[DCA_PRIM_CHANNELS_MAX + 1];
uint8_t dca_buffer[DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE + DCA_BUFFER_PADDING_SIZE];
int dca_buffer_size; ///< how much data is in the dca_buffer
- const int8_t* channel_order_tab; ///< channel reordering table, lfe and non lfe
+ const int8_t *channel_order_tab; ///< channel reordering table, lfe and non lfe
GetBitContext gb;
/* Current position in DCA frame */
int current_subframe;
int current_subsubframe;
+ int core_ext_mask; ///< present extensions in the core substream
+
/* XCh extension information */
- int xch_present;
+ int xch_present; ///< XCh extension present and valid
int xch_base_channel; ///< index of first (only) channel containing XCH data
- /* Other detected extensions in the core substream */
- int xxch_present;
- int x96_present;
-
/* ExSS header parser */
int static_fields; ///< static fields present
int mix_metadata; ///< mixing metadata present
FFTContext imdct;
SynthFilterContext synth;
DCADSPContext dcadsp;
+ FmtConvertContext fmt_conv;
} DCAContext;
static const uint16_t dca_vlc_offs[] = {
}
for (i = 0; i < 10; i++)
- for (j = 0; j < 7; j++){
- if (!bitalloc_codes[i][j]) break;
- dca_smpl_bitalloc[i+1].offset = bitalloc_offsets[i];
- dca_smpl_bitalloc[i+1].wrap = 1 + (j > 4);
- dca_smpl_bitalloc[i+1].vlc[j].table = &dca_table[dca_vlc_offs[c]];
- dca_smpl_bitalloc[i+1].vlc[j].table_allocated = dca_vlc_offs[c + 1] - dca_vlc_offs[c];
- init_vlc(&dca_smpl_bitalloc[i+1].vlc[j], bitalloc_maxbits[i][j],
+ for (j = 0; j < 7; j++) {
+ if (!bitalloc_codes[i][j])
+ break;
+ dca_smpl_bitalloc[i + 1].offset = bitalloc_offsets[i];
+ dca_smpl_bitalloc[i + 1].wrap = 1 + (j > 4);
+ dca_smpl_bitalloc[i + 1].vlc[j].table = &dca_table[dca_vlc_offs[c]];
+ dca_smpl_bitalloc[i + 1].vlc[j].table_allocated = dca_vlc_offs[c + 1] - dca_vlc_offs[c];
+
+ init_vlc(&dca_smpl_bitalloc[i + 1].vlc[j], bitalloc_maxbits[i][j],
bitalloc_sizes[i],
bitalloc_bits[i][j], 1, 1,
bitalloc_codes[i][j], 2, 2, INIT_VLC_USE_NEW_STATIC);
static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
{
- while(len--)
+ while (len--)
*dst++ = get_bits(gb, bits);
}
-static int dca_parse_audio_coding_header(DCAContext * s, int base_channel)
+static int dca_parse_audio_coding_header(DCAContext *s, int base_channel)
{
int i, j;
static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };
static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
- static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
+ static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
- s->total_channels = get_bits(&s->gb, 3) + 1 + base_channel;
- s->prim_channels = s->total_channels;
+ s->total_channels = get_bits(&s->gb, 3) + 1 + base_channel;
+ s->prim_channels = s->total_channels;
if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)
s->prim_channels = DCA_PRIM_CHANNELS_MAX;
get_bits(&s->gb, 16);
}
- s->current_subframe = 0;
+ s->current_subframe = 0;
s->current_subsubframe = 0;
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes);
av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels);
- for (i = base_channel; i < s->prim_channels; i++){
- av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n", s->subband_activity[i]);
- av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n", s->vq_start_subband[i]);
- av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n", s->joint_intensity[i]);
- av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n", s->transient_huffman[i]);
- av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n", s->scalefactor_huffman[i]);
- av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n", s->bitalloc_huffman[i]);
+ for (i = base_channel; i < s->prim_channels; i++) {
+ av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n",
+ s->subband_activity[i]);
+ av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n",
+ s->vq_start_subband[i]);
+ av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n",
+ s->joint_intensity[i]);
+ av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n",
+ s->transient_huffman[i]);
+ av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n",
+ s->scalefactor_huffman[i]);
+ av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n",
+ s->bitalloc_huffman[i]);
av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:");
for (j = 0; j < 11; j++)
- av_log(s->avctx, AV_LOG_DEBUG, " %i",
- s->quant_index_huffman[i][j]);
+ av_log(s->avctx, AV_LOG_DEBUG, " %i", s->quant_index_huffman[i][j]);
av_log(s->avctx, AV_LOG_DEBUG, "\n");
av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:");
for (j = 0; j < 11; j++)
}
#endif
- return 0;
+ return 0;
}
-static int dca_parse_frame_header(DCAContext * s)
+static int dca_parse_frame_header(DCAContext *s)
{
init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
/* Sync code */
- get_bits(&s->gb, 32);
