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663 lines
21 KiB
C
663 lines
21 KiB
C
/*
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* ALAC (Apple Lossless Audio Codec) decoder
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* Copyright (c) 2005 David Hammerton
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file alac.c
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* ALAC (Apple Lossless Audio Codec) decoder
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* @author 2005 David Hammerton
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*
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* For more information on the ALAC format, visit:
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* http://crazney.net/programs/itunes/alac.html
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*
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* Note: This decoder expects a 36- (0x24-)byte QuickTime atom to be
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* passed through the extradata[_size] fields. This atom is tacked onto
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* the end of an 'alac' stsd atom and has the following format:
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* bytes 0-3 atom size (0x24), big-endian
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* bytes 4-7 atom type ('alac', not the 'alac' tag from start of stsd)
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* bytes 8-35 data bytes needed by decoder
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*
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* Extradata:
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* 32bit size
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* 32bit tag (=alac)
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* 32bit zero?
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* 32bit max sample per frame
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* 8bit ?? (zero?)
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* 8bit sample size
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* 8bit history mult
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* 8bit initial history
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* 8bit kmodifier
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* 8bit channels?
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* 16bit ??
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* 32bit max coded frame size
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* 32bit bitrate?
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* 32bit samplerate
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*/
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#include "avcodec.h"
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#include "bitstream.h"
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#include "bytestream.h"
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#define ALAC_EXTRADATA_SIZE 36
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#define MAX_CHANNELS 2
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typedef struct {
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AVCodecContext *avctx;
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GetBitContext gb;
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/* init to 0; first frame decode should initialize from extradata and
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* set this to 1 */
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int context_initialized;
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int samplesize;
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int numchannels;
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int bytespersample;
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/* buffers */
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int32_t *predicterror_buffer[MAX_CHANNELS];
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int32_t *outputsamples_buffer[MAX_CHANNELS];
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/* stuff from setinfo */
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uint32_t setinfo_max_samples_per_frame; /* 0x1000 = 4096 */ /* max samples per frame? */
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uint8_t setinfo_7a; /* 0x00 */
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uint8_t setinfo_sample_size; /* 0x10 */
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uint8_t setinfo_rice_historymult; /* 0x28 */
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uint8_t setinfo_rice_initialhistory; /* 0x0a */
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uint8_t setinfo_rice_kmodifier; /* 0x0e */
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uint8_t setinfo_7f; /* 0x02 */
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uint16_t setinfo_80; /* 0x00ff */
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uint32_t setinfo_82; /* 0x000020e7 */ /* max sample size?? */
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uint32_t setinfo_86; /* 0x00069fe4 */ /* bit rate (average)?? */
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uint32_t setinfo_8a_rate; /* 0x0000ac44 */
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/* end setinfo stuff */
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} ALACContext;
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static void allocate_buffers(ALACContext *alac)
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{
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int chan;
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for (chan = 0; chan < MAX_CHANNELS; chan++) {
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alac->predicterror_buffer[chan] =
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av_malloc(alac->setinfo_max_samples_per_frame * 4);
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alac->outputsamples_buffer[chan] =
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av_malloc(alac->setinfo_max_samples_per_frame * 4);
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}
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}
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static int alac_set_info(ALACContext *alac)
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{
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unsigned char *ptr = alac->avctx->extradata;
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ptr += 4; /* size */
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ptr += 4; /* alac */
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ptr += 4; /* 0 ? */
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if(AV_RB32(ptr) >= UINT_MAX/4){
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av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n");
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return -1;
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}
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/* buffer size / 2 ? */
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alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr);
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alac->setinfo_7a = *ptr++;
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alac->setinfo_sample_size = *ptr++;
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alac->setinfo_rice_historymult = *ptr++;
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alac->setinfo_rice_initialhistory = *ptr++;
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alac->setinfo_rice_kmodifier = *ptr++;
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/* channels? */
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alac->setinfo_7f = *ptr++;
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alac->setinfo_80 = bytestream_get_be16(&ptr);
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/* max coded frame size */
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alac->setinfo_82 = bytestream_get_be32(&ptr);
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/* bitrate ? */
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alac->setinfo_86 = bytestream_get_be32(&ptr);
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/* samplerate */
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alac->setinfo_8a_rate = bytestream_get_be32(&ptr);
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allocate_buffers(alac);
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return 0;
