* Copyright (C) 2004 Benjamin Zores
* Copyright (C) 2006 Benjamin Larsson
* Copyright (C) 2007 Konstantin Shishkov
+ * Copyright (C) 2012 Paul B Mahol
+ * Copyright (C) 2014 Niels Möller
*
* This file is part of Libav.
*
#include <stddef.h>
#include <stdio.h>
+#include "libavutil/attributes.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "avcodec.h"
#include "dca.h"
+#include "dca_syncwords.h"
#include "dcadata.h"
#include "dcadsp.h"
#include "dcahuff.h"
# include "arm/dca.h"
#endif
-//#define TRACE
-
-#define DCA_PRIM_CHANNELS_MAX (7)
-#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_CHANNEL,
DCA_4F2R
};
-/* these are unconfirmed but should be mostly correct */
-enum DCAExSSSpeakerMask {
- DCA_EXSS_FRONT_CENTER = 0x0001,
- DCA_EXSS_FRONT_LEFT_RIGHT = 0x0002,
- DCA_EXSS_SIDE_REAR_LEFT_RIGHT = 0x0004,
- DCA_EXSS_LFE = 0x0008,
- DCA_EXSS_REAR_CENTER = 0x0010,
- DCA_EXSS_FRONT_HIGH_LEFT_RIGHT = 0x0020,
- DCA_EXSS_REAR_LEFT_RIGHT = 0x0040,
- DCA_EXSS_FRONT_HIGH_CENTER = 0x0080,
- DCA_EXSS_OVERHEAD = 0x0100,
- DCA_EXSS_CENTER_LEFT_RIGHT = 0x0200,
- DCA_EXSS_WIDE_LEFT_RIGHT = 0x0400,
- DCA_EXSS_SIDE_LEFT_RIGHT = 0x0800,
- DCA_EXSS_LFE2 = 0x1000,
- DCA_EXSS_SIDE_HIGH_LEFT_RIGHT = 0x2000,
- DCA_EXSS_REAR_HIGH_CENTER = 0x4000,
- DCA_EXSS_REAR_HIGH_LEFT_RIGHT = 0x8000,
-};
-
-enum DCAExtensionMask {
- DCA_EXT_CORE = 0x001, ///< core in core substream
- DCA_EXT_XXCH = 0x002, ///< XXCh channels extension in core substream
- DCA_EXT_X96 = 0x004, ///< 96/24 extension in core substream
- DCA_EXT_XCH = 0x008, ///< XCh channel extension in core substream
- DCA_EXT_EXSS_CORE = 0x010, ///< core in ExSS (extension substream)
- DCA_EXT_EXSS_XBR = 0x020, ///< extended bitrate extension in ExSS
- DCA_EXT_EXSS_XXCH = 0x040, ///< XXCh channels extension in ExSS
- DCA_EXT_EXSS_X96 = 0x080, ///< 96/24 extension in ExSS
- DCA_EXT_EXSS_LBR = 0x100, ///< low bitrate component in ExSS
- 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,
};
-/* 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.
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
-};
-
-static const int8_t dca_channel_reorder_lfe[][9] = {
- { 0, -1, -1, -1, -1, -1, -1, -1, -1 },
- { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
- { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
- { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
- { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
- { 2, 0, 1, -1, -1, -1, -1, -1, -1 },
- { 0, 1, 3, -1, -1, -1, -1, -1, -1 },
- { 2, 0, 1, 4, -1, -1, -1, -1, -1 },
- { 0, 1, 3, 4, -1, -1, -1, -1, -1 },
- { 2, 0, 1, 4, 5, -1, -1, -1, -1 },
- { 3, 4, 0, 1, 5, 6, -1, -1, -1 },
- { 2, 0, 1, 4, 5, 6, -1, -1, -1 },
- { 0, 6, 4, 5, 2, 3, -1, -1, -1 },
- { 4, 2, 5, 0, 1, 6, 7, -1, -1 },
- { 5, 6, 0, 1, 7, 3, 8, 4, -1 },
- { 4, 2, 5, 0, 1, 6, 8, 7, -1 },
-};
-
-static const int8_t dca_channel_reorder_lfe_xch[][9] = {
- { 0, 2, -1, -1, -1, -1, -1, -1, -1 },
- { 0, 1, 3, -1, -1, -1, -1, -1, -1 },
- { 0, 1, 3, -1, -1, -1, -1, -1, -1 },
- { 0, 1, 3, -1, -1, -1, -1, -1, -1 },
- { 0, 1, 3, -1, -1, -1, -1, -1, -1 },
- { 2, 0, 1, 4, -1, -1, -1, -1, -1 },
- { 0, 1, 3, 4, -1, -1, -1, -1, -1 },
- { 2, 0, 1, 4, 5, -1, -1, -1, -1 },
- { 0, 1, 4, 5, 3, -1, -1, -1, -1 },
- { 2, 0, 1, 5, 6, 4, -1, -1, -1 },
- { 3, 4, 0, 1, 6, 7, 5, -1, -1 },
- { 2, 0, 1, 4, 5, 6, 7, -1, -1 },
- { 0, 6, 4, 5, 2, 3, 7, -1, -1 },
- { 4, 2, 5, 0, 1, 7, 8, 6, -1 },
- { 5, 6, 0, 1, 8, 3, 9, 4, 7 },
- { 4, 2, 5, 0, 1, 6, 9, 8, 7 },
-};
-
-static const int8_t dca_channel_reorder_nolfe[][9] = {
- { 0, -1, -1, -1, -1, -1, -1, -1, -1 },
- { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
- { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
- { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
- { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
- { 2, 0, 1, -1, -1, -1, -1, -1, -1 },
- { 0, 1, 2, -1, -1, -1, -1, -1, -1 },
- { 2, 0, 1, 3, -1, -1, -1, -1, -1 },
- { 0, 1, 2, 3, -1, -1, -1, -1, -1 },
- { 2, 0, 1, 3, 4, -1, -1, -1, -1 },
- { 2, 3, 0, 1, 4, 5, -1, -1, -1 },
- { 2, 0, 1, 3, 4, 5, -1, -1, -1 },
- { 0, 5, 3, 4, 1, 2, -1, -1, -1 },
- { 3, 2, 4, 0, 1, 5, 6, -1, -1 },
- { 4, 5, 0, 1, 6, 2, 7, 3, -1 },
- { 3, 2, 4, 0, 1, 5, 7, 6, -1 },
-};
-
-static const int8_t dca_channel_reorder_nolfe_xch[][9] = {
- { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
- { 0, 1, 2, -1, -1, -1, -1, -1, -1 },
- { 0, 1, 2, -1, -1, -1, -1, -1, -1 },
- { 0, 1, 2, -1, -1, -1, -1, -1, -1 },
- { 0, 1, 2, -1, -1, -1, -1, -1, -1 },
- { 2, 0, 1, 3, -1, -1, -1, -1, -1 },
- { 0, 1, 2, 3, -1, -1, -1, -1, -1 },
- { 2, 0, 1, 3, 4, -1, -1, -1, -1 },
- { 0, 1, 3, 4, 2, -1, -1, -1, -1 },
- { 2, 0, 1, 4, 5, 3, -1, -1, -1 },
- { 2, 3, 0, 1, 5, 6, 4, -1, -1 },
- { 2, 0, 1, 3, 4, 5, 6, -1, -1 },
- { 0, 5, 3, 4, 1, 2, 6, -1, -1 },
- { 3, 2, 4, 0, 1, 6, 7, 5, -1 },
- { 4, 5, 0, 1, 7, 2, 8, 3, 6 },
- { 3, 2, 4, 0, 1, 5, 8, 7, 6 },
-};
-
#define DCA_DOLBY 101 /* FIXME */
#define DCA_CHANNEL_BITS 6
#define HEADER_SIZE 14
-#define DCA_MAX_FRAME_SIZE 16384
-#define DCA_MAX_EXSS_HEADER_SIZE 4096
-
-#define DCA_BUFFER_PADDING_SIZE 1024
-
#define DCA_NSYNCAUX 0x9A1105A0
+#define SAMPLES_PER_SUBBAND 8 // number of samples per subband per subsubframe
+
/** Bit allocation */
-typedef struct {
+typedef struct BitAlloc {
