X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fdca.c;h=e8627a23d95830872d3cb867b4eb31b35b6ccfcc;hb=dd5f3238c13e2a6a2caf4d550113219d73122fde;hp=8dfec8e9a913df08d1d635445eea7a52377321d8;hpb=7a00bbad2100367481240e62876b941b5c4befdc;p=ffmpeg diff --git a/libavcodec/dca.c b/libavcodec/dca.c index 8dfec8e9a91..e8627a23d95 100644 --- a/libavcodec/dca.c +++ b/libavcodec/dca.c @@ -22,27 +22,31 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -/** - * @file libavcodec/dca.c - */ - #include #include #include +#include "libavutil/intmath.h" +#include "libavutil/intreadwrite.h" #include "avcodec.h" #include "dsputil.h" -#include "bitstream.h" +#include "fft.h" +#include "get_bits.h" +#include "put_bits.h" #include "dcadata.h" #include "dcahuff.h" #include "dca.h" +#include "synth_filter.h" +#include "dcadsp.h" //#define TRACE -#define DCA_PRIM_CHANNELS_MAX (5) +#define DCA_PRIM_CHANNELS_MAX (7) #define DCA_SUBBANDS (32) #define DCA_ABITS_MAX (32) /* Should be 28 */ -#define DCA_SUBSUBFAMES_MAX (4) +#define DCA_SUBSUBFRAMES_MAX (4) +#define DCA_SUBFRAMES_MAX (16) +#define DCA_BLOCKS_MAX (16) #define DCA_LFE_MAX (3) enum DCAMode { @@ -92,44 +96,81 @@ 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[][8] = { - { 0, -1, -1, -1, -1, -1, -1, -1}, - { 0, 1, -1, -1, -1, -1, -1, -1}, - { 0, 1, -1, -1, -1, -1, -1, -1}, - { 0, 1, -1, -1, -1, -1, -1, -1}, - { 0, 1, -1, -1, -1, -1, -1, -1}, - { 2, 0, 1, -1, -1, -1, -1, -1}, - { 0, 1, 3, -1, -1, -1, -1, -1}, - { 2, 0, 1, 4, -1, -1, -1, -1}, - { 0, 1, 3, 4, -1, -1, -1, -1}, - { 2, 0, 1, 4, 5, -1, -1, -1}, - { 3, 4, 0, 1, 5, 6, -1, -1}, - { 2, 0, 1, 4, 5, 6, -1, -1}, - { 0, 6, 4, 5, 2, 3, -1, -1}, - { 4, 2, 5, 0, 1, 6, 7, -1}, - { 5, 6, 0, 1, 7, 3, 8, 4}, - { 4, 2, 5, 0, 1, 6, 8, 7}, +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_nolfe[][8] = { - { 0, -1, -1, -1, -1, -1, -1, -1}, - { 0, 1, -1, -1, -1, -1, -1, -1}, - { 0, 1, -1, -1, -1, -1, -1, -1}, - { 0, 1, -1, -1, -1, -1, -1, -1}, - { 0, 1, -1, -1, -1, -1, -1, -1}, - { 2, 0, 1, -1, -1, -1, -1, -1}, - { 0, 1, 2, -1, -1, -1, -1, -1}, - { 2, 0, 1, 3, -1, -1, -1, -1}, - { 0, 1, 2, 3, -1, -1, -1, -1}, - { 2, 0, 1, 3, 4, -1, -1, -1}, - { 2, 3, 0, 1, 4, 5, -1, -1}, - { 2, 0, 1, 3, 4, 5, -1, -1}, - { 0, 5, 3, 4, 1, 2, -1, -1}, - { 3, 2, 4, 0, 1, 5, 6, -1}, - { 4, 5, 0, 1, 6, 2, 7, 3}, - { 3, 2, 4, 0, 1, 5, 7, 6}, +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 */ @@ -206,8 +247,8 @@ typedef struct { 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 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 @@ -220,22 +261,23 @@ typedef struct { 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 + float lfe_data[2 * DCA_LFE_MAX * (DCA_BLOCKS_MAX + 4)]; ///< Low frequency effect data int lfe_scale_factor; /* Subband samples history (for ADPCM) */ float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4]; - DECLARE_ALIGNED_16(float, subband_fir_hist[DCA_PRIM_CHANNELS_MAX][512]); - float subband_fir_noidea[DCA_PRIM_CHANNELS_MAX][32]; + DECLARE_ALIGNED(16, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512]; + DECLARE_ALIGNED(16, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32]; int hist_index[DCA_PRIM_CHANNELS_MAX]; + DECLARE_ALIGNED(16, float, raXin)[32]; int output; ///< type of output float add_bias; ///< output bias float scale_bias; ///< output scale - DECLARE_ALIGNED_16(float, samples[1536]); /* 6 * 256 = 1536, might only need 5 */ - const float *samples_chanptr[6]; + DECLARE_ALIGNED(16, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8]; + DECLARE_ALIGNED(16, float, samples)[(DCA_PRIM_CHANNELS_MAX+1)*256]; + const float *samples_chanptr[DCA_PRIM_CHANNELS_MAX+1]; uint8_t dca_buffer[DCA_MAX_FRAME_SIZE]; int dca_buffer_size; ///< how much data is in the dca_buffer @@ -246,47 +288,75 @@ typedef struct { int current_subframe; int current_subsubframe; + /* XCh