diff --git a/Changelog b/Changelog index 0d465aa1ed..7821e72c87 100644 --- a/Changelog +++ b/Changelog @@ -41,6 +41,7 @@ version : - Atrac1 decoder - MD STUDIO audio demuxer - RF64 support in WAV demuxer +- MPEG-4 Audio Lossless Coding (ALS) decoder diff --git a/doc/general.texi b/doc/general.texi index e1402f50dd..d5f39f36f5 100644 --- a/doc/general.texi +++ b/doc/general.texi @@ -569,6 +569,7 @@ following image formats are supported: @item MP2 (MPEG audio layer 2) @tab IX @tab IX @item MP3 (MPEG audio layer 3) @tab E @tab IX @tab encoding supported through external library LAME, ADU MP3 and MP3onMP4 also supported +@item MPEG-4 Audio Lossless Coding (ALS) @tab @tab X @item Musepack SV7 @tab @tab X @item Musepack SV8 @tab @tab X @item Nellymoser Asao @tab X @tab X diff --git a/libavcodec/Makefile b/libavcodec/Makefile index 21cba37b58..dc572e8cf4 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -51,6 +51,7 @@ OBJS-$(CONFIG_AC3_DECODER) += ac3dec.o ac3tab.o \ OBJS-$(CONFIG_AC3_ENCODER) += ac3enc.o ac3tab.o ac3.o OBJS-$(CONFIG_ALAC_DECODER) += alac.o OBJS-$(CONFIG_ALAC_ENCODER) += alacenc.o +OBJS-$(CONFIG_ALS_DECODER) += alsdec.o OBJS-$(CONFIG_AMV_DECODER) += sp5xdec.o mjpegdec.o mjpeg.o OBJS-$(CONFIG_APE_DECODER) += apedec.o OBJS-$(CONFIG_ASV1_DECODER) += asv1.o mpeg12data.o diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c index 32c86cd523..6a2eec26f2 100644 --- a/libavcodec/allcodecs.c +++ b/libavcodec/allcodecs.c @@ -200,6 +200,7 @@ void avcodec_register_all(void) REGISTER_ENCDEC (AAC, aac); REGISTER_ENCDEC (AC3, ac3); REGISTER_ENCDEC (ALAC, alac); + REGISTER_DECODER (ALS, als); REGISTER_DECODER (APE, ape); REGISTER_DECODER (ATRAC1, atrac1); REGISTER_DECODER (ATRAC3, atrac3); diff --git a/libavcodec/als_data.h b/libavcodec/als_data.h new file mode 100644 index 0000000000..8c052cde53 --- /dev/null +++ b/libavcodec/als_data.h @@ -0,0 +1,95 @@ +/* + * ALS header file for common data + * Copyright (c) 2009 Thilo Borgmann + * + * This file is part of FFmpeg. + * + * 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. + * + * 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 FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVCODEC_ALS_DATA_H +#define AVCODEC_ALS_DATA_H + +/** + * @file libavcodec/als_data.h + * MPEG-4 ALS common data tables + * @author Thilo Borgmann + */ + + +#include + +/** Rice parameters and corresponding index offsets for decoding the + * indices of scaled PARCOR values. The table choosen is set globally + * by the encoder and stored in ALSSpecificConfig. + */ +static const int8_t parcor_rice_table[3][20][2] = { + { {-52, 4}, {-29, 5}, {-31, 4}, { 19, 4}, {-16, 4}, + { 12, 3}, { -7, 3}, { 9, 3}, { -5, 3}, { 6, 3}, + { -4, 3}, { 3, 3}, { -3, 2}, { 3, 2}, { -2, 2}, + { 3, 2}, { -1, 2}, { 2, 2}, { -1, 2}, { 2, 2} }, + { {-58, 3}, {-42, 4}, {-46, 4}, { 37, 5}, {-36, 4}, + { 29, 4}, {-29, 4}, { 25, 4}, {-23, 4}, { 20, 4}, + {-17, 4}, { 16, 4}, {-12, 4}, { 12, 3}, {-10, 4}, + { 7, 3}, { -4, 4}, { 3, 3}, { -1, 3}, { 1, 3} }, + { {-59, 3}, {-45, 5}, {-50, 4}, { 38, 4}, {-39, 4}, + { 32, 4}, {-30, 4}, { 25, 3}, {-23, 3}, { 20, 3}, + {-20, 3}, { 16, 3}, {-13, 3}, { 10, 3}, { -7, 3}, + { 3, 3}, { 0, 3}, { -1, 3}, { 2, 3}, { -1, 2} } +}; + + +/** Scaled PARCOR values used for the first two PARCOR coefficients. + * To be indexed by the Rice coded indices. + * Generated by: parcor_scaled_values[i] = 32 + ((i * (i+1)) << 7) - (1 << 20) + * Actual values are divided by 32 in order to be stored in 16 bits. + */ +static const int16_t parcor_scaled_values[] = { + -1048544 / 32, -1048288 / 32, -1047776 / 32, -1047008 / 32, + -1045984 / 32, -1044704 / 32, -1043168 / 32, -1041376 / 32, + -1039328 / 32, -1037024 / 32, -1034464 / 32, -1031648 / 32, + -1028576 / 32, -1025248 / 32, -1021664 / 32, -1017824 / 32, + -1013728 / 32, -1009376 / 32, -1004768 / 32, -999904 / 32, + -994784 / 32, -989408 / 32, -983776 / 32, -977888 / 32, + -971744 / 32, -965344 / 32, -958688 / 32, -951776 / 32, + -944608 / 32, -937184 / 32, -929504 / 32, -921568 / 32, + -913376 / 32, -904928 / 32, -896224 / 32, -887264 / 32, + -878048 / 32, -868576 / 32, -858848 / 32, -848864 / 32, + -838624 / 32, -828128 / 32, -817376 / 32, -806368 / 32, + -795104 / 32, -783584 / 32, -771808 / 32, -759776 / 32, + -747488 / 32, -734944 / 32, -722144 / 32, -709088 / 32, + -695776 / 32, -682208 / 32, -668384 / 32, -654304 / 32, + -639968 / 32, -625376 / 32, -610528 / 32, -595424 / 32, + -580064 / 32, -564448 / 32, -548576 / 32, -532448 / 32, + -516064 / 32, -499424 / 32, -482528 / 32, -465376 / 32, + -447968 / 32, -430304 / 32, -412384 / 32, -394208 / 32, + -375776 / 32, -357088 / 32, -338144 / 32, -318944 / 32, + -299488 / 32, -279776 / 32, -259808 / 32, -239584 / 32, + -219104 / 32, -198368 / 32, -177376 / 32, -156128 / 32, + -134624 / 32, -112864 / 32, -90848 / 32, -68576 / 32, + -46048 / 32, -23264 / 32, -224 / 32, 23072 / 32, + 46624 / 32, 70432 / 32, 94496 / 32, 118816 / 32, + 143392 / 32, 168224 / 32, 193312 / 32, 218656 / 32, + 244256 / 32, 270112 / 32, 296224 / 32, 322592 / 32, + 349216 / 32, 376096 / 32, 403232 / 32, 430624 / 32, + 458272 / 32, 486176 / 32, 514336 / 32, 542752 / 32, + 571424 / 32, 600352 / 32, 629536 / 32, 658976 / 32, + 688672 / 32, 718624 / 32, 748832 / 32, 779296 / 32, + 810016 / 32, 840992 / 32, 872224 / 32, 903712 / 32, + 935456 / 32, 967456 / 32, 999712 / 32, 1032224 / 32 +}; + + +#endif /* AVCODEC_ALS_DATA_H */ diff --git a/libavcodec/alsdec.c b/libavcodec/alsdec.c new file mode 100644 index 0000000000..47b708faf9 --- /dev/null +++ b/libavcodec/alsdec.c @@ -0,0 +1,997 @@ +/* + * MPEG-4 ALS decoder + * Copyright (c) 2009 Thilo Borgmann + * + * This file is part of FFmpeg. + * + * 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. + * + * 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 FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file libavcodec/alsdec.c + * MPEG-4 ALS decoder + * @author Thilo Borgmann + */ + + +//#define DEBUG + + +#include "avcodec.h" +#include "get_bits.h" +#include "unary.h" +#include "mpeg4audio.h" +#include "bytestream.h" + +#include "als_data.h" + +enum RA_Flag { + RA_FLAG_NONE, + RA_FLAG_FRAMES, + RA_FLAG_HEADER +}; + + +typedef struct { + uint32_t samples; ///< number of samples, 0xFFFFFFFF if unknown + int resolution; ///< 000 = 8-bit; 001 = 16-bit; 010 = 24-bit; 011 = 32-bit + int floating; ///< 1 = IEEE 32-bit floating-point, 0 = integer + int frame_length; ///< frame length for each frame (last frame may differ) + int ra_distance; ///< distance between RA frames (in frames, 0...255) + enum RA_Flag ra_flag; ///< indicates where the size of ra units is stored + int adapt_order; ///< adaptive order: 1 = on, 0 = off + int coef_table; ///< table index of Rice code parameters + int long_term_prediction; ///< long term prediction (LTP): 1 = on, 0 = off + int max_order; ///< maximum prediction order (0..1023) + int block_switching; ///< number of block switching levels + int bgmc; ///< "Block Gilbert-Moore Code": 1 = on, 0 = off (Rice coding only) + int sb_part; ///< sub-block partition + int joint_stereo; ///< joint stereo: 1 = on, 0 = off + int mc_coding; ///< extended inter-channel coding (multi channel coding): 1 = on, 0 = off + int chan_config; ///< indicates that a chan_config_info field is present + int chan_sort; ///< channel rearrangement: 1 = on, 0 = off + int rlslms; ///< use "Recursive Least Square-Least Mean Square" predictor: 1 = on, 0 = off + int chan_config_info; ///< mapping of channels to loudspeaker locations. Unused until setting channel configuration is implemented. + int *chan_pos; ///< original channel positions + uint32_t header_size; ///< header size of original audio file in bytes, provided for debugging + uint32_t trailer_size; ///< trailer size of original audio file in bytes, provided for debugging +} ALSSpecificConfig; + + +typedef struct { + AVCodecContext *avctx; + ALSSpecificConfig sconf; + GetBitContext gb; + unsigned int cur_frame_length; ///< length of the current frame to decode + unsigned int frame_id; ///< the frame ID / number of the current frame + unsigned int js_switch; ///< if true, joint-stereo decoding is enforced + unsigned int num_blocks; ///< number of blocks used in the current frame + int32_t *quant_cof; ///< quantized parcor coefficients + int32_t *lpc_cof; ///< coefficients of the direct form prediction filter + int32_t *prev_raw_samples; ///< contains unshifted raw samples from the previous block + int32_t **raw_samples; ///< decoded raw samples for each channel + int32_t *raw_buffer; ///< contains all decoded raw samples including carryover samples +} ALSDecContext; + + +static av_cold void dprint_specific_config(ALSDecContext *ctx) +{ +#ifdef DEBUG + AVCodecContext *avctx = ctx->avctx; + ALSSpecificConfig *sconf = &ctx->sconf; + + dprintf(avctx, "resolution = %i\n", sconf->resolution); + dprintf(avctx, "floating = %i\n", sconf->floating); + dprintf(avctx, "frame_length = %i\n", sconf->frame_length); + dprintf(avctx, "ra_distance = %i\n", sconf->ra_distance); + dprintf(avctx, "ra_flag = %i\n", sconf->ra_flag); + dprintf(avctx, "adapt_order = %i\n", sconf->adapt_order); + dprintf(avctx, "coef_table = %i\n", sconf->coef_table); + dprintf(avctx, "long_term_prediction = %i\n", sconf->long_term_prediction); + dprintf(avctx, "max_order = %i\n", sconf->max_order); + dprintf(avctx, "block_switching = %i\n", sconf->block_switching); + dprintf(avctx, "bgmc = %i\n", sconf->bgmc); + dprintf(avctx, "sb_part = %i\n", sconf->sb_part); + dprintf(avctx, "joint_stereo = %i\n", sconf->joint_stereo); + dprintf(avctx, "mc_coding = %i\n", sconf->mc_coding); + dprintf(avctx, "chan_config = %i\n", sconf->chan_config); + dprintf(avctx, "chan_sort = %i\n", sconf->chan_sort); + dprintf(avctx, "RLSLMS = %i\n", sconf->rlslms); + dprintf(avctx, "chan_config_info = %i\n", sconf->chan_config_info); + dprintf(avctx, "header_size = %i\n", sconf->header_size); + dprintf(avctx, "trailer_size = %i\n", sconf->trailer_size); +#endif +} + + +/** Reads an ALSSpecificConfig from a buffer into the output struct. + */ +static av_cold int read_specific_config(ALSDecContext *ctx) +{ + GetBitContext gb; + uint64_t ht_size; + int i, config_offset, crc_enabled; + MPEG4AudioConfig m4ac; + ALSSpecificConfig *sconf = &ctx->sconf; + AVCodecContext *avctx = ctx->avctx; + uint32_t als_id; + + init_get_bits(&gb, avctx->extradata, avctx->extradata_size * 8); + + config_offset = ff_mpeg4audio_get_config(&m4ac, avctx->extradata, + avctx->extradata_size); + + if (config_offset < 0) + return -1; + + skip_bits_long(&gb, config_offset); + + if (get_bits_left(&gb) < (30 << 3)) + return -1; + + // read the fixed items + als_id = get_bits_long(&gb, 32); + avctx->sample_rate = m4ac.sample_rate; + skip_bits_long(&gb, 32); // sample rate already known + sconf->samples = get_bits_long(&gb, 32); + avctx->channels = m4ac.channels; + skip_bits(&gb, 16); // number of channels already knwon + skip_bits(&gb, 3); // skip file_type + sconf->resolution = get_bits(&gb, 3); + sconf->floating = get_bits1(&gb); + skip_bits1(&gb); // skip msb_first + sconf->frame_length = get_bits(&gb, 16) + 1; + sconf->ra_distance = get_bits(&gb, 8); + sconf->ra_flag = get_bits(&gb, 2); + sconf->adapt_order = get_bits1(&gb); + sconf->coef_table = get_bits(&gb, 2); + sconf->long_term_prediction = get_bits1(&gb); + sconf->max_order = get_bits(&gb, 10); + sconf->block_switching = get_bits(&gb, 2); + sconf->bgmc = get_bits1(&gb); + sconf->sb_part = get_bits1(&gb); + sconf->joint_stereo = get_bits1(&gb); + sconf->mc_coding = get_bits1(&gb); + sconf->chan_config = get_bits1(&gb); + sconf->chan_sort = get_bits1(&gb); + crc_enabled = get_bits1(&gb); + sconf->rlslms = get_bits1(&gb); + skip_bits(&gb, 5); // skip 5 reserved bits + skip_bits1(&gb); // skip aux_data_enabled + + + // check for ALSSpecificConfig struct + if (als_id != MKBETAG('A','L','S','\0')) + return -1; + + ctx->cur_frame_length = sconf->frame_length; + + // allocate quantized parcor coefficient buffer + if (!(ctx->quant_cof = av_malloc(sizeof(*ctx->quant_cof) * sconf->max_order)) || + !(ctx->lpc_cof = av_malloc(sizeof(*ctx->lpc_cof) * sconf->max_order))) { + av_log(avctx, AV_LOG_ERROR, "Allocating buffer memory failed.\n"); + return AVERROR(ENOMEM); + } + + // read channel config + if (sconf->chan_config) + sconf->chan_config_info = get_bits(&gb, 16); + // TODO: use this to set avctx->channel_layout + + + // read channel sorting + if (sconf->chan_sort && avctx->channels > 1) { + int chan_pos_bits = av_ceil_log2(avctx->channels); + int bits_needed = avctx->channels * chan_pos_bits + 7; + if (get_bits_left(&gb) < bits_needed) + return -1; + + if (!