diff options
Diffstat (limited to 'libavcodec/dcadec.c')
| -rw-r--r-- | libavcodec/dcadec.c | 1760 |
1 files changed, 277 insertions, 1483 deletions
diff --git a/libavcodec/dcadec.c b/libavcodec/dcadec.c index 9c1f878..4146a85 100644 --- a/libavcodec/dcadec.c +++ b/libavcodec/dcadec.c @@ -1,1606 +1,400 @@ /* - * 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 Libav. + * This file is part of FFmpeg. * - * Libav is free software; you can redistribute it and/or + * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * Libav is distributed in the hope that it will be useful, + * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with Libav; if not, write to the Free Software + * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#include <math.h> -#include <stddef.h> -#include <stdio.h> - -#include "libavutil/attributes.h" -#include "libavutil/channel_layout.h" -#include "libavutil/common.h" -#include "libavutil/float_dsp.h" -#include "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 "dca_syncwords.h" -#include "dcadata.h" -#include "dcadsp.h" +#include "dcadec.h" #include "dcahuff.h" -#include "fft.h" -#include "fmtconvert.h" -#include "get_bits.h" -#include "internal.h" -#include "mathops.h" +#include "dca_syncwords.h" #include "profiles.h" -#include "put_bits.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 -}; - -/* -1 are reserved or unknown */ -static const int dca_ext_audio_descr_mask[] = { - DCA_EXT_XCH, - -1, - DCA_EXT_X96, - DCA_EXT_XCH | DCA_EXT_X96, - -1, - -1, - DCA_EXT_XXCH, - -1, -}; - -/* Tables for mapping dts channel configurations to libavcodec multichannel api. - * Some compromises have been made for special configurations. Most configurations - * are never used so complete accuracy is not needed. - * - * L = left, R = right, C = center, S = surround, F = front, R = rear, T = total, OV = overhead. - * S -> side, when both rear and back are configured move one of them to the side channel - * OV -> center back - * All 2 channel configurations -> AV_CH_LAYOUT_STEREO - */ -static const uint64_t dca_core_channel_layout[] = { - AV_CH_FRONT_CENTER, ///< 1, A - AV_CH_LAYOUT_STEREO, ///< 2, A + B (dual mono) - AV_CH_LAYOUT_STEREO, ///< 2, L + R (stereo) - AV_CH_LAYOUT_STEREO, ///< 2, (L + R) + (L - R) (sum-difference) - AV_CH_LAYOUT_STEREO, ///< 2, LT + RT (left and right total) - AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER, ///< 3, C + L + R - AV_CH_LAYOUT_STEREO | AV_CH_BACK_CENTER, ///< 3, L + R + S - AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER, ///< 4, C + L + R + S - AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT, ///< 4, L + R + SL + SR - - AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_SIDE_LEFT | - AV_CH_SIDE_RIGHT, ///< 5, C + L + R + SL + SR - - AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT | - AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER, ///< 6, CL + CR + L + R + SL + SR - - AV_CH_LAYOUT_STEREO | AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT | - AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER, ///< 6, C + L + R + LR + RR + OV - - AV_CH_FRONT_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER | - AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_BACK_CENTER | - AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT, ///< 6, CF + CR + LF + RF + LR + RR - - AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER | - AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO | - AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT, ///< 7, CL + C + CR + L + R + SL + SR - - AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER | - AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT | - AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT, ///< 8, CL + CR + L + R + SL1 + SL2 + SR1 + SR2 - - AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER | - AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO | - AV_CH_SIDE_LEFT | AV_CH_BACK_CENTER | AV_CH_SIDE_RIGHT, ///< 8, CL + C + CR + L + R + SL + S + SR -}; - -#define DCA_DOLBY 101 /* FIXME */ - -#define DCA_CHANNEL_BITS 6 -#define DCA_CHANNEL_MASK 0x3F - -#define DCA_LFE 0x80 -#define HEADER_SIZE 14 +#define MIN_PACKET_SIZE 16 +#define MAX_PACKET_SIZE 0x104000 -#define DCA_NSYNCAUX 0x9A1105A0 - -/** 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) +int ff_dca_set_channel_layout(AVCodecContext *avctx, int *ch_remap, int dca_mask) { - return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + - ba->offset; -} - -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); + 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; } - 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; + avctx->channels = nchannels; + return nchannels; } -static inline void get_array(GetBitContext *gb, int *dst, int len, int bits) +void ff_dca_downmix_to_stereo_fixed(DCADSPContext *dcadsp, int32_t **samples, + int *coeff_l, int nsamples, int ch_mask) { - while (len--) - *dst++ = get_bits(gb, bits); -} + int pos, spkr, max_spkr = av_log2(ch_mask); + int *coeff_r = coeff_l + av_popcount(ch_mask); -static int dca_parse_audio_coding_header(DCAContext *s, int base_channel) -{ - int i, j; - static const uint8_t adj_table[4] = { 16, 18, 20, 23 }; - 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 }; + av_assert0(DCA_HAS_STEREO(ch_mask)); - s->audio_header.total_channels = get_bits(&s->gb, 3) + 1 + base_channel; - s->audio_header.prim_channels = s->audio_header.total_channels; + // 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); - if (s->audio_header.prim_channels > DCA_PRIM_CHANNELS_MAX) - s->audio_header.prim_channels = DCA_PRIM_CHANNELS_MAX; + // Downmix remaining channels + for (spkr = 0; spkr <= max_spkr; spkr++) { + if (!(ch_mask & (1U << spkr))) + continue; - 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] = 16; - - 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 (s->crc_present) { - /* Audio header CRC check */ - get_bits(&s->gb, 16); - } + if (*coeff_l && spkr != DCA_SPEAKER_L) + dcadsp->dmix_add(samples[DCA_SPEAKER_L], samples[spkr], + *coeff_l, nsamples); - s->current_subframe = 0; - s->current_subsubframe = 0; + if (*coeff_r && spkr != DCA_SPEAKER_R) + dcadsp->dmix_add(samples[DCA_SPEAKER_R], samples[spkr], + *coeff_r, nsamples); - return 0; + coeff_l++; + coeff_r++; + } } -static int dca_parse_frame_header(DCAContext *s) +void ff_dca_downmix_to_stereo_float(AVFloatDSPContext *fdsp, float **samples, + int *coeff_l, int nsamples, int ch_mask) { - 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; + 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); - 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) { - av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE value: %d\n", s->lfe); - return AVERROR_INVALIDDATA; - } + av_assert0(DCA_HAS_STEREO(ch_mask)); - /* TODO: check CRC */ - if (s->crc_present) - s->header_crc = get_bits(&s->gb, 16); + // 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); - 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); + // Downmix remaining channels + for (spkr = 0; spkr <= max_spkr; spkr++) { + if (!(ch_mask & (1U << spkr))) + continue; - /* FIXME: channels mixing levels */ - s->output = s->amode; - if (s->lfe) - s->output |= DCA_LFE; + if (*coeff_l && spkr != DCA_SPEAKER_L) + fdsp->vector_fmac_scalar(samples[DCA_SPEAKER_L], samples[spkr], + *coeff_l * scale, nsamples); - /* Primary audio coding header */ - s->audio_header.subframes = get_bits(&s->gb, 4) + 1; + if (*coeff_r && spkr != DCA_SPEAKER_R) + fdsp->vector_fmac_scalar(samples[DCA_SPEAKER_R], samples[spkr], + *coeff_r * scale, nsamples); - return dca_parse_audio_coding_header(s, 0); -} - -static inline int get_scale(GetBitContext *gb, int level, int value, int log2range) -{ - 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); - } + coeff_l++; + coeff_r++; } - return value; } -static int dca_subframe_header(DCAContext *s, int base_channel, int block_index) +static int dcadec_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { - /* Primary audio coding side information */ - int j, k; - - if (get_bits_left(&s->gb) < 0) + 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; + uint32_t mrk; + + if (input_size < MIN_PACKET_SIZE || input_size > MAX_PACKET_SIZE) { + av_log(avctx, AV_LOG_ERROR, "Invalid packet size\n"); return AVERROR_INVALIDDATA; - - 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->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); - } - } - } + // Convert input to BE format + mrk = AV_RB32(input); + if (mrk != DCA_SYNCWORD_CORE_BE && mrk != DCA_SYNCWORD_SUBSTREAM) { + av_fast_padded_malloc(&s->buffer, &s->buffer_size, input_size); + if (!s->buffer) + return AVERROR(ENOMEM); - /* 