+ skip_bits_long(&s->gb, 32);
/* Frame header */
s->frame_type = get_bits(&s->gb, 1);
s->sample_blocks = get_bits(&s->gb, 7) + 1;
s->frame_size = get_bits(&s->gb, 14) + 1;
if (s->frame_size < 95)
- return -1;
+ return AVERROR_INVALIDDATA;
s->amode = get_bits(&s->gb, 6);
s->sample_rate = dca_sample_rates[get_bits(&s->gb, 4)];
if (!s->sample_rate)
- return -1;
+ return AVERROR_INVALIDDATA;
s->bit_rate_index = get_bits(&s->gb, 5);
s->bit_rate = dca_bit_rates[s->bit_rate_index];
if (!s->bit_rate)
- return -1;
+ return AVERROR_INVALIDDATA;
s->downmix = get_bits(&s->gb, 1);
s->dynrange = get_bits(&s->gb, 1);
/* FIXME: channels mixing levels */
s->output = s->amode;
- if (s->lfe) s->output |= DCA_LFE;
+ if (s->lfe)
+ s->output |= DCA_LFE;
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type);
static inline int get_scale(GetBitContext *gb, int level, int value)
{
- if (level < 5) {
- /* huffman encoded */
- value += get_bitalloc(gb, &dca_scalefactor, level);
- } else if (level < 8)
- value = get_bits(gb, level + 1);
- return value;
+ if (level < 5) {
+ /* huffman encoded */
+ value += get_bitalloc(gb, &dca_scalefactor, level);
+ } else if (level < 8)
+ value = get_bits(gb, level + 1);
+ return value;
}
-static int dca_subframe_header(DCAContext * s, int base_channel, int block_index)
+static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
{
/* Primary audio coding side information */
int j, k;
if (get_bits_left(&s->gb) < 0)
- return -1;
+ return AVERROR_INVALIDDATA;
if (!base_channel) {
- s->subsubframes[s->current_subframe] = get_bits(&s->gb, 2) + 1;
+ s->subsubframes[s->current_subframe] = get_bits(&s->gb, 2) + 1;
s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3);
}
else if (s->bitalloc_huffman[j] == 7) {
av_log(s->avctx, AV_LOG_ERROR,
"Invalid bit allocation index\n");
- return -1;
+ return AVERROR_INVALIDDATA;
} else {
s->bitalloc[j][k] =
get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]);
}
if (s->bitalloc[j][k] > 26) {
-// av_log(s->avctx,AV_LOG_DEBUG,"bitalloc index [%i][%i] too big (%i)\n",
-// j, k, s->bitalloc[j][k]);
- return -1;
+ // av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index [%i][%i] too big (%i)\n",
+ // j, k, s->bitalloc[j][k]);
+ return AVERROR_INVALIDDATA;
}
}
}
}
if (get_bits_left(&s->gb) < 0)
- return -1;
+ return AVERROR_INVALIDDATA;
for (j = base_channel; j < s->prim_channels; j++) {
const uint32_t *scale_table;
int scale_sum;
- memset(s->scale_factor[j], 0, s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2);
+ memset(s->scale_factor[j], 0,
+ s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2);
if (s->scalefactor_huffman[j] == 6)
scale_table = scale_factor_quant7;
}
if (get_bits_left(&s->gb) < 0)
- return -1;
+ return AVERROR_INVALIDDATA;
/* Scale factors for joint subband coding */
for (j = base_channel; j < s->prim_channels; j++) {
}
#ifdef TRACE
- av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n", s->subsubframes[s->current_subframe]);
+ av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n",
+ s->subsubframes[s->current_subframe]);
av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n",
s->partial_samples[s->current_subframe]);
+
for (j = base_channel; j < s->prim_channels; j++) {
av_log(s->avctx, AV_LOG_DEBUG, "prediction mode:");
for (k = 0; k < s->subband_activity[j]; k++)
}
for (j = base_channel; j < s->prim_channels; j++) {
for (k = 0; k < s->subband_activity[j]; k++)
- av_log(s->avctx, AV_LOG_DEBUG,
- "prediction coefs: %f, %f, %f, %f\n",
- (float) adpcm_vb[s->prediction_vq[j][k]][0] / 8192,
- (float) adpcm_vb[s->prediction_vq[j][k]][1] / 8192,
- (float) adpcm_vb[s->prediction_vq[j][k]][2] / 8192,
- (float) adpcm_vb[s->prediction_vq[j][k]][3] / 8192);
+ av_log(s->avctx, AV_LOG_DEBUG,
+ "prediction coefs: %f, %f, %f, %f\n",
+ (float) adpcm_vb[s->prediction_vq[j][k]][0] / 8192,
+ (float) adpcm_vb[s->prediction_vq[j][k]][1] / 8192,
+ (float) adpcm_vb[s->prediction_vq[j][k]][2] / 8192,
+ (float) adpcm_vb[s->prediction_vq[j][k]][3] / 8192);
}
for (j = base_channel; j < s->prim_channels; j++) {
av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index: ");
if (!base_channel && s->prim_channels > 2 && s->downmix) {
av_log(s->avctx, AV_LOG_DEBUG, "Downmix coeffs:\n");
for (j = 0; j < s->prim_channels; j++) {
- av_log(s->avctx, AV_LOG_DEBUG, "Channel 0,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][0]]);
- av_log(s->avctx, AV_LOG_DEBUG, "Channel 1,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][1]]);
+ av_log(s->avctx, AV_LOG_DEBUG, "Channel 0, %d = %f\n", j,
+ dca_downmix_coeffs[s->downmix_coef[j][0]]);
+ av_log(s->avctx, AV_LOG_DEBUG, "Channel 1, %d = %f\n", j,
+ dca_downmix_coeffs[s->downmix_coef[j][1]]);
}
av_log(s->avctx, AV_LOG_DEBUG, "\n");
}
return 0;
}
-static void qmf_32_subbands(DCAContext * s, int chans,
