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audio-resampler: add fallback to mono function

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Remove stereo implementations. Implement fall back to mono functions
when the stereo function is missing.
This commit is contained in:
Wim Taymans 2016-01-21 10:38:17 +01:00
parent 2555317a71
commit 85c77659b9
1 changed files with 36 additions and 89 deletions
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125
gst-libs/gst/audio/audio-resampler.c
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@ -378,24 +378,6 @@ inner_product_gint16_1_c (gint16 * o, const gint16 * a, const gint16 * b,
*o = CLAMP (res, -(1L << 15), (1L << 15) - 1);
}
static inline void
inner_product_gint16_2_c (gint16 * o, const gint16 * a, const gint16 * b,
gint len)
{
gint i;
gint32 r[2] = { 0, 0 };
for (i = 0; i < len; i++) {
r[0] += (gint32) a[2 * i] * (gint32) b[i];
r[1] += (gint32) a[2 * i + 1] * (gint32) b[i];
}
r[0] = (r[0] + (1 << (PRECISION_S16 - 1))) >> PRECISION_S16;
r[1] = (r[1] + (1 << (PRECISION_S16 - 1))) >> PRECISION_S16;
o[0] = CLAMP (r[0], -(1L << 15), (1L << 15) - 1);
o[1] = CLAMP (r[1], -(1L << 15), (1L << 15) - 1);
}
static inline void
inner_product_gint32_1_c (gint32 * o, const gint32 * a, const gint32 * b,
gint len)
@ -423,21 +405,6 @@ inner_product_gfloat_1_c (gfloat * o, const gfloat * a, const gfloat * b,
*o = res;
}
static inline void
inner_product_gfloat_2_c (gfloat * o, const gfloat * a, const gfloat * b,
gint len)
{
gint i;
gfloat r[2] = { 0.0, 0.0 };
for (i = 0; i < len; i++) {
r[0] += a[2 * i] * b[i];
r[1] += a[2 * i + 1] * b[i];
}
o[0] = r[0];
o[1] = r[1];
}
static inline void
inner_product_gdouble_1_c (gdouble * o, const gdouble * a, const gdouble * b,
gint len)
@ -451,21 +418,6 @@ inner_product_gdouble_1_c (gdouble * o, const gdouble * a, const gdouble * b,
*o = res;
}
static inline void
inner_product_gdouble_2_c (gdouble * o, const gdouble * a, const gdouble * b,
gint len)
{
gint i;
gdouble r[2] = { 0.0, 0.0 };
for (i = 0; i < len; i++) {
r[0] += a[2 * i] * b[i];
r[1] += a[2 * i + 1] * b[i];
}
o[0] = r[0];
o[1] = r[1];
}
#define MAKE_RESAMPLE_FUNC(type,channels,arch) \
static void \
resample_ ##type## _ ##channels## _ ##arch (GstAudioResampler * resampler, \
@ -512,18 +464,16 @@ MAKE_RESAMPLE_FUNC (gint16, 1, c);
MAKE_RESAMPLE_FUNC (gint32, 1, c);
MAKE_RESAMPLE_FUNC (gfloat, 1, c);
MAKE_RESAMPLE_FUNC (gdouble, 1, c);
MAKE_RESAMPLE_FUNC (gint16, 2, c);
MAKE_RESAMPLE_FUNC (gfloat, 2, c);
MAKE_RESAMPLE_FUNC (gdouble, 2, c);
static ResampleFunc resample_funcs[] = {
resample_gint16_1_c,
resample_gint32_1_c,
resample_gfloat_1_c,
resample_gdouble_1_c,
resample_gint16_2_c,
resample_gfloat_2_c,
resample_gdouble_2_c,
NULL,
NULL,
NULL,
NULL,
};
#define resample_gint16_1 resample_funcs[0]
@ -531,8 +481,9 @@ static ResampleFunc resample_funcs[] = {
#define resample_gfloat_1 resample_funcs[2]
#define resample_gdouble_1 resample_funcs[3]
#define resample_gint16_2 resample_funcs[4]
#define resample_gfloat_2 resample_funcs[5]
#define resample_gdouble_2 resample_funcs[6]
#define resample_gint32_2 resample_funcs[5]
#define resample_gfloat_2 resample_funcs[6]
#define resample_gdouble_2 resample_funcs[7]
#if defined HAVE_ORC && !defined DISABLE_ORC
# if defined (__i386__) || defined (__x86_64__)
@ -673,6 +624,9 @@ resampler_calculate_taps (GstAudioResampler * resampler)
gint out_rate;
gint in_rate;
gboolean non_interleaved;
DeinterleaveFunc deinterleave;
ResampleFunc resample, resample_2;
switch (resampler->method) {
case GST_AUDIO_RESAMPLER_METHOD_NEAREST:
@ -745,46 +699,39 @@ resampler_calculate_taps (GstAudioResampler * resampler)
resampler->inc = 1;
switch (resampler->format) {
case GST_AUDIO_FORMAT_F64:
if (!non_interleaved && resampler->channels == 2 && n_taps >= 4) {
resampler->resample = resample_gdouble_2;
resampler->deinterleave = deinterleave_copy;
resampler->blocks = 1;
resampler->inc = resampler->channels;;
} else {
resampler->resample = resample_gdouble_1;
resampler->deinterleave = deinterleave_gdouble;
}
break;
case GST_AUDIO_FORMAT_F32:
if (!non_interleaved && resampler->channels == 2 && n_taps >= 4) {
resampler->resample = resample_gfloat_2;
resampler->deinterleave = deinterleave_copy;
resampler->blocks = 1;
resampler->inc = resampler->channels;;
} else {
resampler->resample = resample_gfloat_1;
resampler->deinterleave = deinterleave_gfloat;
}
case GST_AUDIO_FORMAT_S16:
resample = resample_gint16_1;
resample_2 = resample_gint16_2;
deinterleave = deinterleave_gint16;
break;
case GST_AUDIO_FORMAT_S32:
resampler->resample = resample_gint32_1;
resampler->deinterleave = deinterleave_gint32;
resample = resample_gint32_1;
resample_2 = resample_gint32_2;
deinterleave = deinterleave_gint32;
break;
case GST_AUDIO_FORMAT_S16:
if (!non_interleaved && resampler->channels == 2 && n_taps >= 4) {
resampler->resample = resample_gint16_2;
resampler->deinterleave = deinterleave_copy;
resampler->blocks = 1;
resampler->inc = resampler->channels;;
} else {
resampler->resample = resample_gint16_1;
resampler->deinterleave = deinterleave_gint16;
}
case GST_AUDIO_FORMAT_F32:
resample = resample_gfloat_1;
resample_2 = resample_gfloat_2;
deinterleave = deinterleave_gfloat;
break;
case GST_AUDIO_FORMAT_F64:
resample = resample_gdouble_1;
resample_2 = resample_gdouble_2;
deinterleave = deinterleave_gdouble;
break;
default:
g_assert_not_reached ();
break;
}
if (!non_interleaved && resampler->channels == 2 && n_taps >= 4 && resample_2) {
resampler->resample = resample_2;
resampler->deinterleave = deinterleave_copy;
resampler->blocks = 1;
resampler->inc = resampler->channels;;
} else {
resampler->resample = resample;
resampler->deinterleave = deinterleave;
}
}
#define PRINT_TAPS(type,print) \