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first pass at ALAC decoder from David Hammerton; while David's original

Browse Source 
decoder works great, this decoder is not completely and seamlessly
integrated yet with FFmpeg

Originally committed as revision 4008 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Mike Melanson 2005-03-06 00:43:55 +00:00
parent 2a515c08f2
commit 6d6d7970e7
6 changed files with 996 additions and 3 deletions
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1
CREDITS
View File

@ -14,6 +14,7 @@ Brian Foley
Arpad Gereoffy
Philip Gladstone
Vladimir Gneushev
David Hammerton
Wolfgang Hesseler
Falk Hueffner
Zdenek Kabelac

5
libavcodec/Makefile
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@ -1,6 +1,6 @@
#
# libavcodec Makefile
# (c) 2000-2003 Fabrice Bellard
# (c) 2000-2005 Fabrice Bellard
#
include ../config.mak
@ -22,7 +22,8 @@ OBJS= bitstream.o utils.o mem.o allcodecs.o \
smc.o parser.o flicvideo.o truemotion1.o vmdav.o lcl.o qtrle.o g726.o \
flac.o vp3dsp.o integer.o snow.o tscc.o sonic.o ulti.o h264idct.o \
qdrw.o xl.o rangecoder.o png.o pnm.o qpeg.o vc9.o h263.o h261.o \
msmpeg4.o h263dec.o svq1.o rv10.o wmadec.o indeo3.o shorten.o loco.o
msmpeg4.o h263dec.o svq1.o rv10.o wmadec.o indeo3.o shorten.o loco.o \
alac.o
AMROBJS=
ifeq ($(AMR_NB),yes)

970
libavcodec/alac.c Normal file
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@ -0,0 +1,970 @@
/*
* ALAC (Apple Lossless Audio Codec) decoder
* Copyright (c) 2005 David Hammerton
* All rights reserved.
*
* This library 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 of the License, or (at your option) any later version.
*
* This library 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 this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
/**
* @file alac.c
* ALAC (Apple Lossless Audio Codec) decoder
* @author 2005 David Hammerton
*
* For more information on the ALAC format, visit:
* http://crazney.net/programs/itunes/alac.html
*
* Note: This decoder expects a 36- (0x24-)byte QuickTime atom to be
* passed through the extradata[_size] fields. This atom is tacked onto
* the end of an 'alac' stsd atom and has the following format:
* bytes 0-3 atom size (0x24), big-endian
* bytes 4-7 atom type ('alac', not the 'alac' tag from start of stsd)
* bytes 8-35 data bytes needed by decoder
*/
#include "avcodec.h"
#define ALAC_EXTRADATA_SIZE 36
struct alac_file {
unsigned char *input_buffer;
int input_buffer_index;
int input_buffer_size;
int input_buffer_bitaccumulator; /* used so we can do arbitary
bit reads */
int samplesize;
int numchannels;
int bytespersample;
/* buffers */
int32_t *predicterror_buffer_a;
int32_t *predicterror_buffer_b;
int32_t *outputsamples_buffer_a;
int32_t *outputsamples_buffer_b;
/* stuff from setinfo */
uint32_t setinfo_max_samples_per_frame; /* 0x1000 = 4096 */ /* max samples per frame? */
uint8_t setinfo_7a; /* 0x00 */
uint8_t setinfo_sample_size; /* 0x10 */
uint8_t setinfo_rice_historymult; /* 0x28 */
uint8_t setinfo_rice_initialhistory; /* 0x0a */
uint8_t setinfo_rice_kmodifier; /* 0x0e */
uint8_t setinfo_7f; /* 0x02 */
uint16_t setinfo_80; /* 0x00ff */
uint32_t setinfo_82; /* 0x000020e7 */
uint32_t setinfo_86; /* 0x00069fe4 */
uint32_t setinfo_8a_rate; /* 0x0000ac44 */
/* end setinfo stuff */
};
typedef struct alac_file alac_file;