extension information */ + int xch_present; + int xch_base_channel; ///< index of first (only) channel containing XCH data + int debug_flag; ///< used for suppressing repeated error messages output DSPContext dsp; - MDCTContext imdct; + FFTContext imdct; + SynthFilterContext synth; + DCADSPContext dcadsp; } 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; - int i, j; + int i, j, c = 14; + static VLC_TYPE dca_table[23622][2]; if (vlcs_initialized) return; dca_bitalloc_index.offset = 1; dca_bitalloc_index.wrap = 2; - for (i = 0; i < 5; i++) + 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]; 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); + 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++) + 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]; init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129, scales_bits[i], 1, 1, - scales_codes[i], 2, 2, 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++) + 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]; init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4, tmode_bits[i], 1, 1, - tmode_codes[i], 2, 2, 1); + tmode_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC); + } - for(i = 0; i < 10; i++) - for(j = 0; j < 7; j++){ - if(!bitalloc_codes[i][j]) break; + for (i = 0; i < 10; i++) + for (j = 0; j < 7; j++){ + if (!bitalloc_codes[i][j]) break; dca_smpl_bitalloc[i+1].offset = bitalloc_offsets[i]; dca_smpl_bitalloc[i+1].wrap = 1 + (j > 4); + dca_smpl_bitalloc[i+1].vlc[j].table = &dca_table[dca_vlc_offs[c]]; + dca_smpl_bitalloc[i+1].vlc[j].table_allocated = dca_vlc_offs[c + 1] - dca_vlc_offs[c]; init_vlc(&dca_smpl_bitalloc[i+1].vlc[j], bitalloc_maxbits[i][j], bitalloc_sizes[i], bitalloc_bits[i][j], 1, 1, - bitalloc_codes[i][j], 2, 2, 1); + bitalloc_codes[i][j], 2, 2, INIT_VLC_USE_NEW_STATIC); + c++; } vlcs_initialized = 1; } @@ -297,13 +367,87 @@ static inline void get_array(GetBitContext *gb, int *dst, int len, int bits) *dst++ = get_bits(gb, bits); } -static int dca_parse_frame_header(DCAContext * s) +static int dca_parse_audio_coding_header(DCAContext * s, int base_channel) { int i, j; static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 }; static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 }; static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 }; + s->total_channels = get_bits(&s->gb, 3) + 1 + base_channel; + s->prim_channels = s->total_channels; + + if (s->prim_channels > DCA_PRIM_CHANNELS_MAX) + s->prim_channels = DCA_PRIM_CHANNELS_MAX; + + + for (i = base_channel; 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 = base_channel; 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 + base_channel, s->prim_channels - base_channel, 3); + get_array(&s->gb, s->transient_huffman + base_channel, s->prim_channels - base_channel, 2); + get_array(&s->gb, s->scalefactor_huffman + base_channel, s->prim_channels - base_channel, 3); + get_array(&s->gb, s->bitalloc_huffman + base_channel, s->prim_channels - base_channel, 3); + + /* Get codebooks quantization indexes */ + if (!base_channel) + memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman)); + for (j = 1; j < 11; j++) + for (i = base_channel; 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 = base_channel; i < s->prim_channels; i++) + s->scalefactor_adj[i][j] = 1; + + for (j = 1; j < 11; j++) + for (i = base_channel; 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 = 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; +} + +static int dca_parse_frame_header(DCAContext * s) +{ init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8); /* Sync code */ @@ -351,7 +495,7 @@ static int dca_parse_frame_header(DCAContext * s) /* FIXME: channels mixing levels */ s->output = s->amode; - if(s->lfe) s->output |= DCA_LFE; + if (s->lfe) s->output |= DCA_LFE; #ifdef TRACE