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}
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static inline int count_leading_zeros(int32_t input)
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{
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return 31-av_log2(input);
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}
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static void bastardized_rice_decompress(ALACContext *alac,
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int32_t *output_buffer,
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int output_size,
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int readsamplesize, /* arg_10 */
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int rice_initialhistory, /* arg424->b */
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int rice_kmodifier, /* arg424->d */
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int rice_historymult, /* arg424->c */
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int rice_kmodifier_mask /* arg424->e */
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)
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{
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int output_count;
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unsigned int history = rice_initialhistory;
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int sign_modifier = 0;
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for (output_count = 0; output_count < output_size; output_count++) {
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int32_t x = 0;
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int32_t x_modified;
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int32_t final_val;
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/* read x - number of 1s before 0 represent the rice */
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while (x <= 8 && get_bits1(&alac->gb)) {
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x++;
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}
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if (x > 8) { /* RICE THRESHOLD */
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/* use alternative encoding */
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int32_t value;
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value = get_bits(&alac->gb, readsamplesize);
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/* mask value to readsamplesize size */
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if (readsamplesize != 32)
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value &= (0xffffffff >> (32 - readsamplesize));
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x = value;
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} else {
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/* standard rice encoding */
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int extrabits;
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int k; /* size of extra bits */
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/* read k, that is bits as is */
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k = 31 - rice_kmodifier - count_leading_zeros((history >> 9) + 3);
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if (k < 0)
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k += rice_kmodifier;
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else
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k = rice_kmodifier;
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if (k != 1) {
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extrabits = show_bits(&alac->gb, k);
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/* multiply x by 2^k - 1, as part of their strange algorithm */
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x = (x << k) - x;
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if (extrabits > 1) {
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x += extrabits - 1;
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skip_bits(&alac->gb, k);
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} else
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skip_bits(&alac->gb, k - 1);
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}
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}
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x_modified = sign_modifier + x;
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final_val = (x_modified + 1) / 2;
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if (x_modified & 1) final_val *= -1;
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output_buffer[output_count] = final_val;
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sign_modifier = 0;
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/* now update the history */
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history += x_modified * rice_historymult
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- ((history * rice_historymult) >> 9);
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if (x_modified > 0xffff)
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history = 0xffff;
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/* special case: there may be compressed blocks of 0 */
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if ((history < 128) && (output_count+1 < output_size)) {
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int block_size;
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sign_modifier = 1;
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x = 0;
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while (x <= 8 && get_bits1(&alac->gb)) {
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x++;
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}
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if (x > 8) {
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block_size = get_bits(&alac->gb, 16);
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block_size &= 0xffff;
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} else {
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int k;
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int extrabits;
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k = count_leading_zeros(history) + ((history + 16) >> 6 /* / 64 */) - 24;
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extrabits = show_bits(&alac->gb, k);
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block_size = (((1 << k) - 1) & rice_kmodifier_mask) * x
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+ extrabits - 1;
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if (extrabits < 2) {
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x = 1 - extrabits;
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block_size += x;
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skip_bits(&alac->gb, k - 1);
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} else {
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skip_bits(&alac->gb, k);
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}
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}
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if (block_size > 0) {
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memset(&output_buffer[output_count+1], 0, block_size * 4);
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output_count += block_size;
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}
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if (block_size > 0xffff)
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sign_modifier = 0;
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history = 0;
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}
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}
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}
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#define SIGN_EXTENDED32(val, bits) ((val << (32 - bits)) >> (32 - bits))
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#define SIGN_ONLY(v) \
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((v < 0) ? (-1) : \
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((v > 0) ? (1) : \
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(0)))
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static void predictor_decompress_fir_adapt(int32_t *error_buffer,
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int32_t *buffer_out,
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int output_size,
