X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fdcadec.c;h=f3c397250c476f92efe3e05e94d5bd1bc6dd46c5;hb=ddda2cc43c85d466fe53926d7e3c8a78dde2fcda;hp=e9120a19075497c701884b8de54f7bed2862d1c9;hpb=0c7ade547ad8aeaee0e1afd5b6730087b6c97da2;p=ffmpeg diff --git a/libavcodec/dcadec.c b/libavcodec/dcadec.c index e9120a19075..f3c397250c4 100644 --- a/libavcodec/dcadec.c +++ b/libavcodec/dcadec.c @@ -1,11 +1,5 @@ /* - * DCA compatible decoder - * Copyright (C) 2004 Gildas Bazin - * 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 + * Copyright (C) 2016 foo86 * * This file is part of FFmpeg. * @@ -24,2039 +18,400 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#include -#include -#include - -#include "libavutil/attributes.h" -#include "libavutil/channel_layout.h" -#include "libavutil/common.h" -#include "libavutil/float_dsp.h" -#include "libavutil/internal.h" -#include "libavutil/intreadwrite.h" -#include "libavutil/mathematics.h" #include "libavutil/opt.h" -#include "libavutil/samplefmt.h" +#include "libavutil/channel_layout.h" -#include "avcodec.h" -#include "dca.h" +#include "dcadec.h" +#include "dcamath.h" #include "dca_syncwords.h" -#include "dcadata.h" -#include "dcadsp.h" -#include "dcahuff.h" -#include "fft.h" -#include "fmtconvert.h" -#include "get_bits.h" -#include "internal.h" -#include "mathops.h" #include "profiles.h" -#include "synth_filter.h" - -#if ARCH_ARM -# include "arm/dca.h" -#endif - -enum DCAMode { - DCA_MONO = 0, - DCA_CHANNEL, - DCA_STEREO, - DCA_STEREO_SUMDIFF, - DCA_STEREO_TOTAL, - DCA_3F, - DCA_2F1R, - DCA_3F1R, - DCA_2F2R, - DCA_3F2R, - DCA_4F2R -}; - - -enum DCAXxchSpeakerMask { - DCA_XXCH_FRONT_CENTER = 0x0000001, - DCA_XXCH_FRONT_LEFT = 0x0000002, - DCA_XXCH_FRONT_RIGHT = 0x0000004, - DCA_XXCH_SIDE_REAR_LEFT = 0x0000008, - DCA_XXCH_SIDE_REAR_RIGHT = 0x0000010, - DCA_XXCH_LFE1 = 0x0000020, - DCA_XXCH_REAR_CENTER = 0x0000040, - DCA_XXCH_SURROUND_REAR_LEFT = 0x0000080, - DCA_XXCH_SURROUND_REAR_RIGHT = 0x0000100, - DCA_XXCH_SIDE_SURROUND_LEFT = 0x0000200, - DCA_XXCH_SIDE_SURROUND_RIGHT = 0x0000400, - DCA_XXCH_FRONT_CENTER_LEFT = 0x0000800, - DCA_XXCH_FRONT_CENTER_RIGHT = 0x0001000, - DCA_XXCH_FRONT_HIGH_LEFT = 0x0002000, - DCA_XXCH_FRONT_HIGH_CENTER = 0x0004000, - DCA_XXCH_FRONT_HIGH_RIGHT = 0x0008000, - DCA_XXCH_LFE2 = 0x0010000, - DCA_XXCH_SIDE_FRONT_LEFT = 0x0020000, - DCA_XXCH_SIDE_FRONT_RIGHT = 0x0040000, - DCA_XXCH_OVERHEAD = 0x0080000, - DCA_XXCH_SIDE_HIGH_LEFT = 0x0100000, - DCA_XXCH_SIDE_HIGH_RIGHT = 0x0200000, - DCA_XXCH_REAR_HIGH_CENTER = 0x0400000, - DCA_XXCH_REAR_HIGH_LEFT = 0x0800000, - DCA_XXCH_REAR_HIGH_RIGHT = 0x1000000, - DCA_XXCH_REAR_LOW_CENTER = 0x2000000, - DCA_XXCH_REAR_LOW_LEFT = 0x4000000, - DCA_XXCH_REAR_LOW_RIGHT = 0x8000000, -}; - -#define DCA_DOLBY 101 /* FIXME */ - -#define DCA_CHANNEL_BITS 6 -#define DCA_CHANNEL_MASK 0x3F - -#define DCA_LFE 0x80 - -#define HEADER_SIZE 14 - -#define DCA_NSYNCAUX 0x9A1105A0 - -#define SAMPLES_PER_SUBBAND 8 // number of samples per subband per subsubframe - -/** Bit allocation */ -typedef struct BitAlloc { - int offset; ///< code values offset - int maxbits[8]; ///< max bits in VLC - int wrap; ///< wrap for get_vlc2() - VLC vlc[8]; ///< actual codes -} BitAlloc; - -static BitAlloc dca_bitalloc_index; ///< indexes for samples VLC select -static BitAlloc dca_tmode; ///< transition mode VLCs -static BitAlloc dca_scalefactor; ///< scalefactor VLCs -static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs - -static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba, - int idx) -{ - return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + - ba->offset; -} - -static float dca_dmix_code(unsigned code); - -static av_cold void dca_init_vlcs(void) -{ - static int vlcs_initialized = 0; - 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++) { - 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[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[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); - } - - 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[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], - bitalloc_bits[i][j], 1, 1, - bitalloc_codes[i][j], 2, 2, INIT_VLC_USE_NEW_STATIC); - c++; - } - vlcs_initialized = 1; -} -static inline void get_array(GetBitContext *gb, int *dst, int len, int bits) -{ - while (len--) - *dst++ = get_bits(gb, bits); -} +#define MIN_PACKET_SIZE 16 +#define MAX_PACKET_SIZE 0x104000 -static inline int dca_xxch2index(DCAContext *s, int xxch_ch) +int ff_dca_set_channel_layout(AVCodecContext *avctx, int *ch_remap, int dca_mask) { - int i, base, mask; - - /* locate channel set containing the channel */ - for (i = -1, base = 0, mask = (s->xxch_core_spkmask & ~DCA_XXCH_LFE1); - i <= s->xxch_chset && !(mask & xxch_ch); mask = s->xxch_spk_masks[++i]) - base += av_popcount(mask); - - return base + av_popcount(mask & (xxch_ch - 1)); -} - -static int dca_parse_audio_coding_header(DCAContext *s, int base_channel, - int xxch) -{ - 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 }; - int hdr_pos = 0, hdr_size = 0; - float scale_factor; - int this_chans, acc_mask; - int embedded_downmix; - int nchans, mask[8]; - int coeff, ichan; - - /* xxch has arbitrary sized audio coding headers */ - if (xxch) { - hdr_pos = get_bits_count(&s->gb); - hdr_size = get_bits(&s->gb, 7) + 1; - } - - nchans = get_bits(&s->gb, 3) + 1; - if (xxch && nchans >= 3) { - av_log(s->avctx, AV_LOG_ERROR, "nchans %d is too large\n", nchans); - return AVERROR_INVALIDDATA; - } else if (nchans + base_channel > DCA_PRIM_CHANNELS_MAX) { - av_log(s->avctx, AV_LOG_ERROR, "channel sum %d + %d is too large\n", nchans, base_channel); - return AVERROR_INVALIDDATA; - } - - s->audio_header.total_channels = nchans + base_channel; - s->audio_header.prim_channels = s->audio_header.total_channels; - - /* obtain speaker layout mask & downmix coefficients for XXCH */ - if (xxch) { - acc_mask = s->xxch_core_spkmask; - - this_chans = get_bits(&s->gb, s->xxch_nbits_spk_mask - 6) << 6; - s->xxch_spk_masks[s->xxch_chset] = this_chans; - s->xxch_chset_nch[s->xxch_chset] = nchans; - - for (i = 0; i <= s->xxch_chset; i++) - acc_mask |= s->xxch_spk_masks[i]; - - /* check for downmixing information */ - if (get_bits1(&s->gb)) { - embedded_downmix = get_bits1(&s->gb); - coeff = get_bits(&s->gb, 6); - - if (coeff<1 || coeff>61) { - av_log(s->avctx, AV_LOG_ERROR, "6bit coeff %d is out of range\n", coeff); - return AVERROR_INVALIDDATA; - } - - scale_factor = -1.0f / dca_dmix_code((coeff<<2)-3); - - s->xxch_dmix_sf[s->xxch_chset] = scale_factor; - - for (i = base_channel; i < s->audio_header.prim_channels; i++) { - mask[i] = get_bits(&s->gb, s->xxch_nbits_spk_mask); - } - - for (j = base_channel; j < s->audio_header.prim_channels; j++) { - memset(s->xxch_dmix_coeff[j], 0, sizeof(s->xxch_dmix_coeff[0])); - s->xxch_dmix_embedded |= (embedded_downmix << j); - for (i = 0; i < s->xxch_nbits_spk_mask; i++) { - if (mask[j] & (1 << i)) { - if ((1 << i) == DCA_XXCH_LFE1) { - av_log(s->avctx, AV_LOG_WARNING, - "DCA-XXCH: dmix to LFE1 not supported.\n"); - continue; - } - - coeff = get_bits(&s->gb, 7); - ichan = dca_xxch2index(s, 1 << i); - if ((coeff&63)<1 || (coeff&63)>61) { - av_log(s->avctx, AV_LOG_ERROR, "7bit coeff %d is out of range\n", coeff); - return AVERROR_INVALIDDATA; - } - s->xxch_dmix_coeff[j][ichan] = dca_dmix_code((coeff<<2)-3); - } + static const uint8_t dca2wav_norm[28] = { + 2, 0, 1, 9, 10, 3, 8, 4, 5, 9, 10, 6, 7, 12, + 13, 14, 3, 6, 7, 11, 12, 14, 16, 15, 17, 8, 4, 5, + }; + + static const uint8_t dca2wav_wide[28] = { + 2, 0, 1, 4, 5, 3, 8, 4, 5, 9, 10, 6, 7, 12, + 13, 14, 3, 9, 10, 11, 12, 14, 16, 15, 17, 8, 4, 5, + }; + + int dca_ch, wav_ch, nchannels = 0; + + if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) { + for (dca_ch = 0; dca_ch < DCA_SPEAKER_COUNT; dca_ch++) + if (dca_mask & (1U << dca_ch)) + ch_remap[nchannels++] = dca_ch; + avctx->channel_layout = dca_mask; + } else { + int wav_mask = 0; + int wav_map[18]; + const uint8_t *dca2wav; + if (dca_mask == DCA_SPEAKER_LAYOUT_7POINT0_WIDE || + dca_mask == DCA_SPEAKER_LAYOUT_7POINT1_WIDE) + dca2wav = dca2wav_wide; + else + dca2wav = dca2wav_norm; + for (dca_ch = 0; dca_ch < 28; dca_ch++) { + if (dca_mask & (1 << dca_ch)) { + wav_ch = dca2wav[dca_ch]; + if (!