+static void qmf_32_subbands(DCAContext *s, int chans,
float samples_in[32][8], float *samples_out,
- float scale, float bias)
+ float scale)
{
const float *prCoeff;
int i;
int sb_act = s->subband_activity[chans];
int subindex;
- scale *= sqrt(1/8.0);
+ scale *= sqrt(1 / 8.0);
/* Select filter */
if (!s->multirate_inter) /* Non-perfect reconstruction */
else /* Perfect reconstruction */
prCoeff = fir_32bands_perfect;
+ for (i = sb_act; i < 32; i++)
+ s->raXin[i] = 0.0;
+
/* Reconstructed channel sample index */
for (subindex = 0; subindex < 8; subindex++) {
/* Load in one sample from each subband and clear inactive subbands */
- for (i = 0; i < sb_act; i++){
- uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ ((i-1)&2)<<30;
+ for (i = 0; i < sb_act; i++) {
+ unsigned sign = (i - 1) & 2;
+ uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ sign << 30;
AV_WN32A(&s->raXin[i], v);
}
- for (; i < 32; i++)
- s->raXin[i] = 0.0;
s->synth.synth_filter_float(&s->imdct,
- s->subband_fir_hist[chans], &s->hist_index[chans],
- s->subband_fir_noidea[chans], prCoeff,
- samples_out, s->raXin, scale, bias);
- samples_out+= 32;
-
+ s->subband_fir_hist[chans],
+ &s->hist_index[chans],
+ s->subband_fir_noidea[chans], prCoeff,
+ samples_out, s->raXin, scale);
+ samples_out += 32;
}
}
static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
int num_deci_sample, float *samples_in,
- float *samples_out, float scale,
- float bias)
+ float *samples_out, float scale)
{
/* samples_in: An array holding decimated samples.
* Samples in current subframe starts from samples_in[0],
}
/* Interpolation */
for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
- s->dcadsp.lfe_fir(samples_out, samples_in, prCoeff, decifactor,
- scale, bias);
+ s->dcadsp.lfe_fir(samples_out, samples_in, prCoeff, decifactor, scale);
samples_in++;
samples_out += 2 * decifactor;
}
}
/* downmixing routines */
-#define MIX_REAR1(samples, si1, rs, coef) \
- samples[i] += (samples[si1] - add_bias) * coef[rs][0]; \
- samples[i+256] += (samples[si1] - add_bias) * coef[rs][1];
-
-#define MIX_REAR2(samples, si1, si2, rs, coef) \
- samples[i] += (samples[si1] - add_bias) * coef[rs][0] + (samples[si2] - add_bias) * coef[rs+1][0]; \
- samples[i+256] += (samples[si1] - add_bias) * coef[rs][1] + (samples[si2] - add_bias) * coef[rs+1][1];
-
-#define MIX_FRONT3(samples, coef) \
- t = samples[i+c] - add_bias; \
- u = samples[i+l] - add_bias; \
- v = samples[i+r] - add_bias; \
- samples[i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0] + add_bias; \
- samples[i+256] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1] + add_bias;
-
-#define DOWNMIX_TO_STEREO(op1, op2) \
- for (i = 0; i < 256; i++){ \
- op1 \
- op2 \
+#define MIX_REAR1(samples, si1, rs, coef) \
+ samples[i] += samples[si1] * coef[rs][0]; \
+ samples[i+256] += samples[si1] * coef[rs][1];
+
+#define MIX_REAR2(samples, si1, si2, rs, coef) \
+ samples[i] += samples[si1] * coef[rs][0] + samples[si2] * coef[rs + 1][0]; \
+ samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs + 1][1];
+
+#define MIX_FRONT3(samples, coef) \
+ t = samples[i + c]; \
+ u = samples[i + l]; \
+ v = samples[i + r]; \
+ samples[i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \
+ samples[i+256] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];
+
+#define DOWNMIX_TO_STEREO(op1, op2) \
+ for (i = 0; i < 256; i++) { \
+ op1 \
+ op2 \
}
static void dca_downmix(float *samples, int srcfmt,
int downmix_coef[DCA_PRIM_CHANNELS_MAX][2],
- const int8_t *channel_mapping, float add_bias)
+ const int8_t *channel_mapping)
{
- int c,l,r,sl,sr,s;
+ int c, l, r, sl, sr, s;
int i;
float t, u, v;
float coef[DCA_PRIM_CHANNELS_MAX][2];
- for (i=0; i<DCA_PRIM_CHANNELS_MAX; i++) {
+ for (i = 0; i < DCA_PRIM_CHANNELS_MAX; i++) {
coef[i][0] = dca_downmix_coeffs[downmix_coef[i][0]];
coef[i][1] = dca_downmix_coeffs[downmix_coef[i][1]];
}
c = channel_mapping[0] * 256;
l = channel_mapping[1] * 256;
r = channel_mapping[2] * 256;
- DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),);
+ DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );
break;
case DCA_2F1R:
s = channel_mapping[2] * 256;
- DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + s, 2, coef),);
+ DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + s, 2, coef), );
break;
case DCA_3F1R:
c = channel_mapping[0] * 256;
case DCA_2F2R:
sl = channel_mapping[2] * 256;
sr = channel_mapping[3] * 256;
- DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + sl, i + sr, 2, coef),);
+ DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + sl, i + sr, 2, coef), );
break;
case DCA_3F2R:
- c = channel_mapping[0] * 256;
- l = channel_mapping[1] * 256;
- r = channel_mapping[2] * 256;
+ c = channel_mapping[0] * 256;
+ l = channel_mapping[1] * 256;
+ r = channel_mapping[2] * 256;
