*
* 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
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file flac.c
* FLAC (Free Lossless Audio Codec) decoder
* @author Alex Beregszaszi
+ *
+ * For more information on the FLAC format, visit:
+ * http://flac.sourceforge.net/
+ *
+ * This decoder can be used in 1 of 2 ways: Either raw FLAC data can be fed
+ * through, starting from the initial 'fLaC' signature; or by passing the
+ * 34-byte streaminfo structure through avctx->extradata[_size] followed
+ * by data starting with the 0xFFF8 marker.
*/
-
+
#include <limits.h>
-
+
#include "avcodec.h"
+#include "bitstream.h"
#include "golomb.h"
#undef NDEBUG
#define MAX_CHANNELS 8
#define MAX_BLOCKSIZE 65535
+#define FLAC_STREAMINFO_SIZE 34
enum decorrelation_type {
INDEPENDENT,
uint8_t *bitstream;
int bitstream_size;
int bitstream_index;
- int allocated_bitstream_size;
+ unsigned int allocated_bitstream_size;
} FLACContext;
#define METADATA_TYPE_STREAMINFO 0
static int sample_rate_table[] =
{ 0, 0, 0, 0,
8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000,
- 0, 0, 0, 0 };
+ 0, 0, 0, 0 };
-static int sample_size_table[] =
+static int sample_size_table[] =
{ 0, 8, 12, 0, 16, 20, 24, 0 };
static int blocksize_table[] = {
- 0, 192, 576<<0, 576<<1, 576<<2, 576<<3, 0, 0,
-256<<0, 256<<1, 256<<2, 256<<3, 256<<4, 256<<5, 256<<6, 256<<7
+ 0, 192, 576<<0, 576<<1, 576<<2, 576<<3, 0, 0,
+256<<0, 256<<1, 256<<2, 256<<3, 256<<4, 256<<5, 256<<6, 256<<7
};
static const uint8_t table_crc8[256] = {
{
uint64_t val;
int ones=0, bytes;
-
+
while(get_bits1(gb))
ones++;
if (ones==0) bytes=0;
else if(ones==1) return -1;
else bytes= ones - 1;
-
+
val= get_bits(gb, 7-ones);
while(bytes--){
const int tmp = get_bits(gb, 8);
-
+
if((tmp>>6) != 2)
return -1;
val<<=6;
return val;
}
+#if 0
+static int skip_utf8(GetBitContext *gb)
+{
+ int ones=0, bytes;
+
+ while(get_bits1(gb))
+ ones++;
+
+ if (ones==0) bytes=0;
+ else if(ones==1) return -1;
+ else bytes= ones - 1;
+
+ skip_bits(gb, 7-ones);
+ while(bytes--){
+ const int tmp = get_bits(gb, 8);
+
+ if((tmp>>6) != 2)
+ return -1;
+ }
+ return 0;
+}
+#endif
+
static int get_crc8(const uint8_t *buf, int count){
int crc=0;
int i;
-
+
for(i=0; i<count; i++){
crc = table_crc8[crc ^ buf[i]];
}
return crc;
}
+static void metadata_streaminfo(FLACContext *s);
+static void dump_headers(FLACContext *s);
+
static int flac_decode_init(AVCodecContext * avctx)
{
+ FLACContext *s = avctx->priv_data;
+ s->avctx = avctx;
+
+ /* initialize based on the demuxer-supplied streamdata header */
+ if (avctx->extradata_size == FLAC_STREAMINFO_SIZE) {
+ init_get_bits(&s->gb, avctx->extradata, avctx->extradata_size*8);
+ metadata_streaminfo(s);
+ dump_headers(s);
+ }
+
return 0;
}
s->min_framesize = get_bits_long(&s->gb, 24);
s->max_framesize = get_bits_long(&s->gb, 24);
-
+
s->samplerate = get_bits_long(&s->gb, 20);
s->channels = get_bits(&s->gb, 3) + 1;
s->bps = get_bits(&s->gb, 5) + 1;
-
+
s->avctx->channels = s->channels;
s->avctx->sample_rate = s->samplerate;
skip_bits(&s->gb, 36); /* total num of samples */
-
+
skip_bits(&s->gb, 64); /* md5 sum */
skip_bits(&s->gb, 64); /* md5 sum */
-
+
allocate_buffers(s);
}
av_log(s->avctx, AV_LOG_DEBUG, "illegal residual coding method %d\n", method_type);
return -1;
}
-
+
rice_order = get_bits(&s->gb, 4);
samples= s->blocksize >> rice_order;
- sample=
+ sample=
i= pred_order;
for (partition = 0; partition < (1 << rice_order); partition++)
{
// av_log(s->avctx, AV_LOG_DEBUG, "partitions: %d, samples: %d\n", 1 << rice_order, sample);
