flacdec: add support for SAMPLE_FMT_32
Originally committed as revision 16871 to svn://svn.ffmpeg.org/ffmpeg/trunk
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@ -63,6 +63,8 @@ typedef struct FLACContext {
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int blocksize/*, last_blocksize*/;
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int curr_bps;
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int sample_shift; /* shift required to make output samples 16-bit or 32-bit */
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int is32; /* flag to indicate if output should be 32-bit instead of 16-bit */
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enum decorrelation_type decorrelation;
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int32_t *decoded[MAX_CHANNELS];
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@ -168,6 +170,11 @@ void ff_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s,
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avctx->channels = s->channels;
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avctx->sample_rate = s->samplerate;
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avctx->bits_per_raw_sample = s->bps;
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if (s->bps > 16)
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avctx->sample_fmt = SAMPLE_FMT_S32;
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else
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avctx->sample_fmt = SAMPLE_FMT_S16;
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s->samples = get_bits_long(&gb, 32) << 4;
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s->samples |= get_bits_long(&gb, 4);
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@ -466,6 +473,16 @@ static int decode_frame(FLACContext *s, int alloc_data_size)
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sample_size_code);
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return -1;
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}
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if (bps > 16) {
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s->avctx->sample_fmt = SAMPLE_FMT_S32;
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s->sample_shift = 32 - bps;
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s->is32 = 1;
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} else {
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s->avctx->sample_fmt = SAMPLE_FMT_S16;
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s->sample_shift = 16 - bps;
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s->is32 = 0;
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}
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s->bps = s->avctx->bits_per_raw_sample = bps;
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if (get_bits1(&s->gb)) {
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av_log(s->avctx, AV_LOG_ERROR, "broken stream, invalid padding\n");
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@ -546,7 +563,8 @@ static int flac_decode_frame(AVCodecContext *avctx,
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{
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FLACContext *s = avctx->priv_data;
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int tmp = 0, i, j = 0, input_buf_size = 0;
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int16_t *samples = data;
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int16_t *samples_16 = data;
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int32_t *samples_32 = data;
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int alloc_data_size= *data_size;
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*data_size=0;
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@ -608,16 +626,25 @@ static int flac_decode_frame(AVCodecContext *avctx,
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for (i = 0; i < s->blocksize; i++) {\
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int a= s->decoded[0][i];\
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int b= s->decoded[1][i];\
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*samples++ = ((left) << (24 - s->bps)) >> 8;\
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*samples++ = ((right) << (24 - s->bps)) >> 8;\
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if (s->is32) {\
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*samples_32++ = (left) << s->sample_shift;\
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*samples_32++ = (right) << s->sample_shift;\
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} else {\
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*samples_16++ = (left) << s->sample_shift;\
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*samples_16++ = (right) << s->sample_shift;\
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}\
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}\
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break;
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switch (s->decorrelation) {
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case INDEPENDENT:
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for (j = 0; j < s->blocksize; j++) {
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for (i = 0; i < s->channels; i++)
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*samples++ = (s->decoded[i][j] << (24 - s->bps)) >> 8;
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for (i = 0; i < s->channels; i++) {
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if (s->is32)
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*samples_32++ = s->decoded[i][j] << s->sample_shift;
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else
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*samples_16++ = s->decoded[i][j] << s->sample_shift;
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}
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}
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break;
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case LEFT_SIDE:
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@ -628,7 +655,7 @@ static int flac_decode_frame(AVCodecContext *avctx,
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DECORRELATE( (a-=b>>1) + b, a)
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}
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*data_size = (int8_t *)samples - (int8_t *)data;
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*data_size = s->blocksize * s->channels * (s->is32 ? 4 : 2);
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end:
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i= (get_bits_count(&s->gb)+7)/8;
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