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int readsamplesize,
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int16_t *predictor_coef_table,
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int predictor_coef_num,
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int predictor_quantitization)
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{
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int i;
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/* first sample always copies */
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*buffer_out = *error_buffer;
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if (!predictor_coef_num) {
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if (output_size <= 1)
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return;
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memcpy(buffer_out+1, error_buffer+1, (output_size-1) * 4);
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return;
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}
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if (predictor_coef_num == 0x1f) { /* 11111 - max value of predictor_coef_num */
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/* second-best case scenario for fir decompression,
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* error describes a small difference from the previous sample only
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*/
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if (output_size <= 1)
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return;
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for (i = 0; i < output_size - 1; i++) {
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int32_t prev_value;
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int32_t error_value;
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prev_value = buffer_out[i];
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error_value = error_buffer[i+1];
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buffer_out[i+1] =
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SIGN_EXTENDED32((prev_value + error_value), readsamplesize);
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}
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return;
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}
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/* read warm-up samples */
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if (predictor_coef_num > 0)
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for (i = 0; i < predictor_coef_num; i++) {
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int32_t val;
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val = buffer_out[i] + error_buffer[i+1];
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val = SIGN_EXTENDED32(val, readsamplesize);
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buffer_out[i+1] = val;
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}
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#if 0
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/* 4 and 8 are very common cases (the only ones i've seen). these
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* should be unrolled and optimised
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*/
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if (predictor_coef_num == 4) {
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/* FIXME: optimised general case */
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return;
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}
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if (predictor_coef_table == 8) {
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/* FIXME: optimised general case */
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return;
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}
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#endif
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/* general case */
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if (predictor_coef_num > 0) {
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for (i = predictor_coef_num + 1; i < output_size; i++) {
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int j;
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int sum = 0;
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int outval;
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int error_val = error_buffer[i];
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for (j = 0; j < predictor_coef_num; j++) {
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sum += (buffer_out[predictor_coef_num-j] - buffer_out[0]) *
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predictor_coef_table[j];
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}
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outval = (1 << (predictor_quantitization-1)) + sum;
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outval = outval >> predictor_quantitization;
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outval = outval + buffer_out[0] + error_val;
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outval = SIGN_EXTENDED32(outval, readsamplesize);
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buffer_out[predictor_coef_num+1] = outval;
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if (error_val > 0) {
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int predictor_num = predictor_coef_num - 1;
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while (predictor_num >= 0 && error_val > 0) {
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int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num];
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int sign = SIGN_ONLY(val);
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predictor_coef_table[predictor_num] -= sign;
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val *= sign; /* absolute value */
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error_val -= ((val >> predictor_quantitization) *
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(predictor_coef_num - predictor_num));
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predictor_num--;
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}
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} else if (error_val < 0) {
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int predictor_num = predictor_coef_num - 1;
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while (predictor_num >= 0 && error_val < 0) {
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int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num];
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int sign = - SIGN_ONLY(val);
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predictor_coef_table[predictor_num] -= sign;
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val *= sign; /* neg value */
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error_val -= ((val >> predictor_quantitization) *
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(predictor_coef_num - predictor_num));
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predictor_num--;
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}
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}
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buffer_out++;
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}
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}
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}
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static void reconstruct_stereo_16(int32_t *buffer[MAX_CHANNELS],
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int16_t *buffer_out,
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int numchannels, int numsamples,
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uint8_t interlacing_shift,
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uint8_t interlacing_leftweight)
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{
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int i;
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if (numsamples <= 0)
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return;
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/* weighted interlacing */
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if (interlacing_leftweight) {
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for (i = 0; i < numsamples; i++) {
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int32_t a, b;