(sconf->chan_pos = av_malloc(avctx->channels * sizeof(*sconf->chan_pos)))) + return AVERROR(ENOMEM); + + for (i = 0; i < avctx->channels; i++) + sconf->chan_pos[i] = get_bits(&gb, chan_pos_bits); + + align_get_bits(&gb); + // TODO: use this to actually do channel sorting + } else { + sconf->chan_sort = 0; + } + + + // read fixed header and trailer sizes, + // if size = 0xFFFFFFFF then there is no data field! + if (get_bits_left(&gb) < 64) + return -1; + + sconf->header_size = get_bits_long(&gb, 32); + sconf->trailer_size = get_bits_long(&gb, 32); + if (sconf->header_size == 0xFFFFFFFF) + sconf->header_size = 0; + if (sconf->trailer_size == 0xFFFFFFFF) + sconf->trailer_size = 0; + + ht_size = ((int64_t)(sconf->header_size) + (int64_t)(sconf->trailer_size)) << 3; + + + // skip the header and trailer data + if (get_bits_left(&gb) < ht_size) + return -1; + + if (ht_size > INT32_MAX) + return -1; + + skip_bits_long(&gb, ht_size); + + + // skip the crc data + if (crc_enabled) { + if (get_bits_left(&gb) < 32) + return -1; + + skip_bits_long(&gb, 32); + } + + + // no need to read the rest of ALSSpecificConfig (ra_unit_size & aux data) + + dprint_specific_config(ctx); + + return 0; +} + + +/** Checks the ALSSpecificConfig for unsupported features. + */ +static int check_specific_config(ALSDecContext *ctx) +{ + ALSSpecificConfig *sconf = &ctx->sconf; + int error = 0; + + // report unsupported feature and set error value + #define MISSING_ERR(cond, str, errval) \ + { \ + if (cond) { \ + av_log_missing_feature(ctx->avctx, str, 0); \ + error = errval; \ + } \ + } + + MISSING_ERR(sconf->floating, "Floating point decoding", -1); + MISSING_ERR(sconf->long_term_prediction, "Long-term prediction", -1); + MISSING_ERR(sconf->bgmc, "BGMC entropy decoding", -1); + MISSING_ERR(sconf->mc_coding, "Multi-channel correlation", -1); + MISSING_ERR(sconf->rlslms, "Adaptive RLS-LMS prediction", -1); + MISSING_ERR(sconf->chan_sort, "Channel sorting", 0); + + return error; +} + + +/** Parses the bs_info field to extract the block partitioning used in + * block switching mode, refer to ISO/IEC 14496-3, section 11.6.2. + */ +static void parse_bs_info(const uint32_t bs_info, unsigned int n, + unsigned int div, unsigned int **div_blocks, + unsigned int *num_blocks) +{ + if (n < 31 && ((bs_info << n) & 0x40000000)) { + // if the level is valid and the investigated bit n is set + // then recursively check both children at bits (2n+1) and (2n+2) + n *= 2; + div += 1; + parse_bs_info(bs_info, n + 1, div, div_blocks, num_blocks); + parse_bs_info(bs_info, n + 2, div, div_blocks, num_blocks); + } else { + // else the bit is not set or the last level has been reached + // (bit implicitly not set) + **div_blocks = div; + (*div_blocks)++; + (*num_blocks)++; + } +} + + +/** Reads and decodes a Rice codeword. + */ +static int32_t decode_rice(GetBitContext *gb, unsigned int k) +{ + int max = gb->size_in_bits - get_bits_count(gb) - k; + int q = get_unary(gb, 0, max); + int r = k ? get_bits1(gb) : !(q & 1); + + if (k > 1) { + q <<= (k - 1); + q += get_bits_long(gb, k - 1); + } else if (!k) { + q >>= 1; + } + return r ? q : ~q; +} + + +/** Converts PARCOR coefficient k to direct filter coefficient. + */ +static void parcor_to_lpc(unsigned int k, const int32_t *par, int32_t *cof) +{ + int i, j; + + for (i = 0, j = k - 1; i < j; i++, j--) { + int tmp1 = ((MUL64(par[k], cof[j]) + (1 << 19)) >> 20); + cof[j] += ((MUL64(par[k], cof[i]) + (1 << 19)) >> 20); + cof[i] += tmp1; + } + if (i == j) + cof[i] += ((MUL64(par[k], cof[j]) + (1 << 19)) >> 20); + + cof[k] = par[k]; +} + + +/** Reads block switching field if necessary and sets actual block sizes. + * Also assures that the block sizes of the last frame correspond to the + * actual number of samples. + */ +static void get_block_sizes(ALSDecContext *ctx, unsigned int *div_blocks, + uint32_t *bs_info) +{ + ALSSpecificConfig *sconf = &ctx->sconf; + GetBitContext *gb = &ctx->gb; + unsigned int *ptr_div_blocks = div_blocks; + unsigned int b; + + if (sconf->block_switching) { + unsigned int bs_info_len = 1 << (sconf->block_switching + 2); + *bs_info = get_bits_long(gb, bs_info_len); + *bs_info <<= (32 - bs_info_len); + } + + ctx->num_blocks = 0; + parse_bs_info(*bs_info, 0, 0, &ptr_div_blocks, &ctx->num_blocks); + + // The last frame may have an overdetermined block structure given in + // the bitstream. In that case the defined block structure would need + // more samples than available to be consistent. + // The block structure is actually used but the block sizes are adapted + // to fit the actual number of available samples. + // Example: 5 samples, 2nd level block sizes: 2 2 2 2. + // This results in the actual block sizes: 2 2 1 0. + // This is not specified in 14496-3 but actually done by the reference + // codec RM22 revision 2. + // This appears to happen in case of an odd number of samples in the last + // frame which is actually not allowed by the block length switching part + // of 14496-3. + // The ALS conformance files feature an odd number of samples in the last + // frame. + + for (b = 0; b < ctx->num_blocks; b++) + div_blocks[b] = ctx->sconf.frame_length >> div_blocks[b]; + + if (ctx->cur_frame_length != ctx->sconf.frame_length) { + unsigned int remaining = ctx->cur_frame_length; + + for (b = 0; b < ctx->num_blocks; b++) { + if (remaining < div_blocks[b]) { + div_blocks[b] = remaining; + ctx->num_blocks = b + 1; + break; + } + + remaining -= div_blocks[b]; + } + } +} + + +/** Reads the block data for a constant block + */ +static void read_const_block(ALSDecContext *ctx, int32_t *raw_samples, + unsigned int block_length, unsigned int *js_blocks) +{ + ALSSpecificConfig *sconf = &ctx->sconf; + AVCodecContext *avctx = ctx->avctx; + GetBitContext *gb = &ctx->gb; + int32_t const_val = 0; + unsigned int const_block, k; + + const_block = get_bits1(gb); // 1 = constant value, 0 = zero block (silence) + *js_blocks = get_bits1(gb); + + // skip 5 reserved bits + skip_bits(gb, 5); + + if (const_block) { + unsigned int const_val_bits = sconf->floating ? 24 : avctx->bits_per_raw_sample; + const_val = get_sbits_long(gb, const_val_bits); + } + + // write raw samples into buffer + for (k = 0; k < block_length; k++) + raw_samples[k] = const_val; +} + + +/** Reads the block data for a non-constant block + */ +static int read_var_block(ALSDecContext *ctx, unsigned int ra_block, + int32_t *raw_samples, unsigned int block_length, + unsigned int *js_blocks, int32_t *raw_other, + unsigned int *shift_lsbs) +{ + ALSSpecificConfig *sconf = &ctx->sconf; + AVCodecContext *avctx = ctx->avctx; + GetBitContext *gb = &ctx->gb; + unsigned int k; + unsigned int s[8]; + unsigned int sub_blocks, log2_sub_blocks, sb_length; + unsigned int opt_order = 1; + int32_t *quant_cof = ctx->quant_cof; + int32_t *lpc_cof = ctx->lpc_cof; + unsigned int start = 0; + int smp = 0; + int sb, store_prev_samples; + int64_t y; + + *js_blocks = get_bits1(gb); + + // determine the number of subblocks for entropy decoding + if (!sconf->bgmc && !sconf->sb_part) { + log2_sub_blocks = 0; + } else { + if (sconf->bgmc && sconf->sb_part) + log2_sub_blocks = get_bits(gb, 2); + else + log2_sub_blocks = 2 * get_bits1(gb); + } + + sub_blocks = 1 << log2_sub_blocks; + + // do not continue in case of a damaged stream since + // block_length must be evenly divisible by sub_blocks + if (block_length & (sub_blocks - 1)) { + av_log(avctx, AV_LOG_WARNING, + "Block length is not evenly divisible by the number of subblocks.\n"); + return -1; + } + + sb_length = block_length >> log2_sub_blocks; + + + if (sconf->bgmc) { + // TODO: BGMC mode + } else { + s[0] = get_bits(gb, 4 + (sconf->resolution > 1)); + for (k = 1; k < sub_blocks; k++) + s[k] = s[k - 1] + decode_rice(gb, 0); + } + + if (get_bits1(gb)) + *shift_lsbs = get_bits(gb, 4) + 1; + + store_prev_samples = (*js_blocks && raw_other) || *shift_lsbs; + + + if (!sconf->rlslms) { + if (sconf->adapt_order) { + int opt_order_length = av_ceil_log2(av_clip((block_length >> 3) - 1, + 2, sconf->max_order + 1)); + opt_order = get_bits(gb, opt_order_length); + } else { + opt_order = sconf->max_order; + } + + if (opt_order) { + int add_base; + + if (sconf->coef_table == 3) { + add_base = 0x7F; + + // read coefficient 0 + quant_cof[0] = 32 * parcor_scaled_values[get_bits(gb, 7)]; + + // read coefficient 1 + if (opt_order > 1) + quant_cof[1] = -32 * parcor_scaled_values[get_bits(gb, 7)]; + + // read coefficients 2 to opt_order + for (k = 2; k < opt_order; k++) + quant_cof[k] = get_bits(gb, 7); + } else { + int k_max; + add_base = 1; + + // read coefficient 0 to 19 + k_max = FFMIN(opt_order, 20); + for (k = 0; k < k_max; k++) { + int rice_param = parcor_rice_table[sconf->coef_table][k][1]; + int offset = parcor_rice_table[sconf->coef_table][k][0]; + quant_cof[k] = decode_rice(gb, rice_param) + offset; + } + + // read coefficients 20 to 126 + k_max = FFMIN(opt_order, 127); + for (; k < k_max; k++) + quant_cof[k] = decode_rice(gb, 2) + (k & 1); + + // read coefficients 127 to opt_order + for (; k < opt_order; k++) + quant_cof[k] = decode_rice(gb, 1); + + quant_cof[0] = 32 * parcor_scaled_values[quant_cof[0] + 64]; + + if (opt_order > 1) + quant_cof[1] = -32 * parcor_scaled_values[quant_cof[1] + 64]; + } + + for (k = 2; k < opt_order; k++) + quant_cof[k] = (quant_cof[k] << 14) + (add_base << 13); + } + } + + // TODO: LTP mode + + // read first value and residuals in case of a random access block + if (ra_block) { + if (opt_order) + raw_samples[0] = decode_rice(gb, avctx->bits_per_raw_sample - 4); + if (opt_order > 1) + raw_samples[1] = decode_rice(gb, s[0] + 3); + if (opt_order > 2) + raw_samples[2] = decode_rice(gb, s[0] + 1); + + start = FFMIN(opt_order, 3); + } + + // read all residuals + if (sconf->bgmc) { + // TODO: BGMC mode + } else { + int32_t *current_res = raw_samples + start; + + for (sb = 0; sb < sub_blocks; sb++, start = 0) + for (; start < sb_length; start++) + *current_res++ = decode_rice(gb, s[sb]); + } + + // reconstruct all samples from residuals + if (ra_block) { + for (smp = 0; smp < opt_order; smp++) { + y = 1 << 19; + + for (sb = 0; sb < smp; sb++) + y += MUL64(lpc_cof[sb],raw_samples[smp - (sb + 1)]); + + raw_samples[smp] -= y >> 20; + parcor_to_lpc(smp, quant_cof, lpc_cof); + } + } else { + for (k = 0; k < opt_order; k++) + parcor_to_lpc(k, quant_cof, lpc_cof); + + // store previous samples in case that they have to be altered + if (store_prev_samples) + memcpy(ctx->prev_raw_samples, raw_samples - sconf->max_order, + sizeof(*ctx->prev_raw_samples) * sconf->max_order); + + // reconstruct difference signal for prediction (joint-stereo) + if (*js_blocks && raw_other) { + int32_t *left, *right; + + if (raw_other > raw_samples) { // D = R - L + left = raw_samples; + right = raw_other; + } else { // D = R - L + left = raw_other; + right = raw_samples; + } + + for (sb = -1; sb >= -sconf->max_order; sb--) + raw_samples[sb] = right[sb] - left[sb]; + } + + // reconstruct shifted signal + if (*shift_lsbs) + for (sb = -1; sb >= -sconf->max_order; sb--) + raw_samples[sb] >>= *shift_lsbs; + } + + // reconstruct raw samples + for (; smp < block_length; smp++) { + y = 1 << 19; + + for (sb = 0; sb < opt_order; sb++) + y += MUL64(lpc_cof[sb],raw_samples[smp - (sb + 1)]); + + raw_samples[smp] -= y >> 20; + } + + // restore previous samples in case that they have been altered + if (store_prev_samples) + memcpy(raw_samples - sconf->max_order, ctx->prev_raw_samples, + sizeof(*raw_samples) * sconf->max_order); + + return 0; +} + + +/** Reads the block data. + */ +static int read_block_data(ALSDecContext *ctx, unsigned int ra_block, + int32_t *raw_samples, unsigned int block_length, + unsigned int *js_blocks, int32_t *raw_other) +{ + ALSSpecificConfig *sconf = &ctx->sconf; + GetBitContext *gb = &ctx->gb; + unsigned int shift_lsbs = 0; + unsigned int k; + + // read block type flag and read the samples accordingly + if (get_bits1(gb)) { + if (read_var_block(ctx, ra_block, raw_samples, block_length, js_blocks, + raw_other, &shift_lsbs)) + return -1; + } else { + read_const_block(ctx, raw_samples, block_length, js_blocks); + } + + // TODO: read RLSLMS extension data + + if (!sconf->mc_coding || ctx->js_switch) + align_get_bits(gb); + + if (shift_lsbs) + for (k = 0; k < block_length; k++) + raw_samples[k] <<= shift_lsbs; + + return 0; +} + + +/** Computes the number of samples left to decode for the current frame and + * sets these samples to zero. + */ +static void zero_remaining(unsigned int b, unsigned int b_max, + const unsigned int *div_blocks, int32_t *buf) +{ + unsigned int count = 0; + + while (b < b_max) + count += div_blocks[b]; + + memset(buf, 0, sizeof(*buf) * count); +} + + +/** Decodes blocks independently. + */ +static int decode_blocks_ind(ALSDecContext *ctx, unsigned int ra_frame, + unsigned int c, const unsigned int *div_blocks, + unsigned int *js_blocks) +{ + int32_t *raw_sample; + unsigned int b; + raw_sample = ctx->raw_samples[c]; + + for (b = 0; b < ctx->num_blocks; b++) { + if (read_block_data(ctx, ra_frame, raw_sample, + div_blocks[b], &js_blocks[0], NULL)) { + // damaged block, write zero for the rest of the frame + zero_remaining(b, ctx->num_blocks, div_blocks, raw_sample); + return -1; + } + raw_sample += div_blocks[b]; + ra_frame = 0; + } + + return 0; +} + + +/** Decodes blocks dependently. + */ +static int decode_blocks(ALSDecContext *ctx, unsigned int ra_frame, + unsigned int c, const unsigned int *div_blocks, + unsigned int *js_blocks) +{ + ALSSpecificConfig *sconf = &ctx->sconf; + unsigned int offset = 0; + int32_t *raw_samples_R; + int32_t *raw_samples_L; + unsigned int b; + + // decode all blocks + for (b = 0; b < ctx->num_blocks; b++) { + unsigned int s; + raw_samples_L = ctx->raw_samples[c ] + offset; + raw_samples_R = ctx->raw_samples[c + 1] + offset; + if (read_block_data(ctx, ra_frame, raw_samples_L, div_blocks[b], + &js_blocks[0], raw_samples_R) || + read_block_data(ctx, ra_frame, raw_samples_R, div_blocks[b], + &js_blocks[1], raw_samples_L)) { + // damaged block, write zero for the rest of the frame + zero_remaining(b, ctx->num_blocks, div_blocks, raw_samples_L); + zero_remaining(b, ctx->num_blocks, div_blocks, raw_samples_R); + return -1; + } + + // reconstruct joint-stereo blocks + if (js_blocks[0]) { + if (js_blocks[1]) + av_log(ctx->avctx, AV_LOG_WARNING, "Invalid channel pair!\n"); + + for (s = 0; s < div_blocks[b]; s++) + raw_samples_L[s] = raw_samples_R[s] - raw_samples_L[s]; + } else if (js_blocks[1]) { + for (s = 0; s < div_blocks[b]; s++) + raw_samples_R[s] = raw_samples_R[s] + raw_samples_L[s]; + } + + offset += div_blocks[b]; + ra_frame = 0; + } + + // store carryover raw samples, + // the others channel raw samples are stored by the calling function. + memmove(ctx->raw_samples[c] - sconf->max_order, + ctx->raw_samples[c] - sconf->max_order + sconf->frame_length, + sizeof(*ctx->raw_samples[c]) * sconf->max_order); + + return 0; +} + + +/** Reads the frame data. + */ +static int read_frame_data(ALSDecContext *ctx, unsigned int ra_frame) +{ + ALSSpecificConfig *sconf = &ctx->sconf; + AVCodecContext *avctx = ctx->avctx; + GetBitContext *gb = &ctx->gb; + unsigned int div_blocks[32]; ///< block sizes. + unsigned int c; + unsigned int js_blocks[2]; + + uint32_t bs_info = 0; + + // skip the size of the ra unit if present in the frame + if (sconf->ra_flag == RA_FLAG_FRAMES && ra_frame) + skip_bits_long(gb, 32); + + if (sconf->mc_coding && sconf->joint_stereo) { + ctx->js_switch = get_bits1(gb); + align_get_bits(gb); + } + + if (!sconf->mc_coding || ctx->js_switch) { + int independent_bs = !sconf->joint_stereo; + + for (c = 0; c < avctx->channels; c++) { + js_blocks[0] = 0; + js_blocks[1] = 0; + + get_block_sizes(ctx, div_blocks, &bs_info); + + // if joint_stereo and block_switching is set, independent decoding + // is signaled via the first bit of bs_info + if (sconf->joint_stereo && sconf->block_switching) + if (bs_info >> 31) + independent_bs = 2; + + // if this is the last channel, it has to be decoded independently + if (c == avctx->channels - 1) + independent_bs = 1; + + if (independent_bs) { + if (decode_blocks_ind(ctx, ra_frame, c, div_blocks, js_blocks)) + return -1; + + independent_bs--; + } else { + if (decode_blocks(ctx, ra_frame, c, div_blocks, js_blocks)) + return -1; + + c++; + } + + // store carryover raw samples + memmove(ctx->raw_samples[c] - sconf->max_order, + ctx->raw_samples[c] - sconf->max_order + sconf->frame_length, + sizeof(*ctx->raw_samples[c]) * sconf->max_order); + } + } else { // multi-channel coding + get_block_sizes(ctx, div_blocks, &bs_info); + + // TODO: multi channel coding might use a temporary buffer instead as + // the actual channel is not known when read_block-data is called + if (decode_blocks_ind(ctx, ra_frame, 0, div_blocks, js_blocks)) + return -1; + // TODO: read_channel_data + } + + // TODO: read_diff_float_data + + return 0; +} + + +/** Decodes an ALS frame. + */ +static int decode_frame(AVCodecContext *avctx, + void *data, int *data_size, + AVPacket *avpkt) +{ + ALSDecContext *ctx = avctx->priv_data; + ALSSpecificConfig *sconf = &ctx->sconf; + const uint8_t *buffer = avpkt->data; + int buffer_size = avpkt->size; + int invalid_frame, size; + unsigned int c, sample, ra_frame, bytes_read, shift; + + init_get_bits(&ctx->gb, buffer, buffer_size * 8); + + // In the case that the distance between random access frames is set to zero + // (sconf->ra_distance == 0) no frame is treated as a random access frame. + // For the first frame, if prediction is used, all samples used from the + // previous frame are assumed to be zero. + ra_frame = sconf->ra_distance && !