av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type); @@ -393,74 +537,8 @@ static int dca_parse_frame_header(DCAContext * s) /* Primary audio coding header */ s->subframes = get_bits(&s->gb, 4) + 1; - s->total_channels = get_bits(&s->gb, 3) + 1; - s->prim_channels = s->total_channels; - if (s->prim_channels > DCA_PRIM_CHANNELS_MAX) - s->prim_channels = DCA_PRIM_CHANNELS_MAX; /* We only support DTS core */ - - - 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; + return dca_parse_audio_coding_header(s, 0); } @@ -469,25 +547,28 @@ 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) + } else if (level < 8) value = get_bits(gb, level + 1); return value; } -static int dca_subframe_header(DCAContext * s) +static int dca_subframe_header(DCAContext * s, int base_channel, int block_index) { /* 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++) { + if (!base_channel) { + s->subsubframes[s->current_subframe] = get_bits(&s->gb, 2) + 1; + s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3); + } + + for (j = base_channel; 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 (j = base_channel; 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) */ @@ -497,7 +578,7 @@ static int dca_subframe_header(DCAContext * s) } /* Bit allocation index */ - for (j = 0; j < s->prim_channels; j++) { + for (j = base_channel; 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); @@ -521,10 +602,10 @@ static int dca_subframe_header(DCAContext * s) } /* Transition mode */ - for (j = 0; j < s->prim_channels; j++) { + for (j = base_channel; j < s->prim_channels; j++) { for (k = 0; k < s->subband_activity[j]; k++) { s->transition_mode[j][k] = 0; - if (s->subsubframes > 1 && + if (s->subsubframes[s->current_subframe] > 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]); @@ -532,7 +613,7 @@ static int dca_subframe_header(DCAContext * s) } } - for (j = 0; j < s->prim_channels; j++) { + for (j = base_channel; j < s->prim_channels; j++) { const uint32_t *scale_table; int scale_sum; @@ -561,14 +642,14 @@ static int dca_subframe_header(DCAContext * s) } /* Joint subband scale factor codebook select */ - for (j = 0; j < s->prim_channels; j++) { + for (j = base_channel; 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++) { + for (j = base_channel; j < s->prim_channels; j++) { int source_channel; /* Transmitted only if joint subband coding enabled */ @@ -585,7 +666,7 @@ static int dca_subframe_header(DCAContext * s) s->joint_scale_factor[j][k] = scale; /*joint_scale_table[scale]; */ } - if (!s->debug_flag & 0x02) { + if (!(s->debug_flag & 0x02)) { av_log(s->avctx, AV_LOG_DEBUG, "Joint stereo coding not supported\n"); s->debug_flag |= 0x02; @@ -594,15 +675,15 @@ static int dca_subframe_header(DCAContext * s) } /* Stereo downmix coefficients */ - if (s->prim_channels > 2) { - if(s->downmix) { - for (j = 0; j < s->prim_channels; j++) { + if (!base_channel && s->prim_channels > 2) { + if (s->downmix) { + for (j = base_channel; 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++) { + for (j = base_channel; 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]; } @@ -610,7 +691,7 @@ static int dca_subframe_header(DCAContext * s) } /* Dynamic range coefficient */ - if (s->dynrange) + if (!base_channel && s->dynrange) s->dynrange_coef = get_bits(&s->gb, 8); /* Side information CRC check word */ @@ -623,18 +704,19 @@ static int dca_subframe_header(DCAContext * s) */ /* VQ encoded high frequency subbands */ - for (j = 0; j < s->prim_channels; j++) + for (j = base_channel; 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) { + if (!base_channel && s->lfe) { /* LFE samples */ - int lfe_samples = 2 * s->lfe * s->subsubframes; + int lfe_samples = 2 * s->lfe * (4 + block_index); + int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]); float lfe_scale; - for (j = lfe_samples; j < lfe_samples * 2; j++) { + for (j = lfe_samples; j < lfe_end_sample; j++) { /* Signed 8 bits int */ s->lfe_data[j] = get_sbits(&s->gb, 8); } @@ -645,21 +727,21 @@ static int dca_subframe_header(DCAContext * s) /* Quantization step size * scale factor */ lfe_scale = 0.035 * s->lfe_scale_factor; - for (j = lfe_samples; j < lfe_samples * 2; j++) + for (j = lfe_samples; j < lfe_end_sample; 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, "subsubframes: %i\n", s->subsubframes[s->current_subframe]); av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n", - s->partial_samples); - for (j = 0; j < s->prim_channels; j++) { + 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 = 0; j < s->prim_channels; j++) { + 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", @@ -668,19 +750,19 @@ static int dca_subframe_header(DCAContext * s) (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++) { + 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 = 0; j < s->prim_channels; j++) { + 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 = 0; j < s->prim_channels; j++) { + 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) @@ -690,7 +772,7 @@ static int dca_subframe_header(DCAContext * s) } av_log(s->avctx, AV_LOG_DEBUG, "\n"); } - for (j = 0; j < s->prim_channels; j++) { + 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"); @@ -699,7 +781,7 @@ static int dca_subframe_header(DCAContext * s) av_log(s->avctx, AV_LOG_DEBUG, "\n"); } } - if (s->prim_channels > 2 && s->downmix) { + if (!base_channel && s->prim_channels > 2 && s->downmix) { av_log(s->avctx, AV_LOG_DEBUG, "Downmix coeffs:\n"); for (j = 0; j < s->prim_channels; j++) { av_log(s->avctx, AV_LOG_DEBUG, "Channel 0,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][0]]); @@ -707,13 +789,15 @@ static int dca_subframe_header(DCAContext * s) } av_log(s->avctx, AV_LOG_DEBUG, "\n"); } - for (j = 0; j < s->prim_channels; j++) + 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(s->lfe){ - int lfe_samples = 2 * s->lfe * s->subsubframes; + 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_samples * 2; j++) + 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"); } @@ -727,12 +811,9 @@ static void qmf_32_subbands(DCAContext * s, int chans, float scale, float bias) { const float *prCoeff; - int i, j; - DECLARE_ALIGNED_16(float, raXin[32]); - - int hist_index= s->hist_index[chans]; - float *subband_fir_hist2 = s->subband_fir_noidea[chans]; + int i; + int sb_act = s->subband_activity[chans]; int subindex; scale *= sqrt(1/8.0); @@ -745,48 +826,24 @@ static void qmf_32_subbands(DCAContext * s, int chans, /* Reconstructed channel sample index */ for (subindex = 0; subindex < 8; subindex++) { - float *subband_fir_hist = s->subband_fir_hist[chans] + hist_index; /* Load in one sample from each subband and clear inactive subbands */ - for (i = 0; i < s->subband_activity[chans]; i++){ - if((i-1)&2) raXin[i] = -samples_in[i][subindex]; - else raXin[i] = samples_in[i][subindex]; + for (i = 0; i < sb_act; i++){ + uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ ((i-1)&2)<<30; + AV_WN32A(&s->raXin[i], v); } for (; i < 32; i++) - raXin[i] = 0.0; - - ff_imdct_half(&s->imdct, subband_fir_hist, raXin); - - /* Multiply by filter coefficients */ - for (i = 0; i < 16; i++){ - float a= subband_fir_hist2[i ]; - float b= subband_fir_hist2[i+16]; - float c= 0; - float d= 0; - for (j = 0; j < 512-hist_index; j += 64){ - a += prCoeff[i+j ]*(-subband_fir_hist[15-i+j]); - b += prCoeff[i+j+16]*( subband_fir_hist[ i+j]); - c += prCoeff[i+j+32]*( subband_fir_hist[16+i+j]); - d += prCoeff[i+j+48]*( subband_fir_hist[31-i+j]); - } - for ( ; j < 512; j += 64){ - a += prCoeff[i+j ]*(-subband_fir_hist[15-i+j-512]); - b += prCoeff[i+j+16]*( subband_fir_hist[ i+j-512]); - c += prCoeff[i+j+32]*( subband_fir_hist[16+i+j-512]); - d += prCoeff[i+j+48]*( subband_fir_hist[31-i+j-512]); - } - samples_out[i ] = a * scale + bias; - samples_out[i+16] = b * scale + bias; - subband_fir_hist2[i ] = c; - subband_fir_hist2[i+16] = d; - } + s->raXin[i] = 0.0; + + s->synth.synth_filter_float(&s->imdct, + s->subband_fir_hist[chans], &s->hist_index[chans], + s->subband_fir_noidea[chans], prCoeff, + samples_out, s->raXin, scale, bias); samples_out+= 32; - hist_index = (hist_index-32)&511; } - s->hist_index[chans]= hist_index; } -static void lfe_interpolation_fir(int decimation_select, +static void lfe_interpolation_fir(DCAContext *s, int decimation_select, int num_deci_sample, float *samples_in, float *samples_out, float scale, float bias) @@ -799,61 +856,59 @@ static void lfe_interpolation_fir(int decimation_select, * samples_out: An array holding interpolated samples */ - int decifactor, k, j; + int decifactor; const float *prCoeff; - - int interp_index = 0; /* Index to the interpolated samples */ int deciindex; /* Select decimation filter */ if (decimation_select == 1) { - decifactor = 128; + decifactor = 64; prCoeff = lfe_fir_128; } else { - decifactor = 64; + decifactor = 32; 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; - } + s->dcadsp.lfe_fir(samples_out, samples_in, prCoeff, decifactor, + scale, bias); + samples_in++; + samples_out += 2 * decifactor; } } /* downmixing routines */ #define MIX_REAR1(samples, si1, rs, coef) \ - samples[i] += samples[si1] * coef[rs][0]; \ - samples[i+256] += samples[si1] * coef[rs][1]; + samples[i] += (samples[si1] - add_bias) * coef[rs][0]; \ + samples[i+256] += (samples[si1] - add_bias) * coef[rs][1]; #define MIX_REAR2(samples, si1, si2, rs, coef) \ - samples[i] += samples[si1] * coef[rs][0] + samples[si2] * coef[rs+1][0]; \ - samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs+1][1]; + samples[i] += (samples[si1] - add_bias) * coef[rs][0] + (samples[si2] - add_bias) * coef[rs+1][0]; \ + samples[i+256] += (samples[si1] - add_bias) * coef[rs][1] + (samples[si2] - add_bias) * coef[rs+1][1]; #define MIX_FRONT3(samples, coef) \ - t = samples[i]; \ - 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]; + t = samples[i+c] - add_bias; \ + u = samples[i+l] - add_bias; \ + v = samples[i+r] - add_bias; \ + samples[i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0] + add_bias; \ + samples[i+256] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1] + add_bias; #define DOWNMIX_TO_STEREO(op1, op2) \ - for(i = 0; i < 256; i++){ \ + 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 downmix_coef[DCA_PRIM_CHANNELS_MAX][2], + const int8_t *channel_mapping, float add_bias) { + int c,l,r,sl,sr,s; int i; - float t; + float t, u, v; float coef[DCA_PRIM_CHANNELS_MAX][2]; - for(i=0; i> 1; for (i = 0; i < 4; i++) { - values[i] = (code % levels) - offset; - code /= levels; + int div = FASTDIV(code, levels); + values[i] = code - offset - div*levels; + code = div; } if (code == 0) @@ -912,7 +983,7 @@ static int decode_blockcode(int code, int levels, int *values) 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) +static int dca_subsubframe(DCAContext * s, int base_channel, int block_index) { int k, l; int subsubframe = s->current_subsubframe; @@ -920,7 +991,8 @@ static int dca_subsubframe(DCAContext * s) const float *quant_step_table; /* FIXME */ - float subband_samples[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8]; + float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index]; + LOCAL_ALIGNED_16(int, block, [8]); /* * Audio data @@ -932,7 +1004,7 @@ static int dca_subsubframe(DCAContext * s) else quant_step_table = lossy_quant_d; - for (k = 0; k < s->prim_channels; k++) { + for (k = base_channel; k < s->prim_channels; k++) { for (l = 0; l < s->vq_start_subband[k]; l++) { int m; @@ -940,7 +1012,6 @@ static int dca_subsubframe(DCAContext * s) int abits = s->bitalloc[k][l]; float quant_step_size = quant_step_table[abits]; - float rscale; /* * Determine quantization index code book and its type @@ -952,46 +1023,40 @@ static int dca_subsubframe(DCAContext * s) /* * Extract bits from the bit stream */ - if(!abits){ + 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 { + /* Deal with transients */ + int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l]; + float rscale = quant_step_size * s->scale_factor[k][l][sfi] * s->scalefactor_adj[k][sel]; + + if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){ + if (abits <= 7){ + /* Block code */ + int block_code1, block_code2, size, levels; + + size = abits_sizes[abits-1]; + levels = abits_levels[abits-1]; + + block_code1 = get_bits(&s->gb, size); + /* FIXME Should test return value */ + decode_blockcode(block_code1, levels, block); + block_code2 = get_bits(&s->gb, size); + decode_blockcode(block_code2, levels, &block[4]); + }else{ + /* no coding */ + for (m = 0; m < 8; m++) + block[m] = get_sbits(&s->gb, abits - 3); + } }else{ - /* no coding */ + /* Huffman coded */ for (m = 0; m < 8; m++) - subband_samples[k][l][m] = get_sbits(&s->gb, abits - 3); + block[m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel); } - }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; + s->dsp.int32_to_float_fmul_scalar(subband_samples[k][l], + block, rscale, 8); + } /* * Inverse ADPCM if in prediction mode @@ -1036,7 +1101,7 @@ static int dca_subsubframe(DCAContext * s) } /* Check for DSYNC after subsubframe */ - if (s->aspf || subsubframe == s->subsubframes - 1) { + 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"); @@ -1047,11 +1112,19 @@ static int dca_subsubframe(DCAContext * s) } /* Backup predictor history for adpcm */ - for (k = 0; k < s->prim_channels; k++) + for (k = base_channel; k < s->prim_channels; k++) for (l = 0; l < s->vq_start_subband[k]; l++) memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4], 4 * sizeof(subband_samples[0][0][0])); + return 0; +} + +static int dca_filter_channels(DCAContext * s, int block_index) +{ + float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index]; + int k; + /* 32 subbands QMF */ for (k = 0; k < s->prim_channels; k++) { /* static float pcm_to_double[8] = @@ -1062,18 +1135,14 @@ static int dca_subsubframe(DCAContext * s) } /* Down mixing */ - - if (s->prim_channels > dca_channels[s->output & DCA_CHANNEL_MASK]) { - dca_downmix(s->samples, s->amode, s->downmix_coef); + if (s->avctx->request_channels == 2 && s->prim_channels > 2) { + dca_downmix(s->samples, s->amode, s->downmix_coef, s->channel_order_tab, s->add_bias); } /* Generate LFE samples for this subsubframe FIXME!!! */ if (s->output & DCA_LFE) { - int lfe_samples = 2 * s->lfe * s->subsubframes; - - lfe_interpolation_fir(s->lfe, 2 * s->lfe, - s->lfe_data + lfe_samples + - 2 * s->lfe * subsubframe, + lfe_interpolation_fir(s, s->lfe, 2 * s->lfe, + s->lfe_data + 2 * s->lfe * (block_index + 4), &s->samples[256 * dca_lfe_index[s->amode]], (1.0/256.0)*s->scale_bias, s->add_bias); /* Outputs 20bits pcm samples */ @@ -1083,30 +1152,27 @@ static int dca_subsubframe(DCAContext * s) } -static int dca_subframe_footer(DCAContext * s) +static int dca_subframe_footer(DCAContext * s, int base_channel) { 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); + /* presumably optional information only appears in the core? */ + if (!base_channel) { + if (s->timestamp) + get_bits(&s->gb, 32); - for (i = 0; i < aux_data_count; i++) - get_bits(&s->gb, 8); + if (s->aux_data) + aux_data_count = get_bits(&s->gb, 6); - if (s->crc_present && (s->downmix || s->dynrange)) - get_bits(&s->gb, 16); + for (i = 0; i < aux_data_count; i++) + get_bits(&s->gb, 8); - lfe_samples = 2 * s->lfe * s->subsubframes; - for (i = 0; i < lfe_samples; i++) { - s->lfe_data[i] = s->lfe_data[i + lfe_samples]; + if (s->crc_present && (s->downmix || s->dynrange)) + get_bits(&s->gb, 