typedef struct {
AVCodecContext *avctx;
/* init to 0; first frame decode should initialize from extradata and
* set this to 1 */
int context_initialized;
alac_file *alac;
} ALACContext;
static void allocate_buffers(alac_file *alac)
{
alac->predicterror_buffer_a = av_malloc(alac->setinfo_max_samples_per_frame * 4);
alac->predicterror_buffer_b = av_malloc(alac->setinfo_max_samples_per_frame * 4);
alac->outputsamples_buffer_a = av_malloc(alac->setinfo_max_samples_per_frame * 4);
alac->outputsamples_buffer_b = av_malloc(alac->setinfo_max_samples_per_frame * 4);
}
void alac_set_info(alac_file *alac, char *inputbuffer)
{
char *ptr = inputbuffer;
ptr += 4; /* size */
ptr += 4; /* alac */
ptr += 4; /* 0 ? */
alac->setinfo_max_samples_per_frame = BE_32(ptr); /* buffer size / 2 ? */
ptr += 4;
alac->setinfo_7a = *ptr++;
alac->setinfo_sample_size = *ptr++;
alac->setinfo_rice_historymult = *ptr++;
alac->setinfo_rice_initialhistory = *ptr++;
alac->setinfo_rice_kmodifier = *ptr++;
alac->setinfo_7f = *ptr++;
alac->setinfo_80 = BE_16(ptr);
ptr += 2;
alac->setinfo_82 = BE_32(ptr);
ptr += 4;
alac->setinfo_86 = BE_32(ptr);
ptr += 4;
alac->setinfo_8a_rate = BE_32(ptr);
ptr += 4;
allocate_buffers(alac);
}
/* stream reading */
/* supports reading 1 to 16 bits, in big endian format */
static uint32_t readbits_16(alac_file *alac, int bits)
{
uint32_t result;
int new_accumulator;
if (alac->input_buffer_index + 2 >= alac->input_buffer_size) {
av_log(NULL, AV_LOG_INFO, "alac: input buffer went out of bounds (%d >= %d)\n",
alac->input_buffer_index + 2, alac->input_buffer_size);
exit (0);
}
result = (alac->input_buffer[alac->input_buffer_index + 0] << 16) |
(alac->input_buffer[alac->input_buffer_index + 1] << 8) |
(alac->input_buffer[alac->input_buffer_index + 2]);
/* shift left by the number of bits we've already read,
* so that the top 'n' bits of the 24 bits we read will
* be the return bits */
result = result << alac->input_buffer_bitaccumulator;
result = result & 0x00ffffff;
/* and then only want the top 'n' bits from that, where
* n is 'bits' */
result = result >> (24 - bits);
new_accumulator = (alac->input_buffer_bitaccumulator + bits);
/* increase the buffer pointer if we've read over n bytes. */
alac->input_buffer_index += (new_accumulator >> 3);
/* and the remainder goes back into the bit accumulator */
alac->input_buffer_bitaccumulator = (new_accumulator & 7);
return result;
}
/* supports reading 1 to 32 bits, in big endian format */
static uint32_t readbits(alac_file *alac, int bits)
{
int32_t result = 0;
if (bits > 16) {
bits -= 16;
result = readbits_16(alac, 16) << bits;
}
result |= readbits_16(alac, bits);
return result;
}
/* reads a single bit */
static int readbit(alac_file *alac)
{
int result;
int new_accumulator;
if (alac->input_buffer_index >= alac->input_buffer_size) {
av_log(NULL, AV_LOG_INFO, "alac: input buffer went out of bounds (%d >= %d)\n",
alac->input_buffer_index + 2, alac->input_buffer_size);
exit (0);
}
result = alac->input_buffer[alac->input_buffer_index];
result = result << alac->input_buffer_bitaccumulator;
result = result >> 7 & 1;
new_accumulator = (alac->input_buffer_bitaccumulator + 1);
alac->input_buffer_index += (new_accumulator / 8);
alac->input_buffer_bitaccumulator = (new_accumulator % 8);
return result;
}
static void unreadbits(alac_file *alac, int bits)
{
