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authorjm <jm@0101bb08-14d6-0310-b084-bc0e0c8e3800>2007-02-12 17:12:59 +0300
committerjm <jm@0101bb08-14d6-0310-b084-bc0e0c8e3800>2007-02-12 17:12:59 +0300
commit0c397662f6044203da0aab06d34b0bf047f18156 (patch)
tree14c38bcdb2630ece139719152ebae32a7aff0dd0
parentbc657766a852db088c79a1a0ab4dddfb69969683 (diff)
window type changes with quality.
git-svn-id: http://svn.xiph.org/trunk/speex@12453 0101bb08-14d6-0310-b084-bc0e0c8e3800
-rw-r--r--libspeex/resample.c73
1 files changed, 38 insertions, 35 deletions
diff --git a/libspeex/resample.c b/libspeex/resample.c
index 03fbe85..e95186f 100644
--- a/libspeex/resample.c
+++ b/libspeex/resample.c
@@ -82,35 +82,6 @@ void speex_free (void *ptr) {free(ptr);}
#define IMAX(a,b) ((a) > (b) ? (a) : (b))
-struct QualityMapping {
- int base_length;
- int oversample;
- float downsample_bandwidth;
- float upsample_bandwidth;
-};
-
-/* This table maps conversion quality to internal parameters. There are two
- reasons that explain why the up-sampling bandwidth is larger than the
- down-sampling bandwidth:
- 1) When up-sampling, we can assume that the spectrum is already attenuated
- close to the Nyquist rate (from an A/D or a previous resampling filter)
- 2) Any aliasing that occurs very close to the Nyquist rate will be masked
- by the sinusoids/noise just below the Nyquist rate (guaranteed only for
- up-sampling).
-*/
-const struct QualityMapping quality_map[11] = {
- { 8, 4, 0.830f, 0.860f}, /* Q0 */ /* 63.5% cutoff ( ~20 dB stop) 4 */
- { 16, 4, 0.850f, 0.880f}, /* Q1 */ /* 75.3% cutoff ( ~40 dB stop) 4 */
- { 32, 4, 0.882f, 0.910f}, /* Q2 */ /* 82.3% cutoff ( ~60 dB stop) 6 */
- { 48, 8, 0.895f, 0.917f}, /* Q3 */ /* 84.9% cutoff ( ~80 dB stop) 8 */
- { 64, 8, 0.921f, 0.940f}, /* Q4 */ /* 88.7% cutoff ( ~80 dB stop) 8 */
- { 80, 8, 0.922f, 0.940f}, /* Q5 */ /* 89.1% cutoff (~100 dB stop) 10 */
- { 96, 8, 0.940f, 0.945f}, /* Q6 */ /* 91.5% cutoff (~100 dB stop) 10 */
- {128, 16, 0.950f, 0.950f}, /* Q7 */ /* 93.1% cutoff (~100 dB stop) 10 */
- {160, 16, 0.960f, 0.960f}, /* Q8 */ /* 94.5% cutoff (~100 dB stop) 10 */
- {192, 16, 0.968f, 0.968f}, /* Q9 */ /* 95.5% cutoff (~100 dB stop) 10 */
- {256, 16, 0.975f, 0.975f}, /* Q10 */ /* 96.6% cutoff (~100 dB stop) 10 */
-};
typedef enum {SPEEX_RESAMPLER_DIRECT_SINGLE=0, SPEEX_RESAMPLER_INTERPOLATE_SINGLE=1} SpeexSincType;
@@ -194,6 +165,38 @@ static struct FuncDef _KAISER6 = {kaiser6_table, 32};
static struct FuncDef _KAISER4 = {kaiser4_table, 32};
#define KAISER4 (&_KAISER4)
+struct QualityMapping {
+ int base_length;
+ int oversample;
+ float downsample_bandwidth;
+ float upsample_bandwidth;
+ struct FuncDef *window_func;
+};
+
+
+/* This table maps conversion quality to internal parameters. There are two
+ reasons that explain why the up-sampling bandwidth is larger than the
+ down-sampling bandwidth:
+ 1) When up-sampling, we can assume that the spectrum is already attenuated
+ close to the Nyquist rate (from an A/D or a previous resampling filter)
+ 2) Any aliasing that occurs very close to the Nyquist rate will be masked
+ by the sinusoids/noise just below the Nyquist rate (guaranteed only for
+ up-sampling).
