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flacdec: add support for SAMPLE_FMT_32

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Originally committed as revision 16871 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Justin Ruggles 2009-01-31 01:20:40 +00:00
parent 99b3812265
commit 82159ad992
1 changed files with 33 additions and 6 deletions
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39
libavcodec/flacdec.c
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@ -63,6 +63,8 @@ typedef struct FLACContext {
int blocksize/*, last_blocksize*/; int blocksize/*, last_blocksize*/;
int curr_bps; int curr_bps;
int sample_shift; /* shift required to make output samples 16-bit or 32-bit */
int is32; /* flag to indicate if output should be 32-bit instead of 16-bit */
enum decorrelation_type decorrelation; enum decorrelation_type decorrelation;
int32_t *decoded[MAX_CHANNELS]; int32_t *decoded[MAX_CHANNELS];
@ -168,6 +170,11 @@ void ff_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s,
avctx->channels = s->channels; avctx->channels = s->channels;
avctx->sample_rate = s->samplerate; avctx->sample_rate = s->samplerate;
avctx->bits_per_raw_sample = s->bps;
if (s->bps > 16)
avctx->sample_fmt = SAMPLE_FMT_S32;
else
avctx->sample_fmt = SAMPLE_FMT_S16;
s->samples = get_bits_long(&gb, 32) << 4; s->samples = get_bits_long(&gb, 32) << 4;
s->samples |= get_bits_long(&gb, 4); s->samples |= get_bits_long(&gb, 4);
@ -466,6 +473,16 @@ static int decode_frame(FLACContext *s, int alloc_data_size)
sample_size_code); sample_size_code);
return -1; return -1;
} }
if (bps > 16) {
s->avctx->sample_fmt = SAMPLE_FMT_S32;
s->sample_shift = 32 - bps;
s->is32 = 1;
} else {
s->avctx->sample_fmt = SAMPLE_FMT_S16;
s->sample_shift = 16 - bps;
s->is32 = 0;
}
s->bps = s->avctx->bits_per_raw_sample = bps;
if (get_bits1(&s->gb)) { if (get_bits1(&s->gb)) {
av_log(s->avctx, AV_LOG_ERROR, "broken stream, invalid padding\n"); av_log(s->avctx, AV_LOG_ERROR, "broken stream, invalid padding\n");
@ -546,7 +563,8 @@ static int flac_decode_frame(AVCodecContext *avctx,
{ {
FLACContext *s = avctx->priv_data; FLACContext *s = avctx->priv_data;
int tmp = 0, i, j = 0, input_buf_size = 0; int tmp = 0, i, j = 0, input_buf_size = 0;
int16_t *samples = data; int16_t *samples_16 = data;
int32_t *samples_32 = data;
int alloc_data_size= *data_size; int alloc_data_size= *data_size;
*data_size=0; *data_size=0;
@ -608,16 +626,25 @@ static int flac_decode_frame(AVCodecContext *avctx,
for (i = 0; i < s->blocksize; i++) {\ for (i = 0; i < s->blocksize; i++) {\
int a= s->decoded[0][i];\ int a= s->decoded[0][i];\
int b= s->decoded[1][i];\ int b= s->decoded[1][i];\
*samples++ = ((left) << (24 - s->bps)) >> 8;\ if (s->is32) {\
*samples++ = ((right) << (24 - s->bps)) >> 8;\ *samples_32++ = (left) << s->sample_shift;\
*samples_32++ = (right) << s->sample_shift;\
} else {\
*samples_16++ = (left) << s->sample_shift;\
*samples_16++ = (right) << s->sample_shift;\
}\
}\ }\
break; break;
switch (s->decorrelation) { switch (s->decorrelation) {
case INDEPENDENT: case INDEPENDENT:
for (j = 0; j < s->blocksize; j++) { for (j = 0; j < s->blocksize; j++) {
for (i = 0; i < s->channels; i++) for (i = 0; i < s->channels; i++) {
*samples++ = (s->decoded[i][j] << (24 - s->bps)) >> 8; if (s->is32)
*samples_32++ = s->decoded[i][j] << s->sample_shift;
else
*samples_16++ = s->decoded[i][j] << s->sample_shift;
}
} }
break; break;
case LEFT_SIDE: case LEFT_SIDE:
@ -628,7 +655,7 @@ static int flac_decode_frame(AVCodecContext *avctx,
DECORRELATE( (a-=b>>1) + b, a) DECORRELATE( (a-=b>>1) + b, a)
} }
*data_size = (int8_t *)samples - (int8_t *)data; *data_size = s->blocksize * s->channels * (s->is32 ? 4 : 2);
end: end:
i= (get_bits_count(&s->gb)+7)/8; i= (get_bits_count(&s->gb)+7)/8;