sl = channel_mapping[3] * 256;
sr = channel_mapping[4] * 256;
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
}
+#ifndef decode_blockcodes
/* Very compact version of the block code decoder that does not use table
* look-up but is slightly slower */
static int decode_blockcode(int code, int levels, int *values)
for (i = 0; i < 4; i++) {
int div = FASTDIV(code, levels);
- values[i] = code - offset - div*levels;
+ values[i] = code - offset - div * levels;
code = div;
}
- if (code == 0)
- return 0;
- else {
- av_log(NULL, AV_LOG_ERROR, "ERROR: block code look-up failed\n");
- return -1;
- }
+ return code;
}
-static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
-static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };
+static int decode_blockcodes(int code1, int code2, int levels, int *values)
+{
+ return decode_blockcode(code1, levels, values) |
+ decode_blockcode(code2, levels, values + 4);
+}
+#endif
-static int dca_subsubframe(DCAContext * s, int base_channel, int block_index)
+static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
+static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };
+
+#ifndef int8x8_fmul_int32
+static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)
+{
+ float fscale = scale / 16.0;
+ int i;
+ for (i = 0; i < 8; i++)
+ dst[i] = src[i] * fscale;
+}
+#endif
+
+static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
{
int k, l;
int subsubframe = s->current_subsubframe;
for (k = base_channel; k < s->prim_channels; k++) {
if (get_bits_left(&s->gb) < 0)
- return -1;
+ return AVERROR_INVALIDDATA;
for (l = 0; l < s->vq_start_subband[k]; l++) {
int m;
/*
* Extract bits from the bit stream
*/
- if (!abits){
+ if (!abits) {
memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));
} else {
/* Deal with transients */
int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];
- float rscale = quant_step_size * s->scale_factor[k][l][sfi] * s->scalefactor_adj[k][sel];
+ float rscale = quant_step_size * s->scale_factor[k][l][sfi] *
+ s->scalefactor_adj[k][sel];
- if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){
- if (abits <= 7){
+ if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {
+ if (abits <= 7) {
/* Block code */
- int block_code1, block_code2, size, levels;
+ int block_code1, block_code2, size, levels, err;
- size = abits_sizes[abits-1];
- levels = abits_levels[abits-1];
+ size = abits_sizes[abits - 1];
+ levels = abits_levels[abits - 1];
block_code1 = get_bits(&s->gb, size);
- /* FIXME Should test return value */
- decode_blockcode(block_code1, levels, block);
block_code2 = get_bits(&s->gb, size);
- decode_blockcode(block_code2, levels, &block[4]);
- }else{
+ err = decode_blockcodes(block_code1, block_code2,
+ levels, block);
+ if (err) {
+ av_log(s->avctx, AV_LOG_ERROR,
+ "ERROR: block code look-up failed\n");
+ return AVERROR_INVALIDDATA;
+ }
+ } else {
/* no coding */
for (m = 0; m < 8; m++)
block[m] = get_sbits(&s->gb, abits - 3);
}
- }else{
+ } else {
/* Huffman coded */
for (m = 0; m < 8; m++)
- block[m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);
+ block[m] = get_bitalloc(&s->gb,
+ &dca_smpl_bitalloc[abits], sel);
}
- s->dsp.int32_to_float_fmul_scalar(subband_samples[k][l],
- block, rscale, 8);
+ s->fmt_conv.int32_to_float_fmul_scalar(subband_samples[k][l],
+ block, rscale, 8);
}
/*
else if (s->predictor_history)
subband_samples[k][l][m] +=
(adpcm_vb[s->prediction_vq[k][l]][n - 1] *
- s->subband_samples_hist[k][l][m - n +
- 4] / 8192);
+ s->subband_samples_hist[k][l][m - n + 4] / 8192);
}
}
}
for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {
/* 1 vector -> 32 samples but we only need the 8 samples
* for this subsubframe. */
- int m;
+ int hfvq = s->high_freq_vq[k][l];
if (!s->debug_flag & 0x01) {
- av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n");
+ av_log(s->avctx, AV_LOG_DEBUG,
+ "Stream with high frequencies VQ coding\n");
s->debug_flag |= 0x01;
}
- for (m = 0; m < 8; m++) {
- subband_samples[k][l][m] =
- high_freq_vq[s->high_freq_vq[k][l]][subsubframe * 8 +
- m]
- * (float) s->scale_factor[k][l][0] / 16.0;
- }
+ int8x8_fmul_int32(subband_samples[k][l],
+ &high_freq_vq[hfvq][subsubframe * 8],
+ s->scale_factor[k][l][0]);
}
}
/* Backup predictor history for adpcm */
for (k = base_channel; k < s->prim_channels; k++)
for (l = 0; l < s->vq_start_subband[k]; l++)
- memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4],
- 4 * sizeof(subband_samples[0][0][0]));
+ memcpy(s->subband_samples_hist[k][l],
+ &subband_samples[k][l][4],
+ 4 * sizeof(subband_samples[0][0][0]));
return 0;
}
-static int dca_filter_channels(DCAContext * s, int block_index)
+static int dca_filter_channels(DCAContext *s, int block_index)
{
float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
int k;
/* 32 subbands QMF */
for (k = 0; k < s->prim_channels; k++) {