int new_accumulator = (alac->input_buffer_bitaccumulator - bits);
alac->input_buffer_index += (new_accumulator >> 3);
alac->input_buffer_bitaccumulator = (new_accumulator & 7);
if (alac->input_buffer_bitaccumulator < 0)
alac->input_buffer_bitaccumulator *= -1;
}
/* hideously inefficient. could use a bitmask search,
* alternatively bsr on x86,
*/
static int count_leading_zeros(int32_t input)
{
int i = 0;
while (!(0x80000000 & input) && i < 32) {
i++;
input = input << 1;
}
return i;
}
void bastardized_rice_decompress(alac_file *alac,
int32_t *output_buffer,
int output_size,
int readsamplesize, /* arg_10 */
int rice_initialhistory, /* arg424->b */
int rice_kmodifier, /* arg424->d */
int rice_historymult, /* arg424->c */
int rice_kmodifier_mask /* arg424->e */
)
{
int output_count;
unsigned int history = rice_initialhistory;
int sign_modifier = 0;
for (output_count = 0; output_count < output_size; output_count++) {
int32_t x = 0;
int32_t x_modified;
int32_t final_val;
/* read x - number of 1s before 0 represent the rice */
while (x <= 8 && readbit(alac)) {
x++;
}
if (x > 8) { /* RICE THRESHOLD */
/* use alternative encoding */
int32_t value;
value = readbits(alac, readsamplesize);
/* mask value to readsamplesize size */
if (readsamplesize != 32)
value &= (0xffffffff >> (32 - readsamplesize));
x = value;
} else {
/* standard rice encoding */
int extrabits;
int k; /* size of extra bits */
/* read k, that is bits as is */
k = 31 - rice_kmodifier - count_leading_zeros((history >> 9) + 3);
if (k < 0)
k += rice_kmodifier;
else
k = rice_kmodifier;
if (k != 1) {
extrabits = readbits(alac, k);
/* multiply x by 2^k - 1, as part of their strange algorithm */
x = (x << k) - x;
if (extrabits > 1) {
x += extrabits - 1;
} else
unreadbits(alac, 1);
}
}
x_modified = sign_modifier + x;
final_val = (x_modified + 1) / 2;
if (x_modified & 1) final_val *= -1;
output_buffer[output_count] = final_val;
sign_modifier = 0;
/* now update the history */
history += (x_modified * rice_historymult)
- ((history * rice_historymult) >> 9);
if (x_modified > 0xffff)
history = 0xffff;
/* special case: there may be compressed blocks of 0 */
if ((history < 128) && (output_count+1 < output_size)) {
int block_size;
sign_modifier = 1;
x = 0;
while (x <= 8 && readbit(alac)) {
x++;
}
if (x > 8) {
block_size = readbits(alac, 16);
block_size &= 0xffff;
} else {
int k;
int extrabits;
k = count_leading_zeros(history) + ((history + 16) >> 6 /* / 64 */) - 24;
extrabits = readbits(alac, k);
block_size = (((1 << k) - 1) & rice_kmodifier_mask) * x
+ extrabits - 1;
if (extrabits < 2) {
x = 1 - extrabits;
block_size += x;
unreadbits(alac, 1);
}
}
if (block_size > 0) {
memset(&output_buffer[output_count+1], 0, block_size * 4);
output_count += block_size;
}
if (block_size > 0xffff)
sign_modifier = 0;
history = 0;
}
}
}
#define SIGN_EXTENDED32(val, bits) ((val << (32 - bits)) >> (32 - bits))
#define SIGN_ONLY(v) \
((v < 0) ? (-1) : \
((v > 0) ? (1) : \
(0)))
static void predictor_decompress_fir_adapt(int32_t *error_buffer,
int32_t *buffer_out,
int output_size,
int readsamplesize,
int16_t *predictor_coef_table,
int predictor_coef_num,
int predictor_quantitization)
{
int i;
/* first sample always copies */
*buffer_out = *error_buffer;
if (!predictor_coef_num) {
if (output_size <= 1) return;
memcpy(buffer_out+1, error_buffer+1, (output_size-1) * 4);
