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Commit 992f7db0 authored by Laurent Aimar's avatar Laurent Aimar Committed by Kostya Shishkov
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Add floating point audio decoding to WavPack decoder. 

Patch by Laurent Aimar (fenrir at `antonym of 'audio'+antonym of 'WAN'` dot org)

Originally committed as revision 18754 to svn://svn.ffmpeg.org/ffmpeg/trunk
parent cabb8111
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libavcodec/wavpack.c
+ 131
− 8
View file @ 992f7db0
......@@ -37,6 +37,12 @@
#define WV_HYBRID_BITRATE 0x00000200
#define WV_HYBRID_BALANCE 0x00000400
#define WV_FLT_SHIFT_ONES 0x01
#define WV_FLT_SHIFT_SAME 0x02
#define WV_FLT_SHIFT_SENT 0x04
#define WV_FLT_ZERO_SENT 0x08
#define WV_FLT_ZERO_SIGN 0x10
enum WP_ID_Flags{
WP_IDF_MASK = 0x1F,
WP_IDF_IGNORE = 0x20,
......@@ -97,6 +103,9 @@ typedef struct WavpackContext {
int and, or, shift;
int post_shift;
int hybrid, hybrid_bitrate;
int float_flag;
int float_shift;
int float_max_exp;
WvChannel ch[2];
} WavpackContext;
......@@ -357,6 +366,79 @@ static inline int wv_get_value_integer(WavpackContext *s, uint32_t *crc, int S)
return (((S + bit) << s->shift) - bit) << s->post_shift;
}
static float wv_get_value_float(WavpackContext *s, uint32_t *crc, int S)
{
union {
float f;
uint32_t u;
} value;
int sign;
int exp = s->float_max_exp;
if(s->got_extra_bits){
const int max_bits = 1 + 23 + 8 + 1;
const int left_bits = s->gb_extra_bits.size_in_bits - get_bits_count(&s->gb_extra_bits);
if(left_bits + 8 * FF_INPUT_BUFFER_PADDING_SIZE < max_bits)
return 0.0;
}
if(S){
S <<= s->float_shift;
sign = S < 0;
if(sign)
S = -S;
if(S >= 0x1000000){
if(s->got_extra_bits && get_bits1(&s->gb_extra_bits)){
S = get_bits(&s->gb_extra_bits, 23);
}else{
S = 0;
}
exp = 255;
}else if(exp){
int shift = 23 - av_log2(S);
exp = s->float_max_exp;
if(exp <= shift){
shift = --exp;
}
exp -= shift;
if(shift){
S <<= shift;
if((s->float_flag & WV_FLT_SHIFT_ONES) ||
(s->got_extra_bits && (s->float_flag & WV_FLT_SHIFT_SAME) && get_bits1(&s->gb_extra_bits)) ){
S |= (1 << shift) - 1;
} else if(s->got_extra_bits && (s->float_flag & WV_FLT_SHIFT_SENT)){
S |= get_bits(&s->gb_extra_bits, shift);
}
}
}else{
exp = s->float_max_exp;
}
S &= 0x7fffff;
}else{
sign = 0;
exp = 0;
if(s->got_extra_bits && (s->float_flag & WV_FLT_ZERO_SENT)){
if(get_bits1(&s->gb_extra_bits)){
S = get_bits(&s->gb_extra_bits, 23);
if(s->float_max_exp >= 25)
exp = get_bits(&s->gb_extra_bits, 8);
sign = get_bits1(&s->gb_extra_bits);
}else{
if(s->float_flag & WV_FLT_ZERO_SIGN)
sign = get_bits1(&s->gb_extra_bits);
}
}
}
*crc = *crc * 27 + S * 9 + exp * 3 + sign;
value.u = (sign << 31) | (exp << 23) | S;
return value.f;
}
static inline int wv_unpack_stereo(WavpackContext *s, GetBitContext *gb, void *dst, const int type)
{
int i, j, count = 0;
......@@ -367,6 +449,7 @@ static inline int wv_unpack_stereo(WavpackContext *s, GetBitContext *gb, void *d
uint32_t crc_extra_bits = 0xFFFFFFFF;
int16_t *dst16 = dst;
int32_t *dst32 = dst;
float *dstfl = dst;
s->one = s->zero = s->zeroes = 0;
do{
......@@ -445,7 +528,10 @@ static inline int wv_unpack_stereo(WavpackContext *s, GetBitContext *gb, void *d
L += (R -= (L >> 1));
crc = (crc * 3 + L) * 3 + R;
if(type == SAMPLE_FMT_S32){
if(type == SAMPLE_FMT_FLT){
*dstfl++ = wv_get_value_float(s, &crc_extra_bits, L);
*dstfl++ = wv_get_value_float(s, &crc_extra_bits, R);
} else if(type == SAMPLE_FMT_S32){
*dst32++ = wv_get_value_integer(s, &crc_extra_bits, L);
*dst32++ = wv_get_value_integer(s, &crc_extra_bits, R);
} else {
......@@ -476,6 +562,7 @@ static inline int wv_unpack_mono(WavpackContext *s, GetBitContext *gb, void *dst
uint32_t crc_extra_bits = 0xFFFFFFFF;
int16_t *dst16 = dst;
int32_t *dst32 = dst;
float *dstfl = dst;
s->one = s->zero = s->zeroes = 0;
do{
......@@ -505,7 +592,9 @@ static inline int wv_unpack_mono(WavpackContext *s, GetBitContext *gb, void *dst
pos = (pos + 1) & 7;
crc = crc * 3 + S;
if(type == SAMPLE_FMT_S32)
if(type == SAMPLE_FMT_FLT)