(wav_mask & (1 << wav_ch))) { + wav_map[wav_ch] = dca_ch; + wav_mask |= 1 << wav_ch; } } } + for (wav_ch = 0; wav_ch < 18; wav_ch++) + if (wav_mask & (1 << wav_ch)) + ch_remap[nchannels++] = wav_map[wav_ch]; + avctx->channel_layout = wav_mask; } - if (s->audio_header.prim_channels > DCA_PRIM_CHANNELS_MAX) - s->audio_header.prim_channels = DCA_PRIM_CHANNELS_MAX; - - for (i = base_channel; i < s->audio_header.prim_channels; i++) { - s->audio_header.subband_activity[i] = get_bits(&s->gb, 5) + 2; - if (s->audio_header.subband_activity[i] > DCA_SUBBANDS) - s->audio_header.subband_activity[i] = DCA_SUBBANDS; - } - for (i = base_channel; i < s->audio_header.prim_channels; i++) { - s->audio_header.vq_start_subband[i] = get_bits(&s->gb, 5) + 1; - if (s->audio_header.vq_start_subband[i] > DCA_SUBBANDS) - s->audio_header.vq_start_subband[i] = DCA_SUBBANDS; - } - get_array(&s->gb, s->audio_header.joint_intensity + base_channel, - s->audio_header.prim_channels - base_channel, 3); - get_array(&s->gb, s->audio_header.transient_huffman + base_channel, - s->audio_header.prim_channels - base_channel, 2); - get_array(&s->gb, s->audio_header.scalefactor_huffman + base_channel, - s->audio_header.prim_channels - base_channel, 3); - get_array(&s->gb, s->audio_header.bitalloc_huffman + base_channel, - s->audio_header.prim_channels - base_channel, 3); - - /* Get codebooks quantization indexes */ - if (!base_channel) - memset(s->audio_header.quant_index_huffman, 0, sizeof(s->audio_header.quant_index_huffman)); - for (j = 1; j < 11; j++) - for (i = base_channel; i < s->audio_header.prim_channels; i++) - s->audio_header.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->audio_header.prim_channels; i++) - s->audio_header.scalefactor_adj[i][j] = 1; - - for (j = 1; j < 11; j++) - for (i = base_channel; i < s->audio_header.prim_channels; i++) - if (s->audio_header.quant_index_huffman[i][j] < thr[j]) - s->audio_header.scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)]; - - if (!xxch) { - if (s->crc_present) { - /* Audio header CRC check */ - get_bits(&s->gb, 16); - } - } else { - /* Skip to the end of the header, also ignore CRC if present */ - i = get_bits_count(&s->gb); - if (hdr_pos + 8 * hdr_size > i) - skip_bits_long(&s->gb, hdr_pos + 8 * hdr_size - i); - } - - s->current_subframe = 0; - s->current_subsubframe = 0; - - return 0; + avctx->channels = nchannels; + return nchannels; } -static int dca_parse_frame_header(DCAContext *s) +static uint16_t crc16(const uint8_t *data, int size) { - init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8); - - /* Sync code */ - skip_bits_long(&s->gb, 32); - - /* Frame header */ - s->frame_type = get_bits(&s->gb, 1); - s->samples_deficit = get_bits(&s->gb, 5) + 1; - s->crc_present = get_bits(&s->gb, 1); - s->sample_blocks = get_bits(&s->gb, 7) + 1; - s->frame_size = get_bits(&s->gb, 14) + 1; - if (s->frame_size < 95) - return AVERROR_INVALIDDATA; - s->amode = get_bits(&s->gb, 6); - s->sample_rate = avpriv_dca_sample_rates[get_bits(&s->gb, 4)]; - if (!s->sample_rate) - return AVERROR_INVALIDDATA; - s->bit_rate_index = get_bits(&s->gb, 5); - s->bit_rate = ff_dca_bit_rates[s->bit_rate_index]; - if (!s->bit_rate) - return AVERROR_INVALIDDATA; + static const uint16_t crctab[16] = { + 0x0000, 0x1021, 0x2042, 0x3063, 0x4084, 0x50a5, 0x60c6, 0x70e7, + 0x8108, 0x9129, 0xa14a, 0xb16b, 0xc18c, 0xd1ad, 0xe1ce, 0xf1ef, + }; - skip_bits1(&s->gb); // always 0 (reserved, cf. ETSI TS 102 114 V1.4.1) - s->dynrange = get_bits(&s->gb, 1); - s->timestamp = get_bits(&s->gb, 1); - s->aux_data = get_bits(&s->gb, 1); - s->hdcd = get_bits(&s->gb, 1); - s->ext_descr = get_bits(&s->gb, 3); - s->ext_coding = get_bits(&s->gb, 1); - s->aspf = get_bits(&s->gb, 1); - s->lfe = get_bits(&s->gb, 2); - s->predictor_history = get_bits(&s->gb, 1); - - if (s->lfe > 2) { - s->lfe = 0; - av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE value: %d\n", s->lfe); - return AVERROR_INVALIDDATA; - } - - /* TODO: check CRC */ - if (s->crc_present) - s->header_crc = get_bits(&s->gb, 16); - - s->multirate_inter = get_bits(&s->gb, 1); - s->version = get_bits(&s->gb, 4); - s->copy_history = get_bits(&s->gb, 2); - s->source_pcm_res = get_bits(&s->gb, 3); - s->front_sum = get_bits(&s->gb, 1); - s->surround_sum = get_bits(&s->gb, 1); - s->dialog_norm = get_bits(&s->gb, 4); - - /* FIXME: channels mixing levels */ - s->output = s->amode; - if (s->lfe) - s->output |= DCA_LFE; + uint16_t res = 0xffff; + int i; - /* Primary audio coding header */ - s->audio_header.subframes = get_bits(&s->gb, 4) + 1; + for (i = 0; i < size; i++) { + res = (res << 4) ^ crctab[(data[i] >> 4) ^ (res >> 12)]; + res = (res << 4) ^ crctab[(data[i] & 15) ^ (res >> 12)]; + } - return dca_parse_audio_coding_header(s, 0, 0); + return res; } -static inline int get_scale(GetBitContext *gb, int level, int value, int log2range) +int ff_dca_check_crc(GetBitContext *s, int p1, int p2) { - if (level < 5) { - /* huffman encoded */ - value += get_bitalloc(gb, &dca_scalefactor, level); - value = av_clip(value, 0, (1 << log2range) - 1); - } else if (level < 8) { - if (level + 1 > log2range) { - skip_bits(gb, level + 1 - log2range); - value = get_bits(gb, log2range); - } else { - value = get_bits(gb, level + 1); - } - } - return value; + if (((p1 | p2) & 7) || p1 < 0 || p2 > s->size_in_bits || p2 - p1 < 16) + return -1; + if (crc16(s->buffer + p1 / 8, (p2 - p1) / 8)) + return -1; + return 0; } -static int dca_subframe_header(DCAContext *s, int base_channel, int block_index) +void ff_dca_downmix_to_stereo_fixed(DCADSPContext *dcadsp, int32_t **samples, + int *coeff_l, int nsamples, int ch_mask) { - /* Primary audio coding side information */ - int j, k; - - if (get_bits_left(&s->gb) < 0) - return AVERROR_INVALIDDATA; - - if (!base_channel) { - s->subsubframes[s->current_subframe] = get_bits(&s->gb, 2) + 1; - if (block_index + s->subsubframes[s->current_subframe] > (s->sample_blocks / SAMPLES_PER_SUBBAND)) { - s->subsubframes[s->current_subframe] = 1; - return AVERROR_INVALIDDATA; - } - s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3); - } - - for (j = base_channel; j < s->audio_header.prim_channels; j++) { - for (k = 0; k < s->audio_header.subband_activity[j]; k++) - s->dca_chan[j].prediction_mode[k] = get_bits(&s->gb, 1); - } - - /* Get prediction codebook */ - for (j = base_channel; j < s->audio_header.prim_channels; j++) { - for (k = 0; k < s->audio_header.subband_activity[j]; k++) { - if (s->dca_chan[j].prediction_mode[k] > 0) { - /* (Prediction coefficient VQ address) */ - s->dca_chan[j].prediction_vq[k] = get_bits(&s->gb, 12); - } - } - } - - /* Bit allocation index */ - for (j = base_channel; j < s->audio_header.prim_channels; j++) { - for (k = 0; k < s->audio_header.vq_start_subband[j]; k++) { - if (s->audio_header.bitalloc_huffman[j] == 6) - s->dca_chan[j].bitalloc[k] = get_bits(&s->gb, 5); - else if (s->audio_header.bitalloc_huffman[j] == 5) - s->dca_chan[j].bitalloc[k] = get_bits(&s->gb, 4); - else if (s->audio_header.bitalloc_huffman[j] == 7) { - av_log(s->avctx, AV_LOG_ERROR, - "Invalid bit allocation index\n"); - return AVERROR_INVALIDDATA; - } else { - s->dca_chan[j].bitalloc[k] = - get_bitalloc(&s->gb, &dca_bitalloc_index, s->audio_header.bitalloc_huffman[j]); - } - - if (s->dca_chan[j].bitalloc[k] > 26) { - ff_dlog(s->avctx, "bitalloc index [%i][%i] too big (%i)\n", - j, k, s->dca_chan[j].bitalloc[k]); - return AVERROR_INVALIDDATA; - } - } - } + int pos, spkr, max_spkr = av_log2(ch_mask); + int *coeff_r = coeff_l + av_popcount(ch_mask); - /* Transition mode */ - for (j = base_channel; j < s->audio_header.prim_channels; j++) { - for (k = 0; k < s->audio_header.subband_activity[j]; k++) { - s->dca_chan[j].transition_mode[k] = 0; - if (s->subsubframes[s->current_subframe] > 1 && - k < s->audio_header.vq_start_subband[j] && s->dca_chan[j].bitalloc[k] > 0) { - s->dca_chan[j].transition_mode[k] = - get_bitalloc(&s->gb, &dca_tmode, s->audio_header.transient_huffman[j]); - } - } - } + av_assert0(DCA_HAS_STEREO(ch_mask)); - if (get_bits_left(&s->gb) < 0) - return AVERROR_INVALIDDATA; + // Scale left and right channels + pos = (ch_mask & DCA_SPEAKER_MASK_C); + dcadsp->dmix_scale(samples[DCA_SPEAKER_L], coeff_l[pos ], nsamples); + dcadsp->dmix_scale(samples[DCA_SPEAKER_R], coeff_r[pos + 1], nsamples); - for (j = base_channel; j < s->audio_header.prim_channels; j++) { - const uint32_t *scale_table; - int scale_sum, log_size; + // Downmix remaining channels + for (spkr = 0; spkr <= max_spkr; spkr++) { + if (!