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authorThilo Borgmann <thilo.borgmann@googlemail.com>2009-11-11 22:36:17 +0000
committerCarl Eugen Hoyos <cehoyos@rainbow.studorg.tuwien.ac.at>2009-11-11 22:36:17 +0000
commit99971952b0998fd110bd670fabb0a2c2e39e6879 (patch)
treea4c9bce77d9367e42b8528b35bf70f34855c32bb
parent24026a2d499eec4d641fb61727e346ac06957e37 (diff)
Add MPEG-4 Audio Lossless Coding (ALS) decoder.
Patch by Thilo Borgmann, thilo D borgmann A googlemail Originally committed as revision 20517 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r--Changelog1
-rw-r--r--doc/general.texi1
-rw-r--r--libavcodec/Makefile1
-rw-r--r--libavcodec/allcodecs.c1
-rw-r--r--libavcodec/als_data.h95
-rw-r--r--libavcodec/alsdec.c997
6 files changed, 1096 insertions, 0 deletions
diff --git a/Changelog b/Changelog
index 0d465aa1ed..7821e72c87 100644
--- a/Changelog
+++ b/Changelog
@@ -41,6 +41,7 @@ version <next>:
- 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 <thilo.borgmann _at_ googlemail.com>
+ *
+ * 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 <thilo.borgmann _at_ googlemail.com>
+ */
+
+
+#include <stdint.h>
+
+/** 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 <thilo.borgmann _at_ googlemail.com>
+ *
+ * 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 <thilo.borgmann _at_ googlemail.com>
+ */
+
+
+//#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)"),
+};
+