042f9d62ca
* qatar/master: configure: Automatically add more flags required on symbian mem.h: switch doxygen parameter order to match function prototype doxygen: replace @sa tag by the more readable but equivalent @see doxygen: use Doxygen markup for authors and web links where appropriate doxygen: do not include license boilerplate in Doxygen documentation ac3enc: Mark AVClasses const ffserver: Replace two loops with one loop. ffmpeg: Fix the check for experimental codecs swscale: extend mmx padding. swscale: clip unscaled colorspace conversion path. doxygen: misc consistency cosmetics doc: remove file name from @file directive in Doxygen usage example doxygen: consistently place brief description doxygen: place empty line between brief description and detailed description avformat_open_input(): Add braces to shut up gcc warning. Conflicts: libavcodec/8svx.c libavcodec/tiff.c libavcodec/tiff.h libavcodec/vaapi_h264.c libavcodec/vorbis.c libavcodec/vorbisdec.c libavcodec/vp6.c libswscale/swscale_unscaled.c libswscale/utils.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
703 lines
23 KiB
C
703 lines
23 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
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* ALAC (Apple Lossless Audio Codec) decoder
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* @author 2005 David Hammerton
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* @see 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 "get_bits.h"
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#include "bytestream.h"
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#include "unary.h"
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#include "mathops.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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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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int32_t *wasted_bits_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_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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/* end setinfo stuff */
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int wasted_bits;
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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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alac->wasted_bits_buffer[chan] = 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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const 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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ptr++; /* ??? */
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alac->setinfo_sample_size = *ptr++;
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if (alac->setinfo_sample_size > 32) {
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av_log(alac->avctx, AV_LOG_ERROR, "setinfo_sample_size too large\n");
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return -1;
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}
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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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ptr++; /* channels? */
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bytestream_get_be16(&ptr); /* ??? */
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bytestream_get_be32(&ptr); /* max coded frame size */
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bytestream_get_be32(&ptr); /* bitrate ? */
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bytestream_get_be32(&ptr); /* samplerate */
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allocate_buffers(alac);
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return 0;
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}
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static inline int decode_scalar(GetBitContext *gb, int k, int limit, int readsamplesize){
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/* read x - number of 1s before 0 represent the rice */
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int x = get_unary_0_9(gb);
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if (x > 8) { /* RICE THRESHOLD */
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/* use alternative encoding */
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x = get_bits(gb, readsamplesize);
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} else {
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if (k >= limit)
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k = limit;
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if (k != 1) {
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int extrabits = show_bits(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(gb, k);
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} else
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skip_bits(gb, k - 1);
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}
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}
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return x;
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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;
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int32_t x_modified;
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int32_t final_val;
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/* standard rice encoding */
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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 = av_log2((history >> 9) + 3);
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x= decode_scalar(&alac->gb, k, rice_kmodifier, readsamplesize);
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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 k;
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unsigned int block_size;
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sign_modifier = 1;
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k = 7 - av_log2(history) + ((history + 16) >> 6 /* / 64 */);
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block_size= decode_scalar(&alac->gb, k, rice_kmodifier, 16);
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if (block_size > 0) {
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if(block_size >= output_size - output_count){
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av_log(alac->avctx, AV_LOG_ERROR, "invalid zero block size of %d %d %d\n", block_size, output_size, output_count);
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block_size= output_size - output_count - 1;
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}
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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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static inline int sign_only(int v)
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{
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return v ? FFSIGN(v) : 0;
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}
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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_extend((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_extend(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 optimized
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*/
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if (predictor_coef_num == 4) {
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/* FIXME: optimized 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: optimized 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_extend(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 void decorrelate_stereo_24(int32_t *buffer[MAX_CHANNELS],
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int32_t *buffer_out,
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int32_t *wasted_bits_buffer[MAX_CHANNELS],
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int wasted_bits,
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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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if (wasted_bits) {
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b = (b << wasted_bits) | wasted_bits_buffer[0][i];