int offset; ///< code values offset
int maxbits[8]; ///< max bits in VLC
int wrap; ///< wrap for get_vlc2()
ba->offset;
}
-typedef struct {
- AVClass *class; ///< class for AVOptions
- 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 bit_rate_index; ///< transmission bit rate index
-
- 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 total_channels; ///< number of channels including extensions
- 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[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 transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< transition mode (transients)
- int32_t 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
- float downmix_coef[DCA_PRIM_CHANNELS_MAX + 1][2]; ///< stereo downmix coefficients
- int dynrange_coef; ///< dynamic range coefficient
-
- /* Core substream's embedded downmix coefficients (cf. ETSI TS 102 114 V1.4.1)
- * Input: primary audio channels (incl. LFE if present)
- * Output: downmix audio channels (up to 4, no LFE) */
- uint8_t core_downmix; ///< embedded downmix coefficients available
- uint8_t core_downmix_amode; ///< audio channel arrangement of embedded downmix
- uint16_t core_downmix_codes[DCA_PRIM_CHANNELS_MAX + 1][4]; ///< embedded downmix coefficients (9-bit codes)
-
- int32_t high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< VQ encoded high frequency subbands
-
- float lfe_data[2 * DCA_LFE_MAX * (DCA_BLOCKS_MAX + 4)]; ///< Low frequency effect data
- int lfe_scale_factor;
-
- /* Subband samples history (for ADPCM) */
- 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(32, float, raXin)[32];
-
- int output; ///< type of output
-
- DECLARE_ALIGNED(32, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
- float *samples_chanptr[DCA_PRIM_CHANNELS_MAX + 1];
- float *extra_channels[DCA_PRIM_CHANNELS_MAX + 1];
- uint8_t *extra_channels_buffer;
- unsigned int extra_channels_buffer_size;
-
- 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
- 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; ///< XCh extension present and valid
- int xch_base_channel; ///< index of first (only) channel containing XCH data
- int xch_disable; ///< whether the XCh extension should be decoded or not
-
- /* ExSS header parser */
- int static_fields; ///< static fields present
- int mix_metadata; ///< mixing metadata present
- int num_mix_configs; ///< number of mix out configurations
- int mix_config_num_ch[4]; ///< number of channels in each mix out configuration
-
- int profile;
-
- int debug_flag; ///< used for suppressing repeated error messages output
- AVFloatDSPContext fdsp;
- FFTContext imdct;
- SynthFilterContext synth;
- DCADSPContext dcadsp;
- FmtConvertContext fmt_conv;
-} DCAContext;
-
-static const uint16_t dca_vlc_offs[] = {
- 0, 512, 640, 768, 1282, 1794, 2436, 3080, 3770, 4454, 5364,
- 5372, 5380, 5388, 5392, 5396, 5412, 5420, 5428, 5460, 5492, 5508,
- 5572, 5604, 5668, 5796, 5860, 5892, 6412, 6668, 6796, 7308, 7564,
- 7820, 8076, 8620, 9132, 9388, 9910, 10166, 10680, 11196, 11726, 12240,
- 12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264,
- 18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622,
-};
-
static av_cold void dca_init_vlcs(void)
{
static int vlcs_initialized = 0;
dca_bitalloc_index.offset = 1;
dca_bitalloc_index.wrap = 2;
for (i = 0; i < 5; i++) {
- dca_bitalloc_index.vlc[i].table = &dca_table[dca_vlc_offs[i]];
- dca_bitalloc_index.vlc[i].table_allocated = dca_vlc_offs[i + 1] - dca_vlc_offs[i];
+ dca_bitalloc_index.vlc[i].table = &dca_table[ff_dca_vlc_offs[i]];
+ dca_bitalloc_index.vlc[i].table_allocated = ff_dca_vlc_offs[i + 1] - ff_dca_vlc_offs[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, INIT_VLC_USE_NEW_STATIC);
dca_scalefactor.offset = -64;
dca_scalefactor.wrap = 2;
for (i = 0; i < 5; i++) {
- dca_scalefactor.vlc[i].table = &dca_table[dca_vlc_offs[i + 5]];
- dca_scalefactor.vlc[i].table_allocated = dca_vlc_offs[i + 6] - dca_vlc_offs[i + 5];
+ dca_scalefactor.vlc[i].table = &dca_table[ff_dca_vlc_offs[i + 5]];
+ dca_scalefactor.vlc[i].table_allocated = ff_dca_vlc_offs[i + 6] - ff_dca_vlc_offs[i + 5];
init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,
scales_bits[i], 1, 1,
scales_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
dca_tmode.offset = 0;
dca_tmode.wrap = 1;
for (i = 0; i < 4; i++) {
- dca_tmode.vlc[i].table = &dca_table[dca_vlc_offs[i + 10]];
- dca_tmode.vlc[i].table_allocated = dca_vlc_offs[i + 11] - dca_vlc_offs[i + 10];
+ dca_tmode.vlc[i].table = &dca_table[ff_dca_vlc_offs[i + 10]];
+ dca_tmode.vlc[i].table_allocated = ff_dca_vlc_offs[i + 11] - ff_dca_vlc_offs[i + 10];
init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
tmode_bits[i], 1, 1,
tmode_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
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];
+ dca_smpl_bitalloc[i + 1].vlc[j].table = &dca_table[ff_dca_vlc_offs[c]];
+ dca_smpl_bitalloc[i + 1].vlc[j].table_allocated = ff_dca_vlc_offs[c + 1] - ff_dca_vlc_offs[c];
init_vlc(&dca_smpl_bitalloc[i + 1].vlc[j], bitalloc_maxbits[i][j],
bitalloc_sizes[i],
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]);
- 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;
}
if (!s->sample_rate)
return AVERROR_INVALIDDATA;
s->bit_rate_index = get_bits(&s->gb, 5);
