Add a resampler to the processing chain when needed.
port the audio resampler to the new audioconverter library
audio-converter.h \
audio-info.h \
audio-quantize.h \
+ audio-resampler.h \
gstaudioringbuffer.h
glib_enum_define = GST_AUDIO
audio-converter.c \
audio-info.c \
audio-quantize.c \
+ audio-resampler.c \
gstaudioringbuffer.c \
gstaudioclock.c \
gstaudiocdsrc.c \
audio-converter.h \
audio-info.h \
audio-quantize.h \
+ audio-resampler.h \
gstaudioringbuffer.h \
gstaudioclock.h \
gstaudiofilter.h \
GstAudioChannelMixer *mix;
AudioChain *mix_chain;
+ /* resample */
+ GstAudioResampler *resampler;
+ AudioChain *resample_chain;
+
/* convert out */
AudioConvertFunc convert_out;
AudioChain *convert_out_chain;
}
static gpointer *
-audio_chain_alloc_samples (AudioChain * chain, gsize num_samples)
+audio_chain_alloc_samples (AudioChain * chain, gsize num_samples, gsize * avail)
{
- return chain->alloc_func (chain, num_samples, chain->alloc_data);
+ return chain->alloc_func (chain, num_samples, avail, chain->alloc_data);
}
static void
audio_chain_set_samples (AudioChain * chain, gpointer * samples,
gsize num_samples)
{
- GST_LOG ("set samples %p %" G_GSIZE_FORMAT, samples, num_samples);
+ if (num_samples == 0)
+ return;
+
+ GST_LOG ("set samples %" G_GSIZE_FORMAT, num_samples);
chain->samples = samples;
chain->num_samples = num_samples;
return res;
}
+#define DEFAULT_OPT_RESAMPLER_METHOD GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL
#define DEFAULT_OPT_DITHER_METHOD GST_AUDIO_DITHER_NONE
#define DEFAULT_OPT_NOISE_SHAPING_METHOD GST_AUDIO_NOISE_SHAPING_NONE
#define DEFAULT_OPT_QUANTIZATION 1
+#define GET_OPT_RESAMPLER_METHOD(c) get_opt_enum(c, \
+ GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD, GST_TYPE_AUDIO_RESAMPLER_METHOD, \
+ DEFAULT_OPT_RESAMPLER_METHOD)
#define GET_OPT_DITHER_METHOD(c) get_opt_enum(c, \
GST_AUDIO_CONVERTER_OPT_DITHER_METHOD, GST_TYPE_AUDIO_DITHER_METHOD, \
DEFAULT_OPT_DITHER_METHOD)
}
} else {
tmp = convert->in_data;
+ num_samples = convert->in_samples;
GST_LOG ("get in samples %p", tmp);
}
audio_chain_set_samples (chain, tmp, num_samples);
}
static gboolean
+do_resample (AudioChain * chain, gpointer user_data)
+{
+ GstAudioConverter *convert = user_data;
+ gpointer *in, *out;
+ gsize in_frames, out_frames, produced, consumed;
+
+ in = audio_chain_get_samples (chain->prev, &in_frames);
+
+ out_frames =
+ gst_audio_resampler_get_out_frames (convert->resampler, in_frames);
+ out =
+ (chain->allow_ip ? in : audio_chain_alloc_samples (chain, out_frames,
+ &out_frames));
+
+ GST_LOG ("resample %p %p,%" G_GSIZE_FORMAT " %" G_GSIZE_FORMAT, in, out,
+ in_frames, out_frames);
+
+ gst_audio_resampler_resample (convert->resampler, in, in_frames, out,
+ out_frames, &produced, &consumed);
+
+ audio_chain_set_samples (chain, out, produced);
+
+ return TRUE;
+}
+
+static gboolean
do_convert_out (AudioChain * chain, gpointer user_data)
{
GstAudioConverter *convert = user_data;
}
static AudioChain *
+chain_resample (GstAudioConverter * convert, AudioChain * prev)
+{
+ GstAudioInfo *in = &convert->in;
+ GstAudioInfo *out = &convert->out;
+ GstAudioResamplerMethod method;
+ GstAudioResamplerFlags flags;