+*/
+const struct QualityMapping quality_map[11] = {
+ { 8, 4, 0.830f, 0.860f, KAISER4 }, /* Q0 */ /* 63.5% cutoff ( ~20 dB stop) 4 */
+ { 16, 4, 0.850f, 0.880f, KAISER4 }, /* Q1 */ /* 75.3% cutoff ( ~40 dB stop) 4 */
+ { 32, 4, 0.882f, 0.910f, KAISER6 }, /* Q2 */ /* 82.3% cutoff ( ~60 dB stop) 6 */
+ { 48, 8, 0.895f, 0.917f, KAISER8 }, /* Q3 */ /* 84.9% cutoff ( ~80 dB stop) 8 */
+ { 64, 8, 0.921f, 0.940f, KAISER8 }, /* Q4 */ /* 88.7% cutoff ( ~80 dB stop) 8 */
+ { 80, 8, 0.922f, 0.940f, KAISER10}, /* Q5 */ /* 89.1% cutoff (~100 dB stop) 10 */
+ { 96, 8, 0.940f, 0.945f, KAISER10}, /* Q6 */ /* 91.5% cutoff (~100 dB stop) 10 */
+ {128, 16, 0.950f, 0.950f, KAISER10}, /* Q7 */ /* 93.1% cutoff (~100 dB stop) 10 */
+ {160, 16, 0.960f, 0.960f, KAISER10}, /* Q8 */ /* 94.5% cutoff (~100 dB stop) 10 */
+ {192, 16, 0.968f, 0.968f, KAISER10}, /* Q9 */ /* 95.5% cutoff (~100 dB stop) 10 */
+ {256, 16, 0.975f, 0.975f, KAISER10}, /* Q10 */ /* 96.6% cutoff (~100 dB stop) 10 */
+};
+
static double compute_func(float x, struct FuncDef *func)
{
float y, frac;
@@ -229,7 +232,7 @@ int main(int argc, char **argv)
#ifdef FIXED_POINT
/* The slow way of computing a sinc for the table. Should improve that some day */
-static spx_word16_t sinc(float cutoff, float x, int N)
+static spx_word16_t sinc(float cutoff, float x, int N, struct FuncDef *window_func)
{
/*fprintf (stderr, "%f ", x);*/
x *= cutoff;
@@ -238,11 +241,11 @@ static spx_word16_t sinc(float cutoff, float x, int N)
else if (fabs(x) > .5f*N)
return 0;
/*FIXME: Can it really be any slower than this? */
- return WORD2INT(32768.*cutoff*sin(M_PI*x)/(M_PI*x) * compute_func(fabs(2.*x/N), KAISER10));
+ return WORD2INT(32768.*cutoff*sin(M_PI*x)/(M_PI*x) * compute_func(fabs(2.*x/N), window_func));
}
#else
/* The slow way of computing a sinc for the table. Should improve that some day */
-static spx_word16_t sinc(float cutoff, float x, int N)
+static spx_word16_t sinc(float cutoff, float x, int N, struct FuncDef *window_func)
{
/*fprintf (stderr, "%f ", x);*/
x *= cutoff;
@@ -251,7 +254,7 @@ static spx_word16_t sinc(float cutoff, float x, int N)
else if (fabs(x) > .5*N)
return 0;
/*FIXME: Can it really be any slower than this? */
- return cutoff*sin(M_PI*x)/(M_PI*x) * compute_func(fabs(2.*x/N), KAISER10);
+ return cutoff*sin(M_PI*x)/(M_PI*x) * compute_func(fabs(2.*x/N), window_func);
}
#endif
@@ -407,7 +410,7 @@ static void update_filter(SpeexResamplerState *st)
int j;
for (j=0;j<st->filt_len;j++)
{
- st->sinc_table[i*st->filt_len+j] = sinc(st->cutoff,((j-st->filt_len/2+1)-((float)i)/st->den_rate), st->filt_len);
+ st->sinc_table[i*st->filt_len+j] = sinc(st->cutoff,((j-st->filt_len/2+1)-((float)i)/st->den_rate), st->filt_len, quality_map[st->quality].window_func);
}
}
st->type = SPEEX_RESAMPLER_DIRECT_SINGLE;
@@ -422,7 +425,7 @@ static void update_filter(SpeexResamplerState *st)
st->sinc_table_length = st->filt_len*st->oversample+8;
}
for (i=-4;i<st->oversample*st->filt_len+4;i++)
- st->sinc_table[i+4] = sinc(st->cutoff,(i/(float)st->oversample - st->filt_len/2), st->filt_len);
+ st->sinc_table[i+4] = sinc(st->cutoff,(i/(float)st->oversample - st->filt_len/2), st->filt_len, quality_map[st->quality].window_func);
st->type = SPEEX_RESAMPLER_INTERPOLATE_SINGLE;
st->resampler_ptr = resampler_basic_interpolate_single;
/*fprintf (stderr, "resampler uses interpolated sinc table and normalised cutoff %f\n", cutoff);*/