- s->bit_rate = dca_bit_rates[s->bit_rate_index];
+ s->bit_rate = ff_dca_bit_rates[s->bit_rate_index];
if (!s->bit_rate)
return AVERROR_INVALIDDATA;
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 Hz\n",
- s->sample_rate);
- av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i bits/s\n",
- s->bit_rate);
- 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;
}
if (s->bitalloc[j][k] > 26) {
- av_dlog(s->avctx, "bitalloc index [%i][%i] too big (%i)\n",
+ ff_dlog(s->avctx, "bitalloc index [%i][%i] too big (%i)\n",
j, k, s->bitalloc[j][k]);
return AVERROR_INVALIDDATA;
}
s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2);
if (s->scalefactor_huffman[j] == 6) {
- scale_table = scale_factor_quant7;
+ scale_table = ff_dca_scale_factor_quant7;
log_size = 7;
} else {
- scale_table = scale_factor_quant6;
+ scale_table = ff_dca_scale_factor_quant6;
log_size = 6;
}
/* Scale factor index */
skip_bits(&s->gb, 1);
- s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 7)];
+ s->lfe_scale_factor = ff_dca_scale_factor_quant7[get_bits(&s->gb, 7)];
/* Quantization step size * scale factor */
lfe_scale = 0.035 * s->lfe_scale_factor;
s->lfe_data[j] *= lfe_scale;
}
-#ifdef TRACE
- 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++)
- av_log(s->avctx, AV_LOG_DEBUG, " %i", s->prediction_mode[j][k]);
- av_log(s->avctx, AV_LOG_DEBUG, "\n");
- }
- 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);
- }
- for (j = base_channel; 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 = base_channel; 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 = base_channel; 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 = base_channel; j < s->prim_channels; j++) {
- if (s->joint_intensity[j] > 0) {
- int source_channel = s->joint_intensity[j] - 1;
- 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");
- }
- }
- for (j = base_channel; 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 (!base_channel && s->lfe) {
- int lfe_samples = 2 * s->lfe * (4 + block_index);
- int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);
-
- av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n");
- for (j = lfe_samples; j < lfe_end_sample; 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 samples_in[32][SAMPLES_PER_SUBBAND], float *samples_out,
float scale)
{
const float *prCoeff;
/* Select filter */
if (!s->multirate_inter) /* Non-perfect reconstruction */
- prCoeff = fir_32bands_nonperfect;
+ prCoeff = ff_dca_fir_32bands_nonperfect;
else /* Perfect reconstruction */
- prCoeff = fir_32bands_perfect;
+ prCoeff = ff_dca_fir_32bands_perfect;
s->dcadsp.qmf_32_subbands(samples_in, sb_act, &s->synth, &s->imdct,
s->subband_fir_hist[chans],
samples_out, s->raXin, scale);
}
-static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
- int num_deci_sample, float *samples_in,
+static QMF64_table *qmf64_precompute(void)
+{
+ unsigned i, j;
+ QMF64_table *table = av_malloc(sizeof(*table));
+ if (!table)
+ return NULL;
+
+ for (i = 0; i < 32; i++)
+ for (j = 0; j < 32; j++)
+ table->dct4_coeff[i][j] = cos((2 * i + 1) * (2 * j + 1) * M_PI / 128);
+ for (i = 0; i < 32; i++)
+ for (j = 0; j < 32; j++)
+ table->dct2_coeff[i][j] = cos((2 * i + 1) * j * M_PI / 64);
+
+ /* FIXME: Is the factor 0.125 = 1/8 right? */
+ for (i = 0; i < 32; i++)
+ table->rcos[i] = 0.125 / cos((2 * i + 1) * M_PI / 256);
+ for (i = 0; i < 32; i++)
+ table->rsin[i] = -0.125 / sin((2 * i + 1) * M_PI / 256);
+
+ return table;
+}
+
+/* FIXME: Totally unoptimized. Based on the reference code and
+ * http://multimedia.cx/mirror/dca-transform.pdf, with guessed tweaks
+ * for doubling the size. */
+static void qmf_64_subbands(DCAContext *s, int chans, float samples_in[64][SAMPLES_PER_SUBBAND],
+ float *samples_out, float scale)
+{
+ float raXin[64];
+ float A[32], B[32];
+ float *raX = s->subband_fir_hist[chans];
+ float *raZ = s->subband_fir_noidea[chans];
+ unsigned i, j, k, subindex;
+
+ for (i = s->subband_activity[chans]; i < 64; i++)
+ raXin[i] = 0.0;
+ for (subindex = 0; subindex < SAMPLES_PER_SUBBAND; subindex++) {
+ for (i = 0; i < s->subband_activity[chans]; i++)
+ raXin[i] = samples_in[i][subindex];
+
+ for (k = 0; k < 32; k++) {
+ A[k] = 0.0;
+ for (i = 0; i < 32; i++)
+ A[k] += (raXin[2 * i] + raXin[2 * i + 1]) * s->qmf64_table->dct4_coeff[k][i];
+ }
+ for (k = 0; k < 32; k++) {
+ B[k] = raXin[0] * s->qmf64_table->dct2_coeff[k][0];
+ for (i = 1; i < 32; i++)
+ B[k] += (raXin[2 * i] + raXin[2 * i - 1]) * s->qmf64_table->dct2_coeff[k][i];
+ }
+ for (k = 0; k < 32; k++) {
+ raX[k] = s->qmf64_table->rcos[k] * (A[k] + B[k]);
+ raX[63 - k] = s->qmf64_table->rsin[k] * (A[k] - B[k]);
+ }
+
+ for (i = 0; i < 64; i++) {
+ float out = raZ[i];
+ for (j = 0; j < 1024; j += 128)
+ out += ff_dca_fir_64bands[j + i] * (raX[j + i] - raX[j + 63 - i]);
+ *samples_out++ = out * scale;
+ }
+
+ for (i = 0; i < 64; i++) {
+ float hist = 0.0;
+ for (j = 0; j < 1024; j += 128)
+ hist += ff_dca_fir_64bands[64 + j + i] * (-raX[i + j] - raX[j + 63 - i]);
+
+ raZ[i] = hist;
+ }
+
+ /* FIXME: Make buffer circular, to avoid this move. */
+ memmove(raX + 64, raX, (1024 - 64) * sizeof(*raX));
+ }
+}
+
+static void lfe_interpolation_fir(DCAContext *s, const float *samples_in,
float *samples_out)
{
/* samples_in: An array holding decimated samples.
