[FFmpeg-cvslog] r8881 - in trunk/libavcodec: Makefile dca.c dca.h dca_parser.c
diego
subversion
Fri May 4 01:50:24 CEST 2007
Author: diego
Date: Fri May 4 01:50:24 2007
New Revision: 8881
Log:
Move dca parser to its own file.
Added:
trunk/libavcodec/dca.h
- copied, changed from r8880, /trunk/libavcodec/dca.c
trunk/libavcodec/dca_parser.c
- copied, changed from r8880, /trunk/libavcodec/dca.c
Modified:
trunk/libavcodec/Makefile
trunk/libavcodec/dca.c
Modified: trunk/libavcodec/Makefile
==============================================================================
--- trunk/libavcodec/Makefile (original)
+++ trunk/libavcodec/Makefile Fri May 4 01:50:24 2007
@@ -276,6 +276,7 @@ OBJS-$(CONFIG_XVID) +
OBJS-$(CONFIG_AAC_PARSER) += parser.o
OBJS-$(CONFIG_AC3_PARSER) += parser.o ac3tab.o
OBJS-$(CONFIG_CAVSVIDEO_PARSER) += cavs.o parser.o
+OBJS-$(CONFIG_DCA_PARSER) += dca_parser.o
OBJS-$(CONFIG_DVBSUB_PARSER) += dvbsubdec.o
OBJS-$(CONFIG_DVDSUB_PARSER) += dvdsubdec.o
OBJS-$(CONFIG_H261_PARSER) += h261.o
Modified: trunk/libavcodec/dca.c
==============================================================================
--- trunk/libavcodec/dca.c (original)
+++ trunk/libavcodec/dca.c Fri May 4 01:50:24 2007
@@ -35,15 +35,7 @@
#include "bitstream.h"
#include "dcadata.h"
#include "dcahuff.h"
-#include "parser.h"
-
-/** DCA syncwords, also used for bitstream type detection */
-//@{
-#define DCA_MARKER_RAW_BE 0x7FFE8001
-#define DCA_MARKER_RAW_LE 0xFE7F0180
-#define DCA_MARKER_14B_BE 0x1FFFE800
-#define DCA_MARKER_14B_LE 0xFF1F00E8
-//@}
+#include "dca.h"
//#define TRACE
@@ -1255,102 +1247,3 @@ AVCodec dca_decoder = {
.init = dca_decode_init,
.decode = dca_decode_frame,
};
-
-#ifdef CONFIG_DCA_PARSER
-
-typedef struct DCAParseContext {
- ParseContext pc;
- uint32_t lastmarker;
-} DCAParseContext;
-
-#define IS_MARKER(state, i, buf, buf_size) \
- ((state == DCA_MARKER_14B_LE && (i < buf_size-2) && (buf[i+1] & 0xF0) == 0xF0 && buf[i+2] == 0x07) \
- || (state == DCA_MARKER_14B_BE && (i < buf_size-2) && buf[i+1] == 0x07 && (buf[i+2] & 0xF0) == 0xF0) \
- || state == DCA_MARKER_RAW_LE || state == DCA_MARKER_RAW_BE)
-
-/**
- * finds the end of the current frame in the bitstream.
- * @return the position of the first byte of the next frame, or -1
- */
-static int dca_find_frame_end(DCAParseContext * pc1, const uint8_t * buf,
- int buf_size)
-{
- int start_found, i;
- uint32_t state;
- ParseContext *pc = &pc1->pc;
-
- start_found = pc->frame_start_found;
- state = pc->state;
-
- i = 0;
- if (!start_found) {
- for (i = 0; i < buf_size; i++) {
- state = (state << 8) | buf[i];
- if (IS_MARKER(state, i, buf, buf_size)) {
- if (pc1->lastmarker && state == pc1->lastmarker) {
- start_found = 1;
- break;
- } else if (!pc1->lastmarker) {
- start_found = 1;
- pc1->lastmarker = state;
- break;
- }
- }
- }
- }
- if (start_found) {
- for (; i < buf_size; i++) {
- state = (state << 8) | buf[i];
- if (state == pc1->lastmarker && IS_MARKER(state, i, buf, buf_size)) {
- pc->frame_start_found = 0;
- pc->state = -1;
- return i - 3;
- }
- }
- }
- pc->frame_start_found = start_found;
- pc->state = state;
- return END_NOT_FOUND;
-}
-
-static int dca_parse_init(AVCodecParserContext * s)
-{
- DCAParseContext *pc1 = s->priv_data;
-
- pc1->lastmarker = 0;
- return 0;
-}
-
-static int dca_parse(AVCodecParserContext * s,
- AVCodecContext * avctx,
- uint8_t ** poutbuf, int *poutbuf_size,
- const uint8_t * buf, int buf_size)
-{
- DCAParseContext *pc1 = s->priv_data;
- ParseContext *pc = &pc1->pc;
- int next;
-
- if (s->flags & PARSER_FLAG_COMPLETE_FRAMES) {
- next = buf_size;
- } else {
- next = dca_find_frame_end(pc1, buf, buf_size);
-
- if (ff_combine_frame(pc, next, (uint8_t **) & buf, &buf_size) < 0) {
- *poutbuf = NULL;
- *poutbuf_size = 0;
- return buf_size;
- }
- }
- *poutbuf = (uint8_t *) buf;
- *poutbuf_size = buf_size;
- return next;
-}
-
-AVCodecParser dca_parser = {
- {CODEC_ID_DTS},
- sizeof(DCAParseContext),
- dca_parse_init,
- dca_parse,
- ff_parse_close,
-};
-#endif /* CONFIG_DCA_PARSER */
Copied: trunk/libavcodec/dca.h (from r8880, /trunk/libavcodec/dca.c)
==============================================================================
--- /trunk/libavcodec/dca.c (original)
+++ trunk/libavcodec/dca.h Fri May 4 01:50:24 2007
@@ -22,1335 +22,8 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-/**
- * @file dca.c
- */
-
-#include <math.h>
-#include <stddef.h>
-#include <stdio.h>
-
-#include "avcodec.h"
-#include "dsputil.h"
-#include "bitstream.h"
-#include "dcadata.h"
-#include "dcahuff.h"
-#include "parser.h"
-
/** DCA syncwords, also used for bitstream type detection */
-//@{
#define DCA_MARKER_RAW_BE 0x7FFE8001
#define DCA_MARKER_RAW_LE 0xFE7F0180
#define DCA_MARKER_14B_BE 0x1FFFE800
#define DCA_MARKER_14B_LE 0xFF1F00E8
-//@}
-
-//#define TRACE
-
-#define DCA_PRIM_CHANNELS_MAX (5)
-#define DCA_SUBBANDS (32)
-#define DCA_ABITS_MAX (32) /* Should be 28 */
-#define DCA_SUBSUBFAMES_MAX (4)
-#define DCA_LFE_MAX (3)
-
-enum DCAMode {
- DCA_MONO = 0,
- DCA_CHANNEL,
- DCA_STEREO,
- DCA_STEREO_SUMDIFF,
- DCA_STEREO_TOTAL,
- DCA_3F,
- DCA_2F1R,
- DCA_3F1R,
- DCA_2F2R,
- DCA_3F2R,
- DCA_4F2R
-};
-
-#define DCA_DOLBY 101 /* FIXME */
-
-#define DCA_CHANNEL_BITS 6
-#define DCA_CHANNEL_MASK 0x3F
-
-#define DCA_LFE 0x80
-
-#define HEADER_SIZE 14
-#define CONVERT_BIAS 384
-
-#define DCA_MAX_FRAME_SIZE 16383
-
-/** Bit allocation */
-typedef struct {
- int offset; ///< code values offset
- int maxbits[8]; ///< max bits in VLC
- int wrap; ///< wrap for get_vlc2()
- VLC vlc[8]; ///< actual codes
-} BitAlloc;
-
-static BitAlloc dca_bitalloc_index; ///< indexes for samples VLC select
-static BitAlloc dca_tmode; ///< transition mode VLCs
-static BitAlloc dca_scalefactor; ///< scalefactor VLCs
-static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs
-
-/** Pre-calculated cosine modulation coefs for the QMF */
-static float cos_mod[544];
-
-static int av_always_inline get_bitalloc(GetBitContext *gb, BitAlloc *ba, int idx)
-{
- return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + ba->offset;
-}
-
-typedef struct {
- AVCodecContext *avctx;
- /* Frame header */
- int frame_type; ///< type of the current frame
- int samples_deficit; ///< deficit sample count
- int crc_present; ///< crc is present in the bitstream
- int sample_blocks; ///< number of PCM sample blocks
- int frame_size; ///< primary frame byte size
- int amode; ///< audio channels arrangement
- int sample_rate; ///< audio sampling rate
- int bit_rate; ///< transmission bit rate
-
- int downmix; ///< embedded downmix enabled
- int dynrange; ///< embedded dynamic range flag
- int timestamp; ///< embedded time stamp flag
- int aux_data; ///< auxiliary data flag
- int hdcd; ///< source material is mastered in HDCD
- int ext_descr; ///< extension audio descriptor flag
- int ext_coding; ///< extended coding flag
- int aspf; ///< audio sync word insertion flag
- int lfe; ///< low frequency effects flag
- int predictor_history; ///< predictor history flag
- int header_crc; ///< header crc check bytes
- int multirate_inter; ///< multirate interpolator switch
- int version; ///< encoder software revision
- int copy_history; ///< copy history
- int source_pcm_res; ///< source pcm resolution
- int front_sum; ///< front sum/difference flag
- int surround_sum; ///< surround sum/difference flag
- int dialog_norm; ///< dialog normalisation parameter
-
- /* Primary audio coding header */
- int subframes; ///< number of subframes
- int prim_channels; ///< number of primary audio channels
- int subband_activity[DCA_PRIM_CHANNELS_MAX]; ///< subband activity count
- int vq_start_subband[DCA_PRIM_CHANNELS_MAX]; ///< high frequency vq start subband
- int joint_intensity[DCA_PRIM_CHANNELS_MAX]; ///< joint intensity coding index
- int transient_huffman[DCA_PRIM_CHANNELS_MAX]; ///< transient mode code book
- int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book
- int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX]; ///< bit allocation quantizer select
- int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select
- float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment
-
- /* Primary audio coding side information */
- int subsubframes; ///< number of subsubframes
- int partial_samples; ///< 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 transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< transition mode (transients)
- int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2]; ///< scale factors (2 if transient)