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a = buffer[0][i];
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b = buffer[1][i];
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a -= (b * interlacing_leftweight) >> interlacing_shift;
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b += a;
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buffer_out[i*numchannels] = b;
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buffer_out[i*numchannels + 1] = a;
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}
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return;
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}
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/* otherwise basic interlacing took place */
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for (i = 0; i < numsamples; i++) {
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int16_t left, right;
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left = buffer[0][i];
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right = buffer[1][i];
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buffer_out[i*numchannels] = left;
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buffer_out[i*numchannels + 1] = right;
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}
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}
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static int alac_decode_frame(AVCodecContext *avctx,
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void *outbuffer, int *outputsize,
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uint8_t *inbuffer, int input_buffer_size)
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{
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ALACContext *alac = avctx->priv_data;
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int channels;
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int32_t outputsamples;
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int hassize;
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int readsamplesize;
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int wasted_bytes;
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int isnotcompressed;
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uint8_t interlacing_shift;
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uint8_t interlacing_leftweight;
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/* short-circuit null buffers */
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if (!inbuffer || !input_buffer_size)
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return input_buffer_size;
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/* initialize from the extradata */
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if (!alac->context_initialized) {
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if (alac->avctx->extradata_size != ALAC_EXTRADATA_SIZE) {
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av_log(avctx, AV_LOG_ERROR, "alac: expected %d extradata bytes\n",
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ALAC_EXTRADATA_SIZE);
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return input_buffer_size;
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}
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if (alac_set_info(alac)) {
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av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n");
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return input_buffer_size;
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}
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alac->context_initialized = 1;
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}
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init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8);
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channels = get_bits(&alac->gb, 3) + 1;
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if (channels > MAX_CHANNELS) {
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av_log(avctx, AV_LOG_ERROR, "channels > %d not supported\n",
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MAX_CHANNELS);
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return input_buffer_size;
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}
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/* 2^result = something to do with output waiting.
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* perhaps matters if we read > 1 frame in a pass?
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*/
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skip_bits(&alac->gb, 4);
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skip_bits(&alac->gb, 12); /* unknown, skip 12 bits */
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/* the output sample size is stored soon */
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hassize = get_bits1(&alac->gb);
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wasted_bytes = get_bits(&alac->gb, 2); /* unknown ? */
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/* whether the frame is compressed */
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isnotcompressed = get_bits1(&alac->gb);
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if (hassize) {
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/* now read the number of samples as a 32bit integer */
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outputsamples = get_bits(&alac->gb, 32);
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} else
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outputsamples = alac->setinfo_max_samples_per_frame;
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*outputsize = outputsamples * alac->bytespersample;
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readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8) + channels - 1;
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if (!isnotcompressed) {
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/* so it is compressed */
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int16_t predictor_coef_table[channels][32];
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int predictor_coef_num[channels];
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int prediction_type[channels];
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int prediction_quantitization[channels];
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int ricemodifier[channels];
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int i, chan;
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interlacing_shift = get_bits(&alac->gb, 8);
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interlacing_leftweight = get_bits(&alac->gb, 8);
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for (chan = 0; chan < channels; chan++) {
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prediction_type[chan] = get_bits(&alac->gb, 4);
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prediction_quantitization[chan] = get_bits(&alac->gb, 4);
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ricemodifier[chan] = get_bits(&alac->gb, 3);
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predictor_coef_num[chan] = get_bits(&alac->gb, 5);
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/* read the predictor table */
|
|
for (i = 0; i < predictor_coef_num[chan]; i++)
|
|
predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb, 16);
|
|
}
|
|
|
|
if (wasted_bytes)
|
|
av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of wasted_bytes\n");
|
|
|
|
for (chan = 0; chan < channels; chan++) {
|
|
bastardized_rice_decompress(alac,
|
|
alac->predicterror_buffer[chan],
|
|
outputsamples,
|
|
readsamplesize,
|
|
alac->setinfo_rice_initialhistory,
|
|
alac->setinfo_rice_kmodifier,
|
|
ricemodifier[chan] * alac->setinfo_rice_historymult / 4,
|
|
(1 << alac->setinfo_rice_kmodifier) - 1);
|
|
|
|
if (prediction_type[chan] == 0) {
|
|
/* adaptive fir */
|
|
predictor_decompress_fir_adapt(alac->predicterror_buffer[chan],
|
|
alac->outputsamples_buffer[chan],
|
|
outputsamples,
|
|
readsamplesize,
|
|
predictor_coef_table[chan],
|
|
predictor_coef_num[chan],
|
|
prediction_quantitization[chan]);
|
|
} else {
|
|
av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[chan]);
|
|
/* I think the only other prediction type (or perhaps this is
|
|
* just a boolean?) runs adaptive fir twice.. like:
|
|
* predictor_decompress_fir_adapt(predictor_error, tempout, ...)