+ GstAudioFormat format = convert->current_format;
+ gint channels = convert->current_channels;
+
+ if (in->rate != out->rate) {
+ method = GET_OPT_RESAMPLER_METHOD (convert);
+
+ flags = 0;
+ if (convert->current_layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED)
+ flags |= GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED;
+
+ convert->resampler =
+ gst_audio_resampler_new (method, flags, format, channels, in->rate,
+ out->rate, convert->config);
+
+ prev = convert->resample_chain = audio_chain_new (prev, convert);
+ prev->allow_ip = FALSE;
+ prev->pass_alloc = FALSE;
+ audio_chain_set_make_func (prev, do_resample, convert, NULL);
+ }
+ return prev;
+}
+
+static AudioChain *
chain_convert_out (GstAudioConverter * convert, AudioChain * prev)
{
gboolean in_int, out_int;
g_return_val_if_fail (in_info != NULL, FALSE);
g_return_val_if_fail (out_info != NULL, FALSE);
- g_return_val_if_fail (in_info->rate == out_info->rate, FALSE);
g_return_val_if_fail (in_info->layout == GST_AUDIO_LAYOUT_INTERLEAVED, FALSE);
g_return_val_if_fail (in_info->layout == out_info->layout, FALSE);
prev = chain_convert_in (convert, prev);
/* step 3, channel mix */
prev = chain_mix (convert, prev);
- /* step 4, optional convert for quantize */
+ /* step 4, resample */
+ prev = chain_resample (convert, prev);
+ /* step 5, optional convert for quantize */
prev = chain_convert_out (convert, prev);
- /* step 5, optional quantize */
+ /* step 6, optional quantize */
prev = chain_quantize (convert, prev);
- /* step 6, pack */
+ /* step 7, pack */
convert->pack_chain = chain_pack (convert, prev);
/* optimize */
typedef struct _GstAudioConverter GstAudioConverter;
/**
+ * GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD:
+ *
+ * #GST_TYPE_AUDIO_RESAMPLER_METHOD, The resampler method to use when
+ * changing sample rates.
+ * Default is #GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL.
+ */
+#define GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD "GstAudioConverter.resampler-method"
+
+/**
* GST_AUDIO_CONVERTER_OPT_DITHER_METHOD:
*
* #GST_TYPE_AUDIO_DITHER_METHOD, The dither method to use when
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2015> Wim Taymans <wim.taymans@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+
+#define PRECISION_S16 15
+#define PRECISION_S32 30
+
+#ifdef HAVE_EMMINTRIN_H
+#include <emmintrin.h>
+#endif
+
+static inline void
+inner_product_gdouble (gdouble * o, const gdouble * a, const gdouble * b,
+ gint len)
+{
+ gint i = 0;
+ gdouble res;
+#ifdef HAVE_EMMINTRIN_H
+ __m128d sum = _mm_setzero_pd ();
+
+ for (; i < len - 7; i += 8) {
+ sum =
+ _mm_add_pd (sum, _mm_mul_pd (_mm_loadu_pd (a + i + 0),
+ _mm_loadu_pd (b + i + 0)));
+ sum =
+ _mm_add_pd (sum, _mm_mul_pd (_mm_loadu_pd (a + i + 2),
+ _mm_loadu_pd (b + i + 2)));
+ sum =
+ _mm_add_pd (sum, _mm_mul_pd (_mm_loadu_pd (a + i + 4),
+ _mm_loadu_pd (b + i + 4)));
+ sum =
+ _mm_add_pd (sum, _mm_mul_pd (_mm_loadu_pd (a + i + 6),
+ _mm_loadu_pd (b + i + 6)));
+ }
+ sum = _mm_add_sd (sum, _mm_unpackhi_pd (sum, sum));
+ _mm_store_sd (&res, sum);
+#else
+ res = 0.0;
+#endif
+
+ for (; i < len; i++)
+ res += a[i] * b[i];
+
+ *o = res;
+}
+