- int joint_huff[DCA_PRIM_CHANNELS_MAX]; ///< joint subband scale factors codebook
- int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors
- int downmix_coef[DCA_PRIM_CHANNELS_MAX][2]; ///< stereo downmix coefficients
- int dynrange_coef; ///< dynamic range coefficient
-
- int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< VQ encoded high frequency subbands
-
- float lfe_data[2 * DCA_SUBSUBFAMES_MAX * DCA_LFE_MAX *
- 2 /*history */ ]; ///< Low frequency effect data
- int lfe_scale_factor;
-
- /* Subband samples history (for ADPCM) */
- float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
- float subband_fir_hist[DCA_PRIM_CHANNELS_MAX][512];
- float subband_fir_noidea[DCA_PRIM_CHANNELS_MAX][64];
-
- int output; ///< type of output
- int bias; ///< output bias
-
- DECLARE_ALIGNED_16(float, samples[1536]); /* 6 * 256 = 1536, might only need 5 */
- DECLARE_ALIGNED_16(int16_t, tsamples[1536]);
-
- uint8_t dca_buffer[DCA_MAX_FRAME_SIZE];
- int dca_buffer_size; ///< how much data is in the dca_buffer
-
- GetBitContext gb;
- /* Current position in DCA frame */
- int current_subframe;
- int current_subsubframe;
-
- int debug_flag; ///< used for suppressing repeated error messages output
- DSPContext dsp;
-} DCAContext;
-
-static void dca_init_vlcs(void)
-{
- static int vlcs_inited = 0;
- int i, j;
-
- if (vlcs_inited)
- return;
-
- dca_bitalloc_index.offset = 1;
- dca_bitalloc_index.wrap = 1;
- for (i = 0; i < 5; i++)
- init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12,
- bitalloc_12_bits[i], 1, 1,
- bitalloc_12_codes[i], 2, 2, 1);
- dca_scalefactor.offset = -64;
- dca_scalefactor.wrap = 2;
- for (i = 0; i < 5; i++)
- init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,
- scales_bits[i], 1, 1,
- scales_codes[i], 2, 2, 1);
- dca_tmode.offset = 0;
- dca_tmode.wrap = 1;
- for (i = 0; i < 4; i++)
- init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
- tmode_bits[i], 1, 1,
- tmode_codes[i], 2, 2, 1);
-
- 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);
- 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, 1);
- }
- vlcs_inited = 1;
-}
-
-static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
-{
- while(len--)
- *dst++ = get_bits(gb, bits);
-}
-
-static int dca_parse_frame_header(DCAContext * s)
-{
- 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 };
-
- s->bias = CONVERT_BIAS;
-
- init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
-
- /* Sync code */
- get_bits(&s->gb, 32);
-
- /* Frame header */
- s->frame_type = get_bits(&s->gb, 1);
- s->samples_deficit = get_bits(&s->gb, 5) + 1;
- s->crc_present = 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;
- s->amode = get_bits(&s->gb, 6);
- s->sample_rate = dca_sample_rates[get_bits(&s->gb, 4)];
- if (!s->sample_rate)
- return -1;
- s->bit_rate = dca_bit_rates[get_bits(&s->gb, 5)];
- if (!s->bit_rate)
- return -1;
-
- s->downmix = get_bits(&s->gb, 1);
- s->dynrange = get_bits(&s->gb, 1);
- s->timestamp = get_bits(&s->gb, 1);
- s->aux_data = get_bits(&s->gb, 1);
- s->hdcd = get_bits(&s->gb, 1);
- s->ext_descr = get_bits(&s->gb, 3);
- s->ext_coding = get_bits(&s->gb, 1);
- s->aspf = get_bits(&s->gb, 1);
- s->lfe = get_bits(&s->gb, 2);
- s->predictor_history = get_bits(&s->gb, 1);
-
- /* TODO: check CRC */
- if (s->crc_present)
- s->header_crc = get_bits(&s->gb, 16);
-
- s->multirate_inter = get_bits(&s->gb, 1);
- s->version = get_bits(&s->gb, 4);
- s->copy_history = get_bits(&s->gb, 2);
- s->source_pcm_res = get_bits(&s->gb, 3);
- s->front_sum = get_bits(&s->gb, 1);
- s->surround_sum = get_bits(&s->gb, 1);
- s->dialog_norm = get_bits(&s->gb, 4);
-
- /* FIXME: channels mixing levels */
- s->output = s->amode;
- if(s->lfe) s->output |= DCA_LFE;
-
-#ifdef TRACE
- av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type);
- av_log(s->avctx, AV_LOG_DEBUG, "samples deficit: %i\n", s->samples_deficit);
- av_log(s->avctx, AV_LOG_DEBUG, "crc present: %i\n", s->crc_present);
- av_log(s->avctx, AV_LOG_DEBUG, "sample blocks: %i (%i samples)\n",
- s->sample_blocks, s->sample_blocks * 32);
- av_log(s->avctx, AV_LOG_DEBUG, "frame size: %i bytes\n", s->frame_size);
- av_log(s->avctx, AV_LOG_DEBUG, "amode: %i (%i channels)\n",
- s->amode, dca_channels[s->amode]);
- av_log(s->avctx, AV_LOG_DEBUG, "sample rate: %i (%i Hz)\n",
- s->sample_rate, dca_sample_rates[s->sample_rate]);
- av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i (%i bits/s)\n",
- s->bit_rate, dca_bit_rates[s->bit_rate]);
- av_log(s->avctx, AV_LOG_DEBUG, "downmix: %i\n", s->downmix);
- av_log(s->avctx, AV_LOG_DEBUG, "dynrange: %i\n", s->dynrange);
- av_log(s->avctx, AV_LOG_DEBUG, "timestamp: %i\n", s->timestamp);
- av_log(s->avctx, AV_LOG_DEBUG, "aux_data: %i\n", s->aux_data);
- av_log(s->avctx, AV_LOG_DEBUG, "hdcd: %i\n", s->hdcd);
- av_log(s->avctx, AV_LOG_DEBUG, "ext descr: %i\n", s->ext_descr);
- av_log(s->avctx, AV_LOG_DEBUG, "ext coding: %i\n", s->ext_coding);
- av_log(s->avctx, AV_LOG_DEBUG, "aspf: %i\n", s->aspf);
- av_log(s->avctx, AV_LOG_DEBUG, "lfe: %i\n", s->lfe);
- av_log(s->avctx, AV_LOG_DEBUG, "predictor history: %i\n",
- s->predictor_history);
- av_log(s->avctx, AV_LOG_DEBUG, "header crc: %i\n", s->header_crc);
- av_log(s->avctx, AV_LOG_DEBUG, "multirate inter: %i\n",
- s->multirate_inter);
- av_log(s->avctx, AV_LOG_DEBUG, "version number: %i\n", s->version);
- av_log(s->avctx, AV_LOG_DEBUG, "copy history: %i\n", s->copy_history);
- av_log(s->avctx, AV_LOG_DEBUG,
- "source pcm resolution: %i (%i bits/sample)\n",
- s->source_pcm_res, dca_bits_per_sample[s->source_pcm_res]);
- av_log(s->avctx, AV_LOG_DEBUG, "front sum: %i\n", s->front_sum);
- av_log(s->avctx, AV_LOG_DEBUG, "surround sum: %i\n", s->surround_sum);
- av_log(s->avctx, AV_LOG_DEBUG, "dialog norm: %i\n", s->dialog_norm);
- av_log(s->avctx, AV_LOG_DEBUG, "\n");
-#endif
-
- /* Primary audio coding header */
- s->subframes = get_bits(&s->gb, 4) + 1;
- s->prim_channels = get_bits(&s->gb, 3) + 1;
-
-
- for (i = 0; i < s->prim_channels; i++) {
- s->subband_activity[i] = get_bits(&s->gb, 5) + 2;
- if (s->subband_activity[i] > DCA_SUBBANDS)
- s->subband_activity[i] = DCA_SUBBANDS;
- }
- for (i = 0; i < s->prim_channels; i++) {
- s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
- if (s->vq_start_subband[i] > DCA_SUBBANDS)
- s->vq_start_subband[i] = DCA_SUBBANDS;
- }
- get_array(&s->gb, s->joint_intensity, s->prim_channels, 3);
- get_array(&s->gb, s->transient_huffman, s->prim_channels, 2);
- get_array(&s->gb, s->scalefactor_huffman, s->prim_channels, 3);
- get_array(&s->gb, s->bitalloc_huffman, s->prim_channels, 3);
-
- /* Get codebooks quantization indexes */
- memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
- for (j = 1; j < 11; j++)
- for (i = 0; i < s->prim_channels; i++)
- s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
-
- /* Get scale factor adjustment */
- for (j = 0; j < 11; j++)
- for (i = 0; i < s->prim_channels; i++)
- s->scalefactor_adj[i][j] = 1;
-
- for (j = 1; j < 11; j++)
- for (i = 0; i < s->prim_channels; i++)
- if (s->quant_index_huffman[i][j] < thr[j])
- s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
-
- if (s->crc_present) {
- /* Audio header CRC check */
- get_bits(&s->gb, 16);
- }
-
- 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 = 0; 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, "\n");
- av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:");
- for (j = 0; j < 11; j++)
- av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]);
- av_log(s->avctx, AV_LOG_DEBUG, "\n");
- }
-#endif
-
- return 0;
-}
-
-
-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;
-}
-
-static int dca_subframe_header(DCAContext * s)
-{
- /* Primary audio coding side information */
- int j, k;
-
- s->subsubframes = get_bits(&s->gb, 2) + 1;
- s->partial_samples = get_bits(&s->gb, 3);
- for (j = 0; j < s->prim_channels; j++) {
- for (k = 0; k < s->subband_activity[j]; k++)
- s->prediction_mode[j][k] = get_bits(&s->gb, 1);
- }
-
- /* Get prediction codebook */
- for (j = 0; j < s->prim_channels; j++) {
- for (k = 0; k < s->subband_activity[j]; k++) {
- if (s->prediction_mode[j][k] > 0) {
- /* (Prediction coefficient VQ address) */
- s->prediction_vq[j][k] = get_bits(&s->gb, 12);
- }
- }
- }
-
- /* Bit allocation index */
- for (j = 0; j < s->prim_channels; j++) {
- for (k = 0; k < s->vq_start_subband[j]; k++) {
- if (s->bitalloc_huffman[j] == 6)
- s->bitalloc[j][k] = get_bits(&s->gb, 5);
- else if (s->bitalloc_huffman[j] == 5)