-/* static float pcm_to_double[8] =
- {32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/
- qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * s->channel_order_tab[k]],
- M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ ,
- s->add_bias );
+/* static float pcm_to_double[8] = { 32768.0, 32768.0, 524288.0, 524288.0,
+ 0, 8388608.0, 8388608.0 };*/
+ qmf_32_subbands(s, k, subband_samples[k],
+ &s->samples[256 * s->channel_order_tab[k]],
+ M_SQRT1_2 * s->scale_bias /* pcm_to_double[s->source_pcm_res] */);
}
/* Down mixing */
if (s->avctx->request_channels == 2 && s->prim_channels > 2) {
- dca_downmix(s->samples, s->amode, s->downmix_coef, s->channel_order_tab, s->add_bias);
+ dca_downmix(s->samples, s->amode, s->downmix_coef, s->channel_order_tab);
}
/* Generate LFE samples for this subsubframe FIXME!!! */
lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,
s->lfe_data + 2 * s->lfe * (block_index + 4),
&s->samples[256 * dca_lfe_index[s->amode]],
- (1.0/256.0)*s->scale_bias, s->add_bias);
+ (1.0 / 256.0) * s->scale_bias);
/* Outputs 20bits pcm samples */
}
}
-static int dca_subframe_footer(DCAContext * s, int base_channel)
+static int dca_subframe_footer(DCAContext *s, int base_channel)
{
int aux_data_count = 0, i;
/* presumably optional information only appears in the core? */
if (!base_channel) {
if (s->timestamp)
- get_bits(&s->gb, 32);
+ skip_bits_long(&s->gb, 32);
if (s->aux_data)
aux_data_count = get_bits(&s->gb, 6);
* @param s pointer to the DCAContext
*/
-static int dca_decode_block(DCAContext * s, int base_channel, int block_index)
+static int dca_decode_block(DCAContext *s, int base_channel, int block_index)
{
+ int ret;
/* Sanity check */
if (s->current_subframe >= s->subframes) {
av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
s->current_subframe, s->subframes);
- return -1;
+ return AVERROR_INVALIDDATA;
}
if (!s->current_subsubframe) {
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");
#endif
/* Read subframe header */
- if (dca_subframe_header(s, base_channel, block_index))
- return -1;
+ if ((ret = dca_subframe_header(s, base_channel, block_index)))
+ return ret;
}
/* Read subsubframe */
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");
#endif
- if (dca_subsubframe(s, base_channel, block_index))
- return -1;
+ if ((ret = dca_subsubframe(s, base_channel, block_index)))
+ return ret;
/* Update state */
s->current_subsubframe++;
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");
#endif
/* Read subframe footer */
- if (dca_subframe_footer(s, base_channel))
- return -1;
+ if ((ret = dca_subframe_footer(s, base_channel)))
+ return ret;
}
return 0;
/**
* Convert bitstream to one representation based on sync marker
*/
-static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * dst,
- int max_size)
+static int dca_convert_bitstream(const uint8_t *src, int src_size, uint8_t *dst,
+ int max_size)
{
uint32_t mrk;
int i, tmp;
uint16_t *sdst = (uint16_t *) dst;
PutBitContext pb;
- if ((unsigned)src_size > (unsigned)max_size) {
-// av_log(NULL, AV_LOG_ERROR, "Input frame size larger then DCA_MAX_FRAME_SIZE!\n");
+ if ((unsigned) src_size > (unsigned) max_size) {
+// av_log(NULL, AV_LOG_ERROR, "Input frame size larger than DCA_MAX_FRAME_SIZE!\n");
// return -1;
src_size = max_size;
}
flush_put_bits(&pb);
return (put_bits_count(&pb) + 7) >> 3;
default:
- return -1;
+ return AVERROR_INVALIDDATA;
}
}
static int dca_exss_mask2count(int mask)
{
/* count bits that mean speaker pairs twice */
- return av_popcount(mask)
- + av_popcount(mask & (
- DCA_EXSS_CENTER_LEFT_RIGHT
- | DCA_EXSS_FRONT_LEFT_RIGHT
- | DCA_EXSS_FRONT_HIGH_LEFT_RIGHT
- | DCA_EXSS_WIDE_LEFT_RIGHT
- | DCA_EXSS_SIDE_LEFT_RIGHT
- | DCA_EXSS_SIDE_HIGH_LEFT_RIGHT
- | DCA_EXSS_SIDE_REAR_LEFT_RIGHT
- | DCA_EXSS_REAR_LEFT_RIGHT
- | DCA_EXSS_REAR_HIGH_LEFT_RIGHT
- ));
+ return av_popcount(mask) +
+ av_popcount(mask & (DCA_EXSS_CENTER_LEFT_RIGHT |
+ DCA_EXSS_FRONT_LEFT_RIGHT |
+ DCA_EXSS_FRONT_HIGH_LEFT_RIGHT |
+ DCA_EXSS_WIDE_LEFT_RIGHT |
+ DCA_EXSS_SIDE_LEFT_RIGHT |
+ DCA_EXSS_SIDE_HIGH_LEFT_RIGHT |
+ DCA_EXSS_SIDE_REAR_LEFT_RIGHT |
+ DCA_EXSS_REAR_LEFT_RIGHT |
+ DCA_EXSS_REAR_HIGH_LEFT_RIGHT));
}
/**
*/
static void dca_exss_skip_mix_coeffs(GetBitContext *gb, int channels, int out_ch)
{
- for (int i = 0; i < channels; i++) {
+ int i;
+
+ for (i = 0; i < channels; i++) {
int mix_map_mask = get_bits(gb, out_ch);
int num_coeffs = av_popcount(mix_map_mask);
skip_bits_long(gb, num_coeffs * 6);
int header_size;
int channels;
int embedded_stereo = 0;
- int embedded_6ch = 0;
+ int embedded_6ch = 0;
int drc_code_present;
int extensions_mask;
int i, j;
if (!(extensions_mask & DCA_EXT_CORE))
av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n");
- if (!!(extensions_mask & DCA_EXT_XCH) != s->xch_present)