return;
}
if (predictor_coef_num == 0x1f) { /* 11111 - max value of predictor_coef_num */
/* second-best case scenario for fir decompression,
* error describes a small difference from the previous sample only
*/
if (output_size <= 1) return;
for (i = 0; i < output_size - 1; i++) {
int32_t prev_value;
int32_t error_value;
prev_value = buffer_out[i];
error_value = error_buffer[i+1];
buffer_out[i+1] = SIGN_EXTENDED32((prev_value + error_value), readsamplesize);
}
return;
}
/* read warm-up samples */
if (predictor_coef_num > 0) {
int i;
for (i = 0; i < predictor_coef_num; i++) {
int32_t val;
val = buffer_out[i] + error_buffer[i+1];
val = SIGN_EXTENDED32(val, readsamplesize);
buffer_out[i+1] = val;
}
}
#if 0
/* 4 and 8 are very common cases (the only ones i've seen). these
* should be unrolled and optimised
*/
if (predictor_coef_num == 4) {
/* FIXME: optimised general case */
return;
}
if (predictor_coef_table == 8) {
/* FIXME: optimised general case */
return;
}
#endif
/* general case */
if (predictor_coef_num > 0) {
for (i = predictor_coef_num + 1;
i < output_size;
i++) {
int j;
int sum = 0;
int outval;
int error_val = error_buffer[i];
for (j = 0; j < predictor_coef_num; j++) {
sum += (buffer_out[predictor_coef_num-j] - buffer_out[0]) *
predictor_coef_table[j];
}
outval = (1 << (predictor_quantitization-1)) + sum;
outval = outval >> predictor_quantitization;
outval = outval + buffer_out[0] + error_val;
outval = SIGN_EXTENDED32(outval, readsamplesize);
buffer_out[predictor_coef_num+1] = outval;
if (error_val > 0) {
int predictor_num = predictor_coef_num - 1;
while (predictor_num >= 0 && error_val > 0) {
int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num];
int sign = SIGN_ONLY(val);
predictor_coef_table[predictor_num] -= sign;
val *= sign; /* absolute value */
error_val -= ((val >> predictor_quantitization) *
(predictor_coef_num - predictor_num));
predictor_num--;
}
} else if (error_val < 0) {
int predictor_num = predictor_coef_num - 1;
while (predictor_num >= 0 && error_val < 0) {
int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num];
int sign = - SIGN_ONLY(val);
predictor_coef_table[predictor_num] -= sign;
val *= sign; /* neg value */
error_val -= ((val >> predictor_quantitization) *
(predictor_coef_num - predictor_num));
predictor_num--;
}
}
buffer_out++;
}
}
}
void deinterlace_16(int32_t *buffer_a, int32_t *buffer_b,
int16_t *buffer_out,
int numchannels, int numsamples,
uint8_t interlacing_shift,
uint8_t interlacing_leftweight) {
int i;
if (numsamples <= 0) return;
/* weighted interlacing */
if (interlacing_leftweight) {
for (i = 0; i < numsamples; i++) {
int32_t difference, midright;
int16_t left;
int16_t right;
midright = buffer_a[i];
difference = buffer_b[i];
right = midright - ((difference * interlacing_leftweight) >> interlacing_shift);
left = (midright - ((difference * interlacing_leftweight) >> interlacing_shift))
+ difference;
/* output is always little endian */
/*
if (host_bigendian) {
be2me_16(left);
be2me_16(right);
}
*/
buffer_out[i*numchannels] = left;
buffer_out[i*numchannels + 1] = right;
}
return;
}
/* otherwise basic interlacing took place */
for (i = 0; i < numsamples; i++) {
int16_t left, right;
left = buffer_a[i];
right = buffer_b[i];
/* output is always little endian */
/*
if (host_bigendian) {
be2me_16(left);
be2me_16(right);