- s->bitalloc[j][k] = get_bits(&s->gb, 4);
- 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;
- }
- }
- }
-
- /* Transition mode */
- for (j = 0; j < s->prim_channels; j++) {
- for (k = 0; k < s->subband_activity[j]; k++) {
- s->transition_mode[j][k] = 0;
- if (s->subsubframes > 1 &&
- k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) {
- s->transition_mode[j][k] =
- get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]);
- }
- }
- }
-
- for (j = 0; j < s->prim_channels; j++) {
- 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);
-
- if (s->scalefactor_huffman[j] == 6)
- scale_table = (uint32_t *) scale_factor_quant7;
- else
- scale_table = (uint32_t *) scale_factor_quant6;
-
- /* When huffman coded, only the difference is encoded */
- scale_sum = 0;
-
- for (k = 0; k < s->subband_activity[j]; k++) {
- if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) {
- scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum);
- s->scale_factor[j][k][0] = scale_table[scale_sum];
- }
-
- if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) {
- /* Get second scale factor */
- scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum);
- s->scale_factor[j][k][1] = scale_table[scale_sum];
- }
- }
- }
-
- /* Joint subband scale factor codebook select */
- for (j = 0; j < s->prim_channels; j++) {
- /* Transmitted only if joint subband coding enabled */
- if (s->joint_intensity[j] > 0)
- s->joint_huff[j] = get_bits(&s->gb, 3);
- }
-
- /* Scale factors for joint subband coding */
- for (j = 0; j < s->prim_channels; j++) {
- int source_channel;
-
- /* Transmitted only if joint subband coding enabled */
- if (s->joint_intensity[j] > 0) {
- int scale = 0;
- source_channel = s->joint_intensity[j] - 1;
-
- /* When huffman coded, only the difference is encoded
- * (is this valid as well for joint scales ???) */
-
- for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) {
- scale = get_scale(&s->gb, s->joint_huff[j], 0);
- scale += 64; /* bias */
- s->joint_scale_factor[j][k] = scale; /*joint_scale_table[scale]; */
- }
-
- if (!s->debug_flag & 0x02) {
- av_log(s->avctx, AV_LOG_DEBUG,
- "Joint stereo coding not supported\n");
- s->debug_flag |= 0x02;
- }
- }
- }
-
- /* Stereo downmix coefficients */
- if (s->prim_channels > 2) {
- if(s->downmix) {
- for (j = 0; j < s->prim_channels; j++) {
- s->downmix_coef[j][0] = get_bits(&s->gb, 7);
- s->downmix_coef[j][1] = get_bits(&s->gb, 7);
- }
- } else {
- int am = s->amode & DCA_CHANNEL_MASK;
- for (j = 0; j < s->prim_channels; j++) {
- s->downmix_coef[j][0] = dca_default_coeffs[am][j][0];
- s->downmix_coef[j][1] = dca_default_coeffs[am][j][1];
- }
- }
- }
-
- /* Dynamic range coefficient */
- if (s->dynrange)
- s->dynrange_coef = get_bits(&s->gb, 8);
-
- /* Side information CRC check word */
- if (s->crc_present) {
- get_bits(&s->gb, 16);
- }
-
- /*
- * Primary audio data arrays
- */
-
- /* VQ encoded high frequency subbands */
- for (j = 0; j < s->prim_channels; j++)
- for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
- /* 1 vector -> 32 samples */
- s->high_freq_vq[j][k] = get_bits(&s->gb, 10);
-
- /* Low frequency effect data */
- if (s->lfe) {
- /* LFE samples */
- int lfe_samples = 2 * s->lfe * s->subsubframes;
- float lfe_scale;
-
- for (j = lfe_samples; j < lfe_samples * 2; j++) {
- /* Signed 8 bits int */
- s->lfe_data[j] = get_sbits(&s->gb, 8);
- }
-
- /* Scale factor index */
- s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 8)];
-
- /* Quantization step size * scale factor */
- lfe_scale = 0.035 * s->lfe_scale_factor;
-
- for (j = lfe_samples; j < lfe_samples * 2; j++)
- s->lfe_data[j] *= lfe_scale;
- }
-
-#ifdef TRACE
- av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n", s->subsubframes);
- av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n",
- s->partial_samples);
- for (j = 0; j < s->prim_channels; j++) {
- av_log(s->avctx, AV_LOG_DEBUG, "prediction mode:");
- for (k = 0; k < s->subband_activity[j]; k++)
- av_log(s->avctx, AV_LOG_DEBUG, " %i", s->prediction_mode[j][k]);
- av_log(s->avctx, AV_LOG_DEBUG, "\n");
- }
- for (j = 0; 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);
- }
- for (j = 0; j < s->prim_channels; j++) {
- av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index: ");
- for (k = 0; k < s->vq_start_subband[j]; k++)
- av_log(s->avctx, AV_LOG_DEBUG, "%2.2i ", s->bitalloc[j][k]);
- av_log(s->avctx, AV_LOG_DEBUG, "\n");
- }
- for (j = 0; j < s->prim_channels; j++) {
- av_log(s->avctx, AV_LOG_DEBUG, "Transition mode:");
- for (k = 0; k < s->subband_activity[j]; k++)
- av_log(s->avctx, AV_LOG_DEBUG, " %i", s->transition_mode[j][k]);
- av_log(s->avctx, AV_LOG_DEBUG, "\n");
- }
- for (j = 0; j < s->prim_channels; j++) {
- av_log(s->avctx, AV_LOG_DEBUG, "Scale factor:");
- for (k = 0; k < s->subband_activity[j]; k++) {
- if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0)
- av_log(s->avctx, AV_LOG_DEBUG, " %i", s->scale_factor[j][k][0]);
- if (k < s->vq_start_subband[j] && s->transition_mode[j][k])
- av_log(s->avctx, AV_LOG_DEBUG, " %i(t)", s->scale_factor[j][k][1]);
- }
- av_log(s->avctx, AV_LOG_DEBUG, "\n");
- }
- for (j = 0; j < s->prim_channels; j++) {
- if (s->joint_intensity[j] > 0) {
- av_log(s->avctx, AV_LOG_DEBUG, "Joint scale factor index:\n");
- for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++)
- av_log(s->avctx, AV_LOG_DEBUG, " %i", s->joint_scale_factor[j][k]);
- av_log(s->avctx, AV_LOG_DEBUG, "\n");
- }
- }
- if (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, "\n");
- }
- for (j = 0; j < s->prim_channels; j++)
- for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
- av_log(s->avctx, AV_LOG_DEBUG, "VQ index: %i\n", s->high_freq_vq[j][k]);
- if(s->lfe){
- av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n");
- for (j = lfe_samples; j < lfe_samples * 2; j++)
- av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]);
- av_log(s->avctx, AV_LOG_DEBUG, "\n");
- }
-#endif
-
- return 0;
-}
-
-static void qmf_32_subbands(DCAContext * s, int chans,
- float samples_in[32][8], float *samples_out,
- float scale, float bias)
-{
- float *prCoeff;
- int i, j, k;
- float praXin[33], *raXin = &praXin[1];
-
- float *subband_fir_hist = s->subband_fir_hist[chans];
- float *subband_fir_hist2 = s->subband_fir_noidea[chans];
-
- int chindex = 0, subindex;
-
- praXin[0] = 0.0;
-
- /* Select filter */
- if (!s->multirate_inter) /* Non-perfect reconstruction */
- prCoeff = (float *) fir_32bands_nonperfect;
- else /* Perfect reconstruction */
- prCoeff = (float *) fir_32bands_perfect;
-
- /* Reconstructed channel sample index */
- for (subindex = 0; subindex < 8; subindex++) {
- float t1, t2, sum[16], diff[16];
-
- /* Load in one sample from each subband and clear inactive subbands */
- for (i = 0; i < s->subband_activity[chans]; i++)
- raXin[i] = samples_in[i][subindex];
- for (; i < 32; i++)
- raXin[i] = 0.0;
-
- /* Multiply by cosine modulation coefficients and
- * create temporary arrays SUM and DIFF */
- for (j = 0, k = 0; k < 16; k++) {
- t1 = 0.0;
- t2 = 0.0;
- for (i = 0; i < 16; i++, j++){
- t1 += (raXin[2 * i] + raXin[2 * i + 1]) * cos_mod[j];
- t2 += (raXin[2 * i] + raXin[2 * i - 1]) * cos_mod[j + 256];
- }
- sum[k] = t1 + t2;
- diff[k] = t1 - t2;
- }
-
- j = 512;
- /* Store history */
- for (k = 0; k < 16; k++)
- subband_fir_hist[k] = cos_mod[j++] * sum[k];
- for (k = 0; k < 16; k++)
- subband_fir_hist[32-k-1] = cos_mod[j++] * diff[k];
-
- /* Multiply by filter coefficients */
- for (k = 31, i = 0; i < 32; i++, k--)
- for (j = 0; j < 512; j += 64){
- subband_fir_hist2[i] += prCoeff[i+j] * ( subband_fir_hist[i+j] - subband_fir_hist[j+k]);
- subband_fir_hist2[i+32] += prCoeff[i+j+32]*(-subband_fir_hist[i+j] - subband_fir_hist[j+k]);
- }
-
- /* Create 32 PCM output samples */
- for (i = 0; i < 32; i++)
- samples_out[chindex++] = subband_fir_hist2[i] * scale + bias;
-
- /* Update working arrays */
- memmove(&subband_fir_hist[32], &subband_fir_hist[0], (512 - 32) * sizeof(float));
- memmove(&subband_fir_hist2[0], &subband_fir_hist2[32], 32 * sizeof(float));
- memset(&subband_fir_hist2[32], 0, 32 * sizeof(float));
- }
-}
-
-static void lfe_interpolation_fir(int decimation_select,
- int num_deci_sample, float *samples_in,
- float *samples_out, float scale,
- float bias)
-{
- /* samples_in: An array holding decimated samples.
- * Samples in current subframe starts from samples_in[0],
- * while samples_in[-1], samples_in[-2], ..., stores samples
- * from last subframe as history.