- av_log(s->avctx, AV_LOG_WARNING, "DTS XCh detection mismatch.\n");
- if (!!(extensions_mask & DCA_EXT_XXCH) != s->xxch_present)
- av_log(s->avctx, AV_LOG_WARNING, "DTS XXCh detection mismatch.\n");
- if (!!(extensions_mask & DCA_EXT_X96) != s->x96_present)
- av_log(s->avctx, AV_LOG_WARNING, "DTS X96 detection mismatch.\n");
+ if ((extensions_mask & DCA_CORE_EXTS) != s->core_ext_mask)
+ av_log(s->avctx, AV_LOG_WARNING,
+ "DTS extensions detection mismatch (%d, %d)\n",
+ extensions_mask & DCA_CORE_EXTS, s->core_ext_mask);
return 0;
}
{
int ss_index;
int blownup;
- int header_size;
- int hd_size;
int num_audiop = 1;
int num_assets = 1;
int active_ss_mask[8];
ss_index = get_bits(&s->gb, 2);
blownup = get_bits1(&s->gb);
- header_size = get_bits(&s->gb, 8 + 4 * blownup) + 1;
- hd_size = get_bits_long(&s->gb, 16 + 4 * blownup) + 1;
+ skip_bits(&s->gb, 8 + 4 * blownup); // header_size
+ skip_bits(&s->gb, 16 + 4 * blownup); // hd_size
s->static_fields = get_bits1(&s->gb);
if (s->static_fields) {
int mix_out_mask_size;
skip_bits(&s->gb, 2); // adjustment level
- mix_out_mask_size = (get_bits(&s->gb, 2) + 1) << 2;
- s->num_mix_configs = get_bits(&s->gb, 2) + 1;
+ mix_out_mask_size = (get_bits(&s->gb, 2) + 1) << 2;
+ s->num_mix_configs = get_bits(&s->gb, 2) + 1;
for (i = 0; i < s->num_mix_configs; i++) {
- int mix_out_mask = get_bits(&s->gb, mix_out_mask_size);
+ int mix_out_mask = get_bits(&s->gb, mix_out_mask_size);
s->mix_config_num_ch[i] = dca_exss_mask2count(mix_out_mask);
}
}
}
for (i = 0; i < num_assets; i++)
- skip_bits_long(&s->gb, 16 + 4 * blownup); // asset size
+ skip_bits_long(&s->gb, 16 + 4 * blownup); // asset size
for (i = 0; i < num_assets; i++) {
if (dca_exss_parse_asset_header(s))
* Main frame decoding function
* FIXME add arguments
*/
-static int dca_decode_frame(AVCodecContext * avctx,
- void *data, int *data_size,
- AVPacket *avpkt)
+static int dca_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int lfe_samples;
int num_core_channels = 0;
- int i;
- int16_t *samples = data;
+ int i, ret;
+ float *samples_flt;
+ int16_t *samples_s16;
DCAContext *s = avctx->priv_data;
int channels;
int core_ss_end;
s->xch_present = 0;
- s->x96_present = 0;
- s->xxch_present = 0;
s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer,
DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);
- if (s->dca_buffer_size == -1) {
+ if (s->dca_buffer_size == AVERROR_INVALIDDATA) {
av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
- return -1;
+ return AVERROR_INVALIDDATA;
}
init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
- if (dca_parse_frame_header(s) < 0) {
+ if ((ret = dca_parse_frame_header(s)) < 0) {
//seems like the frame is corrupt, try with the next one
- *data_size=0;
- return buf_size;
+ return ret;
}
//set AVCodec values with parsed data
avctx->sample_rate = s->sample_rate;
- avctx->bit_rate = s->bit_rate;
+ avctx->bit_rate = s->bit_rate;
+ avctx->frame_size = s->sample_blocks * 32;
s->profile = FF_PROFILE_DTS;
for (i = 0; i < (s->sample_blocks / 8); i++) {
- dca_decode_block(s, 0, i);
+ if ((ret = dca_decode_block(s, 0, i))) {
+ av_log(avctx, AV_LOG_ERROR, "error decoding block\n");
+ return ret;
+ }
}
/* record number of core channels incase less than max channels are requested */
num_core_channels = s->prim_channels;
- /* extensions start at 32-bit boundaries into bitstream */
- skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
+ if (s->ext_coding)
+ s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];
+ else
+ s->core_ext_mask = 0;
core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;
- while(core_ss_end - get_bits_count(&s->gb) >= 32) {
- uint32_t bits = get_bits_long(&s->gb, 32);
+ /* only scan for extensions if ext_descr was unknown or indicated a
+ * supported XCh extension */
+ if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) {
- switch(bits) {
- case 0x5a5a5a5a: {
- int ext_amode, xch_fsize;
+ /* if ext_descr was unknown, clear s->core_ext_mask so that the
+ * extensions scan can fill it up */
+ s->core_ext_mask = FFMAX(s->core_ext_mask, 0);
- s->xch_base_channel = s->prim_channels;
+ /* extensions start at 32-bit boundaries into bitstream */
+ skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
- /* validate sync word using XCHFSIZE field */
- xch_fsize = show_bits(&s->gb, 10);
- if((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&
- (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))
- continue;
+ while (core_ss_end - get_bits_count(&s->gb) >= 32) {
+ uint32_t bits = get_bits_long(&s->gb, 32);
- /* skip length-to-end-of-frame field for the moment */
- skip_bits(&s->gb, 10);
+ switch (bits) {
+ case 0x5a5a5a5a: {
+ int ext_amode, xch_fsize;
- s->profile = FFMAX(s->profile, FF_PROFILE_DTS_ES);
+ s->xch_base_channel = s->prim_channels;