+static inline void
+inner_product_gfloat (gfloat * o, const gfloat * a, const gfloat * b, gint len)
+{
+ gint i = 0;
+ gfloat res;
+#ifdef HAVE_EMMINTRIN_H
+ __m128 sum = _mm_setzero_ps ();
+
+ for (; i < len - 7; i += 8) {
+ sum =
+ _mm_add_ps (sum, _mm_mul_ps (_mm_loadu_ps (a + i + 0),
+ _mm_loadu_ps (b + i + 0)));
+ sum =
+ _mm_add_ps (sum, _mm_mul_ps (_mm_loadu_ps (a + i + 4),
+ _mm_loadu_ps (b + i + 4)));
+ }
+ sum = _mm_add_ps (sum, _mm_movehl_ps (sum, sum));
+ sum = _mm_add_ss (sum, _mm_shuffle_ps (sum, sum, 0x55));
+ _mm_store_ss (&res, sum);
+#else
+ res = 0.0;
+#endif
+
+ for (; i < len; i++)
+ res += a[i] * b[i];
+
+ *o = res;
+}
+
+static inline void
+inner_product_gint32 (gint32 * o, const gint32 * a, const gint32 * b, gint len)
+{
+ gint i = 0;
+ gint64 res = 0;
+
+ for (; i < len; i++)
+ res += (gint64) a[i] * (gint64) b[i];
+
+ res = (res + (1 << (PRECISION_S32 - 1))) >> PRECISION_S32;
+ *o = CLAMP (res, -(1L << 31), (1L << 31) - 1);
+}
+
+static inline void
+inner_product_gint16 (gint16 * o, const gint16 * a, const gint16 * b, gint len)
+{
+ gint i = 0;
+ gint32 res = 0;
+#ifdef HAVE_EMMINTRIN_H
+ __m128i sum[2], ta, tb;
+ __m128i t1[2];
+
+ sum[0] = _mm_setzero_si128 ();
+ sum[1] = _mm_setzero_si128 ();
+
+ for (; i < len - 7; i += 8) {
+ ta = _mm_loadu_si128 ((__m128i *) (a + i));
+ tb = _mm_loadu_si128 ((__m128i *) (b + i));
+
+ t1[0] = _mm_mullo_epi16 (ta, tb);
+ t1[1] = _mm_mulhi_epi16 (ta, tb);
+
+ sum[0] = _mm_add_epi32 (sum[0], _mm_unpacklo_epi16 (t1[0], t1[1]));
+ sum[1] = _mm_add_epi32 (sum[1], _mm_unpackhi_epi16 (t1[0], t1[1]));
+ }
+ sum[0] = _mm_add_epi32 (sum[0], sum[1]);
+ sum[0] =
+ _mm_add_epi32 (sum[0], _mm_shuffle_epi32 (sum[0], _MM_SHUFFLE (2, 3, 2,
+ 3)));
+ sum[0] =
+ _mm_add_epi32 (sum[0], _mm_shuffle_epi32 (sum[0], _MM_SHUFFLE (1, 1, 1,
+ 1)));
+ res = _mm_cvtsi128_si32 (sum[0]);
+#else
+ res = 0;
+#endif
+
+ for (; i < len; i++)
+ res += (gint32) a[i] * (gint32) b[i];
+
+ res = (res + (1 << (PRECISION_S16 - 1))) >> PRECISION_S16;
+ *o = CLAMP (res, -(1L << 15), (1L << 15) - 1);
+}
+
+static inline void
+inner_product_gdouble_2 (gdouble * o, const gdouble * a, const gdouble * b,
+ gint len)
+{
+ gint i = 0;
+ gdouble r[2];
+#ifdef HAVE_EMMINTRIN_H
+ __m128d sum = _mm_setzero_pd (), t;
+
+ for (; i < len - 3; i += 4) {
+ t = _mm_loadu_pd (b + i);
+ sum =
+ _mm_add_pd (sum, _mm_mul_pd (_mm_loadu_pd (a + 2 * i),
+ _mm_unpacklo_pd (t, t)));
+ sum =
+ _mm_add_pd (sum, _mm_mul_pd (_mm_loadu_pd (a + 2 * i + 2),
+ _mm_unpackhi_pd (t, t)));
+
+ t = _mm_loadu_pd (b + i + 2);
+ sum =
+ _mm_add_pd (sum, _mm_mul_pd (_mm_loadu_pd (a + 2 * i + 4),
+ _mm_unpacklo_pd (t, t)));
+ sum =
+ _mm_add_pd (sum, _mm_mul_pd (_mm_loadu_pd (a + 2 * i + 6),
+ _mm_unpackhi_pd (t, t)));
+ }
+ _mm_store_pd (r, sum);
+#else
+ r[0] = 0.0;
+ r[1] = 0.0;
+#endif
+
+ for (; 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_gint16_2 (gint16 * o, const gint16 * a, const gint16 * b, gint len)
+{
+ gint i = 0;
+ gint32 r[2];
+#ifdef HAVE_EMMINTRIN_H
+ guint64 r64;
+ __m128i sum[2], ta, tb;
+ __m128i t1[2];
+
+ sum[0] = _mm_setzero_si128 ();
+ sum[1] = _mm_setzero_si128 ();
+
+ for (; i < len - 7; i += 8) {
