* ALAC (Apple Lossless Audio Codec) decoder
* Copyright (c) 2005 David Hammerton
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
- * @file alac.c
+ * @file
* 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
+ * @see 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
#include "avcodec.h"
-#include "bitstream.h"
+#include "get_bits.h"
#include "bytestream.h"
#include "unary.h"
+#include "mathops.h"
#define ALAC_EXTRADATA_SIZE 36
#define MAX_CHANNELS 2
AVCodecContext *avctx;
GetBitContext gb;
- /* init to 0; first frame decode should initialize from extradata and
- * set this to 1 */
- int context_initialized;
- int samplesize;
int numchannels;
int bytespersample;
int32_t *outputsamples_buffer[MAX_CHANNELS];
+ int32_t *wasted_bits_buffer[MAX_CHANNELS];
+
/* 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 */ /* max sample size?? */
- uint32_t setinfo_86; /* 0x00069fe4 */ /* bit rate (average)?? */
- uint32_t setinfo_8a_rate; /* 0x0000ac44 */
/* end setinfo stuff */
+ int wasted_bits;
} ALACContext;
static void allocate_buffers(ALACContext *alac)
alac->outputsamples_buffer[chan] =
av_malloc(alac->setinfo_max_samples_per_frame * 4);
+
+ alac->wasted_bits_buffer[chan] = av_malloc(alac->setinfo_max_samples_per_frame * 4);
}
}
/* buffer size / 2 ? */
alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr);
- alac->setinfo_7a = *ptr++;
+ ptr++; /* ??? */
alac->setinfo_sample_size = *ptr++;
+ if (alac->setinfo_sample_size > 32) {
+ av_log(alac->avctx, AV_LOG_ERROR, "setinfo_sample_size too large\n");
+ return -1;
+ }
alac->setinfo_rice_historymult = *ptr++;
alac->setinfo_rice_initialhistory = *ptr++;
alac->setinfo_rice_kmodifier = *ptr++;
- /* channels? */
- alac->setinfo_7f = *ptr++;
- alac->setinfo_80 = bytestream_get_be16(&ptr);
- /* max coded frame size */
- alac->setinfo_82 = bytestream_get_be32(&ptr);
- /* bitrate ? */
- alac->setinfo_86 = bytestream_get_be32(&ptr);
- /* samplerate */
- alac->setinfo_8a_rate = bytestream_get_be32(&ptr);
+ ptr++; /* channels? */
+ bytestream_get_be16(&ptr); /* ??? */
+ bytestream_get_be32(&ptr); /* max coded frame size */
+ bytestream_get_be32(&ptr); /* bitrate ? */
+ bytestream_get_be32(&ptr); /* samplerate */
allocate_buffers(alac);
/* special case: there may be compressed blocks of 0 */
if ((history < 128) && (output_count+1 < output_size)) {
- int block_size, k;
+ int k;
+ unsigned int block_size;
sign_modifier = 1;
block_size= decode_scalar(&alac->gb, k, rice_kmodifier, 16);
if (block_size > 0) {
+ if(block_size >= output_size - output_count){
+ av_log(alac->avctx, AV_LOG_ERROR, "invalid zero block size of %d %d %d\n", block_size, output_size, output_count);
+ block_size= output_size - output_count - 1;
+ }
memset(&output_buffer[output_count+1], 0, block_size * 4);
output_count += block_size;
}
}
}
-static inline int32_t extend_sign32(int32_t val, int bits)
-{
- return (val << (32 - bits)) >> (32 - bits);
-}
-
static inline int sign_only(int v)
{
return v ? FFSIGN(v) : 0;
prev_value = buffer_out[i];
error_value = error_buffer[i+1];
buffer_out[i+1] =
- extend_sign32((prev_value + error_value), readsamplesize);
+ sign_extend((prev_value + error_value), readsamplesize);
}
return;
}
int32_t val;
val = buffer_out[i] + error_buffer[i+1];
- val = extend_sign32(val, readsamplesize);
+ val = sign_extend(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 optimized
*/
- if (predictor_coef_num == 4) {
- /* FIXME: optimized general case */
- return;
- }
-
- if (predictor_coef_table == 8) {
- /* FIXME: optimized general case */
- return;
- }
-#endif
/* general case */
if (predictor_coef_num > 0) {
outval = (1 << (predictor_quantitization-1)) + sum;
outval = outval >> predictor_quantitization;
outval = outval + buffer_out[0] + error_val;
- outval = extend_sign32(outval, readsamplesize);
+ outval = sign_extend(outval, readsamplesize);
buffer_out[predictor_coef_num+1] = outval;
}
}
+static void decorrelate_stereo_24(int32_t *buffer[MAX_CHANNELS],
+ int32_t *buffer_out,
+ int32_t *wasted_bits_buffer[MAX_CHANNELS],
+ int wasted_bits,
+ 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 a, b;
+
+ a = buffer[0][i];
+ b = buffer[1][i];
+
+ a -= (b * interlacing_leftweight) >> interlacing_shift;
+ b += a;
+
+ if (wasted_bits) {
+ b = (b << wasted_bits) | wasted_bits_buffer[0][i];
+ a = (a << wasted_bits) | wasted_bits_buffer[1][i];
+ }
+
+ buffer_out[i * numchannels] = b << 8;
+ buffer_out[i * numchannels + 1] = a << 8;
+ }
+ } else {
+ for (i = 0; i < numsamples; i++) {
+ int32_t left, right;
+
+ left = buffer[0][i];
+ right = buffer[1][i];
+
+ if (wasted_bits) {
+ left = (left << wasted_bits) | wasted_bits_buffer[0][i];
+ right = (right << wasted_bits) | wasted_bits_buffer[1][i];
+ }
+
+ buffer_out[i * numchannels] = left << 8;
+ buffer_out[i * numchannels + 1] = right << 8;
+ }
+ }
