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author | Justin Ruggles <justin.ruggles@gmail.com> | 2009-01-31 01:20:40 +0000 |
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committer | Justin Ruggles <justin.ruggles@gmail.com> | 2009-01-31 01:20:40 +0000 |
commit | 82159ad992929a4bed36cafd1901cf38abd86dc7 (patch) | |
tree | b8cf6672f1e8a14571a988ebe7cc35ec8cc49a96 | |
parent | 99b3812265a718a64c0eb6ac0dc2f6ae0afb06e5 (diff) | |
download | ffmpeg-82159ad992929a4bed36cafd1901cf38abd86dc7.tar.gz |
flacdec: add support for SAMPLE_FMT_32
Originally committed as revision 16871 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r-- | libavcodec/flacdec.c | 39 |
1 files changed, 33 insertions, 6 deletions
diff --git a/libavcodec/flacdec.c b/libavcodec/flacdec.c index ee3a3f2ca5..82456dc494 100644 --- a/libavcodec/flacdec.c +++ b/libavcodec/flacdec.c @@ -63,6 +63,8 @@ typedef struct FLACContext { int blocksize/*, last_blocksize*/; 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; int32_t *decoded[MAX_CHANNELS]; @@ -168,6 +170,11 @@ void ff_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s, avctx->channels = s->channels; 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, 4); @@ -466,6 +473,16 @@ static int decode_frame(FLACContext *s, int alloc_data_size) sample_size_code); 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)) { 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; 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; *data_size=0; @@ -608,16 +626,25 @@ static int flac_decode_frame(AVCodecContext *avctx, for (i = 0; i < s->blocksize; i++) {\ int a= s->decoded[0][i];\ int b= s->decoded[1][i];\ - *samples++ = ((left) << (24 - s->bps)) >> 8;\ - *samples++ = ((right) << (24 - s->bps)) >> 8;\ + if (s->is32) {\ + *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; switch (s->decorrelation) { case INDEPENDENT: for (j = 0; j < s->blocksize; j++) { - for (i = 0; i < s->channels; i++) - *samples++ = (s->decoded[i][j] << (24 - s->bps)) >> 8; + for (i = 0; i < s->channels; i++) { + if (s->is32) + *samples_32++ = s->decoded[i][j] << s->sample_shift; + else + *samples_16++ = s->decoded[i][j] << s->sample_shift; + } } break; case LEFT_SIDE: @@ -628,7 +655,7 @@ static int flac_decode_frame(AVCodecContext *avctx, 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: i= (get_bits_count(&s->gb)+7)/8; |