+ tb = _mm_loadu_si128 ((__m128i *) (b + i));
+
+ t1[1] = _mm_unpacklo_epi16 (tb, tb);
+
+ ta = _mm_loadu_si128 ((__m128i *) (a + 2 * i));
+ t1[0] = _mm_mullo_epi16 (ta, t1[1]);
+ t1[1] = _mm_mulhi_epi16 (ta, t1[1]);
+
+ sum[0] = _mm_add_epi32 (sum[0], _mm_unpacklo_epi16 (t1[0], t1[1]));
+ sum[1] = _mm_add_epi32 (sum[1], _mm_unpackhi_epi16 (t1[0], t1[1]));
+
+ t1[1] = _mm_unpackhi_epi16 (tb, tb);
+
+ ta = _mm_loadu_si128 ((__m128i *) (a + 2 * i + 8));
+ t1[0] = _mm_mullo_epi16 (ta, t1[1]);
+ t1[1] = _mm_mulhi_epi16 (ta, t1[1]);
+
+ sum[0] = _mm_add_epi32 (sum[0], _mm_unpacklo_epi16 (t1[0], t1[1]));
+ sum[1] = _mm_add_epi32 (sum[1], _mm_unpackhi_epi16 (t1[0], t1[1]));
+ }
+ sum[0] = _mm_add_epi32 (sum[0], sum[1]);
+ sum[0] =
+ _mm_add_epi32 (sum[0], _mm_shuffle_epi32 (sum[0], _MM_SHUFFLE (2, 3, 2,
+ 3)));
+ r64 = _mm_cvtsi128_si64 (sum[0]);
+ r[0] = r64 >> 32;
+ r[1] = r64 & 0xffffffff;
+#else
+ r[0] = 0;
+ r[1] = 0;
+#endif
+
+ for (; 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);
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2015> Wim Taymans <wim.taymans@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include <string.h>
+#include <stdio.h>
+#include <math.h>
+
+#include "audio-resampler.h"
+
+typedef struct _Tap
+{
+ gpointer taps;
+
+ gint sample_inc;
+ gint next_phase;
+ gint size;
+} Tap;
+
+typedef void (*MakeTapsFunc) (GstAudioResampler * resampler, Tap * t, gint j);
+typedef void (*ResampleFunc) (GstAudioResampler * resampler, gpointer in[],
+ gsize in_len, gpointer out[], gsize out_len, gsize * consumed,
+ gsize * produced, gboolean move);
+typedef void (*DeinterleaveFunc) (GstAudioResampler * resampler,
+ gpointer * sbuf, gpointer in[], gsize in_frames);
+typedef void (*MirrorFunc) (GstAudioResampler * resampler, gpointer * sbuf);
+
+struct _GstAudioResampler
+{
+ GstAudioResamplerMethod method;
+ GstAudioResamplerFlags flags;
+ GstAudioFormat format;
+ GstStructure *options;
+ guint channels;
+ gint in_rate;
+ gint out_rate;
+ gint bps, bpf;
+ gint ostride;
+
+ gdouble cutoff;
+ gdouble kaiser_beta;
+ /* for cubic */
+ gdouble b, c;
+
+ guint n_taps;
+ Tap *taps;
+ gpointer coeff;
+ gpointer tmpcoeff;
+
+ DeinterleaveFunc deinterleave;
+ MirrorFunc mirror;
+ ResampleFunc resample;
+
+ gboolean filling;
+ gint samp_inc;
+ gint samp_frac;
+ gint samp_index;
+ gint samp_phase;
+ gint skip;
+
+ gpointer samples;
+ gsize samples_len;
+ gsize samples_avail;
+ gpointer *sbuf;
+};
+
+#ifndef GST_DISABLE_GST_DEBUG
+#define GST_CAT_DEFAULT ensure_debug_category()
+static GstDebugCategory *
+ensure_debug_category (void)
+{
+ static gsize cat_gonce = 0;
+
+ if (g_once_init_enter (&cat_gonce)) {
+ gsize cat_done;
+
+ cat_done = (gsize) _gst_debug_category_new ("audio-resampler", 0,
+ "audio-resampler object");
+
+ g_once_init_leave (&cat_gonce, cat_done);
+ }
+
+ return (GstDebugCategory *) cat_gonce;
+}
+#else
+#define ensure_debug_category() /* NOOP */
+#endif /* GST_DISABLE_GST_DEBUG */
+
+/**
+ * SECTION:gstaudioresampler
+ * @short_description: Utility structure for resampler information
+ *
+ * #GstAudioResampler is a structure which holds the information
+ * required to perform various kinds of resampling filtering.