int deciindex;
/* Select decimation filter */
- if (decimation_select == 1) {
+ if (s->lfe == 1) {
idx = 1;
- prCoeff = lfe_fir_128;
+ prCoeff = ff_dca_lfe_fir_128;
} else {
- idx = 0;
- prCoeff = lfe_fir_64;
+ idx = 0;
+ if (s->exss_ext_mask & DCA_EXT_EXSS_XLL)
+ prCoeff = ff_dca_lfe_xll_fir_64;
+ else
+ prCoeff = ff_dca_lfe_fir_64;
}
/* Interpolation */
- for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
+ for (deciindex = 0; deciindex < 2 * s->lfe; deciindex++) {
s->dcadsp.lfe_fir[idx](samples_out, samples_in, prCoeff);
samples_in++;
samples_out += 2 * 32 * (1 + idx);
break;
}
if (lfe_present) {
- int lf_buf = dca_lfe_index[srcfmt];
- int lf_idx = dca_channels[srcfmt];
+ int lf_buf = ff_dca_lfe_index[srcfmt];
+ int lf_idx = ff_dca_channels[srcfmt];
for (i = 0; i < 256; i++) {
samples[0][i] += samples[lf_buf][i] * coef[lf_idx][0];
samples[1][i] += samples[lf_buf][i] * coef[lf_idx][1];
const float *quant_step_table;
/* FIXME */
- float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
- LOCAL_ALIGNED_16(int32_t, block, [8 * DCA_SUBBANDS]);
+ float (*subband_samples)[DCA_SUBBANDS][SAMPLES_PER_SUBBAND] = s->subband_samples[block_index];
+ LOCAL_ALIGNED_16(int32_t, block, [SAMPLES_PER_SUBBAND * DCA_SUBBANDS]);
/*
* Audio data
/* Select quantization step size table */
if (s->bit_rate_index == 0x1f)
- quant_step_table = lossless_quant_d;
+ quant_step_table = ff_dca_lossless_quant_d;
else
- quant_step_table = lossy_quant_d;
+ quant_step_table = ff_dca_lossy_quant_d;
for (k = base_channel; k < s->prim_channels; k++) {
float rscale[DCA_SUBBANDS];
*/
if (!abits) {
rscale[l] = 0;
- memset(block + 8 * l, 0, 8 * sizeof(block[0]));
+ memset(block + SAMPLES_PER_SUBBAND * l, 0, SAMPLES_PER_SUBBAND * sizeof(block[0]));
} else {
/* Deal with transients */
int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];
block_code1 = get_bits(&s->gb, size);
block_code2 = get_bits(&s->gb, size);
err = decode_blockcodes(block_code1, block_code2,
- levels, block + 8 * l);
+ levels, block + SAMPLES_PER_SUBBAND * l);
if (err) {
av_log(s->avctx, AV_LOG_ERROR,
"ERROR: block code look-up failed\n");
}
} else {
/* no coding */
- for (m = 0; m < 8; m++)
- block[8 * l + m] = get_sbits(&s->gb, abits - 3);
+ for (m = 0; m < SAMPLES_PER_SUBBAND; m++)
+ block[SAMPLES_PER_SUBBAND * l + m] = get_sbits(&s->gb, abits - 3);
}
} else {
/* Huffman coded */
- for (m = 0; m < 8; m++)
- block[8 * l + m] = get_bitalloc(&s->gb,
+ for (m = 0; m < SAMPLES_PER_SUBBAND; m++)
+ block[SAMPLES_PER_SUBBAND * l + m] = get_bitalloc(&s->gb,
&dca_smpl_bitalloc[abits], sel);
}
}
}
s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[k][0],
- block, rscale, 8 * s->vq_start_subband[k]);
+ block, rscale, SAMPLES_PER_SUBBAND * s->vq_start_subband[k]);
for (l = 0; l < s->vq_start_subband[k]; l++) {
int m;
if (s->prediction_mode[k][l]) {
int n;
if (s->predictor_history)
- subband_samples[k][l][0] += (adpcm_vb[s->prediction_vq[k][l]][0] *
+ subband_samples[k][l][0] += (ff_dca_adpcm_vb[s->prediction_vq[k][l]][0] *
s->subband_samples_hist[k][l][3] +
- adpcm_vb[s->prediction_vq[k][l]][1] *
+ ff_dca_adpcm_vb[s->prediction_vq[k][l]][1] *
s->subband_samples_hist[k][l][2] +
- adpcm_vb[s->prediction_vq[k][l]][2] *
+ ff_dca_adpcm_vb[s->prediction_vq[k][l]][2] *
s->subband_samples_hist[k][l][1] +
- adpcm_vb[s->prediction_vq[k][l]][3] *
+ ff_dca_adpcm_vb[s->prediction_vq[k][l]][3] *
s->subband_samples_hist[k][l][0]) *
(1.0f / 8192);
- for (m = 1; m < 8; m++) {
- float sum = adpcm_vb[s->prediction_vq[k][l]][0] *
+ for (m = 1; m < SAMPLES_PER_SUBBAND; m++) {
+ float sum = ff_dca_adpcm_vb[s->prediction_vq[k][l]][0] *
subband_samples[k][l][m - 1];
for (n = 2; n <= 4; n++)
if (m >= n)
- sum += adpcm_vb[s->prediction_vq[k][l]][n - 1] *
+ sum += ff_dca_adpcm_vb[s->prediction_vq[k][l]][n - 1] *
subband_samples[k][l][m - n];
else if (s->predictor_history)
- sum += adpcm_vb[s->prediction_vq[k][l]][n - 1] *
+ sum += ff_dca_adpcm_vb[s->prediction_vq[k][l]][n - 1] *
s->subband_samples_hist[k][l][m - n + 4];
subband_samples[k][l][m] += sum * 1.0f / 8192;
}
s->debug_flag |= 0x01;
}
s->dcadsp.decode_hf(subband_samples[k], s->high_freq_vq[k],
- high_freq_vq, subsubframe * 8,
+ ff_dca_high_freq_vq, subsubframe * SAMPLES_PER_SUBBAND,
s->scale_factor[k], s->vq_start_subband[k],
s->subband_activity[k]);
}
/* Check for DSYNC after subsubframe */
if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {
- if (0xFFFF == get_bits(&s->gb, 16)) { /* 0xFFFF */
-#ifdef TRACE
- av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");
-#endif
- } else {
+ if (get_bits(&s->gb, 16) != 0xFFFF) {
av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");
return AVERROR_INVALIDDATA;
}
return 0;
}
-static int dca_filter_channels(DCAContext *s, int block_index)
+static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
{
- float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