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]); - } + for (i = 0, ret = AVERROR_INVALIDDATA; i < input_size - MIN_PACKET_SIZE + 1 && ret < 0; i++) + ret = avpriv_dca_convert_bitstream(input + i, input_size - i, s->buffer, s->buffer_size); - 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; - } + if (ret < 0) { + av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n"); + return ret; } - } - /* 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]); - } - } + input = s->buffer; + input_size = ret; } - if (get_bits_left(&s->gb) < 0) - return AVERROR_INVALIDDATA; - - for (j = base_channel; j < s->audio_header.prim_channels; j++) { - const uint32_t *scale_table; - int scale_sum, log_size; - - memset(s->dca_chan[j].scale_factor, 0, - s->audio_header.subband_activity[j] * sizeof(s->dca_chan[j].scale_factor[0][0]) * 2); + s->packet = 0; - 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; - } + // Parse backward compatible core sub-stream + if (AV_RB32(input) == DCA_SYNCWORD_CORE_BE) { + int frame_size; - /* When huffman coded, only the difference is encoded */ - scale_sum = 0; + if ((ret = ff_dca_core_parse(&s->core, input, input_size)) < 0) + return ret; - 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]; - } + s->packet |= DCA_PACKET_CORE; - 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]; - } + // 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; } } - /* 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 (!s->core_only) { + DCAExssAsset *asset = NULL; - 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; + // 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]; } } - } - - /* 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) { - /* 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 */ - skip_bits(&s->gb, 1); - s->lfe_scale_factor = ff_dca_scale_factor_quant7[get_bits(&s->gb, 7)]; - - /* Quantization step size * scale factor */ - lfe_scale = 0.035 * s->lfe_scale_factor; - - 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[DCA_SUBBANDS][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) -{ - unsigned i, j; - QMF64_table *table = av_malloc(sizeof(*table)); - if (!table) - return NULL; - - for (i = 0; i < 32; i++) - for (j = 0; j < 32; j++) - table->dct4_coeff[i][j] = cos((2 * i + 1) * (2 * j + 1) * M_PI / 128); - for (i = 0; i < 32; i++) - for (j = 0; j < 32; j++) - table->dct2_coeff[i][j] = cos((2 * i + 1) * j * M_PI / 64); - - /* FIXME: Is the factor 0.125 = 1/8 right? */ - for (i = 0; i < 32; i++) - table->rcos[i] = 0.125 / cos((2 * i + 1) * M_PI / 256); - for (i = 0; i < 32; i++) - table->rsin[i] = -0.125 / sin((2 * i + 1) * M_PI / 256); - - return table; -} - -/* FIXME: Totally unoptimized. Based on the reference code and - * http://multimedia.cx/mirror/dca-transform.pdf, with guessed tweaks - * for doubling the size. */ -static void qmf_64_subbands(DCAContext *s, int chans, - float samples_in[DCA_SUBBANDS_X96K][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 < DCA_SUBBANDS_X96K; 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]); - } - - for (i = 0; i < DCA_SUBBANDS_X96K; i++) { - float out = raZ[i]; - for (j = 0; j < 1024; j += 128) - out += ff_dca_fir_64bands[j + i] * (raX[j + i] - raX[j + 63 - i]); - *samples_out++ = out * scale; - } - - for (i = 0; i < DCA_SUBBANDS_X96K; i++) { - float hist = 0.0; - for (j = 0; j < 1024; j += 128) - hist += ff_dca_fir_64bands[64 + j + i] * (-raX[i + j] - raX[j + 63 - i]); - - raZ[i] = hist; - } - - /* FIXME: Make buffer circular, to avoid this move. */ - memmove(raX + 64, raX, (1024 - 64) * sizeof(*raX)); - } -} - -static void lfe_interpolation_fir(DCAContext *s, const float *samples_in, - float *samples_out) -{ - /* samples_in: An array holding decimated samples. - * 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); - } -} - -/* 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) -{ - int c, l, r, sl, sr, s; - int i; - float t, u, v; - - switch (srcfmt) { - case DCA_MONO: - case DCA_4F2R: - av_log(NULL, 0, "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]; - } - } -} - -#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) -{ - int i; - int offset = (levels - 1) >> 1; - for (i = 0; i < 4; i++) { - int div = FASTDIV(code, levels); - values[i] = code - offset - div * levels; - code = div; - } - - 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 