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update to snapshot spandsp-20080911.tar.gz

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git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@9767 d0543943-73ff-0310-b7d9-9358b9ac24b2
This commit is contained in:
Anthony Minessale
2008-10-01 03:52:36 +00:00
parent 61cd1f8219
commit 1d99b900fd
14 changed files with 2064 additions and 987 deletions
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39
libs/spandsp/src/tone_generate.c
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@@ -22,7 +22,7 @@
* License along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*
* $Id: tone_generate.c,v 1.43 2008/07/02 14:48:26 steveu Exp $
* $Id: tone_generate.c,v 1.46 2008/09/11 15:13:42 steveu Exp $
*/
/*! \file */
@@ -72,15 +72,27 @@ void make_tone_gen_descriptor(tone_gen_descriptor_t *s,
memset(s, 0, sizeof(*s));
if (f1)
{
#if defined(SPANDSP_USE_FIXED_POINT)
s->tone[0].phase_rate = dds_phase_rate((float) f1);
if (f2 < 0)
s->tone[0].phase_rate = -s->tone[0].phase_rate;
s->tone[0].gain = dds_scaling_dbm0((float) l1);
#else
s->tone[0].phase_rate = dds_phase_ratef((float) f1);
if (f2 < 0)
s->tone[0].phase_rate = -s->tone[0].phase_rate;
s->tone[0].gain = dds_scaling_dbm0f((float) l1);
#endif
}
if (f2)
{
#if defined(SPANDSP_USE_FIXED_POINT)
s->tone[1].phase_rate = dds_phase_rate((float) abs(f2));
s->tone[1].gain = (f2 < 0) ? (float) 32767.0f*l2/100.0f : dds_scaling_dbm0((float) l2);
#else
s->tone[1].phase_rate = dds_phase_ratef((float) abs(f2));
s->tone[1].gain = (f2 < 0) ? (float) l2/100.0f : dds_scaling_dbm0f((float) l2);
#endif
}
s->duration[0] = d1*SAMPLE_RATE/1000;
@@ -118,7 +130,11 @@ int tone_gen(tone_gen_state_t *s, int16_t amp[], int max_samples)
{
int samples;
int limit;
#if defined(SPANDSP_USE_FIXED_POINT)
int16_t xamp;
#else
float xamp;
#endif
int i;
if (s->current_section < 0)
@@ -141,30 +157,49 @@ int tone_gen(tone_gen_state_t *s, int16_t amp[], int max_samples)
{
if (s->tone[0].phase_rate < 0)
{
/* Modulated tone */
for ( ; samples < limit; samples++)
{
/* There must be two, and only two tones */
/* There must be two, and only two, tones */
#if defined(SPANDSP_USE_FIXED_POINT)
xamp = ((int32_t) dds_mod(&s->phase[0], -s->tone[0].phase_rate, s->tone[0].gain, 0)
*(32767 + (int32_t) dds_mod(&s->phase[1], s->tone[1].phase_rate, s->tone[1].gain, 0))) >> 15;
amp[samples] = xamp;
#else
xamp = dds_modf(&s->phase[0], -s->tone[0].phase_rate, s->tone[0].gain, 0)
*(1.0f + dds_modf(&s->phase[1], s->tone[1].phase_rate, s->tone[1].gain, 0));
amp[samples] = (int16_t) lrintf(xamp);
#endif
}
}
else
{
for ( ; samples < limit; samples++)
{
#if defined(SPANDSP_USE_FIXED_POINT)
xamp = 0;
#else
xamp = 0.0f;
#endif
for (i = 0; i < 4; i++)
{
if (s->tone[i].phase_rate == 0)
break;
#if defined(SPANDSP_USE_FIXED_POINT)
xamp += dds_mod(&s->phase[i], s->tone[i].phase_rate, s->tone[i].gain, 0);
#else
xamp += dds_modf(&s->phase[i], s->tone[i].phase_rate, s->tone[i].gain, 0);
#endif
}
/* Saturation of the answer is the right thing at this point.
However, we are normally generating well controlled tones,
that cannot clip. So, the overhead of doing saturation is
a waste of valuable time. */
#if defined(SPANDSP_USE_FIXED_POINT)
amp[samples] = xamp;
#else
amp[samples] = (int16_t) lrintf(xamp);
#endif
}
}
}