16); } return 0; @@ -1118,7 +1184,7 @@ static int dca_subframe_footer(DCAContext * s) * @param s pointer to the DCAContext */ -static int dca_decode_block(DCAContext * s) +static int dca_decode_block(DCAContext * s, int base_channel, int block_index) { /* Sanity check */ @@ -1133,7 +1199,7 @@ static int dca_decode_block(DCAContext * s) av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n"); #endif /* Read subframe header */ - if (dca_subframe_header(s)) + if (dca_subframe_header(s, base_channel, block_index)) return -1; } @@ -1141,12 +1207,12 @@ static int dca_decode_block(DCAContext * s) #ifdef TRACE av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n"); #endif - if (dca_subsubframe(s)) + if (dca_subsubframe(s, base_channel, block_index)) return -1; /* Update state */ s->current_subsubframe++; - if (s->current_subsubframe >= s->subsubframes) { + if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) { s->current_subsubframe = 0; s->current_subframe++; } @@ -1155,7 +1221,7 @@ static int dca_decode_block(DCAContext * s) av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n"); #endif /* Read subframe footer */ - if (dca_subframe_footer(s)) + if (dca_subframe_footer(s, base_channel)) return -1; } @@ -1174,7 +1240,7 @@ static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * ds uint16_t *sdst = (uint16_t *) dst; PutBitContext pb; - if((unsigned)src_size > (unsigned)max_size) { + if ((unsigned)src_size > (unsigned)max_size) { // av_log(NULL, AV_LOG_ERROR, "Input frame size larger then DCA_MAX_FRAME_SIZE!\n"); // return -1; src_size = max_size; @@ -1187,7 +1253,7 @@ static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * ds return src_size; case DCA_MARKER_RAW_LE: for (i = 0; i < (src_size + 1) >> 1; i++) - *sdst++ = bswap_16(*ssrc++); + *sdst++ = av_bswap16(*ssrc++); return src_size; case DCA_MARKER_14B_BE: case DCA_MARKER_14B_LE: @@ -1214,12 +1280,15 @@ static int dca_decode_frame(AVCodecContext * avctx, const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; + int lfe_samples; + int num_core_channels = 0; int i; int16_t *samples = data; DCAContext *s = avctx->priv_data; int channels; + s->xch_present = 0; 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 valid DCA frame\n"); @@ -1236,18 +1305,91 @@ static int dca_decode_frame(AVCodecContext * avctx, avctx->sample_rate = s->sample_rate; avctx->bit_rate = s->bit_rate; + for (i = 0; i < (s->sample_blocks / 8); i++) { + dca_decode_block(s, 0, i); + } + + /* record number of core channels incase less than max channels are requested */ + num_core_channels = s->prim_channels; + + /* extensions start at 32-bit boundaries into bitstream */ + skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31); + + while(get_bits_left(&s->gb) >= 32) { + uint32_t bits = get_bits_long(&s->gb, 32); + + switch(bits) { + case 0x5a5a5a5a: { + int ext_amode, xch_fsize; + + s->xch_base_channel = s->prim_channels; + + /* validate sync word using XCHFSIZE field */ + xch_fsize = show_bits(&s->gb, 10); + if((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) && + (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1)) + continue; + + /* skip length-to-end-of-frame field for the moment */ + skip_bits(&s->gb, 10); + + /* extension amode should == 1, number of channels in extension */ + /* AFAIK XCh is not used for more channels */ + if ((ext_amode = get_bits(&s->gb, 4)) != 1) { + av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not" + " supported!