(ch_mask & (1U << spkr))) + continue; - memset(s->dca_chan[j].scale_factor, 0, - s->audio_header.subband_activity[j] * sizeof(s->dca_chan[j].scale_factor[0][0]) * 2); + if (*coeff_l && spkr != DCA_SPEAKER_L) + dcadsp->dmix_add(samples[DCA_SPEAKER_L], samples[spkr], + *coeff_l, nsamples); - if (s->audio_header.scalefactor_huffman[j] == 6) { - scale_table = ff_dca_scale_factor_quant7; - log_size = 7; - } else { - scale_table = ff_dca_scale_factor_quant6; - log_size = 6; - } + if (*coeff_r && spkr != DCA_SPEAKER_R) + dcadsp->dmix_add(samples[DCA_SPEAKER_R], samples[spkr], + *coeff_r, nsamples); - /* When huffman coded, only the difference is encoded */ - scale_sum = 0; - - for (k = 0; k < s->audio_header.subband_activity[j]; k++) { - if (k >= s->audio_header.vq_start_subband[j] || s->dca_chan[j].bitalloc[k] > 0) { - scale_sum = get_scale(&s->gb, s->audio_header.scalefactor_huffman[j], scale_sum, log_size); - s->dca_chan[j].scale_factor[k][0] = scale_table[scale_sum]; - } - - if (k < s->audio_header.vq_start_subband[j] && s->dca_chan[j].transition_mode[k]) { - /* Get second scale factor */ - scale_sum = get_scale(&s->gb, s->audio_header.scalefactor_huffman[j], scale_sum, log_size); - s->dca_chan[j].scale_factor[k][1] = scale_table[scale_sum]; - } - } + coeff_l++; + coeff_r++; } - - /* Joint subband scale factor codebook select */ - for (j = base_channel; j < s->audio_header.prim_channels; j++) { - /* Transmitted only if joint subband coding enabled */ - if (s->audio_header.joint_intensity[j] > 0) - s->dca_chan[j].joint_huff = get_bits(&s->gb, 3); - } - - if (get_bits_left(&s->gb) < 0) - return AVERROR_INVALIDDATA; - - /* Scale factors for joint subband coding */ - for (j = base_channel; j < s->audio_header.prim_channels; j++) { - int source_channel; - - /* Transmitted only if joint subband coding enabled */ - if (s->audio_header.joint_intensity[j] > 0) { - int scale = 0; - source_channel = s->audio_header.joint_intensity[j] - 1; - - /* When huffman coded, only the difference is encoded - * (is this valid as well for joint scales ???) */ - - for (k = s->audio_header.subband_activity[j]; - k < s->audio_header.subband_activity[source_channel]; k++) { - scale = get_scale(&s->gb, s->dca_chan[j].joint_huff, 64 /* bias */, 7); - s->dca_chan[j].joint_scale_factor[k] = scale; /*joint_scale_table[scale]; */ - } - - if (!(s->debug_flag & 0x02)) { - av_log(s->avctx, AV_LOG_DEBUG, - "Joint stereo coding not supported\n"); - s->debug_flag |= 0x02; - } - } - } - - /* Dynamic range coefficient */ - if (!base_channel && s->dynrange) - s->dynrange_coef = get_bits(&s->gb, 8); - - /* Side information CRC check word */ - if (s->crc_present) { - get_bits(&s->gb, 16); - } - - /* - * Primary audio data arrays - */ - - /* VQ encoded high frequency subbands */ - for (j = base_channel; j < s->audio_header.prim_channels; j++) - for (k = s->audio_header.vq_start_subband[j]; k < s->audio_header.subband_activity[j]; k++) - /* 1 vector -> 32 samples */ - s->dca_chan[j].high_freq_vq[k] = get_bits(&s->gb, 10); - - /* Low frequency effect data */ - if (!base_channel && s->lfe) { - int quant7; - /* LFE samples */ - 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_end_sample; j++) { - /* Signed 8 bits int */ - s->lfe_data[j] = get_sbits(&s->gb, 8); - } - - /* Scale factor index */ - quant7 = get_bits(&s->gb, 8); - if (quant7 > 127) { - avpriv_request_sample(s->avctx, "LFEScaleIndex larger than 127"); - return AVERROR_INVALIDDATA; - } - s->lfe_scale_factor = ff_dca_scale_factor_quant7[quant7]; - - /* Quantization step size * scale factor */ - lfe_scale = 0.035 * s->lfe_scale_factor; - - for (j = lfe_samples; j < lfe_end_sample; j++) - s->lfe_data[j] *= lfe_scale; - } - - return 0; -} - -static void qmf_32_subbands(DCAContext *s, int chans, - float samples_in[32][SAMPLES_PER_SUBBAND], float *samples_out, - float scale) -{ - const float *prCoeff; - - int sb_act = s->audio_header.subband_activity[chans]; - - scale *= sqrt(1 / 8.0); - - /* Select filter */ - if (!s->multirate_inter) /* Non-perfect reconstruction */ - prCoeff = ff_dca_fir_32bands_nonperfect; - else /* Perfect reconstruction */ - prCoeff = ff_dca_fir_32bands_perfect; - - s->dcadsp.qmf_32_subbands(samples_in, sb_act, &s->synth, &s->imdct, - s->dca_chan[chans].subband_fir_hist, - &s->dca_chan[chans].hist_index, - s->dca_chan[chans].subband_fir_noidea, prCoeff, - samples_out, s->raXin, scale); } -static QMF64_table *qmf64_precompute(void) +void ff_dca_downmix_to_stereo_float(AVFloatDSPContext *fdsp, float **samples, + int *coeff_l, int nsamples, int ch_mask) { - 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; -} + int pos, spkr, max_spkr = av_log2(ch_mask); + int *coeff_r = coeff_l + av_popcount(ch_mask); + const float scale = 1.0f / (1 << 15); -/* 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->dca_chan[chans].subband_fir_hist; - float *raZ = s->dca_chan[chans].subband_fir_noidea; - unsigned i, j, k, subindex; - - for (i = s->audio_header.subband_activity[chans]; i < 64; i++) - raXin[i] = 0.0; - for (subindex = 0; subindex < SAMPLES_PER_SUBBAND; subindex++) { - for (i = 0; i < s->audio_header.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]); - } + av_assert0(DCA_HAS_STEREO(ch_mask)); - 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; - } + // Scale left and right channels + pos = (ch_mask & DCA_SPEAKER_MASK_C); + fdsp->vector_fmul_scalar(samples[DCA_SPEAKER_L], samples[DCA_SPEAKER_L], + coeff_l[pos ] * scale, nsamples); + fdsp->vector_fmul_scalar(samples[DCA_SPEAKER_R], samples[DCA_SPEAKER_R], + coeff_r[pos + 1] * scale, nsamples); - 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]); + // Downmix remaining channels + for (spkr = 0; spkr <= max_spkr; spkr++) { + if (!(ch_mask & (1U << spkr))) + continue; - raZ[i] = hist; - } + if (*coeff_l && spkr != DCA_SPEAKER_L) + fdsp->vector_fmac_scalar(samples[DCA_SPEAKER_L], samples[spkr], + *coeff_l * scale, nsamples); - /* FIXME: Make buffer circular, to avoid this move. */ - memmove(raX + 64, raX, (1024 - 64) * sizeof(*raX)); - } -} + if (*coeff_r && spkr != DCA_SPEAKER_R) + fdsp->vector_fmac_scalar(samples[DCA_SPEAKER_R], samples[spkr], + *coeff_r * scale, nsamples); -static void lfe_interpolation_fir(DCAContext *s, const float *samples_in, - float *samples_out) -{ - /* samples_in: An array holding decimated samples. - * Samples in current subframe starts from samples_in[0], - * while samples_in[-1], samples_in[-2], ..., stores samples - * from last subframe as history. - * - * samples_out: An array holding interpolated samples - */ - - int idx; - const float *prCoeff; - int deciindex; - - /* Select decimation filter */ - if (s->lfe == 1) { - idx = 1; - prCoeff = ff_dca_lfe_fir_128; - } else { - 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 < 2 * s->lfe; deciindex++) { - s->dcadsp.lfe_fir[idx](samples_out, samples_in, prCoeff); - samples_in++; - samples_out += 2 * 32 * (1 + idx); + coeff_l++; + coeff_r++; } } -/* downmixing routines */ -#define MIX_REAR1(samples, s1, rs, coef) \ - samples[0][i] += samples[s1][i] * coef[rs][0]; \ - samples[1][i] += samples[s1][i] * coef[rs][1]; - -#define MIX_REAR2(samples, s1, s2, rs, coef) \ - samples[0][i] += samples[s1][i] * coef[rs][0] + samples[s2][i] * coef[rs + 1][0]; \ - samples[1][i] += samples[s1][i] * coef[rs][1] + samples[s2][i] * coef[rs + 1][1]; - -#define MIX_FRONT3(samples, coef) \ - t = samples[c][i]; \ - u = samples[l][i]; \ - v = samples[r][i]; \ - samples[0][i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \ - samples[1][i] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1]; - -#define DOWNMIX_TO_STEREO(op1, op2) \ - for (i = 0; i < 256; i++) { \ - op1 \ - op2 \ - } - -static void dca_downmix(float **samples, int srcfmt, int lfe_present, - float coef[DCA_PRIM_CHANNELS_MAX + 1][2], - const int8_t *channel_mapping) +static int convert_bitstream(const uint8_t *src, int src_size, uint8_t *dst, int max_size) { - int c, l, r, sl, sr, s; - int i; - float t, u, v; - - switch (srcfmt) { - case DCA_MONO: - case DCA_4F2R: - av_log(NULL, AV_LOG_ERROR, "Not implemented!