+ *
+ */
+
+typedef struct
+{
+ gdouble cutoff;
+ gdouble downsample_cutoff_factor;
+ gdouble stopband_attenuation;
+ gdouble transition_bandwidth;
+} KaiserQualityMap;
+
+static const KaiserQualityMap kaiser_qualities[] = {
+ {0.860, 0.96511, 60, 0.7}, /* 8 taps */
+ {0.880, 0.96591, 65, 0.29}, /* 16 taps */
+ {0.910, 0.96923, 70, 0.145}, /* 32 taps */
+ {0.920, 0.97600, 80, 0.105}, /* 48 taps */
+ {0.940, 0.97979, 85, 0.087}, /* 64 taps default quality */
+ {0.940, 0.98085, 95, 0.077}, /* 80 taps */
+ {0.945, 0.99471, 100, 0.068}, /* 96 taps */
+ {0.950, 1.0, 105, 0.055}, /* 128 taps */
+ {0.960, 1.0, 110, 0.045}, /* 160 taps */
+ {0.968, 1.0, 115, 0.039}, /* 192 taps */
+ {0.975, 1.0, 120, 0.0305} /* 256 taps */
+};
+
+typedef struct
+{
+ guint n_taps;
+ gdouble cutoff;
+} BlackmanQualityMap;
+
+static const BlackmanQualityMap blackman_qualities[] = {
+ {8, 0.5,},
+ {16, 0.6,},
+ {24, 0.72,},
+ {32, 0.8,},
+ {48, 0.85,}, /* default */
+ {64, 0.90,},
+ {80, 0.92,},
+ {96, 0.933,},
+ {128, 0.950,},
+ {148, 0.955,},
+ {160, 0.960,}
+};
+
+#define DEFAULT_QUALITY GST_AUDIO_RESAMPLER_QUALITY_DEFAULT
+#define DEFAULT_OPT_CUBIC_B 1.0
+#define DEFAULT_OPT_CUBIC_C 0.0
+
+static gdouble
+get_opt_double (GstStructure * options, const gchar * name, gdouble def)
+{
+ gdouble res;
+ if (!options || !gst_structure_get_double (options, name, &res))
+ res = def;
+ return res;
+}
+
+static gint
+get_opt_int (GstStructure * options, const gchar * name, gint def)
+{
+ gint res;
+ if (!options || !gst_structure_get_int (options, name, &res))
+ res = def;
+ return res;
+}
+
+#define GET_OPT_CUTOFF(options,def) get_opt_double(options, \
+ GST_AUDIO_RESAMPLER_OPT_CUTOFF,def)
+#define GET_OPT_DOWN_CUTOFF_FACTOR(options,def) get_opt_double(options, \
+ GST_AUDIO_RESAMPLER_OPT_DOWN_CUTOFF_FACTOR, def)
+#define GET_OPT_STOP_ATTENUATION(options,def) get_opt_double(options, \
+ GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION, def)
+#define GET_OPT_TRANSITION_BANDWIDTH(options,def) get_opt_double(options, \
+ GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH, def)
+#define GET_OPT_CUBIC_B(options) get_opt_double(options, \
+ GST_AUDIO_RESAMPLER_OPT_CUBIC_B, DEFAULT_OPT_CUBIC_B)
+#define GET_OPT_CUBIC_C(options) get_opt_double(options, \
+ GST_AUDIO_RESAMPLER_OPT_CUBIC_C, DEFAULT_OPT_CUBIC_C)
+#define GET_OPT_N_TAPS(options,def) get_opt_int(options, \
+ GST_AUDIO_RESAMPLER_OPT_N_TAPS, def)
+
+#include "dbesi0.c"
+#define bessel dbesi0
+
+static inline gdouble
+get_nearest_tap (GstAudioResampler * resampler, gdouble x)
+{
+ gdouble a = fabs (x);
+
+ if (a < 0.5)
+ return 1.0;
+ else
+ return 0.0;
+}
+
+static inline gdouble
+get_linear_tap (GstAudioResampler * resampler, gdouble x)
+{
+ gdouble a;
+
+ a = fabs (x) / resampler->n_taps;
+
+ if (a < 1.0)