- /* extension amode should == 1, number of channels in extension */
- /* AFAIK XCh is not used for more channels */
- if ((ext_amode = get_bits(&s->gb, 4)) != 1) {
- av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not"
- " supported!\n",ext_amode);
- continue;
- }
+ /* validate sync word using XCHFSIZE field */
+ xch_fsize = show_bits(&s->gb, 10);
+ if ((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&
+ (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))
+ continue;
- /* much like core primary audio coding header */
- dca_parse_audio_coding_header(s, s->xch_base_channel);
+ /* skip length-to-end-of-frame field for the moment */
+ skip_bits(&s->gb, 10);
- for (i = 0; i < (s->sample_blocks / 8); i++) {
- dca_decode_block(s, s->xch_base_channel, i);
+ s->core_ext_mask |= DCA_EXT_XCH;
+
+ /* extension amode(number of channels in extension) should be 1 */
+ /* AFAIK XCh is not used for more channels */
+ if ((ext_amode = get_bits(&s->gb, 4)) != 1) {
+ av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not"
+ " supported!\n", ext_amode);
+ continue;
+ }
+
+ /* much like core primary audio coding header */
+ dca_parse_audio_coding_header(s, s->xch_base_channel);
+
+ for (i = 0; i < (s->sample_blocks / 8); i++)
+ if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {
+ av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");
+ continue;
+ }
+
+ s->xch_present = 1;
+ break;
+ }
+ case 0x47004a03:
+ /* XXCh: extended channels */
+ /* usually found either in core or HD part in DTS-HD HRA streams,
+ * but not in DTS-ES which contains XCh extensions instead */
+ s->core_ext_mask |= DCA_EXT_XXCH;
+ break;
+
+ case 0x1d95f262: {
+ int fsize96 = show_bits(&s->gb, 12) + 1;
+ if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)
+ continue;
+
+ av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n",
+ get_bits_count(&s->gb));
+ skip_bits(&s->gb, 12);
+ av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);
+ av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));
+
+ s->core_ext_mask |= DCA_EXT_X96;
+ break;
+ }
}
- s->xch_present = 1;
- break;
- }
- case 0x47004a03:
- /* XXCh: extended channels */
- /* usually found either in core or HD part in DTS-HD HRA streams,
- * but not in DTS-ES which contains XCh extensions instead */
- s->xxch_present = 1;
- s->profile = FFMAX(s->profile, FF_PROFILE_DTS_ES);
- break;
-
- case 0x1d95f262: {
- int fsize96 = show_bits(&s->gb, 12) + 1;
- if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)
- continue;
-
- av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n", get_bits_count(&s->gb));
- skip_bits(&s->gb, 12);
- av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);
- av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));
-
- s->x96_present = 1;
- s->profile = FFMAX(s->profile, FF_PROFILE_DTS_96_24);
- break;
+ skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
}
- }
-
- skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
+ } else {
+ /* no supported extensions, skip the rest of the core substream */
+ skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));
}
+ if (s->core_ext_mask & DCA_EXT_X96)
+ s->profile = FF_PROFILE_DTS_96_24;
+ else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))
+ s->profile = FF_PROFILE_DTS_ES;
+
/* check for ExSS (HD part) */
- if (s->dca_buffer_size - s->frame_size > 32
- && get_bits_long(&s->gb, 32) == DCA_HD_MARKER)
+ if (s->dca_buffer_size - s->frame_size > 32 &&
+ get_bits_long(&s->gb, 32) == DCA_HD_MARKER)
dca_exss_parse_header(s);
avctx->profile = s->profile;
channels = s->prim_channels + !!s->lfe;
- if (s->amode<16) {
+ if (s->amode < 16) {
avctx->channel_layout = dca_core_channel_layout[s->amode];
if (s->xch_present && (!avctx->request_channels ||
if (channels > !!s->lfe &&
s->channel_order_tab[channels - 1 - !!s->lfe] < 0)
- return -1;
+ return AVERROR_INVALIDDATA;
if (avctx->request_channels == 2 && s->prim_channels > 2) {
channels = 2;
avctx->channel_layout = AV_CH_LAYOUT_STEREO;
}
} else {
- av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n",s->amode);
- return -1;
+ av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode);
+ return AVERROR_INVALIDDATA;
}
/* There is nothing that prevents a dts frame to change channel configuration
- but FFmpeg doesn't support that so only set the channels if it is previously
+ but Libav doesn't support that so only set the channels if it is previously
unset. Ideally during the first probe for channels the crc should be checked
and only set avctx->channels when the crc is ok. Right now the decoder could
set the channels based on a broken first frame.*/
if (avctx->channels != channels) {
av_log(avctx, AV_LOG_ERROR, "DCA decoder does not support number of "
"channels changing in stream. Skipping frame.\n");
- return -1;
+ return AVERROR_PATCHWELCOME;
}