\n"); - break; - case DCA_CHANNEL: - case DCA_STEREO: - case DCA_STEREO_TOTAL: - case DCA_STEREO_SUMDIFF: - break; - case DCA_3F: - c = channel_mapping[0]; - l = channel_mapping[1]; - r = channel_mapping[2]; - DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), ); - break; - case DCA_2F1R: - s = channel_mapping[2]; - DOWNMIX_TO_STEREO(MIX_REAR1(samples, s, 2, coef), ); - break; - case DCA_3F1R: - c = channel_mapping[0]; - l = channel_mapping[1]; - r = channel_mapping[2]; - s = channel_mapping[3]; - DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), - MIX_REAR1(samples, s, 3, coef)); - break; - case DCA_2F2R: - sl = channel_mapping[2]; - sr = channel_mapping[3]; - DOWNMIX_TO_STEREO(MIX_REAR2(samples, sl, sr, 2, coef), ); - break; - case DCA_3F2R: - c = channel_mapping[0]; - l = channel_mapping[1]; - r = channel_mapping[2]; - sl = channel_mapping[3]; - sr = channel_mapping[4]; - DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), - MIX_REAR2(samples, sl, sr, 3, coef)); - break; - } - if (lfe_present) { - 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]; - } + switch (AV_RB32(src)) { + case DCA_SYNCWORD_CORE_BE: + case DCA_SYNCWORD_SUBSTREAM: + memcpy(dst, src, src_size); + return src_size; + case DCA_SYNCWORD_CORE_LE: + case DCA_SYNCWORD_CORE_14B_BE: + case DCA_SYNCWORD_CORE_14B_LE: + return avpriv_dca_convert_bitstream(src, src_size, dst, max_size); + default: + return AVERROR_INVALIDDATA; } } -#ifndef decode_blockcodes -/* Very compact version of the block code decoder that does not use table - * look-up but is slightly slower */ -static int decode_blockcode(int code, int levels, int32_t *values) +static int dcadec_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { - int i; - int offset = (levels - 1) >> 1; + DCAContext *s = avctx->priv_data; + AVFrame *frame = data; + uint8_t *input = avpkt->data; + int input_size = avpkt->size; + int i, ret, prev_packet = s->packet; - for (i = 0; i < 4; i++) { - int div = FASTDIV(code, levels); - values[i] = code - offset - div * levels; - code = div; + if (input_size < MIN_PACKET_SIZE || input_size > MAX_PACKET_SIZE) { + av_log(avctx, AV_LOG_ERROR, "Invalid packet size\n"); + return AVERROR_INVALIDDATA; } - return code; -} - -static int decode_blockcodes(int code1, int code2, int levels, int32_t *values) -{ - return decode_blockcode(code1, levels, values) | - decode_blockcode(code2, levels, values + 4); -} -#endif - -static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 }; -static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 }; - -static int dca_subsubframe(DCAContext *s, int base_channel, int block_index) -{ - int k, l; - int subsubframe = s->current_subsubframe; - - const float *quant_step_table; - - 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 = ff_dca_lossless_quant_d; - else - quant_step_table = ff_dca_lossy_quant_d; - - for (k = base_channel; k < s->audio_header.prim_channels; k++) { - float (*subband_samples)[8] = s->dca_chan[k].subband_samples[block_index]; - float rscale[DCA_SUBBANDS]; - - if (get_bits_left(&s->gb) < 0) - return AVERROR_INVALIDDATA; - - for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) { - int m; - - /* Select the mid-tread linear quantizer */ - int abits = s->dca_chan[k].bitalloc[l]; + av_fast_malloc(&s->buffer, &s->buffer_size, + FFALIGN(input_size, 4096) + DCA_BUFFER_PADDING_SIZE); + if (!s->buffer) + return AVERROR(ENOMEM); - float quant_step_size = quant_step_table[abits]; + for (i = 0, ret = AVERROR_INVALIDDATA; i < input_size - MIN_PACKET_SIZE + 1 && ret < 0; i++) + ret = convert_bitstream(input + i, input_size - i, s->buffer, s->buffer_size); - /* - * Determine quantization index code book and its type - */ + if (ret < 0) + return ret; - /* Select quantization index code book */ - int sel = s->audio_header.quant_index_huffman[k][abits]; + input = s->buffer; + input_size = ret; - /* - * Extract bits from the bit stream - */ - if (!abits) { - rscale[l] = 0; - memset(block + SAMPLES_PER_SUBBAND * l, 0, SAMPLES_PER_SUBBAND * sizeof(block[0])); - } else { - /* Deal with transients */ - int sfi = s->dca_chan[k].transition_mode[l] && - subsubframe >= s->dca_chan[k].transition_mode[l]; - rscale[l] = quant_step_size * s->dca_chan[k].scale_factor[l][sfi] * - s->audio_header.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, err; - - size = abits_sizes[abits - 1]; - levels = abits_levels[abits - 1]; - - block_code1 = get_bits(&s->gb, size); - block_code2 = get_bits(&s->gb, size); - err = decode_blockcodes(block_code1, block_code2, - levels, block + SAMPLES_PER_SUBBAND * l); - if (err) { - av_log(s->avctx, AV_LOG_ERROR, - "ERROR: block code look-up failed\n"); - return AVERROR_INVALIDDATA; - } - } else { - /* no coding */ - for (m = 0; m < SAMPLES_PER_SUBBAND; m++) - block[SAMPLES_PER_SUBBAND * l + m] = get_sbits(&s->gb, abits - 3); - } - } else { - /* Huffman coded */ - for (m = 0; m < SAMPLES_PER_SUBBAND; m++) - block[SAMPLES_PER_SUBBAND * l + m] = get_bitalloc(&s->gb, - &dca_smpl_bitalloc[abits], sel); - } - } - } + s->packet = 0; - s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[0], - block, rscale, SAMPLES_PER_SUBBAND * s->audio_header.vq_start_subband[k]); - - for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) { - int m; - /* - * Inverse ADPCM if in prediction mode - */ - if (s->dca_chan[k].prediction_mode[l]) { - int n; - if (s->predictor_history) - subband_samples[l][0] += (ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] * - s->dca_chan[k].subband_samples_hist[l][3] + - ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][1] * - s->dca_chan[k].subband_samples_hist[l][2] + - ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][2] * - s->dca_chan[k].subband_samples_hist[l][1] + - ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][3] * - s->dca_chan[k].subband_samples_hist[l][0]) * - (1.0f / 8192); - for (m = 1; m < SAMPLES_PER_SUBBAND; m++) { - float sum = ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] * - subband_samples[l][m - 1]; - for (n = 2; n <= 4; n++) - if (m >= n) - sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] * - subband_samples[l][m - n]; - else if (s->predictor_history) - sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] * - s->dca_chan[k].subband_samples_hist[l][m - n + 4]; - subband_samples[l][m] += sum * (1.0f / 8192); - } - } + // Parse backward compatible core sub-stream + if (AV_RB32(input) == DCA_SYNCWORD_CORE_BE) { + int frame_size; + if ((ret = ff_dca_core_parse(&s->core, input, input_size)) < 0) { + s->core_residual_valid = 0; + return ret; } - /* Backup predictor history for adpcm */ - for (l = 0; l < DCA_SUBBANDS; l++) - AV_COPY128(s->dca_chan[k].subband_samples_hist[l], &subband_samples[l][4]); - - - /* - * Decode VQ encoded high frequencies - */ - if (s->audio_header.subband_activity[k] > s->audio_header.vq_start_subband[k]) { - if (!