}
*/
buffer_out[i*numchannels] = left;
buffer_out[i*numchannels + 1] = right;
}
}
int decode_frame(ALACContext *s, alac_file *alac,
unsigned char *inbuffer,
int input_buffer_size,
void *outbuffer, int *outputsize){
int channels;
int32_t outputsamples = alac->setinfo_max_samples_per_frame;
/* initialize from the extradata */
if (!s->context_initialized) {
if (s->avctx->extradata_size != ALAC_EXTRADATA_SIZE) {
av_log(NULL, AV_LOG_ERROR, "alac: expected %d extradata bytes\n",
ALAC_EXTRADATA_SIZE);
return input_buffer_size;
}
alac_set_info(s->alac, s->avctx->extradata);
s->context_initialized = 1;
}
/* setup the stream */
alac->input_buffer = inbuffer;
alac->input_buffer_index = 0;
alac->input_buffer_size = input_buffer_size;
alac->input_buffer_bitaccumulator = 0;
channels = readbits(alac, 3);
*outputsize = outputsamples * alac->bytespersample;
switch(channels) {
case 0: { /* 1 channel */
int hassize;
int isnotcompressed;
int readsamplesize;
int wasted_bytes;
int ricemodifier;
/* 2^result = something to do with output waiting.
* perhaps matters if we read > 1 frame in a pass?
*/
readbits(alac, 4);
readbits(alac, 12); /* unknown, skip 12 bits */
hassize = readbits(alac, 1); /* the output sample size is stored soon */
wasted_bytes = readbits(alac, 2); /* unknown ? */
isnotcompressed = readbits(alac, 1); /* whether the frame is compressed */
if (hassize) {
/* now read the number of samples,
* as a 32bit integer */
outputsamples = readbits(alac, 32);
*outputsize = outputsamples * alac->bytespersample;
}
readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8);
if (!isnotcompressed) {
/* so it is compressed */
int16_t predictor_coef_table[32];
int predictor_coef_num;
int prediction_type;
int prediction_quantitization;
int i;
/* skip 16 bits, not sure what they are. seem to be used in
* two channel case */
readbits(alac, 8);
readbits(alac, 8);
prediction_type = readbits(alac, 4);
prediction_quantitization = readbits(alac, 4);
ricemodifier = readbits(alac, 3);
predictor_coef_num = readbits(alac, 5);
/* read the predictor table */
for (i = 0; i < predictor_coef_num; i++) {
predictor_coef_table[i] = (int16_t)readbits(alac, 16);
}
if (wasted_bytes) {
/* these bytes seem to have something to do with
* > 2 channel files.
*/
av_log(NULL, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of wasted_bytes\n");
}
bastardized_rice_decompress(alac,
alac->predicterror_buffer_a,
outputsamples,
readsamplesize,
alac->setinfo_rice_initialhistory,
alac->setinfo_rice_kmodifier,
ricemodifier * alac->setinfo_rice_historymult / 4,
(1 << alac->setinfo_rice_kmodifier) - 1);
if (prediction_type == 0) {
/* adaptive fir */
predictor_decompress_fir_adapt(alac->predicterror_buffer_a,
alac->outputsamples_buffer_a,
outputsamples,
readsamplesize,
predictor_coef_table,
predictor_coef_num,
prediction_quantitization);
} else {
av_log(NULL, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type);
/* 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 (readsamplesize <= 16) {
int i;
for (i = 0; i < outputsamples; i++) {
int32_t audiobits = readbits(alac, readsamplesize);
audiobits = SIGN_EXTENDED32(audiobits, readsamplesize);
alac->outputsamples_buffer_a[i] = audiobits;
}
} else {
int i;
for (i = 0; i < outputsamples; i++) {
int32_t audiobits;
audiobits = readbits(alac, 16);
/* special case of sign extension..