uint32_t *quant_step_table; - - /* - * Audio data - */ - - /* Select quantization step size table */ - if (s->bit_rate_index == 0x1f) - quant_step_table = ff_dca_lossless_quant; - else - quant_step_table = ff_dca_lossy_quant; - - for (k = base_channel; k < s->audio_header.prim_channels; k++) { - int32_t (*subband_samples)[8] = s->dca_chan[k].subband_samples[block_index]; - - 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]; - - uint32_t quant_step_size = quant_step_table[abits]; - - /* - * Extract bits from the bit stream - */ - if (!abits) - memset(subband_samples[l], 0, SAMPLES_PER_SUBBAND * - sizeof(subband_samples[l][0])); - else { - uint32_t rscale; - /* Deal with transients */ - int sfi = s->dca_chan[k].transition_mode[l] && - subsubframe >= s->dca_chan[k].transition_mode[l]; - /* Determine quantization index code book and its type. - Select quantization index code book */ - int sel = s->audio_header.quant_index_huffman[k][abits]; - - rscale = (s->dca_chan[k].scale_factor[l][sfi] * - s->audio_header.scalefactor_adj[k][sel] + 8) >> 4; - - 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, subband_samples[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++) - subband_samples[l][m] = get_sbits(&s->gb, abits - 3); - } - } else { - /* Huffman coded */ - for (m = 0; m < SAMPLES_PER_SUBBAND; m++) - subband_samples[l][m] = get_bitalloc(&s->gb, - &dca_smpl_bitalloc[abits], sel); - } - s->dcadsp.dequantize(subband_samples[l], quant_step_size, rscale); + // 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; } } - 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] * - (int64_t)s->dca_chan[k].subband_samples_hist[l][3] + - ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][1] * - (int64_t)s->dca_chan[k].subband_samples_hist[l][2] + - ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][2] * - (int64_t)s->dca_chan[k].subband_samples_hist[l][1] + - ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][3] * - (int64_t)s->dca_chan[k].subband_samples_hist[l][0]) + - (1 << 12) >> 13; - for (m = 1; m < SAMPLES_PER_SUBBAND; m++) { - int64_t sum = ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] * - (int64_t)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] * - (int64_t)subband_samples[l][m - n]; - else if (s->predictor_history) - sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] * - (int64_t)s->dca_chan[k].subband_samples_hist[l][m - n + 4]; - subband_samples[l][m] += (int32_t)(sum + (1 << 12) >> 13); - } - } - - } - /* 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; + // Parse LBR component in EXSS + if (asset && (asset->extension_mask & DCA_EXSS_LBR)) { + if ((ret = ff_dca_lbr_parse(&s->lbr, input, asset)) < 0) { + if (ret == AVERROR(ENOMEM) || (avctx->err_recognition & AV_EF_EXPLODE)) + return ret; + } else { + s->packet |= DCA_PACKET_LBR; } - - 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]); } - } - /* 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; - } + // 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; } - return 0; -} - -static int dca_filter_channels(DCAContext *s, int block_index, int upsample, int downmix) -{ - int k; - - if (upsample) { - LOCAL_ALIGNED(32, float, samples, [DCA_SUBBANDS_X96K], [SAMPLES_PER_SUBBAND]); + // Filter the frame + if (s->packet & DCA_PACKET_LBR) { + if ((ret = ff_dca_lbr_filter_frame(&s->lbr, frame)) < 0) + return ret; + } else if (s->packet & DCA_PACKET_XLL) { + if (s->packet & DCA_PACKET_CORE) { + int x96_synth = -1; + + // Enable X96 synthesis if needed + if (s->xll.chset[0].freq == 96000 && s->core.sample_rate == 48000) + x96_synth = 1; + + if ((ret = ff_dca_core_filter_fixed(&s->core, x96_synth)) < 0) + return ret; + + // 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 (!