(s->debug_flag & 0x01)) { - av_log(s->avctx, AV_LOG_DEBUG, - "Stream with high frequencies VQ coding\n"); - s->debug_flag |= 0x01; - } - s->dcadsp.decode_hf(subband_samples, s->dca_chan[k].high_freq_vq, - ff_dca_high_freq_vq, subsubframe * SAMPLES_PER_SUBBAND, - s->dca_chan[k].scale_factor, - s->audio_header.vq_start_subband[k], - s->audio_header.subband_activity[k]); - } - } + s->packet |= DCA_PACKET_CORE; - /* Check for DSYNC after subsubframe */ - if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) { - if (get_bits(&s->gb, 16) != 0xFFFF) { - av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n"); - return AVERROR_INVALIDDATA; + // EXXS data must be aligned on 4-byte boundary + frame_size = FFALIGN(s->core.frame_size, 4); + if (input_size - 4 > frame_size) { + input += frame_size; + input_size -= frame_size; } } - return 0; -} + if (!s->core_only) { + DCAExssAsset *asset = NULL; -static int dca_filter_channels(DCAContext *s, int block_index, int upsample) -{ - int k; - - 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->audio_header.prim_channels; k++) { - float (*subband_samples)[SAMPLES_PER_SUBBAND] = s->dca_chan[k].subband_samples[block_index]; - - if (s->channel_order_tab[k] >= 0) - qmf_64_subbands(s, k, subband_samples, - 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->audio_header.prim_channels; k++) { - float (*subband_samples)[SAMPLES_PER_SUBBAND] = s->dca_chan[k].subband_samples[block_index]; - - if (s->channel_order_tab[k] >= 0) - qmf_32_subbands(s, k, subband_samples, - s->samples_chanptr[s->channel_order_tab[k]], - M_SQRT1_2 / 32768.0); - } - } - - /* Generate LFE samples for this subsubframe FIXME!!! */ - if (s->lfe) { - float *samples = s->samples_chanptr[s->lfe_index]; - lfe_interpolation_fir(s, - s->lfe_data + 2 * s->lfe * (block_index + 4), - samples); - if (upsample) { - unsigned i; - /* Should apply the filter in Table 6-11 when upsampling. For - * now, just duplicate. */ - for (i = 255; i > 0; i--) { - samples[2 * i] = - samples[2 * i + 1] = samples[i]; + // Parse extension sub-stream (EXSS) + if (AV_RB32(input) == DCA_SYNCWORD_SUBSTREAM) { + if ((ret = ff_dca_exss_parse(&s->exss, input, input_size)) < 0) { + if (avctx->err_recognition & AV_EF_EXPLODE) + return ret; + } else { + s->packet |= DCA_PACKET_EXSS; + asset = &s->exss.assets[0]; } - 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->audio_header.prim_channels + !!s->lfe > 2 && - s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) { - dca_downmix(s->samples_chanptr, s->amode, !!s->lfe, s->downmix_coef, - s->channel_order_tab); - } - - return 0; -} - -static int dca_subframe_footer(DCAContext *s, int base_channel) -{ - int in, out, aux_data_count, aux_data_end, reserved; - uint32_t nsyncaux; - - /* - * Unpack optional information - */ - - /* presumably optional information only appears in the core? */ - if (!base_channel) { - if (s->timestamp) - skip_bits_long(&s->gb, 32); - - if (s->aux_data) { - aux_data_count = get_bits(&s->gb, 6); - - // align (32-bit) - skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31); - - aux_data_end = 8 * aux_data_count + get_bits_count(&s->gb); - - if ((nsyncaux = get_bits_long(&s->gb, 32)) != DCA_NSYNCAUX) { - av_log(s->avctx, AV_LOG_ERROR, "nSYNCAUX mismatch %#"PRIx32"\n", - nsyncaux); - return AVERROR_INVALIDDATA; - } - - if (get_bits1(&s->gb)) { // bAUXTimeStampFlag - avpriv_request_sample(s->avctx, - "Auxiliary Decode Time Stamp Flag"); - // align (4-bit) - skip_bits(&s->gb, (-get_bits_count(&s->gb)) & 4); - // 44 bits: nMSByte (8), nMarker (4), nLSByte (28), nMarker (4) - skip_bits_long(&s->gb, 44); - } - - if ((s->core_downmix = get_bits1(&s->gb))) { - int am = get_bits(&s->gb, 3); - switch (am) { - case 0: - s->core_downmix_amode = DCA_MONO; - break; - case 1: - s->core_downmix_amode = DCA_STEREO; - break; - case 2: - s->core_downmix_amode = DCA_STEREO_TOTAL; - break; - case 3: - s->core_downmix_amode = DCA_3F; - break; - case 4: - s->core_downmix_amode = DCA_2F1R; - break; - case 5: - s->core_downmix_amode = DCA_2F2R; - break; - case 6: - s->core_downmix_amode = DCA_3F1R; - break; - default: - av_log(s->avctx, AV_LOG_ERROR, - "Invalid mode %d for embedded downmix coefficients\n", - am); - return AVERROR_INVALIDDATA; - } - for (out = 0; out < ff_dca_channels[s->core_downmix_amode]; out++) { - for (in = 0; in < s->audio_header.prim_channels + !!s->lfe; in++) { - uint16_t tmp = get_bits(&s->gb, 9); - if ((tmp & 0xFF) > 241) { - av_log(s->avctx, AV_LOG_ERROR, - "Invalid downmix coefficient code %"PRIu16"\n", - tmp); - return AVERROR_INVALIDDATA; - } - s->core_downmix_codes[in][out] = tmp; - } - } - } - - align_get_bits(&s->gb); // byte align - skip_bits(&s->gb, 16); // nAUXCRC16 - - /* - * additional data (reserved, cf. ETSI TS 102 114 V1.4.1) - * - * Note: don't check for overreads, aux_data_count can't be trusted. - */ - if ((reserved = (aux_data_end - get_bits_count(&s->gb))) > 0) { - avpriv_request_sample(s->avctx, - "Core auxiliary data reserved content"); - skip_bits_long(&s->gb, reserved); + // Parse XLL component in EXSS + if (asset && (asset->extension_mask & DCA_EXSS_XLL)) { + if ((ret = ff_dca_xll_parse(&s->xll, input, asset)) < 0) { + // Conceal XLL synchronization error + if (ret == AVERROR(EAGAIN) + && (prev_packet & DCA_PACKET_XLL) + && (s->packet & DCA_PACKET_CORE)) + s->packet |= DCA_PACKET_XLL | DCA_PACKET_RECOVERY; + else if (ret == AVERROR(ENOMEM) || (avctx->err_recognition & AV_EF_EXPLODE)) + return ret; + } else { + s->packet |= DCA_PACKET_XLL; } } - if (s->crc_present && s->dynrange) - get_bits(&s->gb, 16); - } - - return 0; -} - -/** - * Decode a dca frame block - * - * @param s pointer to the DCAContext - */ - -static int dca_decode_block(DCAContext *s, int base_channel, int block_index) -{ - int ret; - - /* Sanity check */ - if (s->current_subframe >= s->audio_header.subframes) { - av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i", - s->current_subframe, s->audio_header.subframes); - return AVERROR_INVALIDDATA; - } - - if (!s->current_subsubframe) { - /* Read subframe header */ - if ((ret = dca_subframe_header(s, base_channel, block_index))) + // Parse core extensions in EXSS or backward compatible core sub-stream + if ((s->packet & DCA_PACKET_CORE) + && (ret = ff_dca_core_parse_exss(&s->core, input, asset)) < 0) return ret; } - /* Read subsubframe */ - if ((ret = dca_subsubframe(s, base_channel, block_index))) - return ret; - - /* Update state */ - s->current_subsubframe++; - if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) { - s->current_subsubframe = 0; - s->current_subframe++; - } - if (s->current_subframe >= s->audio_header.subframes) { - /* Read subframe footer */ - if ((ret = dca_subframe_footer(s, base_channel))) - return ret; - } + // Filter the frame + if (s->packet & DCA_PACKET_XLL) { + if (s->packet & DCA_PACKET_CORE) { + int x96_synth = -1; - return 0; -} + // Enable X96 synthesis if needed + if (s->xll.chset[0].freq == 96000 && s->core.sample_rate == 48000) + x96_synth = 1; -int ff_dca_xbr_parse_frame(DCAContext *s) -{ - int scale_table_high[DCA_CHSET_CHANS_MAX][DCA_SUBBANDS][2]; - int active_bands[DCA_CHSETS_MAX][DCA_CHSET_CHANS_MAX]; - int abits_high[DCA_CHSET_CHANS_MAX][DCA_SUBBANDS]; - int anctemp[DCA_CHSET_CHANS_MAX]; - int chset_fsize[DCA_CHSETS_MAX]; - int n_xbr_ch[DCA_CHSETS_MAX]; - int hdr_size, num_chsets, xbr_tmode, hdr_pos; - int i, j, k, l, chset, chan_base; - - av_log(s->avctx, AV_LOG_DEBUG, "DTS-XBR: decoding XBR extension\n"); - - /* get bit position of sync header */ - hdr_pos = get_bits_count(&s->gb) - 32; - - hdr_size = get_bits(&s->gb, 6) + 1; - num_chsets = get_bits(&s->gb, 2) + 1; - - for(i = 0; i < num_chsets; i++) - chset_fsize[i] = get_bits(&s->gb, 14) + 1; - - xbr_tmode = get_bits1(&s->gb); - - for(i = 0; i < num_chsets; i++) { - n_xbr_ch[i] = get_bits(&s->gb, 3) + 1; - k = get_bits(&s->gb, 2) + 5; - for(j = 0; j < n_xbr_ch[i]; j++) { - active_bands[i][j] = get_bits(&s->gb, k) + 1; - if (active_bands[i][j] > DCA_SUBBANDS) { - av_log(s->avctx, AV_LOG_ERROR, "too many active subbands (%d)\n", active_bands[i][j]); - return AVERROR_INVALIDDATA; + if ((ret = ff_dca_core_filter_fixed(&s->core, x96_synth)) < 0) { + s->core_residual_valid = 0; + return ret; } - } - } - - /* skip to the end of the header */ - i = get_bits_count(&s->gb); - if(hdr_pos + hdr_size * 8 > i) - skip_bits_long(&s->gb, hdr_pos + hdr_size * 8 - i); - - /* loop over the channel data sets */ - /* only decode as many channels as we've decoded base data for */ - for(chset = 0, chan_base = 0; - chset < num_chsets && chan_base + n_xbr_ch[chset] <= s->audio_header.prim_channels; - chan_base += n_xbr_ch[chset++]) { - int start_posn = get_bits_count(&s->gb); - int subsubframe = 0; - int subframe = 0; - - /* loop over subframes */ - for (k = 0; k < (s->sample_blocks / 8); k++) { - /* parse header if we're on first subsubframe of a block */ - if(subsubframe == 0) { - /* Parse subframe header */ - for(i = 0; i < n_xbr_ch[chset]; i++) { - anctemp[i] = get_bits(&s->gb, 2) + 2; - } - for(i = 0; i < n_xbr_ch[chset]; i++) { - get_array(&s->gb, abits_high[i], active_bands[chset][i], anctemp[i]); - } - - for(i = 0; i < n_xbr_ch[chset]; i++) { - anctemp[i] = get_bits(&s->gb, 3); - if(anctemp[i] < 1) { - av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: SYNC ERROR\n"); - return AVERROR_INVALIDDATA; - } - } - - /* generate scale factors */ - for(i = 0; i < n_xbr_ch[chset]; i++) { - const uint32_t *scale_table; - int nbits; - int scale_table_size; - - if (s->audio_header.scalefactor_huffman[chan_base+i] == 6) { - scale_table = ff_dca_scale_factor_quant7; - scale_table_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7); - } else { - scale_table = ff_dca_scale_factor_quant6; - scale_table_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant6); - } - - nbits = anctemp[i]; - - for(j = 0; j < active_bands[chset][i]; j++) { - if(abits_high[i][j] > 0) { - int index = get_bits(&s->gb, nbits); - if (index >= scale_table_size) { - av_log(s->avctx, AV_LOG_ERROR, "scale table index %d invalid\n", index); - return AVERROR_INVALIDDATA; - } - scale_table_high[i][j][0] = scale_table[index]; - - if(xbr_tmode && s->dca_chan[i].transition_mode[j]) { - int index = get_bits(&s->gb, nbits); - if (index >= scale_table_size) { - av_log(s->avctx, AV_LOG_ERROR, "scale table index %d invalid\n", index); - return AVERROR_INVALIDDATA; - } - scale_table_high[i][j][1] = scale_table[index]; - } - } - } - } - } - - /* decode audio array for this block */ - for(i = 0; i < n_xbr_ch[chset]; i++) { - for(j = 0; j < active_bands[chset][i]; j++) { - const int xbr_abits = abits_high[i][j]; - const float quant_step_size = ff_dca_lossless_quant_d[xbr_abits]; - const int sfi = xbr_tmode && s->dca_chan[i].transition_mode[j] && subsubframe >= s->dca_chan[i].transition_mode[j]; - const float rscale = quant_step_size * scale_table_high[i][j][sfi]; - float *subband_samples = s->dca_chan[chan_base+i].subband_samples[k][j]; - int block[8]; - - if(xbr_abits <= 0) - continue; - - if(xbr_abits > 7) { - get_array(&s->gb, block, 8, xbr_abits - 3); - } else { - int block_code1, block_code2, size, levels, err; - - size = abits_sizes[xbr_abits - 1]; - levels = abits_levels[xbr_abits - 1]; - - block_code1 = get_bits(&s->gb, size); - block_code2 = get_bits(&s->gb, size); - err = decode_blockcodes(block_code1, block_code2, - levels, block); - if (err) { - av_log(s->avctx, AV_LOG_ERROR, - "ERROR: DTS-XBR: block code look-up failed\n"); - return AVERROR_INVALIDDATA; - } - } - - /* scale & sum into subband */ - for(l = 0; l < 8; l++) - subband_samples[l] += (float)block[l] * rscale; - } - } - - /* check DSYNC marker */ - if(s->aspf || subsubframe == s->subsubframes[subframe] - 1) { - if(get_bits(&s->gb, 16) != 0xffff) { - av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: Didn't get subframe DSYNC\n"); - return AVERROR_INVALIDDATA; - } - } - - /* advance sub-sub-frame index */ - if(++subsubframe >= s->subsubframes[subframe]) { - subsubframe = 0; - subframe++; + // Force lossy downmixed output on the first core frame filtered. + // This prevents audible clicks when seeking and is consistent with + // what reference decoder does when there are multiple channel sets. + if (!s->core_residual_valid) { + if (s->xll.nreschsets > 0 && s->xll.nchsets > 1) + s->packet |= DCA_PACKET_RECOVERY; + s->core_residual_valid = 1; } } - /* skip to next channel set */ - i = get_bits_count(&s->gb); - if(start_posn + chset_fsize[chset] * 8 != i) { - j = start_posn + chset_fsize[chset] * 8 - i; - if(j < 0 || j >= 8) - av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: end of channel set," - " skipping further than expected (%d bits)\n", j); - skip_bits_long(&s->gb, j); - } - } - - return 0; -} - - -/* parse initial header for XXCH and dump details */ -int ff_dca_xxch_decode_frame(DCAContext *s) -{ - int hdr_size, spkmsk_bits, num_chsets, core_spk, hdr_pos; - int i, chset, base_channel, chstart, fsize[8]; - - /* assume header word has already been parsed */ - hdr_pos = get_bits_count(&s->gb) - 32; - hdr_size = get_bits(&s->gb, 6) + 1; - /*chhdr_crc =*/ skip_bits1(&s->gb); - spkmsk_bits = get_bits(&s->gb, 5) + 1; - num_chsets = get_bits(&s->gb, 2) + 1; - - for (i = 0; i < num_chsets; i++) - fsize[i] = get_bits(&s->gb, 14) + 1; - - core_spk = get_bits(&s->gb, spkmsk_bits); - s->xxch_core_spkmask = core_spk; - s->xxch_nbits_spk_mask = spkmsk_bits; - s->xxch_dmix_embedded = 0; - - /* skip to the end of the header */ - i = get_bits_count(&s->gb); - if (hdr_pos + hdr_size * 8 > i) - skip_bits_long(&s->gb, hdr_pos + hdr_size * 8 - i); - - for (chset = 0; chset < num_chsets; chset++) { - chstart = get_bits_count(&s->gb); - base_channel = s->audio_header.prim_channels; - s->xxch_chset = chset; - - /* XXCH and Core headers differ, see 6.4.2 "XXCH Channel Set Header" vs. - 5.3.2 "Primary Audio Coding Header", DTS Spec 1.3.1 */ - dca_parse_audio_coding_header(s, base_channel, 1); - - /* decode channel data */ - for (i = 0; i < (s->sample_blocks / 8); i++) { - if (dca_decode_block(s, base_channel, i)) { - av_log(s->avctx, AV_LOG_ERROR, - "Error decoding DTS-XXCH extension\n"); - continue; + if ((ret = ff_dca_xll_filter_frame(&s->xll, frame)) < 0) { + // Fall back to core unless hard error + if (!(s->packet & DCA_PACKET_CORE)) + return ret; + if (ret != AVERROR_INVALIDDATA || (avctx->err_recognition & AV_EF_EXPLODE)) + return ret; + if ((ret = ff_dca_core_filter_frame(&s->core, frame)) < 0) { + s->core_residual_valid = 0; + return ret; } } - - /* skip to end of this section */ - i = get_bits_count(&s->gb); - if (chstart + fsize[chset] * 8 > i) - skip_bits_long(&s->gb, chstart + fsize[chset] * 8 - i); - } - s->xxch_chset = num_chsets; - - return 0; -} - -static float dca_dmix_code(unsigned code) -{ - int sign = (code >> 8) - 1; - code &= 0xff; - return ((ff_dca_dmixtable[code] ^ sign) - sign) * (1.0 / (1 << 15)); -} - -static int scan_for_extensions(AVCodecContext *avctx) -{ - DCAContext *s = avctx->priv_data; - int core_ss_end, ret = 0; - - core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8; - - /* only scan for extensions if ext_descr was unknown or indicated a - * supported XCh extension */ - if (s->core_ext_mask < 0 || s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH)) { - /* if ext_descr was unknown, clear s->core_ext_mask so that the - * extensions scan can fill it up */ - s->core_ext_mask = FFMAX(s->core_ext_mask, 0); - - /* extensions start at 32-bit boundaries into bitstream */ - skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31); - - while (core_ss_end - get_bits_count(&s->gb) >= 32) { - uint32_t bits = get_bits_long(&s->gb, 32); - int i; - - switch (bits) { - case DCA_SYNCWORD_XCH: { - int ext_amode, xch_fsize; - - s->xch_base_channel = s->audio_header.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); - - s->core_ext_mask |= DCA_EXT_XCH; - - /* extension amode(number of channels in extension) should be 1 */ - /* AFAIK XCh is not used for more channels */ - if ((ext_amode = get_bits(&s->gb, 4)) != 1) { - av_log(avctx, AV_LOG_ERROR, - "XCh extension amode %d not supported!