* as we'll be ORing the low 16bits into this */
audiobits = audiobits << 16;
audiobits = audiobits >> (32 - readsamplesize);
audiobits |= readbits(alac, readsamplesize - 16);
alac->outputsamples_buffer_a[i] = audiobits;
}
}
/* wasted_bytes = 0; // unused */
}
switch(alac->setinfo_sample_size) {
case 16: {
int i;
for (i = 0; i < outputsamples; i++) {
int16_t sample = alac->outputsamples_buffer_a[i];
be2me_16(sample);
((int16_t*)outbuffer)[i * alac->numchannels] = sample;
}
break;
}
case 20:
case 24:
case 32:
av_log(NULL, AV_LOG_ERROR, "FIXME: unimplemented sample size %i\n", alac->setinfo_sample_size);
break;
default:
break;
}
break;
}
case 1: { /* 2 channels */
int hassize;
int isnotcompressed;
int readsamplesize;
int wasted_bytes;
uint8_t interlacing_shift;
uint8_t interlacing_leftweight;
/* 2^result = something to do with output waiting.
* perhaps matters if we read > 1 frame in a pass?
*/
readbits(alac, 4);
readbits(alac, 12); /* unknown, skip 12 bits */
hassize = readbits(alac, 1); /* the output sample size is stored soon */
wasted_bytes = readbits(alac, 2); /* unknown ? */
isnotcompressed = readbits(alac, 1); /* whether the frame is compressed */
if (hassize) {
/* now read the number of samples,
* as a 32bit integer */
outputsamples = readbits(alac, 32);
*outputsize = outputsamples * alac->bytespersample;
}
readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8) + 1;
if (!isnotcompressed) {
/* compressed */
int16_t predictor_coef_table_a[32];
int predictor_coef_num_a;
int prediction_type_a;
int prediction_quantitization_a;
int ricemodifier_a;
int16_t predictor_coef_table_b[32];
int predictor_coef_num_b;
int prediction_type_b;
int prediction_quantitization_b;
int ricemodifier_b;
int i;
interlacing_shift = readbits(alac, 8);
interlacing_leftweight = readbits(alac, 8);
/******** channel 1 ***********/
prediction_type_a = readbits(alac, 4);
prediction_quantitization_a = readbits(alac, 4);
ricemodifier_a = readbits(alac, 3);
predictor_coef_num_a = readbits(alac, 5);
/* read the predictor table */
for (i = 0; i < predictor_coef_num_a; i++) {
predictor_coef_table_a[i] = (int16_t)readbits(alac, 16);
}
/******** channel 2 *********/
prediction_type_b = readbits(alac, 4);
prediction_quantitization_b = readbits(alac, 4);
ricemodifier_b = readbits(alac, 3);
predictor_coef_num_b = readbits(alac, 5);
/* read the predictor table */
for (i = 0; i < predictor_coef_num_b; i++) {
predictor_coef_table_b[i] = (int16_t)readbits(alac, 16);
}
/*********************/
if (wasted_bytes) {
/* see mono case */
av_log(NULL, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of wasted_bytes\n");
}
/* channel 1 */
bastardized_rice_decompress(alac,
alac->predicterror_buffer_a,
outputsamples,
readsamplesize,
alac->setinfo_rice_initialhistory,
alac->setinfo_rice_kmodifier,
ricemodifier_a * alac->setinfo_rice_historymult / 4,
(1 << alac->setinfo_rice_kmodifier) - 1);
if (prediction_type_a == 0) {
/* adaptive fir */