- *
- * samples_out: An array holding interpolated samples
- */
-
- int decifactor, k, j;
- const float *prCoeff;
-
- int interp_index = 0; /* Index to the interpolated samples */
- int deciindex;
-
- /* Select decimation filter */
- if (decimation_select == 1) {
- decifactor = 128;
- prCoeff = lfe_fir_128;
- } else {
- decifactor = 64;
- prCoeff = lfe_fir_64;
- }
- /* Interpolation */
- for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
- /* One decimated sample generates decifactor interpolated ones */
- for (k = 0; k < decifactor; k++) {
- float rTmp = 0.0;
- //FIXME the coeffs are symetric, fix that
- for (j = 0; j < 512 / decifactor; j++)
- rTmp += samples_in[deciindex - j] * prCoeff[k + j * decifactor];
- samples_out[interp_index++] = rTmp / scale + bias;
- }
- }
-}
-
-/* downmixing routines */
-#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]; \
- samples[i] = t * coef[0][0] + samples[i+256] * coef[1][0] + samples[i+512] * coef[2][0]; \
- samples[i+256] = t * coef[0][1] + samples[i+256] * coef[1][1] + samples[i+512] * 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])
-{
- int i;
- float t;
- float coef[DCA_PRIM_CHANNELS_MAX][2];
-
- 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]];
- }
-
- switch (srcfmt) {
- case DCA_MONO:
- case DCA_CHANNEL:
- case DCA_STEREO_TOTAL:
- case DCA_STEREO_SUMDIFF:
- case DCA_4F2R:
- av_log(NULL, 0, "Not implemented!\n");
- break;
- case DCA_STEREO:
- break;
- case DCA_3F:
- DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),);
- break;
- case DCA_2F1R:
- DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + 512, 2, coef),);
- break;
- case DCA_3F1R:
- DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
- MIX_REAR1(samples, i + 768, 3, coef));
- break;
- case DCA_2F2R:
- DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + 512, i + 768, 2, coef),);
- break;
- case DCA_3F2R:
- DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
- MIX_REAR2(samples, i + 768, i + 1024, 3, coef));
- break;
- }
-}
-
-
-/* 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)
-{
- int i;
- int offset = (levels - 1) >> 1;
-
- for (i = 0; i < 4; i++) {
- values[i] = (code % levels) - offset;
- code /= levels;
- }
-
- if (code == 0)
- return 0;
- else {
- av_log(NULL, AV_LOG_ERROR, "ERROR: block code look-up failed\n");
- return -1;
- }
-}
-
-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 dca_subsubframe(DCAContext * s)
-{
- int k, l;
- int subsubframe = s->current_subsubframe;
-
- float *quant_step_table;
-
- /* FIXME */
- float subband_samples[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
-
- /*
- * Audio data
- */
-
- /* Select quantization step size table */
- if (s->bit_rate == 0x1f)
- quant_step_table = (float *) lossless_quant_d;
- else
- quant_step_table = (float *) lossy_quant_d;
-
- for (k = 0; k < s->prim_channels; k++) {
- for (l = 0; l < s->vq_start_subband[k]; l++) {
- int m;
-
- /* Select the mid-tread linear quantizer */
- int abits = s->bitalloc[k][l];
-
- float quant_step_size = quant_step_table[abits];
- float rscale;
-
- /*
- * Determine quantization index code book and its type
- */
-
- /* Select quantization index code book */
- int sel = s->quant_index_huffman[k][abits];
-
- /*
- * Extract bits from the bit stream
- */
- if(!abits){
- memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));
- }else if(abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){
- if(abits <= 7){
- /* Block code */
- int block_code1, block_code2, size, levels;
- int block[8];
-
- 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]);
- for (m = 0; m < 8; m++)
- subband_samples[k][l][m] = block[m];
- }else{
- /* no coding */
- for (m = 0; m < 8; m++)
- subband_samples[k][l][m] = get_sbits(&s->gb, abits - 3);
- }
- }else{
- /* Huffman coded */
- for (m = 0; m < 8; m++)
- subband_samples[k][l][m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);
- }
-
- /* Deal with transients */
- if (s->transition_mode[k][l] &&
- subsubframe >= s->transition_mode[k][l])
- rscale = quant_step_size * s->scale_factor[k][l][1];
- else
- rscale = quant_step_size * s->scale_factor[k][l][0];
-
- rscale *= s->scalefactor_adj[k][sel];
-
- for (m = 0; m < 8; m++)
- subband_samples[k][l][m] *= rscale;
-
- /*
- * Inverse ADPCM if in prediction mode
- */
- if (s->prediction_mode[k][l]) {
- int n;
- for (m = 0; m < 8; m++) {
- for (n = 1; n <= 4; n++)
- if (m >= n)
- subband_samples[k][l][m] +=
- (adpcm_vb[s->prediction_vq[k][l]][n - 1] *
- subband_samples[k][l][m - n] / 8192);
- 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);
- }
- }
- }
-
- /*
- * Decode VQ encoded high frequencies
- */
- 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;
-
- if (!s->debug_flag & 0x01) {
- 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;
- }
- }
- }
-
- /* Check for DSYNC after subsubframe */
- if (s->aspf || subsubframe == s->subsubframes - 1) {
- if (0xFFFF == get_bits(&s->gb, 16)) { /* 0xFFFF */
-#ifdef TRACE
- av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");
-#endif
- } else {
- av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");
- }
- }
-
- /* Backup predictor history for adpcm */
- for (k = 0; 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]));
-
- /* 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 * k],
- 2.0 / 3 /*pcm_to_double[s->source_pcm_res] */ ,
- 0 /*s->bias */ );
- }
-
- /* Down mixing */
-
- if (s->prim_channels > dca_channels[s->output & DCA_CHANNEL_MASK]) {
- dca_downmix(s->samples, s->amode, s->downmix_coef);
- }
-
- /* Generate LFE samples for this subsubframe FIXME!!! */
- if (s->output & DCA_LFE) {
- int lfe_samples = 2 * s->lfe * s->subsubframes;
- int i_channels = dca_channels[s->output & DCA_CHANNEL_MASK];
-
- lfe_interpolation_fir(s->lfe, 2 * s->lfe,
- s->lfe_data + lfe_samples +
- 2 * s->lfe * subsubframe,
- &s->samples[256 * i_channels],
- 8388608.0, s->bias);
- /* Outputs 20bits pcm samples */
- }
-
- return 0;
-}
-
-
-static int dca_subframe_footer(DCAContext * s)
-{
- int aux_data_count = 0, i;
- int lfe_samples;
-
- /*
- * Unpack optional information
- */
-
- if (s->timestamp)
- get_bits(&s->gb, 32);
-
- if (s->aux_data)
- aux_data_count = get_bits(&s->gb, 6);
-
- for (i = 0; i < aux_data_count; i++)
- get_bits(&s->gb, 8);
-
- if (s->crc_present && (s->downmix || s->dynrange))
- get_bits(&s->gb, 16);
-
- lfe_samples = 2 * s->lfe * s->subsubframes;
- for (i = 0; i < lfe_samples; i++) {
- s->lfe_data[i] = s->lfe_data[i + lfe_samples];
- }
-
- return 0;
-}
-
-/**
- * Decode a dca frame block
- *
- * @param s pointer to the DCAContext
- */
-
-static int dca_decode_block(DCAContext * s)
-{
-
- /* 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;
- }
-
- if (!s->current_subsubframe) {
-#ifdef TRACE
- av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");
-#endif
- /* Read subframe header */
- if (dca_subframe_header(s))
- return -1;
- }
-
- /* Read subsubframe */
-#ifdef TRACE
- av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");
-#endif
- if (dca_subsubframe(s))
- return -1;
-
- /* Update state */
- s->current_subsubframe++;
- if (s->current_subsubframe >= s->subsubframes) {
- s->current_subsubframe = 0;
- s->current_subframe++;
- }
- if (s->current_subframe >= s->subframes) {
-#ifdef TRACE
- av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");
-#endif
- /* Read subframe footer */
- if (dca_subframe_footer(s))
- return -1;
- }
-
- return 0;
-}
-
-/**
- * Convert bitstream to one representation based on sync marker
- */
-static int dca_convert_bitstream(uint8_t * src, int src_size, uint8_t * dst,
- int max_size)
-{
- uint32_t mrk;
- int i, tmp;
- uint16_t *ssrc = (uint16_t *) src, *sdst = (uint16_t *) dst;
- PutBitContext pb;
-
- if((unsigned)src_size > (unsigned)max_size)
- return -1;
-
- mrk = AV_RB32(src);
- switch (mrk) {
- case DCA_MARKER_RAW_BE:
- memcpy(dst, src, FFMIN(src_size, max_size));
- return FFMIN(src_size, max_size);
- case DCA_MARKER_RAW_LE:
- for (i = 0; i < (FFMIN(src_size, max_size) + 1) >> 1; i++)
- *sdst++ = bswap_16(*ssrc++);
- return FFMIN(src_size, max_size);
- case DCA_MARKER_14B_BE:
- case DCA_MARKER_14B_LE:
- init_put_bits(&pb, dst, max_size);
- for (i = 0; i < (src_size + 1) >> 1; i++, src += 2) {
- tmp = ((mrk == DCA_MARKER_14B_BE) ? AV_RB16(src) : AV_RL16(src)) & 0x3FFF;
- put_bits(&pb, 14, tmp);
- }
- flush_put_bits(&pb);
- return (put_bits_count(&pb) + 7) >> 3;
- default:
- return -1;
- }
-}
-
-/**
- * Main frame decoding function
- * FIXME add arguments
- */
-static int dca_decode_frame(AVCodecContext * avctx,
- void *data, int *data_size,
- uint8_t * buf, int buf_size)
-{
-
- int i, j, k;
- int16_t *samples = data;
- DCAContext *s = avctx->priv_data;
- int channels;
-
-
- s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer, DCA_MAX_FRAME_SIZE);
- if (s->dca_buffer_size == -1) {
- av_log(avctx, AV_LOG_ERROR, "Not a DCA frame\n");
- return -1;
- }
-
- init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
- if (dca_parse_frame_header(s) < 0) {
- //seems like the frame is corrupt, try with the next one
- return buf_size;
- }
- //set AVCodec values with parsed data
- avctx->sample_rate = s->sample_rate;
- avctx->bit_rate = s->bit_rate;
-
- channels = s->prim_channels + !!s->lfe;
- if(avctx->channels == 0) {
- avctx->channels = channels;
- } else if(channels < avctx->channels) {
- av_log(avctx, AV_LOG_WARNING, "DTS source channels are less than "
- "specified: output to %d channels.\n", channels);
- avctx->channels = channels;
- }
- if(avctx->channels == 2) {
- s->output = DCA_STEREO;
- } else if(avctx->channels != channels) {
- av_log(avctx, AV_LOG_ERROR, "Cannot downmix DTS to %d channels.\n",
- avctx->channels);
- return -1;
- }
-
- channels = avctx->channels;
- if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
- return -1;
- *data_size = 0;
- for (i = 0; i < (s->sample_blocks / 8); i++) {
- dca_decode_block(s);
- s->dsp.float_to_int16(s->tsamples, s->samples, 256 * channels);
- /* interleave samples */
- for (j = 0; j < 256; j++) {
- for (k = 0; k < channels; k++)
- samples[k] = s->tsamples[j + k * 256];
- samples += channels;
- }
- *data_size += 256 * sizeof(int16_t) * channels;
- }
-
- return buf_size;
-}
-
-
-
-/**
- * Build the cosine modulation tables for the QMF
- *
- * @param s pointer to the DCAContext
- */
-
-static void pre_calc_cosmod(DCAContext * s)
-{
- int i, j, k;
- static int cosmod_inited = 0;
-
- if(cosmod_inited) return;
- for (j = 0, k = 0; k < 16; k++)
- for (i = 0; i < 16; i++)
- 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++)
- cos_mod[j++] = cos((i) * (2 * k + 1) * M_PI / 32);
-
- for (k = 0; k < 16; k++)
- cos_mod[j++] = 0.25 / (2 * cos((2 * k + 1) * M_PI / 128));
-
- for (k = 0; k < 16; k++)
- cos_mod[j++] = -0.25 / (2.0 * sin((2 * k + 1) * M_PI / 128));
-
- cosmod_inited = 1;
-}
-
-
-/**
- * DCA initialization
- *
- * @param avctx pointer to the AVCodecContext
- */
-
-static int dca_decode_init(AVCodecContext * avctx)
-{
- DCAContext *s = avctx->priv_data;
-
- s->avctx = avctx;
- dca_init_vlcs();
- pre_calc_cosmod(s);
-
- dsputil_init(&s->dsp, avctx);
- return 0;
-}
-
-
-AVCodec dca_decoder = {
- .name = "dca",
- .type = CODEC_TYPE_AUDIO,
- .id = CODEC_ID_DTS,
- .priv_data_size = sizeof(DCAContext),
- .init = dca_decode_init,
- .decode = dca_decode_frame,
-};
-
-#ifdef CONFIG_DCA_PARSER
-
-typedef struct DCAParseContext {
- ParseContext pc;
- uint32_t lastmarker;
-} DCAParseContext;
-
-#define IS_MARKER(state, i, buf, buf_size) \
- ((state == DCA_MARKER_14B_LE && (i < buf_size-2) && (buf[i+1] & 0xF0) == 0xF0 && buf[i+2] == 0x07) \
- || (state == DCA_MARKER_14B_BE && (i < buf_size-2) && buf[i+1] == 0x07 && (buf[i+2] & 0xF0) == 0xF0) \
- || state == DCA_MARKER_RAW_LE || state == DCA_MARKER_RAW_BE)
-
-/**
- * finds the end of the current frame in the bitstream.