+ float (*subband_samples)[DCA_SUBBANDS][SAMPLES_PER_SUBBAND] = s->subband_samples[block_index];
int k;
- /* 32 subbands QMF */
- for (k = 0; k < s->prim_channels; k++) {
- if (s->channel_order_tab[k] >= 0)
- qmf_32_subbands(s, k, subband_samples[k],
- s->samples_chanptr[s->channel_order_tab[k]],
- M_SQRT1_2 / 32768.0);
+ if (upsample) {
+ if (!s->qmf64_table) {
+ s->qmf64_table = qmf64_precompute();
+ if (!s->qmf64_table)
+ return AVERROR(ENOMEM);
+ }
+
+ /* 64 subbands QMF */
+ for (k = 0; k < s->prim_channels; k++) {
+ if (s->channel_order_tab[k] >= 0)
+ qmf_64_subbands(s, k, subband_samples[k],
+ s->samples_chanptr[s->channel_order_tab[k]],
+ /* Upsampling needs a factor 2 here. */
+ M_SQRT2 / 32768.0);
+ }
+ } else {
+ /* 32 subbands QMF */
+ for (k = 0; k < s->prim_channels; k++) {
+ if (s->channel_order_tab[k] >= 0)
+ qmf_32_subbands(s, k, subband_samples[k],
+ s->samples_chanptr[s->channel_order_tab[k]],
+ M_SQRT1_2 / 32768.0);
+ }
}
/* Generate LFE samples for this subsubframe FIXME!!! */
if (s->lfe) {
- lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,
+ float *samples = s->samples_chanptr[ff_dca_lfe_index[s->amode]];
+ lfe_interpolation_fir(s,
s->lfe_data + 2 * s->lfe * (block_index + 4),
- s->samples_chanptr[dca_lfe_index[s->amode]]);
- /* Outputs 20bits pcm samples */
+ samples);
+ if (upsample) {
+ unsigned i;
+ /* Should apply the filter in Table 6-11 when upsampling. For
+ * now, just duplicate. */
+ for (i = 511; i > 0; i--) {
+ samples[2 * i] =
+ samples[2 * i + 1] = samples[i];
+ }
+ samples[1] = samples[0];
+ }
}
+ /* FIXME: This downmixing is probably broken with upsample.
+ * Probably totally broken also with XLL in general. */
/* Downmixing to Stereo */
if (s->prim_channels + !!s->lfe > 2 &&
s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
am);
return AVERROR_INVALIDDATA;
}
- for (out = 0; out < dca_channels[s->core_downmix_amode]; out++) {
+ for (out = 0; out < ff_dca_channels[s->core_downmix_amode]; out++) {
for (in = 0; in < s->prim_channels + !!s->lfe; in++) {
uint16_t tmp = get_bits(&s->gb, 9);
if ((tmp & 0xFF) > 241) {
}
if (!s->current_subsubframe) {
-#ifdef TRACE
- av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");
-#endif
/* Read subframe header */
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 ((ret = dca_subsubframe(s, base_channel, block_index)))
return ret;
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 ((ret = dca_subframe_footer(s, base_channel)))
return ret;
return 0;
}
-/**
- * Return the number of channels in an ExSS speaker mask (HD)
- */
-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));
-}
-
-/**
- * Skip mixing coefficients of a single mix out configuration (HD)
- */
-static void dca_exss_skip_mix_coeffs(GetBitContext *gb, int channels, int out_ch)
-{
- 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);
- }
-}
-
-/**
- * Parse extension substream asset header (HD)
- */
-static int dca_exss_parse_asset_header(DCAContext *s)
-{
- int header_pos = get_bits_count(&s->gb);
- int header_size;
- int channels = 0;
- int embedded_stereo = 0;
- int embedded_6ch = 0;
- int drc_code_present;
- int extensions_mask = 0;
- int i, j;
-
- if (get_bits_left(&s->gb) < 16)
- return -1;
-
- /* We will parse just enough to get to the extensions bitmask with which
- * we can set the profile value. */
-
- header_size = get_bits(&s->gb, 9) + 1;
- skip_bits(&s->gb, 3); // asset index
-
- if (s->static_fields) {
- if (get_bits1(&s->gb))
- skip_bits(&s->gb, 4); // asset type descriptor
- if (get_bits1(&s->gb))
- skip_bits_long(&s->gb, 24); // language descriptor
-
- if (get_bits1(&s->gb)) {
- /* How can one fit 1024 bytes of text here if the maximum value
- * for the asset header size field above was 512 bytes? */
- int text_length = get_bits(&s->gb, 10) + 1;
- if (get_bits_left(&s->gb) < text_length * 8)
- return -1;
- skip_bits_long(&s->gb, text_length * 8); // info text
- }
-
- skip_bits(&s->gb, 5); // bit resolution - 1
- skip_bits(&s->gb, 4); // max sample rate code
- channels = get_bits(&s->gb, 8) + 1;
-
- if (get_bits1(&s->gb)) { // 1-to-1 channels to speakers
- int spkr_remap_sets;
- int spkr_mask_size = 16;
- int num_spkrs[7];
-
- if (channels > 2)
- embedded_stereo = get_bits1(&s->gb);
- if (channels > 6)
- embedded_6ch = get_bits1(&s->gb);
-
- if (get_bits1(&s->gb)) {
- spkr_mask_size = (get_bits(&s->gb, 2) + 1) << 2;
- skip_bits(&s->gb, spkr_mask_size); // spkr activity mask
- }
-
- spkr_remap_sets = get_bits(&s->gb, 3);
-
- for (i = 0; i < spkr_remap_sets; i++) {
- /* std layout mask for each remap set */
- num_spkrs[i] = dca_exss_mask2count(get_bits(&s->gb, spkr_mask_size));
- }
-
- for (i = 0; i < spkr_remap_sets; i++) {
- int num_dec_ch_remaps = get_bits(&s->gb, 5) + 1;
- if (get_bits_left(&s->gb) < 0)
- return -1;
-
- for (j = 0; j < num_spkrs[i]; j++) {
- int remap_dec_ch_mask = get_bits_long(&s->gb, num_dec_ch_remaps);
- int num_dec_ch = av_popcount(remap_dec_ch_mask);