predictor_decompress_fir_adapt(alac->predicterror_buffer_a,
alac->outputsamples_buffer_a,
outputsamples,
readsamplesize,
predictor_coef_table_a,
predictor_coef_num_a,
prediction_quantitization_a);
} else {
/* see mono case */
av_log(NULL, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type_a);
}
/* channel 2 */
bastardized_rice_decompress(alac,
alac->predicterror_buffer_b,
outputsamples,
readsamplesize,
alac->setinfo_rice_initialhistory,
alac->setinfo_rice_kmodifier,
ricemodifier_b * alac->setinfo_rice_historymult / 4,
(1 << alac->setinfo_rice_kmodifier) - 1);
if (prediction_type_b == 0) {
/* adaptive fir */
predictor_decompress_fir_adapt(alac->predicterror_buffer_b,
alac->outputsamples_buffer_b,
outputsamples,
readsamplesize,
predictor_coef_table_b,
predictor_coef_num_b,
prediction_quantitization_b);
} else {
av_log(NULL, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type_b);
}
} else {
/* not compressed, easy case */
if (alac->setinfo_sample_size <= 16) {
int i;
for (i = 0; i < outputsamples; i++) {
int32_t audiobits_a, audiobits_b;
audiobits_a = readbits(alac, alac->setinfo_sample_size);
audiobits_b = readbits(alac, alac->setinfo_sample_size);
audiobits_a = SIGN_EXTENDED32(audiobits_a, alac->setinfo_sample_size);
audiobits_b = SIGN_EXTENDED32(audiobits_b, alac->setinfo_sample_size);
alac->outputsamples_buffer_a[i] = audiobits_a;
alac->outputsamples_buffer_b[i] = audiobits_b;
}
} else {
int i;
for (i = 0; i < outputsamples; i++) {
int32_t audiobits_a, audiobits_b;
audiobits_a = readbits(alac, 16);
audiobits_a = audiobits_a << 16;
audiobits_a = audiobits_a >> (32 - alac->setinfo_sample_size);
audiobits_a |= readbits(alac, alac->setinfo_sample_size - 16);
audiobits_b = readbits(alac, 16);
audiobits_b = audiobits_b << 16;
audiobits_b = audiobits_b >> (32 - alac->setinfo_sample_size);
audiobits_b |= readbits(alac, alac->setinfo_sample_size - 16);
alac->outputsamples_buffer_a[i] = audiobits_a;
alac->outputsamples_buffer_b[i] = audiobits_b;
}
}
/* wasted_bytes = 0; */
interlacing_shift = 0;
interlacing_leftweight = 0;
}
switch(alac->setinfo_sample_size) {
case 16: {
deinterlace_16(alac->outputsamples_buffer_a,
alac->outputsamples_buffer_b,
(int16_t*)outbuffer,
alac->numchannels,
outputsamples,
interlacing_shift,
interlacing_leftweight);
break;
}
case 20:
case 24:
case 32:
av_log(NULL, AV_LOG_ERROR, "FIXME: unimplemented sample size %i\n", alac->setinfo_sample_size);
break;
default:
break;
}
break;
}
}
av_log(NULL, AV_LOG_INFO, "buf size = %d, consumed %d\n",
input_buffer_size, alac->input_buffer_index);
/* avoid infinite loop: if decoder consumed 0 bytes; report all bytes
* consumed */
// if (alac->input_buffer_index)
// return alac->input_buffer_index;
// else
return input_buffer_size;
}
static int alac_decode_init(AVCodecContext * avctx)
{
ALACContext *s = avctx->priv_data;
s->avctx = avctx;
s->context_initialized = 0;
s->alac = av_malloc(sizeof(alac_file));
s->alac->samplesize = s->avctx->bits_per_sample;
s->alac->numchannels = s->avctx->channels;