- * @return the position of the first byte of the next frame, or -1
- */
-static int dca_find_frame_end(DCAParseContext * pc1, const uint8_t * buf,
- int buf_size)
-{
- int start_found, i;
- uint32_t state;
- ParseContext *pc = &pc1->pc;
-
- start_found = pc->frame_start_found;
- state = pc->state;
-
- i = 0;
- if (!start_found) {
- for (i = 0; i < buf_size; i++) {
- state = (state << 8) | buf[i];
- if (IS_MARKER(state, i, buf, buf_size)) {
- if (pc1->lastmarker && state == pc1->lastmarker) {
- start_found = 1;
- break;
- } else if (!pc1->lastmarker) {
- start_found = 1;
- pc1->lastmarker = state;
- break;
- }
- }
- }
- }
- if (start_found) {
- for (; i < buf_size; i++) {
- state = (state << 8) | buf[i];
- if (state == pc1->lastmarker && IS_MARKER(state, i, buf, buf_size)) {
- pc->frame_start_found = 0;
- pc->state = -1;
- return i - 3;
- }
- }
- }
- pc->frame_start_found = start_found;
- pc->state = state;
- return END_NOT_FOUND;
-}
-
-static int dca_parse_init(AVCodecParserContext * s)
-{
- DCAParseContext *pc1 = s->priv_data;
-
- pc1->lastmarker = 0;
- return 0;
-}
-
-static int dca_parse(AVCodecParserContext * s,
- AVCodecContext * avctx,
- uint8_t ** poutbuf, int *poutbuf_size,
- const uint8_t * buf, int buf_size)
-{
- DCAParseContext *pc1 = s->priv_data;
- ParseContext *pc = &pc1->pc;
- int next;
-
- if (s->flags & PARSER_FLAG_COMPLETE_FRAMES) {
- next = buf_size;
- } else {
- next = dca_find_frame_end(pc1, buf, buf_size);
-
- if (ff_combine_frame(pc, next, (uint8_t **) & buf, &buf_size) < 0) {
- *poutbuf = NULL;
- *poutbuf_size = 0;
- return buf_size;
- }
- }
- *poutbuf = (uint8_t *) buf;
- *poutbuf_size = buf_size;
- return next;
-}
-
-AVCodecParser dca_parser = {
- {CODEC_ID_DTS},
- sizeof(DCAParseContext),
- dca_parse_init,
- dca_parse,
- ff_parse_close,
-};
-#endif /* CONFIG_DCA_PARSER */
Copied: trunk/libavcodec/dca_parser.c (from r8880, /trunk/libavcodec/dca.c)
==============================================================================
--- /trunk/libavcodec/dca.c (original)
+++ trunk/libavcodec/dca_parser.c Fri May 4 01:50:24 2007
@@ -1,5 +1,5 @@
/*
- * DCA compatible decoder
+ * DCA parser
* Copyright (C) 2004 Gildas Bazin
* Copyright (C) 2004 Benjamin Zores
* Copyright (C) 2006 Benjamin Larsson
@@ -23,1240 +23,12 @@
*/
/**
- * @file dca.c
+ * @file dca_parser.c
*/
-#include <math.h>
-#include <stddef.h>
-#include <stdio.h>
-
#include "avcodec.h"
-#include "dsputil.h"
-#include "bitstream.h"
-#include "dcadata.h"
-#include "dcahuff.h"
#include "parser.h"
-
-/** DCA syncwords, also used for bitstream type detection */
-//@{
-#define DCA_MARKER_RAW_BE 0x7FFE8001
-#define DCA_MARKER_RAW_LE 0xFE7F0180
-#define DCA_MARKER_14B_BE 0x1FFFE800
-#define DCA_MARKER_14B_LE 0xFF1F00E8
-//@}
-
-//#define TRACE
-
-#define DCA_PRIM_CHANNELS_MAX (5)
-#define DCA_SUBBANDS (32)
-#define DCA_ABITS_MAX (32) /* Should be 28 */
-#define DCA_SUBSUBFAMES_MAX (4)
-#define DCA_LFE_MAX (3)
-
-enum DCAMode {
- DCA_MONO = 0,
- DCA_CHANNEL,
- DCA_STEREO,
- DCA_STEREO_SUMDIFF,
- DCA_STEREO_TOTAL,
- DCA_3F,
- DCA_2F1R,
- DCA_3F1R,
- DCA_2F2R,
- DCA_3F2R,
- DCA_4F2R
-};
-
-#define DCA_DOLBY 101 /* FIXME */
-
-#define DCA_CHANNEL_BITS 6
-#define DCA_CHANNEL_MASK 0x3F
-
-#define DCA_LFE 0x80
-
-#define HEADER_SIZE 14
-#define CONVERT_BIAS 384
-
-#define DCA_MAX_FRAME_SIZE 16383
-
-/** Bit allocation */
-typedef struct {
- int offset; ///< code values offset
- int maxbits[8]; ///< max bits in VLC
- int wrap; ///< wrap for get_vlc2()
- VLC vlc[8]; ///< actual codes
-} BitAlloc;
-
-static BitAlloc dca_bitalloc_index; ///< indexes for samples VLC select
-static BitAlloc dca_tmode; ///< transition mode VLCs
-static BitAlloc dca_scalefactor; ///< scalefactor VLCs
-static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs
-
-/** Pre-calculated cosine modulation coefs for the QMF */
-static float cos_mod[544];
-
-static int av_always_inline get_bitalloc(GetBitContext *gb, BitAlloc *ba, int idx)
-{
- return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + ba->offset;
-}
-
-typedef struct {
- AVCodecContext *avctx;
- /* Frame header */
- int frame_type; ///< type of the current frame
- int samples_deficit; ///< deficit sample count
- int crc_present; ///< crc is present in the bitstream
- int sample_blocks; ///< number of PCM sample blocks
- int frame_size; ///< primary frame byte size
- int amode; ///< audio channels arrangement
- int sample_rate; ///< audio sampling rate
- int bit_rate; ///< transmission bit rate
-
- int downmix; ///< embedded downmix enabled
- int dynrange; ///< embedded dynamic range flag
- int timestamp; ///< embedded time stamp flag
- int aux_data; ///< auxiliary data flag
- int hdcd; ///< source material is mastered in HDCD
- int ext_descr; ///< extension audio descriptor flag
- int ext_coding; ///< extended coding flag
- int aspf; ///< audio sync word insertion flag
- int lfe; ///< low frequency effects flag
- int predictor_history; ///< predictor history flag
- int header_crc; ///< header crc check bytes
- int multirate_inter; ///< multirate interpolator switch
- int version; ///< encoder software revision
- int copy_history; ///< copy history
- int source_pcm_res; ///< source pcm resolution
- int front_sum; ///< front sum/difference flag
- int surround_sum; ///< surround sum/difference flag
- int dialog_norm; ///< dialog normalisation parameter
-
- /* Primary audio coding header */
- int subframes; ///< number of subframes
- int prim_channels; ///< number of primary audio channels
- int subband_activity[DCA_PRIM_CHANNELS_MAX]; ///< subband activity count
- int vq_start_subband[DCA_PRIM_CHANNELS_MAX]; ///< high frequency vq start subband
- int joint_intensity[DCA_PRIM_CHANNELS_MAX]; ///< joint intensity coding index
- int transient_huffman[DCA_PRIM_CHANNELS_MAX]; ///< transient mode code book
- int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book
- int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX]; ///< bit allocation quantizer select
- int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select
- float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment
-
- /* Primary audio coding side information */
- int subsubframes; ///< number of subsubframes
- int partial_samples; ///< 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 transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< transition mode (transients)
- int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2]; ///< scale factors (2 if transient)
- int joint_huff[DCA_PRIM_CHANNELS_MAX]; ///< joint subband scale factors codebook
- int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors
- int downmix_coef[DCA_PRIM_CHANNELS_MAX][2]; ///< stereo downmix coefficients
- int dynrange_coef; ///< dynamic range coefficient
-
- int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< VQ encoded high frequency subbands
-
- float lfe_data[2 * DCA_SUBSUBFAMES_MAX * DCA_LFE_MAX *
- 2 /*history */ ]; ///< Low frequency effect data
- int lfe_scale_factor;
-
- /* Subband samples history (for ADPCM) */
- float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
- float subband_fir_hist[DCA_PRIM_CHANNELS_MAX][512];
- float subband_fir_noidea[DCA_PRIM_CHANNELS_MAX][64];
-
- int output; ///< type of output
- int bias; ///< output bias
-
- DECLARE_ALIGNED_16(float, samples[1536]); /* 6 * 256 = 1536, might only need 5 */
- DECLARE_ALIGNED_16(int16_t, tsamples[1536]);
-
- uint8_t dca_buffer[DCA_MAX_FRAME_SIZE];
- int dca_buffer_size; ///< how much data is in the dca_buffer
-
- GetBitContext gb;
- /* Current position in DCA frame */
- int current_subframe;
- int current_subsubframe;
-
- int debug_flag; ///< used for suppressing repeated error messages output
- DSPContext dsp;
-} DCAContext;
-
-static void dca_init_vlcs(void)
-{
- static int vlcs_inited = 0;
- int i, j;
-