- skip_bits_long(&s->gb, num_dec_ch * 5); // remap codes
- }
- }
- } else {
- skip_bits(&s->gb, 3); // representation type
- }
- }
-
- drc_code_present = get_bits1(&s->gb);
- if (drc_code_present)
- get_bits(&s->gb, 8); // drc code
-
- if (get_bits1(&s->gb))
- skip_bits(&s->gb, 5); // dialog normalization code
-
- if (drc_code_present && embedded_stereo)
- get_bits(&s->gb, 8); // drc stereo code
-
- if (s->mix_metadata && get_bits1(&s->gb)) {
- skip_bits(&s->gb, 1); // external mix
- skip_bits(&s->gb, 6); // post mix gain code
-
- if (get_bits(&s->gb, 2) != 3) // mixer drc code
- skip_bits(&s->gb, 3); // drc limit
- else
- skip_bits(&s->gb, 8); // custom drc code
-
- if (get_bits1(&s->gb)) // channel specific scaling
- for (i = 0; i < s->num_mix_configs; i++)
- skip_bits_long(&s->gb, s->mix_config_num_ch[i] * 6); // scale codes
- else
- skip_bits_long(&s->gb, s->num_mix_configs * 6); // scale codes
-
- for (i = 0; i < s->num_mix_configs; i++) {
- if (get_bits_left(&s->gb) < 0)
- return -1;
- dca_exss_skip_mix_coeffs(&s->gb, channels, s->mix_config_num_ch[i]);
- if (embedded_6ch)
- dca_exss_skip_mix_coeffs(&s->gb, 6, s->mix_config_num_ch[i]);
- if (embedded_stereo)
- dca_exss_skip_mix_coeffs(&s->gb, 2, s->mix_config_num_ch[i]);
- }
- }
-
- switch (get_bits(&s->gb, 2)) {
- case 0:
- extensions_mask = get_bits(&s->gb, 12);
- break;
- case 1:
- extensions_mask = DCA_EXT_EXSS_XLL;
- break;
- case 2:
- extensions_mask = DCA_EXT_EXSS_LBR;
- break;
- case 3:
- extensions_mask = 0; /* aux coding */
- break;
- }
-
- /* not parsed further, we were only interested in the extensions mask */
-
- if (get_bits_left(&s->gb) < 0)
- return -1;
-
- if (get_bits_count(&s->gb) - header_pos > header_size * 8) {
- av_log(s->avctx, AV_LOG_WARNING, "Asset header size mismatch.\n");
- return -1;
- }
- skip_bits_long(&s->gb, header_pos + header_size * 8 - get_bits_count(&s->gb));
-
- if (extensions_mask & DCA_EXT_EXSS_XLL)
- s->profile = FF_PROFILE_DTS_HD_MA;
- else if (extensions_mask & (DCA_EXT_EXSS_XBR | DCA_EXT_EXSS_X96 |
- DCA_EXT_EXSS_XXCH))
- s->profile = FF_PROFILE_DTS_HD_HRA;
-
- if (!(extensions_mask & DCA_EXT_CORE))
- av_log(s->avctx, AV_LOG_WARNING, "DTS core 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;
-}
-
-/**
- * Parse extension substream header (HD)
- */
-static void dca_exss_parse_header(DCAContext *s)
-{
- int ss_index;
- int blownup;
- int num_audiop = 1;
- int num_assets = 1;
- int active_ss_mask[8];
- int i, j;
-
- if (get_bits_left(&s->gb) < 52)
- return;
-
- skip_bits(&s->gb, 8); // user data
- ss_index = get_bits(&s->gb, 2);
-
- blownup = get_bits1(&s->gb);
- 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) {
- skip_bits(&s->gb, 2); // reference clock code
- skip_bits(&s->gb, 3); // frame duration code
-
- if (get_bits1(&s->gb))
- skip_bits_long(&s->gb, 36); // timestamp
-
- /* a single stream can contain multiple audio assets that can be
- * combined to form multiple audio presentations */
-
- num_audiop = get_bits(&s->gb, 3) + 1;
- if (num_audiop > 1) {
- avpriv_request_sample(s->avctx,
- "Multiple DTS-HD audio presentations");
- /* ignore such streams for now */
- return;
- }
-
- num_assets = get_bits(&s->gb, 3) + 1;
- if (num_assets > 1) {
- avpriv_request_sample(s->avctx, "Multiple DTS-HD audio assets");
- /* ignore such streams for now */
- return;
- }
-
- for (i = 0; i < num_audiop; i++)
- active_ss_mask[i] = get_bits(&s->gb, ss_index + 1);
-
- for (i = 0; i < num_audiop; i++)
- for (j = 0; j <= ss_index; j++)
- if (active_ss_mask[i] & (1 << j))
- skip_bits(&s->gb, 8); // active asset mask
-
- s->mix_metadata = get_bits1(&s->gb);
- if (s->mix_metadata) {
- 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;
-
- for (i = 0; i < s->num_mix_configs; i++) {
- 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
-
- for (i = 0; i < num_assets; i++) {
- if (dca_exss_parse_asset_header(s))
- return;
- }
-
- /* not parsed further, we were only interested in the extensions mask
- * from the asset header */
-}
-
static float dca_dmix_code(unsigned code)
{
int sign = (code >> 8) - 1;
code &= 0xff;
- return ((dca_dmixtable[code] ^ sign) - sign) * (1.0 / (1U << 15));
+ return ((ff_dca_dmixtable[code] ^ sign) - sign) * (1.0 / (1U << 15));
}
-/**
- * Main frame decoding function
- * FIXME add arguments
- */
-static int dca_decode_frame(AVCodecContext *avctx, void *data,
- int *got_frame_ptr, AVPacket *avpkt)
+static int scan_for_extensions(AVCodecContext *avctx)
{
- AVFrame *frame = data;
- const uint8_t *buf = avpkt->data;
- int buf_size = avpkt->size;
-
- int lfe_samples;
- int num_core_channels = 0;
- int i, ret;
- float **samples_flt;
DCAContext *s = avctx->priv_data;
- int channels, full_channels;
- int core_ss_end;
-
- s->xch_present = 0;
-
- s->dca_buffer_size = ff_dca_convert_bitstream(buf, buf_size, s->dca_buffer,
- DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);
- if (s->dca_buffer_size == AVERROR_INVALIDDATA) {