s->alac->bytespersample = (s->alac->samplesize / 8) * s->alac->numchannels;
return 0;
}
static int alac_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
uint8_t *buf, int buf_size)
{
ALACContext *s = avctx->priv_data;
int bytes_consumed = buf_size;
if (buf)
bytes_consumed = decode_frame(s, s->alac, buf, buf_size,
data, data_size);
return bytes_consumed;
}
static int alac_decode_close(AVCodecContext *avctx)
{
ALACContext *s = avctx->priv_data;
av_free(s->alac->predicterror_buffer_a);
av_free(s->alac->predicterror_buffer_b);
av_free(s->alac->outputsamples_buffer_a);
av_free(s->alac->outputsamples_buffer_b);
return 0;
}
AVCodec alac_decoder = {
"alac",
CODEC_TYPE_AUDIO,
CODEC_ID_ALAC,
sizeof(ALACContext),
alac_decode_init,
NULL,
alac_decode_close,
alac_decode_frame,
};

1
libavcodec/allcodecs.c
View File

@ -196,6 +196,7 @@ void avcodec_register_all(void)
register_avcodec(&qtrle_decoder);
register_avcodec(&flac_decoder);
register_avcodec(&shorten_decoder);
register_avcodec(&alac_decoder);
#endif /* CONFIG_DECODERS */
#ifdef AMR_NB

4
libavcodec/avcodec.h
View File

@ -17,7 +17,7 @@ extern "C" {
#define FFMPEG_VERSION_INT 0x000409
#define FFMPEG_VERSION "0.4.9-pre1"
#define LIBAVCODEC_BUILD 4744
#define LIBAVCODEC_BUILD 4745
#define LIBAVCODEC_VERSION_INT FFMPEG_VERSION_INT
#define LIBAVCODEC_VERSION FFMPEG_VERSION
@ -170,6 +170,7 @@ enum CodecID {
CODEC_ID_MPEG2TS= 0x20000, /* _FAKE_ codec to indicate a raw MPEG2 transport
stream (only used by libavformat) */
CODEC_ID_ALAC,
};
/* CODEC_ID_MP3LAME is absolete */
@ -2011,6 +2012,7 @@ extern AVCodec xl_decoder;
extern AVCodec qpeg_decoder;
extern AVCodec shorten_decoder;
extern AVCodec loco_decoder;
extern AVCodec alac_decoder;
/* pcm codecs */
#define PCM_CODEC(id, name) \

18
libavformat/mov.c
View File

@ -142,6 +142,7 @@ static const CodecTag mov_audio_tags[] = {
{ CODEC_ID_AMR_NB, MKTAG('s', 'a', 'm', 'r') }, /* AMR-NB 3gp */
{ CODEC_ID_AMR_WB, MKTAG('s', 'a', 'w', 'b') }, /* AMR-WB 3gp */
{ CODEC_ID_AC3, MKTAG('m', 's', 0x20, 0x00) }, /* Dolby AC-3 */
{ CODEC_ID_ALAC,MKTAG('a', 'l', 'a', 'c') }, /* Apple Lossless */
{ CODEC_ID_NONE, 0 },
};
@ -1101,6 +1102,23 @@ static int mov_read_stsd(MOVContext *c, ByteIOContext *pb, MOV_atom_t atom)
st->codec.channels = (px[1] >> 3) & 15;
}
}
else if( st->codec.codec_tag == MKTAG( 'a', 'l', 'a', 'c' ))
{
/* Handle alac audio tag + special extradata */
get_be32(pb); /* version */
get_be32(pb);
st->codec.channels = get_be16(pb); /* channels */
st->codec.bits_per_sample = get_be16(pb); /* bits per sample */
get_be32(pb);
st->codec.sample_rate = get_be16(pb);
get_be16(pb);
/* fetch the 36-byte extradata needed for alac decoding */
st->codec.extradata_size = 36;
st->codec.extradata = (uint8_t*)
av_mallocz(st->codec.extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
get_buffer(pb, st->codec.extradata, st->codec.extradata_size);
}
else if(size>=(16+20))
{//16 bytes read, reading atleast 20 more
uint16_t version;