+ return 1.0 - a;
+ else
+ return 0.0;
+}
+
+static inline gdouble
+get_cubic_tap (GstAudioResampler * resampler, gdouble x)
+{
+ gdouble a, a2, a3, b, c;
+
+ a = fabs (x * 4.0) / resampler->n_taps;
+ a2 = a * a;
+ a3 = a2 * a;
+
+ b = resampler->b;
+ c = resampler->c;
+
+ if (a <= 1.0)
+ return ((12.0 - 9.0 * b - 6.0 * c) * a3 +
+ (-18.0 + 12.0 * b + 6.0 * c) * a2 + (6.0 - 2.0 * b)) / 6.0;
+ else if (a <= 2.0)
+ return ((-b - 6.0 * c) * a3 +
+ (6.0 * b + 30.0 * c) * a2 +
+ (-12.0 * b - 48.0 * c) * a + (8.0 * b + 24.0 * c)) / 6.0;
+ else
+ return 0.0;
+}
+
+static inline gdouble
+get_blackman_nuttall_tap (GstAudioResampler * resampler, gdouble x)
+{
+ gdouble s, y, w, Fc = resampler->cutoff;
+
+ y = G_PI * x;
+ s = (y == 0.0 ? Fc : sin (y * Fc) / y);
+
+ w = 2.0 * y / resampler->n_taps + G_PI;
+ return s * (0.3635819 - 0.4891775 * cos (w) + 0.1365995 * cos (2 * w) -
+ 0.0106411 * cos (3 * w));
+}
+
+static inline gdouble
+get_kaiser_tap (GstAudioResampler * resampler, gdouble x)
+{
+ gdouble s, y, w, Fc = resampler->cutoff;
+
+ y = G_PI * x;
+ s = (y == 0.0 ? Fc : sin (y * Fc) / y);
+
+ w = 2.0 * x / resampler->n_taps;
+ return s * bessel (resampler->kaiser_beta * sqrt (MAX (1 - w * w, 0)));
+}
+
+#define CONVERT_TAPS(type, precision) \
+G_STMT_START { \
+ type *taps = t->taps = (type *) resampler->coeff + j * n_taps; \
+ gdouble multiplier = (1 << precision); \
+ gint i, j; \
+ gdouble offset, l_offset, h_offset; \
+ gboolean exact = FALSE; \
+ /* Round to integer, but with an adjustable bias that we use to */ \
+ /* eliminate the DC error. */ \
+ l_offset = 0.0; \
+ h_offset = 1.0; \
+ offset = 0.5; \
+ for (i = 0; i < 32; i++) { \
+ gint64 sum = 0; \
+ for (j = 0; j < n_taps; j++) \
+ sum += taps[j] = floor (offset + tmpcoeff[j] * multiplier / weight); \
+ if (sum == (1 << precision)) { \
+ exact = TRUE; \
+ break; \
+ } \
+ if (l_offset == h_offset) \
+ break; \
+ if (sum < (1 << precision)) { \
+ if (offset > l_offset) \
+ l_offset = offset; \
+ offset += (h_offset - l_offset) / 2; \
+ } else { \
+ if (offset < h_offset) \
+ h_offset = offset; \
+ offset -= (h_offset - l_offset) / 2; \
+ } \
+ } \
+ if (!exact) \
+ GST_WARNING ("can't find exact taps"); \
+} G_STMT_END
+
+#include "audio-resampler-core.h"
+
+static void
+make_taps (GstAudioResampler * resampler, Tap * t, gint j)
+{
+ gint n_taps = resampler->n_taps;
+ gdouble x, weight = 0.0;
+ gdouble *tmpcoeff = resampler->tmpcoeff;
+ gint tap_offs = n_taps / 2;
+ gint out_rate = resampler->out_rate;
+ gint l;
+
+ x = ((double) (1.0 - tap_offs) - (double) j / out_rate);
+
+ switch (resampler->method) {
+ case GST_AUDIO_RESAMPLER_METHOD_NEAREST:
+ for (l = 0; l < n_taps; l++, x += 1.0)
+ weight += tmpcoeff[l] = get_nearest_tap (resampler, x);
+ break;
+
+ case GST_AUDIO_RESAMPLER_METHOD_LINEAR:
+ for (l = 0; l < n_taps; l++, x += 1.0)
+ weight += tmpcoeff[l] = get_linear_tap (resampler, x);
+ break;
+
+ case GST_AUDIO_RESAMPLER_METHOD_CUBIC:
+ for (l = 0; l < n_taps; l++, x += 1.0)
+ weight += tmpcoeff[l] = get_cubic_tap (resampler, x);
+ break;
+
+ case GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL:
+ for (l = 0; l < n_taps; l++, x += 1.0)
+ weight += tmpcoeff[l] = get_blackman_nuttall_tap (resampler, x);
+ break;
+
+ case GST_AUDIO_RESAMPLER_METHOD_KAISER:
+ for (l = 0; l < n_taps; l++, x += 1.0)
+ weight += tmpcoeff[l] = get_kaiser_tap (resampler, x);
+ break;
+
+ default:
+ break;
+ }
+
+ switch (resampler->format) {
+ case GST_AUDIO_FORMAT_F64:
+ {
+ gdouble *taps = t->taps = (gdouble *) resampler->coeff + j * n_taps;
+ for (l = 0; l < n_taps; l++)
+ taps[l] = tmpcoeff[l] / weight;
+ break;
+ }
+ case GST_AUDIO_FORMAT_F32:
+ {
+ gfloat *taps = t->taps = (gfloat *) resampler->coeff + j * n_taps;
+ for (l = 0; l < n_taps; l++)
+ taps[l] = tmpcoeff[l] / weight;
+ break;
+ }
+ case GST_AUDIO_FORMAT_S32:
+ CONVERT_TAPS (gint32, PRECISION_S32);
+ break;
+ case GST_AUDIO_FORMAT_S16:
+ CONVERT_TAPS (gint16, PRECISION_S16);
+ break;
+ default:
+ break;
+ }
+}
+
+#define MAKE_RESAMPLE_FUNC(type) \
+static void \
+resample_ ##type (GstAudioResampler * resampler, gpointer in[], gsize in_len, \
+ gpointer out[], gsize out_len, gsize * consumed, gsize * produced, \
+ gboolean move) \
+{ \
+ gint c, di = 0; \
+ gint n_taps = resampler->n_taps; \
+ gint channels = resampler->channels; \
+ gint ostride = resampler->ostride; \
+ gint samp_index = 0; \
+ gint samp_phase = 0; \
+ \
+ for (c = 0; c < channels; c++) { \
+ type *ip = in[c]; \
+ type *op = ostride == 1 ? out[c] : (type *)out[0] + c; \
+ \
+ samp_index = resampler->samp_index; \
+ samp_phase = resampler->samp_phase; \
+ \
+ for (di = 0; di < out_len; di++) { \
+ Tap *t = &resampler->taps[samp_phase]; \
+ type *ipp = &ip[samp_index]; \
+ \
+ if (t->taps == NULL) \
+ make_taps (resampler, t, samp_phase); \
+ \
+ inner_product_ ##type (op, ipp, t->taps, n_taps); \
+ op += ostride; \
+ \
+ samp_phase = t->next_phase; \
+ samp_index += t->sample_inc; \
+ } \
+ if (move) \
+ memmove (ip, &ip[samp_index], (in_len - samp_index) * sizeof(type)); \
+ } \
+ *consumed = samp_index - resampler->samp_index; \
+ *produced = di; \
+ \
+ resampler->samp_index = move ? 0 : samp_index; \
+ resampler->samp_phase = samp_phase; \
+}
+
+MAKE_RESAMPLE_FUNC (gdouble);
+MAKE_RESAMPLE_FUNC (gfloat);
+MAKE_RESAMPLE_FUNC (gint32);
+MAKE_RESAMPLE_FUNC (gint16);
+
+#define MAKE_RESAMPLE_INTERLEAVED_FUNC(type,channels) \
+static void \
+resample_interleaved_ ##type##_##channels (GstAudioResampler * resampler, gpointer in[],\
+ gsize in_len, gpointer out[], gsize out_len, gsize * consumed, gsize * produced, \