- if (vlcs_inited)
- return;
-
- dca_bitalloc_index.offset = 1;
- dca_bitalloc_index.wrap = 1;
- for (i = 0; i < 5; i++)
- init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12,
- bitalloc_12_bits[i], 1, 1,
- bitalloc_12_codes[i], 2, 2, 1);
- dca_scalefactor.offset = -64;
- dca_scalefactor.wrap = 2;
- for (i = 0; i < 5; i++)
- init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,
- scales_bits[i], 1, 1,
- scales_codes[i], 2, 2, 1);
- dca_tmode.offset = 0;
- dca_tmode.wrap = 1;
- for (i = 0; i < 4; i++)
- init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
- tmode_bits[i], 1, 1,
- tmode_codes[i], 2, 2, 1);
-
- 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);
- 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, 1);
- }
- vlcs_inited = 1;
-}
-
-static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
-{
- while(len--)
- *dst++ = get_bits(gb, bits);
-}
-
-static int dca_parse_frame_header(DCAContext * s)
-{
- 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 };
-
- s->bias = CONVERT_BIAS;
-
- init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
-
- /* Sync code */
- get_bits(&s->gb, 32);
-
- /* Frame header */
- s->frame_type = get_bits(&s->gb, 1);
- s->samples_deficit = get_bits(&s->gb, 5) + 1;
- s->crc_present = 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;
- s->amode = get_bits(&s->gb, 6);
- s->sample_rate = dca_sample_rates[get_bits(&s->gb, 4)];
- if (!s->sample_rate)
- return -1;
- s->bit_rate = dca_bit_rates[get_bits(&s->gb, 5)];
- if (!s->bit_rate)
- return -1;
-
- s->downmix = get_bits(&s->gb, 1);
- s->dynrange = get_bits(&s->gb, 1);
- s->timestamp = get_bits(&s->gb, 1);
- s->aux_data = get_bits(&s->gb, 1);
- s->hdcd = get_bits(&s->gb, 1);
- s->ext_descr = get_bits(&s->gb, 3);
- s->ext_coding = get_bits(&s->gb, 1);
- s->aspf = get_bits(&s->gb, 1);
- s->lfe = get_bits(&s->gb, 2);
- s->predictor_history = get_bits(&s->gb, 1);
-
- /* TODO: check CRC */
- if (s->crc_present)
- s->header_crc = get_bits(&s->gb, 16);
-
- s->multirate_inter = get_bits(&s->gb, 1);
- s->version = get_bits(&s->gb, 4);
- s->copy_history = get_bits(&s->gb, 2);
- s->source_pcm_res = get_bits(&s->gb, 3);
- s->front_sum = get_bits(&s->gb, 1);
- s->surround_sum = get_bits(&s->gb, 1);
- s->dialog_norm = get_bits(&s->gb, 4);
-
- /* FIXME: channels mixing levels */
- s->output = s->amode;
- if(s->lfe) s->output |= DCA_LFE;
-
-#ifdef TRACE
- av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type);
- av_log(s->avctx, AV_LOG_DEBUG, "samples deficit: %i\n", s->samples_deficit);
- av_log(s->avctx, AV_LOG_DEBUG, "crc present: %i\n", s->crc_present);
- av_log(s->avctx, AV_LOG_DEBUG, "sample blocks: %i (%i samples)\n",
- s->sample_blocks, s->sample_blocks * 32);
- av_log(s->avctx, AV_LOG_DEBUG, "frame size: %i bytes\n", s->frame_size);
- av_log(s->avctx, AV_LOG_DEBUG, "amode: %i (%i channels)\n",
- s->amode, dca_channels[s->amode]);
- av_log(s->avctx, AV_LOG_DEBUG, "sample rate: %i (%i Hz)\n",
- s->sample_rate, dca_sample_rates[s->sample_rate]);
- av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i (%i bits/s)\n",
- s->bit_rate, dca_bit_rates[s->bit_rate]);
- av_log(s->avctx, AV_LOG_DEBUG, "downmix: %i\n", s->downmix);
- av_log(s->avctx, AV_LOG_DEBUG, "dynrange: %i\n", s->dynrange);
- av_log(s->avctx, AV_LOG_DEBUG, "timestamp: %i\n", s->timestamp);
- av_log(s->avctx, AV_LOG_DEBUG, "aux_data: %i\n", s->aux_data);
- av_log(s->avctx, AV_LOG_DEBUG, "hdcd: %i\n", s->hdcd);
- av_log(s->avctx, AV_LOG_DEBUG, "ext descr: %i\n", s->ext_descr);
- av_log(s->avctx, AV_LOG_DEBUG, "ext coding: %i\n", s->ext_coding);
- av_log(s->avctx, AV_LOG_DEBUG, "aspf: %i\n", s->aspf);
- av_log(s->avctx, AV_LOG_DEBUG, "lfe: %i\n", s->lfe);
- av_log(s->avctx, AV_LOG_DEBUG, "predictor history: %i\n",
- s->predictor_history);
- av_log(s->avctx, AV_LOG_DEBUG, "header crc: %i\n", s->header_crc);
- av_log(s->avctx, AV_LOG_DEBUG, "multirate inter: %i\n",
- s->multirate_inter);
- av_log(s->avctx, AV_LOG_DEBUG, "version number: %i\n", s->version);
- av_log(s->avctx, AV_LOG_DEBUG, "copy history: %i\n", s->copy_history);
- av_log(s->avctx, AV_LOG_DEBUG,
- "source pcm resolution: %i (%i bits/sample)\n",
- s->source_pcm_res, dca_bits_per_sample[s->source_pcm_res]);
- av_log(s->avctx, AV_LOG_DEBUG, "front sum: %i\n", s->front_sum);
- av_log(s->avctx, AV_LOG_DEBUG, "surround sum: %i\n", s->surround_sum);
- av_log(s->avctx, AV_LOG_DEBUG, "dialog norm: %i\n", s->dialog_norm);
- av_log(s->avctx, AV_LOG_DEBUG, "\n");
-#endif
-
- /* Primary audio coding header */
- s->subframes = get_bits(&s->gb, 4) + 1;
- s->prim_channels = get_bits(&s->gb, 3) + 1;
-
-
- for (i = 0; i < s->prim_channels; i++) {
- s->subband_activity[i] = get_bits(&s->gb, 5) + 2;
- if (s->subband_activity[i] > DCA_SUBBANDS)
- s->subband_activity[i] = DCA_SUBBANDS;
- }
- for (i = 0; i < s->prim_channels; i++) {
- s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
- if (s->vq_start_subband[i] > DCA_SUBBANDS)
- s->vq_start_subband[i] = DCA_SUBBANDS;
- }
- get_array(&s->gb, s->joint_intensity, s->prim_channels, 3);
- get_array(&s->gb, s->transient_huffman, s->prim_channels, 2);
- get_array(&s->gb, s->scalefactor_huffman, s->prim_channels, 3);
- get_array(&s->gb, s->bitalloc_huffman, s->prim_channels, 3);
-
- /* Get codebooks quantization indexes */
- memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
- for (j = 1; j < 11; j++)
- for (i = 0; i < s->prim_channels; i++)
- s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
-
- /* Get scale factor adjustment */
- for (j = 0; j < 11; j++)
- for (i = 0; i < s->prim_channels; i++)
- s->scalefactor_adj[i][j] = 1;
-
- for (j = 1; j < 11; j++)
- for (i = 0; i < s->prim_channels; i++)
- if (s->quant_index_huffman[i][j] < thr[j])
- s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
-
- if (s->crc_present) {
- /* Audio header CRC check */
- get_bits(&s->gb, 16);
- }
-
- 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 = 0; 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, "\n");
- av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:");
- for (j = 0; j < 11; j++)
- av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]);
- av_log(s->avctx, AV_LOG_DEBUG, "\n");
- }
-#endif
-
- return 0;
-}
-
-
-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;
-}
-
-static int dca_subframe_header(DCAContext * s)
-{
- /* Primary audio coding side information */
- int j, k;
-
- s->subsubframes = get_bits(&s->gb, 2) + 1;
- s->partial_samples = get_bits(&s->gb, 3);
- for (j = 0; j < s->prim_channels; j++) {
- for (k = 0; k < s->subband_activity[j]; k++)
- s->prediction_mode[j][k] = get_bits(&s->gb, 1);
- }
-
- /* Get prediction codebook */
- for (j = 0; j < s->prim_channels; j++) {
- for (k = 0; k < s->subband_activity[j]; k++) {
- if (s->prediction_mode[j][k] > 0) {
- /* (Prediction coefficient VQ address) */
- s->prediction_vq[j][k] = get_bits(&s->gb, 12);
- }
- }
- }
-
- /* Bit allocation index */
- for (j = 0; j < s->prim_channels; j++) {
- for (k = 0; k < s->vq_start_subband[j]; k++) {
- if (s->bitalloc_huffman[j] == 6)
- s->bitalloc[j][k] = get_bits(&s->gb, 5);
- else if (s->bitalloc_huffman[j] == 5)
- s->bitalloc[j][k] = get_bits(&s->gb, 4);
- 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;
- }
- }
- }
-
- /* Transition mode */
- for (j = 0; j < s->prim_channels; j++) {
- for (k = 0; k < s->subband_activity[j]; k++) {
- s->transition_mode[j][k] = 0;
- if (s->subsubframes > 1 &&
- k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) {
- s->transition_mode[j][k] =
- get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]);
- }
- }
- }
-
- for (j = 0; j < s->prim_channels; j++) {
- 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);
-
- if (s->scalefactor_huffman[j] == 6)
- scale_table = (uint32_t *) scale_factor_quant7;
- else