*dstfl++ = wv_get_value_float(s, &crc_extra_bits, S);
else if(type == SAMPLE_FMT_S32)
*dst32++ = wv_get_value_integer(s, &crc_extra_bits, S);
else
*dst16++ = wv_get_value_integer(s, &crc_extra_bits, S);
......@@ -547,7 +636,7 @@ static int wavpack_decode_frame(AVCodecContext *avctx,
WavpackContext *s = avctx->priv_data;
void *samples = data;
int samplecount;
int got_terms = 0, got_weights = 0, got_samples = 0, got_entropy = 0, got_bs = 0;
int got_terms = 0, got_weights = 0, got_samples = 0, got_entropy = 0, got_bs = 0, got_float = 0;
int got_hybrid = 0;
const uint8_t* buf_end = buf + buf_size;
int i, j, id, size, ssize, weights, t;
......@@ -570,7 +659,10 @@ static int wavpack_decode_frame(AVCodecContext *avctx,
return buf_size;
}
s->frame_flags = AV_RL32(buf); buf += 4;
if((s->frame_flags&0x03) <= 1){
if(s->frame_flags&0x80){
bpp = sizeof(float);
avctx->sample_fmt = SAMPLE_FMT_FLT;
} else if((s->frame_flags&0x03) <= 1){
bpp = 2;
avctx->sample_fmt = SAMPLE_FMT_S16;
} else {
......@@ -742,6 +834,18 @@ static int wavpack_decode_frame(AVCodecContext *avctx,
}
buf += 4;
break;
case WP_ID_FLOATINFO:
if(size != 4){
av_log(avctx, AV_LOG_ERROR, "Invalid FLOATINFO, size = %i\n", size);
buf += ssize;
continue;
}
s->float_flag = buf[0];
s->float_shift = buf[1];
s->float_max_exp = buf[2];
buf += 4;
got_float = 1;
break;
case WP_ID_DATA:
init_get_bits(&s->gb, buf, size * 8);
s->data_size = size * 8;
......@@ -788,7 +892,11 @@ static int wavpack_decode_frame(AVCodecContext *avctx,
av_log(avctx, AV_LOG_ERROR, "Packed samples not found\n");
return -1;
}
if(s->got_extra_bits){
if(!got_float && avctx->sample_fmt == SAMPLE_FMT_FLT){
av_log(avctx, AV_LOG_ERROR, "Float information not found\n");
return -1;
}
if(s->got_extra_bits && avctx->sample_fmt != SAMPLE_FMT_FLT){
const int size = s->gb_extra_bits.size_in_bits - get_bits_count(&s->gb_extra_bits);
const int wanted = s->samples * s->extra_bits << s->stereo_in;
if(size < wanted){
......@@ -800,13 +908,19 @@ static int wavpack_decode_frame(AVCodecContext *avctx,
if(s->stereo_in){
if(avctx->sample_fmt == SAMPLE_FMT_S16)
samplecount = wv_unpack_stereo(s, &s->gb, samples, SAMPLE_FMT_S16);
else
else if(avctx->sample_fmt == SAMPLE_FMT_S32)
samplecount = wv_unpack_stereo(s, &s->gb, samples, SAMPLE_FMT_S32);
else
samplecount = wv_unpack_stereo(s, &s->gb, samples, SAMPLE_FMT_FLT);
}else{
if(avctx->sample_fmt == SAMPLE_FMT_S16)
samplecount = wv_unpack_mono(s, &s->gb, samples, SAMPLE_FMT_S16);
else
else if(avctx->sample_fmt == SAMPLE_FMT_S32)
samplecount = wv_unpack_mono(s, &s->gb, samples, SAMPLE_FMT_S32);
else
samplecount = wv_unpack_mono(s, &s->gb, samples, SAMPLE_FMT_FLT);
if(s->stereo && avctx->sample_fmt == SAMPLE_FMT_S16){
int16_t *dst = (int16_t*)samples + samplecount * 2;
int16_t *src = (int16_t*)samples + samplecount;
......@@ -816,7 +930,7 @@ static int wavpack_decode_frame(AVCodecContext *avctx,
*--dst = *src;
}
samplecount *= 2;
}else if(s->stereo){ //32-bit output
}else if(s->stereo && avctx->sample_fmt == SAMPLE_FMT_S32){
int32_t *dst = (int32_t*)samples + samplecount * 2;
int32_t *src = (int32_t*)samples + samplecount;
int cnt = samplecount;
......@@ -825,6 +939,15 @@ static int wavpack_decode_frame(AVCodecContext *avctx,
*--dst = *src;
}
samplecount *= 2;
}else if(s->stereo){
float *dst = (float*)samples + samplecount * 2;
float *src = (float*)samples + samplecount;
int cnt = samplecount;
while(cnt--){
*--dst = *--src;
*--dst = *src;
}
samplecount *= 2;
}
}
*data_size = samplecount * bpp;
......
libavformat/wv.c
+ 0
− 5
View file @ 992f7db0
......@@ -96,11 +96,6 @@ static int wv_read_block_header(AVFormatContext *ctx, ByteIOContext *pb)
get_buffer(pb, wc->extra, WV_EXTRA_SIZE);
wc->flags = AV_RL32(wc->extra + 4);
//parse flags
if(wc->flags & WV_FLOAT){
av_log(ctx, AV_LOG_ERROR, "Floating point data is not supported\n");
return -1;
}
bpp = ((wc->flags & 3) + 1) << 3;
chan = 1 + !(wc->flags & WV_MONO);
rate = wv_rates[(wc->flags >> 23) & 0xF];
......
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