- if (*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
- return -1;
- *data_size = 256 / 8 * s->sample_blocks * sizeof(int16_t) * channels;
+ /* get output buffer */
+ s->frame.nb_samples = 256 * (s->sample_blocks / 8);
+ if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
+ }
+ samples_flt = (float *) s->frame.data[0];
+ samples_s16 = (int16_t *) s->frame.data[0];
/* filter to get final output */
for (i = 0; i < (s->sample_blocks / 8); i++) {
/* If this was marked as a DTS-ES stream we need to subtract back- */
/* channel from SL & SR to remove matrixed back-channel signal */
- if((s->source_pcm_res & 1) && s->xch_present) {
- float* back_chan = s->samples + s->channel_order_tab[s->xch_base_channel] * 256;
- float* lt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 2] * 256;
- float* rt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256;
- int j;
- for(j = 0; j < 256; ++j) {
- lt_chan[j] -= (back_chan[j] - s->add_bias) * M_SQRT1_2;
- rt_chan[j] -= (back_chan[j] - s->add_bias) * M_SQRT1_2;
- }
+ if ((s->source_pcm_res & 1) && s->xch_present) {
+ float *back_chan = s->samples + s->channel_order_tab[s->xch_base_channel] * 256;
+ float *lt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 2] * 256;
+ float *rt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256;
+ s->dsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);
+ s->dsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);
}
- s->dsp.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels);
- samples += 256 * channels;
+ if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
+ s->fmt_conv.float_interleave(samples_flt, s->samples_chanptr, 256,
+ channels);
+ samples_flt += 256 * channels;
+ } else {
+ s->fmt_conv.float_to_int16_interleave(samples_s16,
+ s->samples_chanptr, 256,
+ channels);
+ samples_s16 += 256 * channels;
+ }
}
/* update lfe history */
lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);
- for (i = 0; i < 2 * s->lfe * 4; i++) {
+ for (i = 0; i < 2 * s->lfe * 4; i++)
s->lfe_data[i] = s->lfe_data[i + lfe_samples];
- }
+
+ *got_frame_ptr = 1;
+ *(AVFrame *) data = s->frame;
return buf_size;
}
* @param avctx pointer to the AVCodecContext
*/
-static av_cold int dca_decode_init(AVCodecContext * avctx)
+static av_cold int dca_decode_init(AVCodecContext *avctx)
{
DCAContext *s = avctx->priv_data;
int i;
s->avctx = avctx;
dca_init_vlcs();
- dsputil_init(&s->dsp, avctx);
+ ff_dsputil_init(&s->dsp, avctx);
ff_mdct_init(&s->imdct, 6, 1, 1.0);
ff_synth_filter_init(&s->synth);
ff_dcadsp_init(&s->dcadsp);
+ ff_fmt_convert_init(&s->fmt_conv, avctx);
- for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)
+ for (i = 0; i < DCA_PRIM_CHANNELS_MAX + 1; i++)
s->samples_chanptr[i] = s->samples + i * 256;
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
- if (s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
- s->add_bias = 385.0f;
- s->scale_bias = 1.0 / 32768.0;
+ if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+ s->scale_bias = 1.0 / 32768.0;
} else {
- s->add_bias = 0.0f;
- s->scale_bias = 1.0;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ s->scale_bias = 1.0;
+ }
- /* allow downmixing to stereo */
- if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
- avctx->request_channels == 2) {
- avctx->channels = avctx->request_channels;
- }
+ /* allow downmixing to stereo */
+ if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
+ avctx->request_channels == 2) {
+ avctx->channels = avctx->request_channels;
}
+ avcodec_get_frame_defaults(&s->frame);
+ avctx->coded_frame = &s->frame;
return 0;
}
-static av_cold int dca_decode_end(AVCodecContext * avctx)
+static av_cold int dca_decode_end(AVCodecContext *avctx)
{
DCAContext *s = avctx->priv_data;
ff_mdct_end(&s->imdct);
return 0;
}
-AVCodec dca_decoder = {
- .name = "dca",
- .type = AVMEDIA_TYPE_AUDIO,
- .id = CODEC_ID_DTS,
- .priv_data_size = sizeof(DCAContext),
- .init = dca_decode_init,
- .decode = dca_decode_frame,
- .close = dca_decode_end,
- .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
- .capabilities = CODEC_CAP_CHANNEL_CONF,
+static const AVProfile profiles[] = {
+ { FF_PROFILE_DTS, "DTS" },
+ { FF_PROFILE_DTS_ES, "DTS-ES" },
+ { FF_PROFILE_DTS_96_24, "DTS 96/24" },
+ { FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },
+ { FF_PROFILE_DTS_HD_MA, "DTS-HD MA" },
+ { FF_PROFILE_UNKNOWN },
+};
+
+AVCodec ff_dca_decoder = {
+ .name = "dca",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_DTS,
+ .priv_data_size = sizeof(DCAContext),
+ .init = dca_decode_init,
+ .decode = dca_decode_frame,
+ .close = dca_decode_end,
+ .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
+ .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
+ AV_SAMPLE_FMT_S16,
+ AV_SAMPLE_FMT_NONE },
+ .profiles = NULL_IF_CONFIG_SMALL(profiles),
};
projects / ffmpeg / blobdiff