- av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
- return AVERROR_INVALIDDATA;
- }
-
- if ((ret = dca_parse_frame_header(s)) < 0) {
- // seems like the frame is corrupt, try with the next one
- return ret;
- }
- // set AVCodec values with parsed data
- avctx->sample_rate = s->sample_rate;
- avctx->bit_rate = s->bit_rate;
-
- s->profile = FF_PROFILE_DTS;
-
- for (i = 0; i < (s->sample_blocks / 8); 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;
-
- if (s->ext_coding)
- s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];
- else
- s->core_ext_mask = 0;
+ int core_ss_end, ret = 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);
+ int i;
switch (bits) {
- case 0x5a5a5a5a: {
+ case DCA_SYNCWORD_XCH: {
int ext_amode, xch_fsize;
s->xch_base_channel = s->prim_channels;
s->xch_present = 1;
break;
}
- case 0x47004a03:
+ case DCA_SYNCWORD_XXCH:
/* 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 */
/* check for ExSS (HD part) */
if (s->dca_buffer_size - s->frame_size > 32 &&
- get_bits_long(&s->gb, 32) == DCA_HD_MARKER)
- dca_exss_parse_header(s);
+ get_bits_long(&s->gb, 32) == DCA_SYNCWORD_SUBSTREAM)
+ ff_dca_exss_parse_header(s);
- avctx->profile = s->profile;
+ return ret;
+}
- full_channels = channels = s->prim_channels + !!s->lfe;
+static int set_channel_layout(AVCodecContext *avctx, int channels, int num_core_channels)
+{
+ DCAContext *s = avctx->priv_data;
+ int i;
if (s->amode < 16) {
avctx->channel_layout = dca_core_channel_layout[s->amode];
s->xch_disable = 1;
}
-#if FF_API_REQUEST_CHANNELS
-FF_DISABLE_DEPRECATION_WARNINGS
- if (s->xch_present && !s->xch_disable &&
- (!avctx->request_channels ||
- avctx->request_channels > num_core_channels + !!s->lfe)) {
-FF_ENABLE_DEPRECATION_WARNINGS
-#else
if (s->xch_present && !s->xch_disable) {
-#endif
avctx->channel_layout |= AV_CH_BACK_CENTER;
if (s->lfe) {
avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
- s->channel_order_tab = dca_channel_reorder_lfe_xch[s->amode];
+ s->channel_order_tab = ff_dca_channel_reorder_lfe_xch[s->amode];
} else {
- s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode];
+ s->channel_order_tab = ff_dca_channel_reorder_nolfe_xch[s->amode];
}
} else {
channels = num_core_channels + !!s->lfe;
s->xch_present = 0; /* disable further xch processing */
if (s->lfe) {
avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
- s->channel_order_tab = dca_channel_reorder_lfe[s->amode];
+ s->channel_order_tab = ff_dca_channel_reorder_lfe[s->amode];
} else
- s->channel_order_tab = dca_channel_reorder_nolfe[s->amode];
+ s->channel_order_tab = ff_dca_channel_reorder_nolfe[s->amode];
}
if (channels > !!s->lfe &&
s->output = s->core_downmix_amode;
} else {
int am = s->amode & DCA_CHANNEL_MASK;
- if (am >= FF_ARRAY_ELEMS(dca_default_coeffs)) {
+ if (am >= FF_ARRAY_ELEMS(ff_dca_default_coeffs)) {
av_log(s->avctx, AV_LOG_ERROR,
"Invalid channel mode %d\n", am);
return AVERROR_INVALIDDATA;
}
if (num_core_channels + !!s->lfe >
- FF_ARRAY_ELEMS(dca_default_coeffs[0])) {
+ FF_ARRAY_ELEMS(ff_dca_default_coeffs[0])) {
avpriv_request_sample(s->avctx, "Downmixing %d channels",
s->prim_channels + !!s->lfe);
return AVERROR_PATCHWELCOME;
}
for (i = 0; i < num_core_channels + !!s->lfe; i++) {
- s->downmix_coef[i][0] = dca_default_coeffs[am][i][0];
- s->downmix_coef[i][1] = dca_default_coeffs[am][i][1];
+ s->downmix_coef[i][0] = ff_dca_default_coeffs[am][i][0];
+ s->downmix_coef[i][1] = ff_dca_default_coeffs[am][i][1];
}
}
- av_dlog(s->avctx, "Stereo downmix coeffs:\n");
+ ff_dlog(s->avctx, "Stereo downmix coeffs:\n");
for (i = 0; i < num_core_channels + !!s->lfe; i++) {
- av_dlog(s->avctx, "L, input channel %d = %f\n", i,
+ ff_dlog(s->avctx, "L, input channel %d = %f\n", i,
s->downmix_coef[i][0]);
- av_dlog(s->avctx, "R, input channel %d = %f\n", i,
+ ff_dlog(s->avctx, "R, input channel %d = %f\n", i,
s->downmix_coef[i][1]);
}
- av_dlog(s->avctx, "\n");
+ ff_dlog(s->avctx, "\n");
}
} else {
av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode);
return AVERROR_INVALIDDATA;
}
+
+ return 0;
+}
+
+/**
+ * Main frame decoding function
+ * FIXME add arguments
+ */
+static int dca_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
+{
+ AVFrame *frame = data;
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
+
+ int lfe_samples;
+ int num_core_channels = 0;
+ int i, ret;
+ float **samples_flt;
+ DCAContext *s = avctx->priv_data;
+ int channels, full_channels;
+ int upsample = 0;
+
+ s->exss_ext_mask = 0;
+ s->xch_present = 0;
+
+ s->dca_buffer_size = ff_dca_convert_bitstream(buf, buf_size, s->dca_buffer,
+ DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);
+ if (s->dca_buffer_size == AVERROR_INVALIDDATA) {
+ av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ if ((ret = dca_parse_frame_header(s)) < 0) {
+ // seems like the frame is corrupt, try with the next one
+ return ret;
+ }
+ // set AVCodec values with parsed data