|
|
* predictor_decompress_fir_adapt(predictor_error, outputsamples ...)
|
|
* little strange..
|
|
*/
|
|
}
|
|
}
|
|
} else {
|
|
/* not compressed, easy case */
|
|
if (alac->setinfo_sample_size <= 16) {
|
|
int i, chan;
|
|
for (chan = 0; chan < channels; chan++)
|
|
for (i = 0; i < outputsamples; i++) {
|
|
int32_t audiobits;
|
|
|
|
audiobits = get_bits(&alac->gb, alac->setinfo_sample_size);
|
|
audiobits = SIGN_EXTENDED32(audiobits, readsamplesize);
|
|
|
|
alac->outputsamples_buffer[chan][i] = audiobits;
|
|
}
|
|
} else {
|
|
int i, chan;
|
|
for (chan = 0; chan < channels; chan++)
|
|
for (i = 0; i < outputsamples; i++) {
|
|
int32_t audiobits;
|
|
|
|
audiobits = get_bits(&alac->gb, 16);
|
|
/* special case of sign extension..
|
|
* as we'll be ORing the low 16bits into this */
|
|
audiobits = audiobits << 16;
|
|
audiobits = audiobits >> (32 - alac->setinfo_sample_size);
|
|
audiobits |= get_bits(&alac->gb, alac->setinfo_sample_size - 16);
|
|
|
|
alac->outputsamples_buffer[chan][i] = audiobits;
|
|
}
|
|
}
|
|
/* wasted_bytes = 0; */
|
|
interlacing_shift = 0;
|
|
interlacing_leftweight = 0;
|
|
}
|
|
|
|
switch(alac->setinfo_sample_size) {
|
|
case 16:
|
|
if (channels == 2) {
|
|
reconstruct_stereo_16(alac->outputsamples_buffer,
|
|
(int16_t*)outbuffer,
|
|
alac->numchannels,
|
|
outputsamples,
|
|
interlacing_shift,
|
|
interlacing_leftweight);
|
|
} else {
|
|
int i;
|
|
for (i = 0; i < outputsamples; i++) {
|
|
int16_t sample = alac->outputsamples_buffer[0][i];
|
|
((int16_t*)outbuffer)[i * alac->numchannels] = sample;
|
|
}
|
|
}
|
|
break;
|
|
case 20:
|
|
case 24:
|
|
case 32:
|
|
av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented sample size %i\n", alac->setinfo_sample_size);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return input_buffer_size;
|
|
}
|
|
|
|
static int alac_decode_init(AVCodecContext * avctx)
|
|
{
|
|
ALACContext *alac = avctx->priv_data;
|
|
alac->avctx = avctx;
|
|
alac->context_initialized = 0;
|
|
|
|
alac->samplesize = alac->avctx->bits_per_sample;
|
|
alac->numchannels = alac->avctx->channels;
|
|
alac->bytespersample = (alac->samplesize / 8) * alac->numchannels;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int alac_decode_close(AVCodecContext *avctx)
|
|
{
|
|
ALACContext *alac = avctx->priv_data;
|
|
|
|
int chan;
|
|
for (chan = 0; chan < MAX_CHANNELS; chan++) {
|
|
av_free(alac->predicterror_buffer[chan]);
|
|
av_free(alac->outputsamples_buffer[chan]);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
AVCodec alac_decoder = {
|
|
"alac",
|
|
CODEC_TYPE_AUDIO,
|
|
CODEC_ID_ALAC,
|
|
sizeof(ALACContext),
|
|
alac_decode_init,
|
|
NULL,
|
|
alac_decode_close,
|
|
alac_decode_frame,
|
|
};
|