\n",ext_amode); + continue; + } + + /* much like core primary audio coding header */ + dca_parse_audio_coding_header(s, s->xch_base_channel); + + for (i = 0; i < (s->sample_blocks / 8); i++) { + dca_decode_block(s, s->xch_base_channel, i); + } + + s->xch_present = 1; + break; + } + case 0x1d95f262: + av_log(avctx, AV_LOG_DEBUG, "Possible X96 extension found at %d bits\n", get_bits_count(&s->gb)); + av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", get_bits(&s->gb, 12)+1); + av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4)); + break; + } + + skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31); + } + channels = s->prim_channels + !!s->lfe; if (s->amode<16) { avctx->channel_layout = dca_core_channel_layout[s->amode]; - if (s->lfe) { - avctx->channel_layout |= CH_LOW_FREQUENCY; - s->channel_order_tab = dca_channel_reorder_lfe[s->amode]; - } else - s->channel_order_tab = dca_channel_reorder_nolfe[s->amode]; + if (s->xch_present && (!avctx->request_channels || + avctx->request_channels > num_core_channels + !!s->lfe)) { + avctx->channel_layout |= CH_BACK_CENTER; + if (s->lfe) { + avctx->channel_layout |= CH_LOW_FREQUENCY; + s->channel_order_tab = dca_channel_reorder_lfe_xch[s->amode]; + } else { + s->channel_order_tab = 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 |= CH_LOW_FREQUENCY; + s->channel_order_tab = dca_channel_reorder_lfe[s->amode]; + } else + s->channel_order_tab = dca_channel_reorder_nolfe[s->amode]; + } + + if (channels > !!s->lfe && + s->channel_order_tab[channels - 1 - !!s->lfe] < 0) + return -1; - if(avctx->request_channels == 2 && s->prim_channels > 2) { + if (avctx->request_channels == 2 && s->prim_channels > 2) { channels = 2; s->output = DCA_STEREO; avctx->channel_layout = CH_LAYOUT_STEREO; @@ -1263,18 +1405,39 @@ static int dca_decode_frame(AVCodecContext * avctx, unset. Ideally during the first probe for channels the crc should be checked and only set avctx->channels when the crc is ok. Right now the decoder could set the channels based on a broken first frame.*/ - if (!avctx->channels) - avctx->channels = channels; + avctx->channels = channels; - if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels) + if (*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels) return -1; *data_size = 256 / 8 * s->sample_blocks * sizeof(int16_t) * channels; + + /* filter to get final output */ for (i = 0; i < (s->sample_blocks / 8); i++) { - dca_decode_block(s); + dca_filter_channels(s, i); + + /* If this was marked as a DTS-ES stream we need to subtract back- */ + /* channel from SL & SR to remove matrixed back-channel signal */ + if((s->source_pcm_res & 1) && s->xch_present) { + float* back_chan = s->samples + s->channel_order_tab[s->xch_base_channel] * 256; + float* lt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 2] * 256; + float* rt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256; + int j; + for(j = 0; j < 256; ++j) { + lt_chan[j] -= (back_chan[j] - s->add_bias) * M_SQRT1_2; + rt_chan[j] -= (back_chan[j] - s->add_bias) * M_SQRT1_2; + } + } + s->dsp.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels); samples += 256 * channels; } + /* update lfe history */ + lfe_samples = 2 * s->lfe * (s->sample_blocks / 8); + for (i = 0; i < 2 * s->lfe * 4; i++) { + s->lfe_data[i] = s->lfe_data[i + lfe_samples]; + } + return buf_size; } @@ -1295,13 +1458,15 @@ static av_cold int dca_decode_init(AVCodecContext * avctx) dca_init_vlcs(); dsputil_init(&s->dsp, avctx); - ff_mdct_init(&s->imdct, 6, 1); + ff_mdct_init(&s->imdct, 6, 1, 1.0); + ff_synth_filter_init(&s->synth); + ff_dcadsp_init(&s->dcadsp); - for(i = 0; i < 6; i++) + for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++) s->samples_chanptr[i] = s->samples + i * 256; avctx->sample_fmt = SAMPLE_FMT_S16; - if(s->dsp.float_to_int16 == ff_float_to_int16_c) { + if (s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) { s->add_bias = 385.0f; s->scale_bias = 1.0 / 32768.0; } else { @@ -1328,7 +1493,7 @@ static av_cold int dca_decode_end(AVCodecContext * avctx) AVCodec dca_decoder = { .name = "dca", - .type = CODEC_TYPE_AUDIO, + .type = AVMEDIA_TYPE_AUDIO, .id = CODEC_ID_DTS, .priv_data_size = sizeof(DCAContext), .init = dca_decode_init,