(ctx->frame_id % sconf->ra_distance); + + // the last frame to decode might have a different length + if (sconf->samples != 0xFFFFFFFF) + ctx->cur_frame_length = FFMIN(sconf->samples - ctx->frame_id * (uint64_t) sconf->frame_length, + sconf->frame_length); + else + ctx->cur_frame_length = sconf->frame_length; + + // decode the frame data + if ((invalid_frame = read_frame_data(ctx, ra_frame) < 0)) + av_log(ctx->avctx, AV_LOG_WARNING, + "Reading frame data failed. Skipping RA unit.\n"); + + ctx->frame_id++; + + // check for size of decoded data + size = ctx->cur_frame_length * avctx->channels * + (av_get_bits_per_sample_format(avctx->sample_fmt) >> 3); + + if (size > *data_size) { + av_log(avctx, AV_LOG_ERROR, "Decoded data exceeds buffer size.\n"); + return -1; + } + + *data_size = size; + + // transform decoded frame into output format + #define INTERLEAVE_OUTPUT(bps) \ + { \ + int##bps##_t *dest = (int##bps##_t*) data; \ + shift = bps - ctx->avctx->bits_per_raw_sample; \ + for (sample = 0; sample < ctx->cur_frame_length; sample++) \ + for (c = 0; c < avctx->channels; c++) \ + *dest++ = ctx->raw_samples[c][sample] << shift; \ + } + + if (ctx->avctx->bits_per_raw_sample <= 16) { + INTERLEAVE_OUTPUT(16) + } else { + INTERLEAVE_OUTPUT(32) + } + + bytes_read = invalid_frame ? buffer_size : + (get_bits_count(&ctx->gb) + 7) >> 3; + + return bytes_read; +} + + +/** Uninitializes the ALS decoder. + */ +static av_cold int decode_end(AVCodecContext *avctx) +{ + ALSDecContext *ctx = avctx->priv_data; + + av_freep(&ctx->sconf.chan_pos); + + av_freep(&ctx->quant_cof); + av_freep(&ctx->lpc_cof); + av_freep(&ctx->prev_raw_samples); + av_freep(&ctx->raw_samples); + av_freep(&ctx->raw_buffer); + + return 0; +} + + +/** Initializes the ALS decoder. + */ +static av_cold int decode_init(AVCodecContext *avctx) +{ + unsigned int c; + unsigned int channel_size; + ALSDecContext *ctx = avctx->priv_data; + ALSSpecificConfig *sconf = &ctx->sconf; + ctx->avctx = avctx; + + if (!avctx->extradata) { + av_log(avctx, AV_LOG_ERROR, "Missing required ALS extradata.\n"); + return -1; + } + + if (read_specific_config(ctx)) { + av_log(avctx, AV_LOG_ERROR, "Reading ALSSpecificConfig failed.\n"); + decode_end(avctx); + return -1; + } + + if (check_specific_config(ctx)) { + decode_end(avctx); + return -1; + } + + if (sconf->floating) { + avctx->sample_fmt = SAMPLE_FMT_FLT; + avctx->bits_per_raw_sample = 32; + } else { + avctx->sample_fmt = sconf->resolution > 1 + ? SAMPLE_FMT_S32 : SAMPLE_FMT_S16; + avctx->bits_per_raw_sample = (sconf->resolution + 1) * 8; + } + + avctx->frame_size = sconf->frame_length; + channel_size = sconf->frame_length + sconf->max_order; + + ctx->prev_raw_samples = av_malloc (sizeof(*ctx->prev_raw_samples) * sconf->max_order); + ctx->raw_buffer = av_mallocz(sizeof(*ctx-> raw_buffer) * avctx->channels * channel_size); + ctx->raw_samples = av_malloc (sizeof(*ctx-> raw_samples) * avctx->channels); + + // allocate previous raw sample buffer + if (!ctx->prev_raw_samples || !ctx->raw_buffer|| !ctx->raw_samples) { + av_log(avctx, AV_LOG_ERROR, "Allocating buffer memory failed.\n"); + decode_end(avctx); + return AVERROR(ENOMEM); + } + + // assign raw samples buffers + ctx->raw_samples[0] = ctx->raw_buffer + sconf->max_order; + for (c = 1; c < avctx->channels; c++) + ctx->raw_samples[c] = ctx->raw_samples[c - 1] + channel_size; + + return 0; +} + + +/** Flushes (resets) the frame ID after seeking. + */ +static av_cold void flush(AVCodecContext *avctx) +{ + ALSDecContext *ctx = avctx->priv_data; + + ctx->frame_id = 0; +} + + +AVCodec als_decoder = { + "als", + CODEC_TYPE_AUDIO, + CODEC_ID_MP4ALS, + sizeof(ALSDecContext), + decode_init, + NULL, + decode_end, + decode_frame, + .flush = flush, + .capabilities = CODEC_CAP_SUBFRAMES, + .long_name = NULL_IF_CONFIG_SMALL("MPEG-4 Audio Lossless Coding (ALS)"), +}; +