return 0;
-}
+}
static int decode_subframe_fixed(FLACContext *s, int channel, int pred_order)
{
int i;
-
+
// av_log(s->avctx, AV_LOG_DEBUG, " SUBFRAME FIXED\n");
-
+
/* warm up samples */
// av_log(s->avctx, AV_LOG_DEBUG, " warm up samples: %d\n", pred_order);
-
+
for (i = 0; i < pred_order; i++)
{
s->decoded[channel][i] = get_sbits(&s->gb, s->curr_bps);
// av_log(s->avctx, AV_LOG_DEBUG, " %d: %d\n", i, s->decoded[channel][i]);
}
-
+
if (decode_residuals(s, channel, pred_order) < 0)
return -1;
break;
case 3:
for (i = pred_order; i < s->blocksize; i++)
- s->decoded[channel][i] += 3*s->decoded[channel][i-1]
+ s->decoded[channel][i] += 3*s->decoded[channel][i-1]
- 3*s->decoded[channel][i-2]
+ s->decoded[channel][i-3];
break;
case 4:
for (i = pred_order; i < s->blocksize; i++)
- s->decoded[channel][i] += 4*s->decoded[channel][i-1]
+ s->decoded[channel][i] += 4*s->decoded[channel][i-1]
- 6*s->decoded[channel][i-2]
+ 4*s->decoded[channel][i-3]
- s->decoded[channel][i-4];
int sum, i, j;
int coeff_prec, qlevel;
int coeffs[pred_order];
-
+
// av_log(s->avctx, AV_LOG_DEBUG, " SUBFRAME LPC\n");
-
+
/* warm up samples */
// av_log(s->avctx, AV_LOG_DEBUG, " warm up samples: %d\n", pred_order);
-
+
for (i = 0; i < pred_order; i++)
{
s->decoded[channel][i] = get_sbits(&s->gb, s->curr_bps);
// av_log(s->avctx, AV_LOG_DEBUG, " %d: %d\n", i, s->decoded[channel][i]);
}
-
+
coeff_prec = get_bits(&s->gb, 4) + 1;
if (coeff_prec == 16)
{
coeffs[i] = get_sbits(&s->gb, coeff_prec);
// av_log(s->avctx, AV_LOG_DEBUG, " %d: %d\n", i, coeffs[i]);
}
-
+
if (decode_residuals(s, channel, pred_order) < 0)
return -1;
sum += coeffs[j] * s->decoded[channel][i-j-1];
s->decoded[channel][i] += sum >> qlevel;
}
-
+
return 0;
}
{
int type, wasted = 0;
int i, tmp;
-
+
s->curr_bps = s->bps;
if(channel == 0){
if(s->decorrelation == RIGHT_SIDE)
if (get_bits1(&s->gb))
{
- av_log(s->avctx, AV_LOG_DEBUG, "invalid subframe padding\n");
+ av_log(s->avctx, AV_LOG_ERROR, "invalid subframe padding\n");
return -1;
}
type = get_bits(&s->gb, 6);
// wasted = get_bits1(&s->gb);
-
+
// if (wasted)
// {
// while (!get_bits1(&s->gb))
}
else
{
- av_log(s->avctx, AV_LOG_DEBUG, "invalid coding type\n");
+ av_log(s->avctx, AV_LOG_ERROR, "invalid coding type\n");
return -1;
}
-
+
if (wasted)
{
int i;
{
int blocksize_code, sample_rate_code, sample_size_code, assignment, i, crc8;
int decorrelation, bps, blocksize, samplerate;
-
+
blocksize_code = get_bits(&s->gb, 4);
sample_rate_code = get_bits(&s->gb, 4);
-
+
assignment = get_bits(&s->gb, 4); /* channel assignment */
if (assignment < 8 && s->channels == assignment+1)
decorrelation = INDEPENDENT;
decorrelation = LEFT_SIDE + assignment - 8;
else
{
- av_log(s->avctx, AV_LOG_DEBUG, "unsupported channel assignment %d (channels=%d)\n", assignment, s->channels);
+ av_log(s->avctx, AV_LOG_ERROR, "unsupported channel assignment %d (channels=%d)\n", assignment, s->channels);
return -1;
}
-
+
sample_size_code = get_bits(&s->gb, 3);
if(sample_size_code == 0)
bps= s->bps;
else if((sample_size_code != 3) && (sample_size_code != 7))
bps = sample_size_table[sample_size_code];
- else
+ else
{
- av_log(s->avctx, AV_LOG_DEBUG, "invalid sample size code (%d)\n", sample_size_code);
+ av_log(s->avctx, AV_LOG_ERROR, "invalid sample size code (%d)\n", sample_size_code);
return -1;
}
if (get_bits1(&s->gb))
{
- av_log(s->avctx, AV_LOG_DEBUG, "broken stream, invalid padding\n");
+ av_log(s->avctx, AV_LOG_ERROR, "broken stream, invalid padding\n");