(prev_packet & DCA_PACKET_RESIDUAL) && s->xll.nreschsets > 0 + && s->xll.nchsets > 1) { + av_log(avctx, AV_LOG_VERBOSE, "Forcing XLL recovery mode\n"); + s->packet |= DCA_PACKET_RECOVERY; + } - if (!s->qmf64_table) { - s->qmf64_table = qmf64_precompute(); - if (!s->qmf64_table) - return AVERROR(ENOMEM); + // Set 'residual ok' flag for the next frame + s->packet |= DCA_PACKET_RESIDUAL; } - /* 64 subbands QMF */ - for (k = 0; k < s->audio_header.prim_channels; k++) { - int channel = s->channel_order_tab[k]; - int32_t (*subband_samples)[SAMPLES_PER_SUBBAND] = - s->dca_chan[k].subband_samples[block_index]; - - s->fmt_conv.int32_to_float(samples[0], subband_samples[0], - DCA_SUBBANDS_X96K * SAMPLES_PER_SUBBAND); - - if (channel >= 0) - qmf_64_subbands(s, k, samples, - s->samples_chanptr[channel], - /* Upsampling needs a factor 2 here. */ - M_SQRT2 / 32768.0); + 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) + return ret; } + } else if (s->packet & DCA_PACKET_CORE) { + if ((ret = ff_dca_core_filter_frame(&s->core, frame)) < 0) + return ret; + if (s->core.filter_mode & DCA_FILTER_MODE_FIXED) + s->packet |= DCA_PACKET_RESIDUAL; } else { - /* 32 subbands QMF */ - LOCAL_ALIGNED(32, float, samples, [DCA_SUBBANDS], [SAMPLES_PER_SUBBAND]); - - for (k = 0; k < s->audio_header.prim_channels; k++) { - int channel = s->channel_order_tab[k]; - int32_t (*subband_samples)[SAMPLES_PER_SUBBAND] = - s->dca_chan[k].subband_samples[block_index]; - - s->fmt_conv.int32_to_float(samples[0], subband_samples[0], - DCA_SUBBANDS * SAMPLES_PER_SUBBAND); - - if (channel >= 0) - qmf_32_subbands(s, k, samples, - s->samples_chanptr[channel], - M_SQRT1_2 / 32768.0); - } - } - - /* Generate LFE samples for this subsubframe FIXME!!! */ - if (s->lfe) { - float *samples = s->samples_chanptr[ff_dca_lfe_index[s->amode]]; - lfe_interpolation_fir(s, - s->lfe_data + 2 * s->lfe * (block_index + 4), - samples); - if (upsample) { - unsigned i; - /* Should apply the filter in Table 6-11 when upsampling. For - * now, just duplicate. */ - for (i = 511; i > 0; i--) { - samples[2 * i] = - samples[2 * i + 1] = samples[i]; - } - samples[1] = samples[0]; - } - } - - /* FIXME: This downmixing is probably broken with upsample. - * Probably totally broken also with XLL in general. */ - /* Downmixing to Stereo */ - if (downmix) { - 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); - } - } - - 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); + av_log(avctx, AV_LOG_ERROR, "No valid DCA sub-stream found\n"); + if (s->core_only) + av_log(avctx, AV_LOG_WARNING, "Consider disabling 'core_only' option\n"); return AVERROR_INVALIDDATA; } - if (!s->current_subsubframe) { - /* Read subframe header */ - if ((ret = dca_subframe_header(s, base_channel, block_index))) - 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; - } - - return 0; -} + *got_frame_ptr = 1; -static float dca_dmix_code(unsigned code) -{ - int sign = (code >> 8) - 1; - code &= 0xff; - return ((ff_dca_dmixtable[code] ^ sign) - sign) * (1.0 / (1U << 15)); + return avpkt->size; } -static int scan_for_extensions(AVCodecContext *avctx) +static av_cold void dcadec_flush(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) { - /* 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; - } - - /* much like core primary audio coding header */ - dca_parse_audio_coding_header(s, s->xch_base_channel); - for (i = 0; i < (s->sample_blocks / 8); i++) - if ((ret = dca_decode_block(s, s->xch_base_channel, i))) { - av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n"); - continue; - } + ff_dca_core_flush(&s->core); + ff_dca_xll_flush(&s->xll); + ff_dca_lbr_flush(&s->lbr); - 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; - 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 { - /* no supported extensions, skip the rest of the core substream */ - skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb)); - } - - if (s->core_ext_mask & DCA_EXT_X96) - s->profile = FF_PROFILE_DTS_96_24; - else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH)) - s->profile = FF_PROFILE_DTS_ES; - - /* check for ExSS (HD part) */ - if (s->dca_buffer_size - s->frame_size > 32 && - get_bits_long(&s->gb, 32) == DCA_SYNCWORD_SUBSTREAM) - ff_dca_exss_parse_header(s); - - return ret; + s->packet &= DCA_PACKET_MASK; } -static int set_channel_layout(AVCodecContext *avctx, int channels) +static av_cold int dcadec_close(AVCodecContext *avctx) { DCAContext *s = avctx->priv_data; - int num_core_channels = s->audio_header.prim_channels; - int i; - - if (s->amode < 16) { - avctx->channel_layout = 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) { - 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]; - } - } 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]; - } + ff_dca_core_close(&s->core); + ff_dca_xll_close(&s->xll); + ff_dca_lbr_close(&s->lbr); - if (channels < ff_dca_channels[s->amode] + !!s->lfe) - return AVERROR_INVALIDDATA; - - if (channels > !!s->lfe && - s->channel_order_tab[channels - 1 - !!s->lfe] < 0) - 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; - - /* 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"); - } - } else { - av_log(avctx, AV_LOG_ERROR, "Nonstandard configuration %d !