- scale_table = (uint32_t *) scale_factor_quant6;
-
- /* When huffman coded, only the difference is encoded */
- scale_sum = 0;
-
- for (k = 0; k < s->subband_activity[j]; k++) {
- if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) {
- scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum);
- s->scale_factor[j][k][0] = scale_table[scale_sum];
- }
-
- if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) {
- /* Get second scale factor */
- scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum);
- s->scale_factor[j][k][1] = scale_table[scale_sum];
- }
- }
- }
-
- /* Joint subband scale factor codebook select */
- for (j = 0; j < s->prim_channels; j++) {
- /* Transmitted only if joint subband coding enabled */
- if (s->joint_intensity[j] > 0)
- s->joint_huff[j] = get_bits(&s->gb, 3);
- }
-
- /* Scale factors for joint subband coding */
- for (j = 0; j < s->prim_channels; j++) {
- int source_channel;
-
- /* Transmitted only if joint subband coding enabled */
- if (s->joint_intensity[j] > 0) {
- int scale = 0;
- source_channel = s->joint_intensity[j] - 1;
-
- /* When huffman coded, only the difference is encoded
- * (is this valid as well for joint scales ???) */
-
- for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) {
- scale = get_scale(&s->gb, s->joint_huff[j], 0);
- scale += 64; /* bias */
- s->joint_scale_factor[j][k] = scale; /*joint_scale_table[scale]; */
- }
-
- if (!s->debug_flag & 0x02) {
- av_log(s->avctx, AV_LOG_DEBUG,
- "Joint stereo coding not supported\n");
- s->debug_flag |= 0x02;
- }
- }
- }
-
- /* Stereo downmix coefficients */
- if (s->prim_channels > 2) {
- if(s->downmix) {
- for (j = 0; j < s->prim_channels; j++) {
- s->downmix_coef[j][0] = get_bits(&s->gb, 7);
- s->downmix_coef[j][1] = get_bits(&s->gb, 7);
- }
- } else {
- int am = s->amode & DCA_CHANNEL_MASK;
- for (j = 0; j < s->prim_channels; j++) {
- s->downmix_coef[j][0] = dca_default_coeffs[am][j][0];
- s->downmix_coef[j][1] = dca_default_coeffs[am][j][1];
- }
- }
- }
-
- /* Dynamic range coefficient */
- if (s->dynrange)
- s->dynrange_coef = get_bits(&s->gb, 8);
-
- /* Side information CRC check word */
- if (s->crc_present) {
- get_bits(&s->gb, 16);
- }
-
- /*
- * Primary audio data arrays
- */
-
- /* VQ encoded high frequency subbands */
- for (j = 0; j < s->prim_channels; j++)
- for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
- /* 1 vector -> 32 samples */
- s->high_freq_vq[j][k] = get_bits(&s->gb, 10);
-
- /* Low frequency effect data */
- if (s->lfe) {
- /* LFE samples */
- int lfe_samples = 2 * s->lfe * s->subsubframes;
- float lfe_scale;
-
- for (j = lfe_samples; j < lfe_samples * 2; j++) {
- /* Signed 8 bits int */
- s->lfe_data[j] = get_sbits(&s->gb, 8);
- }
-
- /* Scale factor index */
- s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 8)];
-
- /* Quantization step size * scale factor */
- lfe_scale = 0.035 * s->lfe_scale_factor;
-
- for (j = lfe_samples; j < lfe_samples * 2; j++)
- s->lfe_data[j] *= lfe_scale;
- }
-
-#ifdef TRACE
- av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n", s->subsubframes);
- av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n",
- s->partial_samples);
- for (j = 0; j < s->prim_channels; j++) {
- av_log(s->avctx, AV_LOG_DEBUG, "prediction mode:");
- for (k = 0; k < s->subband_activity[j]; k++)
- av_log(s->avctx, AV_LOG_DEBUG, " %i", s->prediction_mode[j][k]);
- av_log(s->avctx, AV_LOG_DEBUG, "\n");
- }
- for (j = 0; 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);
- }
- for (j = 0; j < s->prim_channels; j++) {
- av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index: ");
- for (k = 0; k < s->vq_start_subband[j]; k++)
- av_log(s->avctx, AV_LOG_DEBUG, "%2.2i ", s->bitalloc[j][k]);
- av_log(s->avctx, AV_LOG_DEBUG, "\n");
- }
- for (j = 0; j < s->prim_channels; j++) {
- av_log(s->avctx, AV_LOG_DEBUG, "Transition mode:");
- for (k = 0; k < s->subband_activity[j]; k++)
- av_log(s->avctx, AV_LOG_DEBUG, " %i", s->transition_mode[j][k]);
- av_log(s->avctx, AV_LOG_DEBUG, "\n");
- }
- for (j = 0; j < s->prim_channels; j++) {
- av_log(s->avctx, AV_LOG_DEBUG, "Scale factor:");
- for (k = 0; k < s->subband_activity[j]; k++) {
- if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0)
- av_log(s->avctx, AV_LOG_DEBUG, " %i", s->scale_factor[j][k][0]);
- if (k < s->vq_start_subband[j] && s->transition_mode[j][k])
- av_log(s->avctx, AV_LOG_DEBUG, " %i(t)", s->scale_factor[j][k][1]);
- }
- av_log(s->avctx, AV_LOG_DEBUG, "\n");
- }
- for (j = 0; j < s->prim_channels; j++) {
- if (s->joint_intensity[j] > 0) {
- av_log(s->avctx, AV_LOG_DEBUG, "Joint scale factor index:\n");
- for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++)
- av_log(s->avctx, AV_LOG_DEBUG, " %i", s->joint_scale_factor[j][k]);
- av_log(s->avctx, AV_LOG_DEBUG, "\n");
- }
- }
- if (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, "\n");
- }
- for (j = 0; j < s->prim_channels; j++)
- for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
- av_log(s->avctx, AV_LOG_DEBUG, "VQ index: %i\n", s->high_freq_vq[j][k]);
- if(s->lfe){
- av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n");
- for (j = lfe_samples; j < lfe_samples * 2; j++)
- av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]);
- av_log(s->avctx, AV_LOG_DEBUG, "\n");
- }
-#endif
-
- return 0;
-}
-
-static void qmf_32_subbands(DCAContext * s, int chans,
- float samples_in[32][8], float *samples_out,
- float scale, float bias)
-{
- float *prCoeff;
- int i, j, k;
- float praXin[33], *raXin = &praXin[1];
-
- float *subband_fir_hist = s->subband_fir_hist[chans];
- float *subband_fir_hist2 = s->subband_fir_noidea[chans];
-
- int chindex = 0, subindex;
-
- praXin[0] = 0.0;
-
- /* Select filter */
- if (!s->multirate_inter) /* Non-perfect reconstruction */
- prCoeff = (float *) fir_32bands_nonperfect;
- else /* Perfect reconstruction */
- prCoeff = (float *) fir_32bands_perfect;
-
- /* Reconstructed channel sample index */
- for (subindex = 0; subindex < 8; subindex++) {
- float t1, t2, sum[16], diff[16];
-
- /* Load in one sample from each subband and clear inactive subbands */
- for (i = 0; i < s->subband_activity[chans]; i++)
- raXin[i] = samples_in[i][subindex];
- for (; i < 32; i++)
- raXin[i] = 0.0;
-
- /* Multiply by cosine modulation coefficients and
- * create temporary arrays SUM and DIFF */
- for (j = 0, k = 0; k < 16; k++) {
- t1 = 0.0;
- t2 = 0.0;
- for (i = 0; i < 16; i++, j++){
- t1 += (raXin[2 * i] + raXin[2 * i + 1]) * cos_mod[j];
- t2 += (raXin[2 * i] + raXin[2 * i - 1]) * cos_mod[j + 256];
- }
- sum[k] = t1 + t2;
- diff[k] = t1 - t2;
- }
-
- j = 512;
- /* Store history */
- for (k = 0; k < 16; k++)
- subband_fir_hist[k] = cos_mod[j++] * sum[k];
- for (k = 0; k < 16; k++)
- subband_fir_hist[32-k-1] = cos_mod[j++] * diff[k];
-
- /* Multiply by filter coefficients */
- for (k = 31, i = 0; i < 32; i++, k--)
- for (j = 0; j < 512; j += 64){
- subband_fir_hist2[i] += prCoeff[i+j] * ( subband_fir_hist[i+j] - subband_fir_hist[j+k]);
- subband_fir_hist2[i+32] += prCoeff[i+j+32]*(-subband_fir_hist[i+j] - subband_fir_hist[j+k]);
- }
-
- /* Create 32 PCM output samples */
- for (i = 0; i < 32; i++)
- samples_out[chindex++] = subband_fir_hist2[i] * scale + bias;
-
- /* Update working arrays */
- memmove(&subband_fir_hist[32], &subband_fir_hist[0], (512 - 32) * sizeof(float));
- memmove(&subband_fir_hist2[0], &subband_fir_hist2[32], 32 * sizeof(float));
- memset(&subband_fir_hist2[32], 0, 32 * sizeof(float));
- }
-}
-
-static void lfe_interpolation_fir(int decimation_select,
- int num_deci_sample, float *samples_in,
- float *samples_out, float scale,
- float bias)
-{
- /* samples_in: An array holding decimated samples.
- * Samples in current subframe starts from samples_in[0],
- * while samples_in[-1], samples_in[-2], ..., stores samples
- * from last subframe as history.