return -1;
}
-
+
if(get_utf8(&s->gb) < 0){
av_log(s->avctx, AV_LOG_ERROR, "utf8 fscked\n");
return -1;
}
-#if 0
+#if 0
if (/*((blocksize_code == 6) || (blocksize_code == 7)) &&*/
(s->min_blocksize != s->max_blocksize)){
}else{
}
#endif
-
+
if (blocksize_code == 0)
blocksize = s->min_blocksize;
else if (blocksize_code == 6)
blocksize = get_bits(&s->gb, 8)+1;
else if (blocksize_code == 7)
blocksize = get_bits(&s->gb, 16)+1;
- else
+ else
blocksize = blocksize_table[blocksize_code];
if(blocksize > s->max_blocksize){
skip_bits(&s->gb, 8);
crc8= get_crc8(s->gb.buffer, get_bits_count(&s->gb)/8);
if(crc8){
- av_log(s->avctx, AV_LOG_ERROR, "header crc missmatch crc=%2X\n", crc8);
+ av_log(s->avctx, AV_LOG_ERROR, "header crc mismatch crc=%2X\n", crc8);
return -1;
}
-
+
s->blocksize = blocksize;
s->samplerate = samplerate;
s->bps = bps;
if (decode_subframe(s, i) < 0)
return -1;
}
-
+
align_get_bits(&s->gb);
/* frame footer */
{
FLACContext *s = avctx->priv_data;
int metadata_last, metadata_type, metadata_size;
- int tmp = 0, i, j = 0, input_buf_size;
- int16_t *samples = data, *left, *right;
+ int tmp = 0, i, j = 0, input_buf_size = 0;
+ int16_t *samples = data;
- s->avctx = avctx;
-
if(s->max_framesize == 0){
- s->max_framesize= 8192; // should hopefully be enough for the first header
+ s->max_framesize= 65536; // should hopefully be enough for the first header
s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->max_framesize);
}
if(1 && s->max_framesize){//FIXME truncated
- buf_size= FFMIN(buf_size, s->max_framesize - s->bitstream_size);
+ buf_size= FFMAX(FFMIN(buf_size, s->max_framesize - s->bitstream_size), 0);
input_buf_size= buf_size;
if(s->bitstream_index + s->bitstream_size + buf_size > s->allocated_bitstream_size){
buf= &s->bitstream[s->bitstream_index];
buf_size += s->bitstream_size;
s->bitstream_size= buf_size;
-
+
if(buf_size < s->max_framesize){
// printf("wanna more data ...\n");
return input_buf_size;
}
init_get_bits(&s->gb, buf, buf_size*8);
-
+
/* fLaC signature (be) */
if (show_bits_long(&s->gb, 32) == bswap_32(ff_get_fourcc("fLaC")))
{
metadata_last = get_bits(&s->gb, 1);
metadata_type = get_bits(&s->gb, 7);
metadata_size = get_bits_long(&s->gb, 24);
-
+
av_log(s->avctx, AV_LOG_DEBUG, " metadata block: flag = %d, type = %d, size = %d\n",
metadata_last, metadata_type,
metadata_size);
if(metadata_size){
switch(metadata_type)
{
- case METADATA_TYPE_STREAMINFO:
+ case METADATA_TYPE_STREAMINFO:{
metadata_streaminfo(s);
+
+ /* Buffer might have been reallocated, reinit bitreader */
+ if(buf != &s->bitstream[s->bitstream_index])
+ {
+ int bits_count = get_bits_count(&s->gb);
+ buf= &s->bitstream[s->bitstream_index];
+ init_get_bits(&s->gb, buf, buf_size*8);
+ skip_bits(&s->gb, bits_count);
+ }
+
dump_headers(s);
- break;
+ break;}
default:
for(i=0; i<metadata_size; i++)
skip_bits(&s->gb, 8);
}
else
{
-
+
tmp = show_bits(&s->gb, 16);
if(tmp != 0xFFF8){
av_log(s->avctx, AV_LOG_ERROR, "FRAME HEADER not here\n");
}
}
-
+
#if 0
/* fix the channel order here */
if (s->order == MID_SIDE)
*(samples++) = mid + side;
*(samples++) = mid;
#else
-
+
mid <<= 1;
if (side & 1)
mid++;
s->bitstream_index += i;
s->bitstream_size -= i;
return input_buf_size;
- }else
+ }else
return i;
}
{
FLACContext *s = avctx->priv_data;
int i;
-
+
for (i = 0; i < s->channels; i++)
{
av_freep(&s->decoded[i]);
}
av_freep(&s->bitstream);
-
+
return 0;
}
NULL,
flac_decode_close,
flac_decode_frame,
- .flush= flac_flush,
+ .flush= flac_flush,
};
projects / ffmpeg / blobdiff