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a = (a << wasted_bits) | wasted_bits_buffer[1][i];
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}
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buffer_out[i * numchannels] = b << 8;
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buffer_out[i * numchannels + 1] = a << 8;
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}
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} else {
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for (i = 0; i < numsamples; i++) {
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int32_t left, right;
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left = buffer[0][i];
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right = buffer[1][i];
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if (wasted_bits) {
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left = (left << wasted_bits) | wasted_bits_buffer[0][i];
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right = (right << wasted_bits) | wasted_bits_buffer[1][i];
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}
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buffer_out[i * numchannels] = left << 8;
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buffer_out[i * numchannels + 1] = right << 8;
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}
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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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AVPacket *avpkt)
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{
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const uint8_t *inbuffer = avpkt->data;
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int input_buffer_size = avpkt->size;
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ALACContext *alac = avctx->priv_data;
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int channels;
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unsigned int outputsamples;
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int hassize;
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unsigned int readsamplesize;
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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 -1;
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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 -1;
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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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alac->wasted_bits = get_bits(&alac->gb, 2) << 3;
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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_long(&alac->gb, 32);
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if(outputsamples > alac->setinfo_max_samples_per_frame){
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av_log(avctx, AV_LOG_ERROR, "outputsamples %d > %d\n", outputsamples, alac->setinfo_max_samples_per_frame);
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return -1;
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}
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} else
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outputsamples = alac->setinfo_max_samples_per_frame;
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switch (alac->setinfo_sample_size) {
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case 16: avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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alac->bytespersample = channels << 1;
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break;
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case 24: avctx->sample_fmt = AV_SAMPLE_FMT_S32;
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alac->bytespersample = channels << 2;
|
|
break;
|
|
default: av_log(avctx, AV_LOG_ERROR, "Sample depth %d is not supported.\n",
|
|
alac->setinfo_sample_size);
|
|
return -1;
|
|
}
|
|
|
|
if(outputsamples > *outputsize / alac->bytespersample){
|
|
av_log(avctx, AV_LOG_ERROR, "sample buffer too small\n");
|
|
return -1;
|
|
}
|
|
|
|
*outputsize = outputsamples * alac->bytespersample;
|
|
readsamplesize = alac->setinfo_sample_size - (alac->wasted_bits) + channels - 1;
|
|
if (readsamplesize > MIN_CACHE_BITS) {
|
|
av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n", readsamplesize);
|
|
return -1;
|
|
}
|
|
|
|
if (!isnotcompressed) {
|
|
/* so it is compressed */
|
|
int16_t predictor_coef_table[MAX_CHANNELS][32];
|
|
int predictor_coef_num[MAX_CHANNELS];
|
|
int prediction_type[MAX_CHANNELS];
|
|
int prediction_quantitization[MAX_CHANNELS];
|
|
int ricemodifier[MAX_CHANNELS];
|
|
int i, chan;
|
|
|
|
interlacing_shift = get_bits(&alac->gb, 8);
|
|
interlacing_leftweight = get_bits(&alac->gb, 8);
|
|
|
|
for (chan = 0; chan < channels; chan++) {
|
|
prediction_type[chan] = get_bits(&alac->gb, 4);
|
|
prediction_quantitization[chan] = get_bits(&alac->gb, 4);
|
|
|
|
ricemodifier[chan] = get_bits(&alac->gb, 3);
|
|
predictor_coef_num[chan] = get_bits(&alac->gb, 5);
|
|
|
|
/* 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 (alac->wasted_bits) {
|
|
int i, ch;
|
|
for (i = 0; i < outputsamples; i++) {
|
|
for (ch = 0; ch < channels; ch++)
|
|
alac->wasted_bits_buffer[ch][i] = get_bits(&alac->gb, alac->wasted_bits);
|
|
}
|
|
}
|
|
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 */
|
|
int i, chan;
|
|
if (alac->setinfo_sample_size <= 16) {
|
|
for (i = 0; i < outputsamples; i++)
|
|
for (chan = 0; chan < channels; chan++) {
|
|
int32_t audiobits;
|
|
|
|
audiobits = get_sbits_long(&alac->gb, alac->setinfo_sample_size);
|
|
|
|
alac->outputsamples_buffer[chan][i] = audiobits;
|
|
}
|
|
} else {
|
|
for (i = 0; i < outputsamples; i++) {
|
|
for (chan = 0; chan < channels; chan++) {
|
|
alac->outputsamples_buffer[chan][i] = get_bits(&alac->gb,
|
|
alac->setinfo_sample_size);
|
|
alac->outputsamples_buffer[chan][i] = sign_extend(alac->outputsamples_buffer[chan][i],
|
|
alac->setinfo_sample_size);
|
|
}
|
|
}
|
|
}
|
|
alac->wasted_bits = 0;
|
|
interlacing_shift = 0;
|
|
interlacing_leftweight = 0;
|
|
}
|
|
if (get_bits(&alac->gb, 3) != 7)
|
|
av_log(avctx, AV_LOG_ERROR, "Error : Wrong End Of Frame\n");
|
|
|
|
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*)outbuffer)[i] = alac->outputsamples_buffer[0][i];
|
|
}
|
|
}
|
|
break;
|
|
case 24:
|
|
if (channels == 2) {
|
|
decorrelate_stereo_24(alac->outputsamples_buffer,
|
|
outbuffer,
|
|
alac->wasted_bits_buffer,
|
|
alac->wasted_bits,
|
|
alac->numchannels,
|
|
outputsamples,
|
|
interlacing_shift,
|
|
interlacing_leftweight);
|
|
} else {
|
|
int i;
|
|
for (i = 0; i < outputsamples; i++)
|
|
((int32_t *)outbuffer)[i] = alac->outputsamples_buffer[0][i] << 8;
|
|
}
|
|
break;
|
|
}
|
|
|
|
if (input_buffer_size * 8 - get_bits_count(&alac->gb) > 8)
|
|
av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n", input_buffer_size * 8 - get_bits_count(&alac->gb));
|
|
|
|
return input_buffer_size;
|
|
}
|
|
|
|
static av_cold int alac_decode_init(AVCodecContext * avctx)
|
|
{
|
|
ALACContext *alac = avctx->priv_data;
|
|
alac->avctx = avctx;
|
|
alac->numchannels = alac->avctx->channels;
|
|
|
|
/* initialize from the extradata */
|
|
if (alac->avctx->extradata_size != ALAC_EXTRADATA_SIZE) {
|
|
av_log(avctx, AV_LOG_ERROR, "alac: expected %d extradata bytes\n",
|
|
ALAC_EXTRADATA_SIZE);
|
|
return -1;
|
|
}
|
|
if (alac_set_info(alac)) {
|
|
av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n");
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static av_cold int alac_decode_close(AVCodecContext *avctx)
|
|
{
|
|
ALACContext *alac = avctx->priv_data;
|
|
|
|
int chan;
|
|
for (chan = 0; chan < MAX_CHANNELS; chan++) {
|
|
av_freep(&alac->predicterror_buffer[chan]);
|
|
av_freep(&alac->outputsamples_buffer[chan]);
|
|
av_freep(&alac->wasted_bits_buffer[chan]);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
AVCodec ff_alac_decoder = {
|
|
"alac",
|
|
AVMEDIA_TYPE_AUDIO,
|
|
CODEC_ID_ALAC,
|
|
sizeof(ALACContext),
|
|
alac_decode_init,
|
|
NULL,
|
|
alac_decode_close,
|
|
alac_decode_frame,
|
|
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
|
|
};
|