+ avctx->sample_rate = s->sample_rate;
+ avctx->bit_rate = s->bit_rate;
+
+ s->profile = FF_PROFILE_DTS;
+
+ for (i = 0; i < (s->sample_blocks / SAMPLES_PER_SUBBAND); 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;
+
+ if (s->ext_coding)
+ s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];
+ else
+ s->core_ext_mask = 0;
+
+ ret = scan_for_extensions(avctx);
+
+ avctx->profile = s->profile;
+
+ full_channels = channels = s->prim_channels + !!s->lfe;
+
+ ret = set_channel_layout(avctx, channels, num_core_channels);
+ if (ret < 0)
+ return ret;
avctx->channels = channels;
/* get output buffer */
- frame->nb_samples = 256 * (s->sample_blocks / 8);
+ frame->nb_samples = 256 * (s->sample_blocks / SAMPLES_PER_SUBBAND);
+ if (s->exss_ext_mask & DCA_EXT_EXSS_XLL) {
+ int xll_nb_samples = s->xll_segments * s->xll_smpl_in_seg;
+ /* Check for invalid/unsupported conditions first */
+ if (s->xll_residual_channels > channels) {
+ av_log(s->avctx, AV_LOG_WARNING,
+ "DCA: too many residual channels (%d, core channels %d). Disabling XLL\n",
+ s->xll_residual_channels, channels);
+ s->exss_ext_mask &= ~DCA_EXT_EXSS_XLL;
+ } else if (xll_nb_samples != frame->nb_samples &&
+ 2 * frame->nb_samples != xll_nb_samples) {
+ av_log(s->avctx, AV_LOG_WARNING,
+ "DCA: unsupported upsampling (%d XLL samples, %d core samples). Disabling XLL\n",
+ xll_nb_samples, frame->nb_samples);
+ s->exss_ext_mask &= ~DCA_EXT_EXSS_XLL;
+ } else {
+ if (2 * frame->nb_samples == xll_nb_samples) {
+ av_log(s->avctx, AV_LOG_INFO,
+ "XLL: upsampling core channels by a factor of 2\n");
+ upsample = 1;
+
+ frame->nb_samples = xll_nb_samples;
+ // FIXME: Is it good enough to copy from the first channel set?
+ avctx->sample_rate = s->xll_chsets[0].sampling_frequency;
+ }
+ /* If downmixing to stereo, don't decode additional channels.
+ * FIXME: Using the xch_disable flag for this doesn't seem right. */
+ if (!s->xch_disable)
+ avctx->channels += s->xll_channels - s->xll_residual_channels;
+ }
+ }
+
+ /* FIXME: This is an ugly hack, to just revert to the default
+ * layout if we have additional channels. Need to convert the XLL
+ * channel masks to libav channel_layout mask. */
+ if (av_get_channel_layout_nb_channels(avctx->channel_layout) != avctx->channels)
+ avctx->channel_layout = 0;
+
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
/* filter to get final output */
- for (i = 0; i < (s->sample_blocks / 8); i++) {
+ for (i = 0; i < (s->sample_blocks / SAMPLES_PER_SUBBAND); i++) {
int ch;
-
+ unsigned block = upsample ? 512 : 256;
for (ch = 0; ch < channels; ch++)
- s->samples_chanptr[ch] = samples_flt[ch] + i * 256;
+ s->samples_chanptr[ch] = samples_flt[ch] + i * block;
for (; ch < full_channels; ch++)
- s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * 256;
+ s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * block;
- dca_filter_channels(s, i);
+ dca_filter_channels(s, i, upsample);
/* If this was marked as a DTS-ES stream we need to subtract back- */
/* channel from SL & SR to remove matrixed back-channel signal */
}
/* update lfe history */
- lfe_samples = 2 * s->lfe * (s->sample_blocks / 8);
+ lfe_samples = 2 * s->lfe * (s->sample_blocks / SAMPLES_PER_SUBBAND);
for (i = 0; i < 2 * s->lfe * 4; i++)
s->lfe_data[i] = s->lfe_data[i + lfe_samples];
+ if (s->exss_ext_mask & DCA_EXT_EXSS_XLL) {
+ ret = ff_dca_xll_decode_audio(s, frame);
+ if (ret < 0)
+ return ret;
+ }
/* AVMatrixEncoding
*
* DCA_STEREO_TOTAL (Lt/Rt) is equivalent to Dolby Surround */
s->avctx = avctx;
dca_init_vlcs();
- avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
+ avpriv_float_dsp_init(&s->fdsp, avctx->flags & AV_CODEC_FLAG_BITEXACT);
ff_mdct_init(&s->imdct, 6, 1, 1.0);
ff_synth_filter_init(&s->synth);
ff_dcadsp_init(&s->dcadsp);
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
/* allow downmixing to stereo */
-#if FF_API_REQUEST_CHANNELS
-FF_DISABLE_DEPRECATION_WARNINGS
- if (avctx->request_channels == 2)
- avctx->request_channel_layout = AV_CH_LAYOUT_STEREO;
-FF_ENABLE_DEPRECATION_WARNINGS
-#endif
if (avctx->channels > 2 &&
avctx->request_channel_layout == AV_CH_LAYOUT_STEREO)
avctx->channels = 2;
DCAContext *s = avctx->priv_data;
ff_mdct_end(&s->imdct);
av_freep(&s->extra_channels_buffer);
+ av_freep(&s->xll_sample_buf);
+ av_freep(&s->qmf64_table);
return 0;
}
static const AVOption options[] = {
{ "disable_xch", "disable decoding of the XCh extension", offsetof(DCAContext, xch_disable), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM },
+ { "disable_xll", "disable decoding of the XLL extension", offsetof(DCAContext, xll_disable), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM },
{ NULL },
};
.init = dca_decode_init,
.decode = dca_decode_frame,
.close = dca_decode_end,
- .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
+ .capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.profiles = NULL_IF_CONFIG_SMALL(profiles),
projects / ffmpeg / blobdiff