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Merged in first part of ffmpeg accuracy improvments. No change on my samples, but brings us closer to ffmpeg's source and clears the way for additional bug fixes.

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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@13804 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Michael Giacomelli 2007-07-06 02:44:58 +00:00
parent e0473717e5
commit 2f1da8d24a
2 changed files with 18 additions and 38 deletions
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1
apps/codecs/libwma/wmadec.h
View file

@ -124,6 +124,7 @@ typedef struct WMADecodeContext
int block_pos; /* current position in frame */
uint8_t ms_stereo; /* true if mid/side stereo mode */
uint8_t channel_coded[MAX_CHANNELS]; /* true if channel is coded */
int exponents_bsize[MAX_CHANNELS]; // log2 ratio frame/exp. length
fixed32 exponents[MAX_CHANNELS][BLOCK_MAX_SIZE];
fixed32 max_exponent[MAX_CHANNELS];
int16_t coefs1[MAX_CHANNELS][BLOCK_MAX_SIZE];

51
apps/codecs/libwma/wmadeci.c
View file

@ -1192,6 +1192,7 @@ int wma_decode_init(WMADecodeContext* s, asf_waveformatex_t *wfx)
return 0;
}
#if 0
/* interpolate values for a bigger or smaller block. The block must
have multiple sizes */
static void interpolate_array(fixed32 *scale, int old_size, int new_size)
@ -1227,7 +1228,7 @@ static void interpolate_array(fixed32 *scale, int old_size, int new_size)
}
}
}
#endif
/* compute x^-0.25 with an exponent and mantissa table. We use linear
interpolation to reduce the mantissa table size at a small speed
expense (linear interpolation approximately doubles the number of
@ -1397,7 +1398,7 @@ static int decode_exp_vlc(WMADecodeContext *s, int ch)
static int wma_decode_block(WMADecodeContext *s)
{
int n, v, a, ch, code, bsize;
int coef_nb_bits, total_gain, parse_exponents;
int coef_nb_bits, total_gain;
//static fixed32 window[BLOCK_MAX_SIZE * 2]; //crap can't do this locally on the device! its big as the whole stack
int nb_coefs[MAX_CHANNELS];
fixed32 mdct_norm;
@ -1564,15 +1565,8 @@ static int wma_decode_block(WMADecodeContext *s)
}
}
/* exposant can be interpolated in short blocks. */
parse_exponents = 1;
if (s->block_len_bits != s->frame_len_bits)
{
parse_exponents = get_bits(&s->gb, 1);
}
if (parse_exponents)
{
/* exponents can be reused in short blocks. */
if ((s->block_len_bits == s->frame_len_bits) || get_bits(&s->gb, 1)) {
for(ch = 0; ch < s->nb_channels; ++ch)
{
@ -1589,24 +1583,11 @@ static int wma_decode_block(WMADecodeContext *s)
{
decode_exp_lsp(s, ch);
}
s->exponents_bsize[ch] = bsize;
}
}
}
else
{
for(ch = 0; ch < s->nb_channels; ++ch)
{
if (s->channel_coded[ch])
{
interpolate_array(s->exponents[ch],
1 << s->prev_block_len_bits,
s->block_len);
}
}
}
//ok up to here!
//printf("got here!\n");
//rb->splash(HZ, "in wma_decode_block 2");
/* parse spectral coefficients : just RLE encoding */
for(ch = 0; ch < s->nb_channels; ++ch)
{
@ -1693,8 +1674,6 @@ static int wma_decode_block(WMADecodeContext *s)
}
//rb->splash(HZ, "in wma_decode_block 3");
/* finally compute the MDCT coefficients */
for(ch = 0; ch < s->nb_channels; ++ch)
{
@ -1706,15 +1685,14 @@ static int wma_decode_block(WMADecodeContext *s)
fixed64 mult;
fixed64 mult1;
fixed32 noise;
int i, j, n, n1, last_high_band;
int i, j, n, n1, last_high_band, esize;
fixed32 exp_power[HIGH_BAND_MAX_SIZE];
//double test, mul;
//total_gain, coefs1, mdctnorm are lossless
coefs1 = s->coefs1[ch];
exponents = s->exponents[ch];
esize = s->exponents_bsize[ch];
mult = fixdiv64(pow_table[total_gain],Fixed32To64(s->max_exponent[ch]));
// mul = fixtof64(pow_table[total_gain])/(s->block_len/2)/fixtof64(s->max_exponent[ch]);
@ -1829,13 +1807,14 @@ static int wma_decode_block(WMADecodeContext *s)
atemp = (fixed32)(coefs1[i]*mult>>16);
//atemp= ftofix32(coefs1[i] * fixtof64(exponents[i]) * fixtof64(mult>>16)); //this "works" in the sense that the mdcts converge
*coefs++=fixmul32(atemp,exponents[i]); //this does not work
//this can still overflow in rare cases
//running a full scale value square wave through here does bad things
*coefs++=fixmul32(atemp,exponents[i<<bsize>>esize]);
//atemp = ftofix32( coefs1[i]*mul* fixtof64(exponents[i]) ); //this doesn't seem to help any at all.
// *coefs++=atemp;
} //coefs1 could underflow?
}
n = s->block_len - s->coefs_end[bsize];
for(i = 0;i < n; ++i)
*coefs++ = 0;