+}
+
static int alac_decode_frame(AVCodecContext *avctx,
void *outbuffer, int *outputsize,
- const uint8_t *inbuffer, int input_buffer_size)
+ AVPacket *avpkt)
{
+ const uint8_t *inbuffer = avpkt->data;
+ int input_buffer_size = avpkt->size;
ALACContext *alac = avctx->priv_data;
int channels;
- int32_t outputsamples;
+ unsigned int outputsamples;
int hassize;
- int readsamplesize;
- int wasted_bytes;
+ unsigned int readsamplesize;
int isnotcompressed;
uint8_t interlacing_shift;
uint8_t interlacing_leftweight;
/* short-circuit null buffers */
if (!inbuffer || !input_buffer_size)
- return input_buffer_size;
-
- /* initialize from the extradata */
- if (!alac->context_initialized) {
- if (alac->avctx->extradata_size != ALAC_EXTRADATA_SIZE) {
- av_log(avctx, AV_LOG_ERROR, "alac: expected %d extradata bytes\n",
- ALAC_EXTRADATA_SIZE);
- return input_buffer_size;
- }
- if (alac_set_info(alac)) {
- av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n");
- return input_buffer_size;
- }
- alac->context_initialized = 1;
- }
+ return -1;
init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8);
if (channels > MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR, "channels > %d not supported\n",
MAX_CHANNELS);
- return input_buffer_size;
+ return -1;
}
/* 2^result = something to do with output waiting.
/* the output sample size is stored soon */
hassize = get_bits1(&alac->gb);
- wasted_bytes = get_bits(&alac->gb, 2); /* unknown ? */
+ alac->wasted_bits = get_bits(&alac->gb, 2) << 3;
/* whether the frame is compressed */
isnotcompressed = get_bits1(&alac->gb);
if (hassize) {
/* now read the number of samples as a 32bit integer */
- outputsamples = get_bits(&alac->gb, 32);
+ outputsamples = get_bits_long(&alac->gb, 32);
+ if(outputsamples > alac->setinfo_max_samples_per_frame){
+ av_log(avctx, AV_LOG_ERROR, "outputsamples %d > %d\n", outputsamples, alac->setinfo_max_samples_per_frame);
+ return -1;
+ }
} else
outputsamples = alac->setinfo_max_samples_per_frame;
+ switch (alac->setinfo_sample_size) {
+ case 16: avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ alac->bytespersample = channels << 1;
+ break;
+ case 24: avctx->sample_fmt = AV_SAMPLE_FMT_S32;
+ 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 - (wasted_bytes * 8) + channels - 1;
+ 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[channels][32];
- int predictor_coef_num[channels];
- int prediction_type[channels];
- int prediction_quantitization[channels];
- int ricemodifier[channels];
+ 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);
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");
-
+ 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],
}
} else {
/* not compressed, easy case */
+ int i, chan;
if (alac->setinfo_sample_size <= 16) {
- int i, chan;
- for (chan = 0; chan < channels; chan++)
- for (i = 0; i < outputsamples; i++) {
- int32_t audiobits;
+ for (i = 0; i < outputsamples; i++)
+ for (chan = 0; chan < channels; chan++) {
+ int32_t audiobits;
- audiobits = get_bits(&alac->gb, alac->setinfo_sample_size);
- audiobits = extend_sign32(audiobits, readsamplesize);
+ audiobits = get_sbits_long(&alac->gb, alac->setinfo_sample_size);
- alac->outputsamples_buffer[chan][i] = audiobits;
- }
+ 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;
+ 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);
}
+ }
}
- /* wasted_bytes = 0; */
+ 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:
} else {
int i;
for (i = 0; i < outputsamples; i++) {
- int16_t sample = alac->outputsamples_buffer[0][i];
- ((int16_t*)outbuffer)[i * alac->numchannels] = sample;
+ ((int16_t*)outbuffer)[i] = alac->outputsamples_buffer[0][i];
}
}
break;
- case 20:
case 24:
- // It is not clear if there exist any encoder that creates 24 bit ALAC
- // files. iTunes convert 24 bit raw files to 16 bit before encoding.
- case 32:
- av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented sample size %i\n", alac->setinfo_sample_size);
- break;
- default:
+ 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;
}
{
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;
+
+ /* 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;
}
int chan;
for (chan = 0; chan < MAX_CHANNELS; chan++) {
- av_free(alac->predicterror_buffer[chan]);
- av_free(alac->outputsamples_buffer[chan]);
+ av_freep(&alac->predicterror_buffer[chan]);
+ av_freep(&alac->outputsamples_buffer[chan]);
+ av_freep(&alac->wasted_bits_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,
+AVCodec ff_alac_decoder = {
+ .name = "alac",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_ALAC,
+ .priv_data_size = sizeof(ALACContext),
+ .init = alac_decode_init,
+ .close = alac_decode_close,
+ .decode = alac_decode_frame,
+ .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
};
projects / ffmpeg / blobdiff