- *
- * samples_out: An array holding interpolated samples
- */
-
- int decifactor, k, j;
- const float *prCoeff;
-
- int interp_index = 0; /* Index to the interpolated samples */
- int deciindex;
-
- /* Select decimation filter */
- if (decimation_select == 1) {
- decifactor = 128;
- prCoeff = lfe_fir_128;
- } else {
- decifactor = 64;
- prCoeff = lfe_fir_64;
- }
- /* Interpolation */
- for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
- /* One decimated sample generates decifactor interpolated ones */
- for (k = 0; k < decifactor; k++) {
- float rTmp = 0.0;
- //FIXME the coeffs are symetric, fix that
- for (j = 0; j < 512 / decifactor; j++)
- rTmp += samples_in[deciindex - j] * prCoeff[k + j * decifactor];
- samples_out[interp_index++] = rTmp / scale + bias;
- }
- }
-}
-
-/* downmixing routines */
-#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]; \
- samples[i] = t * coef[0][0] + samples[i+256] * coef[1][0] + samples[i+512] * coef[2][0]; \
- samples[i+256] = t * coef[0][1] + samples[i+256] * coef[1][1] + samples[i+512] * 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])
-{
- int i;
- float t;
- float coef[DCA_PRIM_CHANNELS_MAX][2];
-
- 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]];
- }
-
- switch (srcfmt) {
- case DCA_MONO:
- case DCA_CHANNEL:
- case DCA_STEREO_TOTAL:
- case DCA_STEREO_SUMDIFF:
- case DCA_4F2R:
- av_log(NULL, 0, "Not implemented!\n");
- break;
- case DCA_STEREO:
- break;
- case DCA_3F:
- DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),);
- break;
- case DCA_2F1R:
- DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + 512, 2, coef),);
- break;
- case DCA_3F1R:
- DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
- MIX_REAR1(samples, i + 768, 3, coef));
- break;
- case DCA_2F2R:
- DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + 512, i + 768, 2, coef),);
- break;
- case DCA_3F2R:
- DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
- MIX_REAR2(samples, i + 768, i + 1024, 3, coef));
- break;
- }
-}
-
-
-/* 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)
-{
- int i;
- int offset = (levels - 1) >> 1;
-
- for (i = 0; i < 4; i++) {
- values[i] = (code % levels) - offset;
- code /= levels;
- }
-
- if (code == 0)
- return 0;
- else {
- av_log(NULL, AV_LOG_ERROR, "ERROR: block code look-up failed\n");
- return -1;
- }
-}
-
-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 dca_subsubframe(DCAContext * s)
-{
- int k, l;
- int subsubframe = s->current_subsubframe;
-
- float *quant_step_table;
-
- /* FIXME */
- float subband_samples[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
-
- /*
- * Audio data
- */
-
- /* Select quantization step size table */
- if (s->bit_rate == 0x1f)
- quant_step_table = (float *) lossless_quant_d;
- else
- quant_step_table = (float *) lossy_quant_d;
-
- for (k = 0; k < s->prim_channels; k++) {
- for (l = 0; l < s->vq_start_subband[k]; l++) {
- int m;
-
- /* Select the mid-tread linear quantizer */
- int abits = s->bitalloc[k][l];
-
- float quant_step_size = quant_step_table[abits];
- float rscale;
-
- /*
- * Determine quantization index code book and its type
- */
-
- /* Select quantization index code book */
- int sel = s->quant_index_huffman[k][abits];
-
- /*
- * Extract bits from the bit stream
- */
- if(!abits){
- memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));
- }else if(abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){
- if(abits <= 7){
- /* Block code */
- int block_code1, block_code2, size, levels;
- int block[8];
-
- 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]);
- for (m = 0; m < 8; m++)
- subband_samples[k][l][m] = block[m];
- }else{
- /* no coding */
- for (m = 0; m < 8; m++)
- subband_samples[k][l][m] = get_sbits(&s->gb, abits - 3);
- }
- }else{
- /* Huffman coded */
- for (m = 0; m < 8; m++)
- subband_samples[k][l][m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);
- }
-
- /* Deal with transients */
- if (s->transition_mode[k][l] &&
- subsubframe >= s->transition_mode[k][l])
- rscale = quant_step_size * s->scale_factor[k][l][1];
- else
- rscale = quant_step_size * s->scale_factor[k][l][0];
-
- rscale *= s->scalefactor_adj[k][sel];
-
- for (m = 0; m < 8; m++)
- subband_samples[k][l][m] *= rscale;
-
- /*
- * Inverse ADPCM if in prediction mode
- */
- if (s->prediction_mode[k][l]) {
- int n;
- for (m = 0; m < 8; m++) {
- for (n = 1; n <= 4; n++)
- if (m >= n)
- subband_samples[k][l][m] +=
- (adpcm_vb[s->prediction_vq[k][l]][n - 1] *
- subband_samples[k][l][m - n] / 8192);
- 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);
- }
- }
- }
-
- /*
- * Decode VQ encoded high frequencies
- */
- 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;
-
- if (!s->debug_flag & 0x01) {
- 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;
- }
- }
- }
-
- /* Check for DSYNC after subsubframe */
- if (s->aspf || subsubframe == s->subsubframes - 1) {
- if (0xFFFF == get_bits(&s->gb, 16)) { /* 0xFFFF */
-#ifdef TRACE
- av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");
-#endif
- } else {
- av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");
- }
- }
-
- /* Backup predictor history for adpcm */
- for (k = 0; 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]));
-
- /* 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 * k],
- 2.0 / 3 /*pcm_to_double[s->source_pcm_res] */ ,
- 0 /*s->bias */ );
- }
-
- /* Down mixing */
-
- if (s->prim_channels > dca_channels[s->output & DCA_CHANNEL_MASK]) {
- dca_downmix(s->samples, s->amode, s->downmix_coef);
- }
-
- /* Generate LFE samples for this subsubframe FIXME!!! */
- if (s->output & DCA_LFE) {
- int lfe_samples = 2 * s->lfe * s->subsubframes;
- int i_channels = dca_channels[s->output & DCA_CHANNEL_MASK];
-
- lfe_interpolation_fir(s->lfe, 2 * s->lfe,
- s->lfe_data + lfe_samples +
- 2 * s->lfe * subsubframe,
- &s->samples[256 * i_channels],
- 8388608.0, s->bias);
- /* Outputs 20bits pcm samples */
- }
-
- return 0;
-}
-
-
-static int dca_subframe_footer(DCAContext * s)
-{
- int aux_data_count = 0, i;
- int lfe_samples;
-
- /*
- * Unpack optional information
- */
-
- if (s->timestamp)
- get_bits(&s->gb, 32);
-
- if (s->aux_data)
- aux_data_count = get_bits(&s->gb, 6);
-
- for (i = 0; i < aux_data_count; i++)
- get_bits(&s->gb, 8);
-
- if (s->crc_present && (s->downmix || s->dynrange))
- get_bits(&s->gb, 16);
-
- lfe_samples = 2 * s->lfe * s->subsubframes;
- for (i = 0; i < lfe_samples; i++) {
- s->lfe_data[i] = s->lfe_data[i + lfe_samples];
- }
-
- return 0;
-}
-
-/**
- * Decode a dca frame block
- *
- * @param s pointer to the DCAContext
- */
-
-static int dca_decode_block(DCAContext * s)
-{
-
- /* 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;
- }
-
- if (!s->current_subsubframe) {
-#ifdef TRACE
- av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");
-#endif
- /* Read subframe header */
- if (dca_subframe_header(s))
- return -1;
- }
-
- /* Read subsubframe */
-#ifdef TRACE
- av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");
-#endif
- if (dca_subsubframe(s))
- return -1;
-
- /* Update state */
- s->current_subsubframe++;
- if (s->current_subsubframe >= s->subsubframes) {
- s->current_subsubframe = 0;
- s->current_subframe++;
- }
- if (s->current_subframe >= s->subframes) {
-#ifdef TRACE
- av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");
-#endif
- /* Read subframe footer */
- if (dca_subframe_footer(s))
- return -1;
- }
-
- return 0;
-}
-
-/**
- * Convert bitstream to one representation based on sync marker
- */
-static int dca_convert_bitstream(uint8_t * src, int src_size, uint8_t * dst,
- int max_size)
-{
- uint32_t mrk;
- int i, tmp;
- uint16_t *ssrc = (uint16_t *) src, *sdst = (uint16_t *) dst;
- PutBitContext pb;
-
- if((unsigned)src_size > (unsigned)max_size)
- return -1;
-
- mrk = AV_RB32(src);
- switch (mrk) {
- case DCA_MARKER_RAW_BE:
- memcpy(dst, src, FFMIN(src_size, max_size));
- return FFMIN(src_size, max_size);
- case DCA_MARKER_RAW_LE:
- for (i = 0; i < (FFMIN(src_size, max_size) + 1) >> 1; i++)
- *sdst++ = bswap_16(*ssrc++);
- return FFMIN(src_size, max_size);
- case DCA_MARKER_14B_BE:
- case DCA_MARKER_14B_LE:
- init_put_bits(&pb, dst, max_size);
- for (i = 0; i < (src_size + 1) >> 1; i++, src += 2) {
- tmp = ((mrk == DCA_MARKER_14B_BE) ? AV_RB16(src) : AV_RL16(src)) & 0x3FFF;
- put_bits(&pb, 14, tmp);
- }
- flush_put_bits(&pb);
- return (put_bits_count(&pb) + 7) >> 3;
- default:
- return -1;
- }
-}
-
-/**
- * Main frame decoding function
- * FIXME add arguments
- */
-static int dca_decode_frame(AVCodecContext * avctx,
- void *data, int *data_size,
- uint8_t * buf, int buf_size)
-{
-
- int i, j, k;
- int16_t *samples = data;
- DCAContext *s = avctx->priv_data;
- int channels;
-
-
- s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer, DCA_MAX_FRAME_SIZE);
- if (s->dca_buffer_size == -1) {
- av_log(avctx, AV_LOG_ERROR, "Not a DCA frame\n");
- return -1;
- }
-
- init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
- if (dca_parse_frame_header(s) < 0) {
- //seems like the frame is corrupt, try with the next one
- return buf_size;
- }
- //set AVCodec values with parsed data
- avctx->sample_rate = s->sample_rate;
- avctx->bit_rate = s->bit_rate;
-
- channels = s->prim_channels + !!s->lfe;
- if(avctx->channels == 0) {
- avctx->channels = channels;
- } else if(channels < avctx->channels) {
- av_log(avctx, AV_LOG_WARNING, "DTS source channels are less than "
- "specified: output to %d channels.\n", channels);
- avctx->channels = channels;
- }
- if(avctx->channels == 2) {
- s->output = DCA_STEREO;
- } else if(avctx->channels != channels) {
- av_log(avctx, AV_LOG_ERROR, "Cannot downmix DTS to %d channels.\n",
- avctx->channels);
- return -1;
- }
-
- channels = avctx->channels;
- if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
- return -1;
- *data_size = 0;
- for (i = 0; i < (s->sample_blocks / 8); i++) {
- dca_decode_block(s);
- s->dsp.float_to_int16(s->tsamples, s->samples, 256 * channels);
- /* interleave samples */
- for (j = 0; j < 256; j++) {
- for (k = 0; k < channels; k++)
- samples[k] = s->tsamples[j + k * 256];
- samples += channels;
- }
- *data_size += 256 * sizeof(int16_t) * channels;
- }
-
- return buf_size;
-}
-
-
-
-/**
- * Build the cosine modulation tables for the QMF
- *
- * @param s pointer to the DCAContext
- */
-
-static void pre_calc_cosmod(DCAContext * s)
-{
- int i, j, k;
- static int cosmod_inited = 0;
-
- if(cosmod_inited) return;
- for (j = 0, k = 0; k < 16; k++)
- for (i = 0; i < 16; i++)
- 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++)
- cos_mod[j++] = cos((i) * (2 * k + 1) * M_PI / 32);
-
- for (k = 0; k < 16; k++)
- cos_mod[j++] = 0.25 / (2 * cos((2 * k + 1) * M_PI / 128));
-
- for (k = 0; k < 16; k++)
- cos_mod[j++] = -0.25 / (2.0 * sin((2 * k + 1) * M_PI / 128));
-
- cosmod_inited = 1;
-}
-
-
-/**
- * DCA initialization
- *
- * @param avctx pointer to the AVCodecContext
- */
-
-static int dca_decode_init(AVCodecContext * avctx)
-{
- DCAContext *s = avctx->priv_data;
-
- s->avctx = avctx;
- dca_init_vlcs();
- pre_calc_cosmod(s);
-
- dsputil_init(&s->dsp, avctx);
- return 0;
-}
-
-
-AVCodec dca_decoder = {
- .name = "dca",
- .type = CODEC_TYPE_AUDIO,
- .id = CODEC_ID_DTS,
- .priv_data_size = sizeof(DCAContext),
- .init = dca_decode_init,
- .decode = dca_decode_frame,
-};
-
-#ifdef CONFIG_DCA_PARSER
+#include "dca.h"
typedef struct DCAParseContext {
ParseContext pc;
@@ -1353,4 +125,3 @@ AVCodecParser dca_parser = {
dca_parse,
ff_parse_close,
};
-#endif /* CONFIG_DCA_PARSER */
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