\n", s->amode); - return AVERROR_INVALIDDATA; - } + av_freep(&s->buffer); + s->buffer_size = 0; return 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_init(AVCodecContext *avctx) { - AVFrame *frame = data; - const uint8_t *buf = avpkt->data; - int buf_size = avpkt->size; - - int lfe_samples; - int i, ret; - float **samples_flt; DCAContext *s = avctx->priv_data; - int channels, full_channels; - int upsample = 0; - int downmix; - - s->exss_ext_mask = 0; - s->xch_present = 0; - s->dca_buffer_size = ff_dca_convert_bitstream(buf, buf_size, s->dca_buffer, - DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE); - if (s->dca_buffer_size == AVERROR_INVALIDDATA) { - av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n"); - return AVERROR_INVALIDDATA; - } - - if ((ret = dca_parse_frame_header(s)) < 0) { - // seems like the frame is corrupt, try with the next one - return ret; - } - // set AVCodec values with parsed data - avctx->sample_rate = s->sample_rate; - avctx->bit_rate = s->bit_rate; - - s->profile = FF_PROFILE_DTS; - - for (i = 0; i < (s->sample_blocks / SAMPLES_PER_SUBBAND); i++) { - if ((ret = dca_decode_block(s, 0, i))) { - av_log(avctx, AV_LOG_ERROR, "error decoding block\n"); - return ret; - } - } - - if (s->ext_coding) - s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr]; - else - s->core_ext_mask = 0; - - ret = scan_for_extensions(avctx); - - avctx->profile = s->profile; - - full_channels = channels = s->audio_header.prim_channels + !!s->lfe; - - ret = set_channel_layout(avctx, channels); - if (ret < 0) - return ret; - avctx->channels = channels; - - /* 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) - avctx->channels += s->xll_channels - s->xll_residual_channels; - } - } - - /* FIXME: This is an ugly hack, to just revert to the default - * layout if we have additional channels. Need to convert the XLL - * channel masks to libav channel_layout mask. */ - if (av_get_channel_layout_nb_channels(avctx->channel_layout) != avctx->channels) - avctx->channel_layout = 0; - - if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) { - av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); - return ret; - } - 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; - } - - downmix = s->audio_header.prim_channels > 2 && - avctx->request_channel_layout == AV_CH_LAYOUT_STEREO; - - /* 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, downmix); - - /* 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); - } - } - - /* 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; - - *got_frame_ptr = 1; - - return buf_size; -} + s->avctx = avctx; + s->core.avctx = avctx; + s->exss.avctx = avctx; + s->xll.avctx = avctx; + s->lbr.avctx = avctx; -/** - * DCA initialization - * - * @param avctx pointer to the AVCodecContext - */ + ff_dca_init_vlcs(); -static av_cold int dca_decode_init(AVCodecContext *avctx) -{ - DCAContext *s = avctx->priv_data; + if (ff_dca_core_init(&s->core) < 0) + return AVERROR(ENOMEM); - s->avctx = avctx; - dca_init_vlcs(); + if (ff_dca_lbr_init(&s->lbr) < 0) + return AVERROR(ENOMEM); - avpriv_float_dsp_init(&s->fdsp, avctx->flags & AV_CODEC_FLAG_BITEXACT); - ff_mdct_init(&s->imdct, 6, 1, 1.0); - ff_synth_filter_init(&s->synth); ff_dcadsp_init(&s->dcadsp); - ff_fmt_convert_init(&s->fmt_conv, avctx); + s->core.dcadsp = &s->dcadsp; + s->xll.dcadsp = &s->dcadsp; + s->lbr.dcadsp = &s->dcadsp; - avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; + s->crctab = av_crc_get_table(AV_CRC_16_CCITT); - /* allow downmixing to stereo */ - if (avctx->channels > 2 && - avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) - avctx->channels = 2; + 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; + } 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->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_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM }, - { "disable_xll", "disable decoding of the XLL extension", offsetof(DCAContext, xll_disable), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM }, - { NULL }, +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, }; |