\n", - ext_amode); - continue; - } - - if (s->xch_base_channel < 2) { - avpriv_request_sample(avctx, "XCh with fewer than 2 base channels"); - continue; - } - - /* much like core primary audio coding header */ - dca_parse_audio_coding_header(s, s->xch_base_channel, 0); - - for (i = 0; i < (s->sample_blocks / 8); i++) - if ((ret = dca_decode_block(s, s->xch_base_channel, i))) { - av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n"); - continue; - } - - s->xch_present = 1; - break; - } - case 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 */ - s->core_ext_mask |= DCA_EXT_XXCH; - ff_dca_xxch_decode_frame(s); - break; - - case 0x1d95f262: { - int fsize96 = show_bits(&s->gb, 12) + 1; - if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96) - continue; - - av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n", - get_bits_count(&s->gb)); - skip_bits(&s->gb, 12); - av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96); - av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4)); - - s->core_ext_mask |= DCA_EXT_X96; - break; - } - } - - skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31); + } else if (s->packet & DCA_PACKET_CORE) { + if ((ret = ff_dca_core_filter_frame(&s->core, frame)) < 0) { + s->core_residual_valid = 0; + return ret; } + s->core_residual_valid = !!(s->core.filter_mode & DCA_FILTER_MODE_FIXED); } else { - /* no supported extensions, skip the rest of the core substream */ - skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb)); + return AVERROR_INVALIDDATA; } - if (s->core_ext_mask & DCA_EXT_X96) - s->profile = FF_PROFILE_DTS_96_24; - else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH)) - s->profile = FF_PROFILE_DTS_ES; - - /* check for ExSS (HD part) */ - if (s->dca_buffer_size - s->frame_size > 32 && - get_bits_long(&s->gb, 32) == DCA_SYNCWORD_SUBSTREAM) - ff_dca_exss_parse_header(s); + *got_frame_ptr = 1; - return ret; + return avpkt->size; } -static int set_channel_layout(AVCodecContext *avctx, int *channels, int num_core_channels) +static av_cold void dcadec_flush(AVCodecContext *avctx) { DCAContext *s = avctx->priv_data; - int i, j, chset, mask; - int channel_layout, channel_mask; - int posn, lavc; - - /* If we have XXCH then the channel layout is managed differently */ - /* note that XLL will also have another way to do things */ - if (!(s->core_ext_mask & DCA_EXT_XXCH)) { - /* xxx should also do MA extensions */ - if (s->amode < 16) { - avctx->channel_layout = ff_dca_core_channel_layout[s->amode]; - - if (s->audio_header.prim_channels + !!s->lfe > 2 && - avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) { - /* - * Neither the core's auxiliary data nor our default tables contain - * downmix coefficients for the additional channel coded in the XCh - * extension, so when we're doing a Stereo downmix, don't decode it. - */ - s->xch_disable = 1; - } - if (s->xch_present && !s->xch_disable) { - if (avctx->channel_layout & AV_CH_BACK_CENTER) { - avpriv_request_sample(avctx, "XCh with Back center channel"); - return AVERROR_INVALIDDATA; - } - avctx->channel_layout |= AV_CH_BACK_CENTER; - if (s->lfe) { - avctx->channel_layout |= AV_CH_LOW_FREQUENCY; - s->channel_order_tab = ff_dca_channel_reorder_lfe_xch[s->amode]; - } else { - s->channel_order_tab = ff_dca_channel_reorder_nolfe_xch[s->amode]; - } - if (s->channel_order_tab[s->xch_base_channel] < 0) - return AVERROR_INVALIDDATA; - } 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 = ff_dca_channel_reorder_lfe[s->amode]; - } else - s->channel_order_tab = ff_dca_channel_reorder_nolfe[s->amode]; - } - - if (*channels > !!s->lfe && - s->channel_order_tab[*channels - 1 - !!s->lfe] < 0) - return AVERROR_INVALIDDATA; - - if (av_get_channel_layout_nb_channels(avctx->channel_layout) != *channels) { - av_log(avctx, AV_LOG_ERROR, "Number of channels %d mismatches layout %d\n", *channels, av_get_channel_layout_nb_channels(avctx->channel_layout)); - return AVERROR_INVALIDDATA; - } - - if (num_core_channels + !!s->lfe > 2 && - avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) { - *channels = 2; - s->output = s->audio_header.prim_channels == 2 ? s->amode : DCA_STEREO; - avctx->channel_layout = AV_CH_LAYOUT_STEREO; - } - else if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) { - static const int8_t dca_channel_order_native[9] = { 0, 1, 2, 3, 4, 5, 6, 7, 8 }; - s->channel_order_tab = dca_channel_order_native; - } - s->lfe_index = ff_dca_lfe_index[s->amode]; - } else { - av_log(avctx, AV_LOG_ERROR, - "Non standard configuration %d !\n", s->amode); - return AVERROR_INVALIDDATA; - } - - s->xxch_dmix_embedded = 0; - } else { - /* we only get here if an XXCH channel set can be added to the mix */ - channel_mask = s->xxch_core_spkmask; - - { - *channels = s->audio_header.prim_channels + !!s->lfe; - for (i = 0; i < s->xxch_chset; i++) { - channel_mask |= s->xxch_spk_masks[i]; - } - } - - /* Given the DTS spec'ed channel mask, generate an avcodec version */ - channel_layout = 0; - for (i = 0; i < s->xxch_nbits_spk_mask; ++i) { - if (channel_mask & (1 << i)) { - channel_layout |= ff_dca_map_xxch_to_native[i]; - } - } - - /* make sure that we have managed to get equivalent dts/avcodec channel - * masks in some sense -- unfortunately some channels could overlap */ - if (av_popcount(channel_mask) != av_popcount(channel_layout)) { - av_log(avctx, AV_LOG_DEBUG, - "DTS-XXCH: Inconsistent avcodec/dts channel layouts\n"); - return AVERROR_INVALIDDATA; - } - - avctx->channel_layout = channel_layout; - - if (!(avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE)) { - /* Estimate DTS --> avcodec ordering table */ - for (chset = -1, j = 0; chset < s->xxch_chset; ++chset) { - mask = chset >= 0 ? s->xxch_spk_masks[chset] - : s->xxch_core_spkmask; - for (i = 0; i < s->xxch_nbits_spk_mask; i++) { - if (mask & ~(DCA_XXCH_LFE1 | DCA_XXCH_LFE2) & (1 << i)) { - lavc = ff_dca_map_xxch_to_native[i]; - posn = av_popcount(channel_layout & (lavc - 1)); - s->xxch_order_tab[j++] = posn; - } - } - - } - - s->lfe_index = av_popcount(channel_layout & (AV_CH_LOW_FREQUENCY-1)); - } else { /* native ordering */ - for (i = 0; i < *channels; i++) - s->xxch_order_tab[i] = i; + ff_dca_core_flush(&s->core); + ff_dca_xll_flush(&s->xll); - s->lfe_index = *channels - 1; - } - - s->channel_order_tab = s->xxch_order_tab; - } - - return 0; + s->core_residual_valid = 0; } -/** - * Main frame decoding function - * FIXME add arguments - */ -static int dca_decode_frame(AVCodecContext *avctx, void *data, - int *got_frame_ptr, AVPacket *avpkt) +static av_cold int dcadec_close(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; - float *src_chan; - float *dst_chan; DCAContext *s = avctx->priv_data; - int channels, full_channels; - float scale; - int achan; - int chset; - int mask; - int j, k; - int endch; - int upsample = 0; - - s->exss_ext_mask = 0; - s->xch_present = 0; - - s->dca_buffer_size = AVERROR_INVALIDDATA; - for (i = 0; i < buf_size - 3 && s->dca_buffer_size == AVERROR_INVALIDDATA; i++) - s->dca_buffer_size = avpriv_dca_convert_bitstream(buf + i, buf_size - i, 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; - - 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->audio_header.prim_channels; - - if (s->audio_header.prim_channels + !!s->lfe > 2 && - avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) { - /* Stereo downmix coefficients - * - * The decoder can only downmix to 2-channel, so we need to ensure - * embedded downmix coefficients are actually targeting 2-channel. - */ - if (s->core_downmix && (s->core_downmix_amode == DCA_STEREO || - s->core_downmix_amode == DCA_STEREO_TOTAL)) { - for (i = 0; i < num_core_channels + !!s->lfe; i++) { - /* Range checked earlier */ - s->downmix_coef[i][0] = dca_dmix_code(s->core_downmix_codes[i][0]); - s->downmix_coef[i][1] = dca_dmix_code(s->core_downmix_codes[i][1]); - } - s->output = s->core_downmix_amode; - } else { - int am = s->amode & DCA_CHANNEL_MASK; - 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(ff_dca_default_coeffs[0])) { - avpriv_request_sample(s->avctx, "Downmixing %d channels", - s->audio_header.prim_channels + !!s->lfe); - return AVERROR_PATCHWELCOME; - } - for (i = 0; i < num_core_channels + !!s->lfe; i++) { - s->downmix_coef[i][0] = ff_dca_default_coeffs[am][i][0]; - s->downmix_coef[i][1] = ff_dca_default_coeffs[am][i][1]; - } - } - ff_dlog(s->avctx, "Stereo downmix coeffs:\n"); - for (i = 0; i < num_core_channels + !!s->lfe; i++) { - ff_dlog(s->avctx, "L, input channel %d = %f\n", i, - s->downmix_coef[i][0]); - ff_dlog(s->avctx, "R, input channel %d = %f\n", i, - s->downmix_coef[i][1]); - } - ff_dlog(s->avctx, "\n"); - } - - if (s->ext_coding) - s->core_ext_mask = ff_dca_ext_audio_descr_mask[s->ext_descr]; - else - s->core_ext_mask = 0; - ret = scan_for_extensions(avctx); + ff_dca_core_close(&s->core); + ff_dca_xll_close(&s->xll); - avctx->profile = s->profile; + av_freep(&s->buffer); + s->buffer_size = 0; - full_channels = channels = s->audio_header.prim_channels + !!s->lfe; - - ret = set_channel_layout(avctx, &channels, num_core_channels); - if (ret < 0) - return ret; - - /* get output buffer */ - 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) - channels = s->xll_channels; - } - } - - if (avctx->channels != channels) { - if (avctx->channels) - av_log(avctx, AV_LOG_INFO, "Number of channels changed in DCA decoder (%d -> %d)\n", avctx->channels, channels); - avctx->channels = 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 ffmpeg 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) - return ret; - samples_flt = (float **) frame->extended_data; - - /* allocate buffer for extra channels if downmixing */ - if (avctx->channels < full_channels) { - ret = av_samples_get_buffer_size(NULL, full_channels - channels, - frame->nb_samples, - avctx->sample_fmt, 0); - if (ret < 0) - return ret; - - av_fast_malloc(&s->extra_channels_buffer, - &s->extra_channels_buffer_size, ret); - if (!s->extra_channels_buffer) - return AVERROR(ENOMEM); - - ret = av_samples_fill_arrays((uint8_t **) s->extra_channels, NULL, - s->extra_channels_buffer, - full_channels - channels, - frame->nb_samples, avctx->sample_fmt, 0); - if (ret < 0) - return ret; - } - - /* filter to get final output */ - 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 * block; - for (; ch < full_channels; ch++) - s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * block; - - 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 */ - if ((s->source_pcm_res & 1) && s->xch_present) { - float *back_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel]]; - float *lt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 2]]; - float *rt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 1]]; - s->fdsp->vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256); - s->fdsp->vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256); - } - - /* If stream contains XXCH, we might need to undo an embedded downmix */ - if (s->xxch_dmix_embedded) { - /* Loop over channel sets in turn */ - ch = num_core_channels; - for (chset = 0; chset < s->xxch_chset; chset++) { - endch = ch + s->xxch_chset_nch[chset]; - mask = s->xxch_dmix_embedded; - - /* undo downmix */ - for (j = ch; j < endch; j++) { - if (mask & (1 << j)) { /* this channel has been mixed-out */ - src_chan = s->samples_chanptr[s->channel_order_tab[j]]; - for (k = 0; k < endch; k++) { - achan = s->channel_order_tab[k]; - scale = s->xxch_dmix_coeff[j][k]; - if (scale != 0.0) { - dst_chan = s->samples_chanptr[achan]; - s->fdsp->vector_fmac_scalar(dst_chan, src_chan, - -scale, 256); - } - } - } - } - - /* if a downmix has been embedded then undo the pre-scaling */ - if ((mask & (1 << ch)) && s->xxch_dmix_sf[chset] != 1.0f) { - scale = s->xxch_dmix_sf[chset]; - - for (j = 0; j < ch; j++) { - src_chan = s->samples_chanptr[s->channel_order_tab[j]]; - for (k = 0; k < 256; k++) - src_chan[k] *= scale; - } - - /* LFE channel is always part of core, scale if it exists */ - if (s->lfe) { - src_chan = s->samples_chanptr[s->lfe_index]; - for (k = 0; k < 256; k++) - src_chan[k] *= scale; - } - } - - ch = endch; - } - - } - } - - /* update lfe history */ - 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 */ - ret = ff_side_data_update_matrix_encoding(frame, - (s->output & ~DCA_LFE) == DCA_STEREO_TOTAL ? - AV_MATRIX_ENCODING_DOLBY : AV_MATRIX_ENCODING_NONE); - if (ret < 0) - return ret; - - if ( avctx->profile != FF_PROFILE_DTS_HD_MA - && avctx->profile != FF_PROFILE_DTS_HD_HRA) - avctx->bit_rate = s->bit_rate; - *got_frame_ptr = 1; - - return buf_size; + return 0; } -/** - * DCA initialization - * - * @param avctx pointer to the AVCodecContext - */ - -static av_cold int dca_decode_init(AVCodecContext *avctx) +static av_cold int dcadec_init(AVCodecContext *avctx) { DCAContext *s = avctx->priv_data; s->avctx = avctx; - dca_init_vlcs(); + s->core.avctx = avctx; + s->exss.avctx = avctx; + s->xll.avctx = avctx; - s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT); - if (!s->fdsp) + if (ff_dca_core_init(&s->core) < 0) return AVERROR(ENOMEM); - ff_mdct_init(&s->imdct, 6, 1, 1.0); - ff_synth_filter_init(&s->synth); ff_dcadsp_init(&s->dcadsp); - ff_fmt_convert_init(&s->fmt_conv, avctx); + s->core.dcadsp = &s->dcadsp; + s->xll.dcadsp = &s->dcadsp; - avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; + switch (avctx->request_channel_layout & ~AV_CH_LAYOUT_NATIVE) { + case 0: + s->request_channel_layout = 0; + break; + case AV_CH_LAYOUT_STEREO: + case AV_CH_LAYOUT_STEREO_DOWNMIX: + s->request_channel_layout = DCA_SPEAKER_LAYOUT_STEREO; + break; + case AV_CH_LAYOUT_5POINT0: + s->request_channel_layout = DCA_SPEAKER_LAYOUT_5POINT0; + break; + case AV_CH_LAYOUT_5POINT1: + s->request_channel_layout = DCA_SPEAKER_LAYOUT_5POINT1; + break; + default: + av_log(avctx, AV_LOG_WARNING, "Invalid request_channel_layout\n"); + break; + } - /* allow downmixing to stereo */ - if (avctx->channels > 2 && - avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) - avctx->channels = 2; + avctx->sample_fmt = AV_SAMPLE_FMT_S32P; + avctx->bits_per_raw_sample = 24; return 0; } -static av_cold int dca_decode_end(AVCodecContext *avctx) -{ - DCAContext *s = avctx->priv_data; - ff_mdct_end(&s->imdct); - av_freep(&s->extra_channels_buffer); - av_freep(&s->fdsp); - av_freep(&s->xll_sample_buf); - av_freep(&s->qmf64_table); - return 0; -} +#define OFFSET(x) offsetof(DCAContext, x) +#define PARAM AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM -static const AVOption options[] = { - { "disable_xch", "disable decoding of the XCh extension", offsetof(DCAContext, xch_disable), AV_OPT_TYPE_BOOL, { .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_BOOL, { .i64 = 1 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM }, - { NULL }, +static const AVOption dcadec_options[] = { + { "core_only", "Decode core only without extensions", OFFSET(core_only), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, PARAM }, + { NULL } }; -static const AVClass dca_decoder_class = { +static const AVClass dcadec_class = { .class_name = "DCA decoder", .item_name = av_default_item_name, - .option = options, + .option = dcadec_options, .version = LIBAVUTIL_VERSION_INT, .category = AV_CLASS_CATEGORY_DECODER, }; AVCodec ff_dca_decoder = { - .name = "dca", - .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"), - .type = AVMEDIA_TYPE_AUDIO, - .id = AV_CODEC_ID_DTS, - .priv_data_size = sizeof(DCAContext), - .init = dca_decode_init, - .decode = dca_decode_frame, - .close = dca_decode_end, - .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(ff_dca_profiles), - .priv_class = &dca_decoder_class, + .name = "dca", + .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"), + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_CODEC_ID_DTS, + .priv_data_size = sizeof(DCAContext), + .init = dcadec_init, + .decode = dcadec_decode_frame, + .close = dcadec_close, + .flush = dcadec_flush, + .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF, + .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P, + AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }, + .priv_class = &dcadec_class, + .profiles = NULL_IF_CONFIG_SMALL(ff_dca_profiles), + .caps_internal = FF_CODEC_CAP_INIT_CLEANUP, };