*/
/**
- * @file alac.c
+ * @file libavcodec/alac.c
* ALAC (Apple Lossless Audio Codec) decoder
* @author 2005 David Hammerton
*
#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
}
}
-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;
}
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 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;
unsigned int outputsamples;
int hassize;
- int readsamplesize;
+ unsigned int readsamplesize;
int wasted_bytes;
int isnotcompressed;
uint8_t interlacing_shift;
*outputsize = outputsamples * alac->bytespersample;
readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8) + 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 */
for (chan = 0; chan < channels; chan++) {
int32_t audiobits;
- audiobits = get_bits_long(&alac->gb, alac->setinfo_sample_size);
- audiobits = extend_sign32(audiobits, alac->setinfo_sample_size);
+ audiobits = get_sbits_long(&alac->gb, alac->setinfo_sample_size);
alac->outputsamples_buffer[chan][i] = audiobits;
}
alac->context_initialized = 0;
alac->numchannels = alac->avctx->channels;
- alac->bytespersample = (avctx->bits_per_sample / 8) * alac->numchannels;
+ alac->bytespersample = 2 * alac->numchannels;
avctx->sample_fmt = SAMPLE_FMT_S16;
return 0;