+ gboolean move) \
+{ \
+ gint di = 0; \
+ gint n_taps = resampler->n_taps; \
+ gint ostride = resampler->ostride; \
+ gint samp_index = 0; \
+ gint samp_phase = 0; \
+ \
+ { \
+ type *ip = in[0]; \
+ type *op = out[0]; \
+ \
+ samp_index = resampler->samp_index; \
+ samp_phase = resampler->samp_phase; \
+ \
+ for (di = 0; di < out_len; di++) { \
+ Tap *t = &resampler->taps[samp_phase]; \
+ type *ipp = &ip[samp_index * channels]; \
+ \
+ if (t->taps == NULL) \
+ make_taps (resampler, t, samp_phase); \
+ \
+ inner_product_ ##type## _##channels (op, ipp, t->taps, n_taps); \
+ \
+ op += ostride; \
+ samp_phase = t->next_phase; \
+ samp_index += t->sample_inc; \
+ } \
+ if (move) \
+ memmove (ip, &ip[samp_index * channels], \
+ (in_len - samp_index) * sizeof(type) * channels); \
+ } \
+ *consumed = samp_index - resampler->samp_index; \
+ *produced = di; \
+ \
+ resampler->samp_index = move ? 0 : samp_index; \
+ resampler->samp_phase = samp_phase; \
+}
+
+MAKE_RESAMPLE_INTERLEAVED_FUNC (gdouble, 2);
+MAKE_RESAMPLE_INTERLEAVED_FUNC (gint16, 2);
+
+
+#define MAKE_DEINTERLEAVE_FUNC(type) \
+static void \
+deinterleave_ ##type (GstAudioResampler * resampler, gpointer sbuf[], \
+ gpointer in[], gsize in_frames) \
+{ \
+ guint i, c, channels = resampler->channels; \
+ gsize samples_avail = resampler->samples_avail; \
+ for (c = 0; c < channels; c++) { \
+ type *s = (type *) sbuf[c] + samples_avail; \
+ if (in == NULL) { \
+ for (i = 0; i < in_frames; i++) \
+ s[i] = 0; \
+ } else { \
+ type *ip = (type *) in[0] + c; \
+ for (i = 0; i < in_frames; i++, ip += channels) \
+ s[i] = *ip; \
+ } \
+ } \
+}
+
+MAKE_DEINTERLEAVE_FUNC (gdouble);
+MAKE_DEINTERLEAVE_FUNC (gfloat);
+MAKE_DEINTERLEAVE_FUNC (gint32);
+MAKE_DEINTERLEAVE_FUNC (gint16);
+
+static void
+deinterleave_copy (GstAudioResampler * resampler, gpointer sbuf[],
+ gpointer in[], gsize in_frames)
+{
+ gsize samples_avail = resampler->samples_avail;
+ gint bpf = resampler->bpf;
+
+ if (in == NULL)
+ memset ((guint8 *) sbuf[0] + samples_avail * bpf, 0, in_frames * bpf);
+ else
+ memcpy ((guint8 *) sbuf[0] + samples_avail * bpf, in[0], in_frames * bpf);
+}
+
+static void
+deinterleave_copy_n (GstAudioResampler * resampler, gpointer sbuf[],
+ gpointer in[], gsize in_frames)
+{
+ guint c, channels = resampler->channels;
+ gsize samples_avail = resampler->samples_avail;
+ gint bps = resampler->bps;
+
+ for (c = 0; c < channels; c++) {
+ if (in == NULL)
+ memset ((guint8 *) sbuf[c] + samples_avail * bps, 0, in_frames * bps);
+ else
+ memcpy ((guint8 *) sbuf[c] + samples_avail * bps, in[c], in_frames * bps);
+ }
+}
+
+/* mirror input samples into the history when we have nothing else */
+#define MAKE_MIRROR_FUNC(type) \
+static void \
+mirror_ ##type (GstAudioResampler * resampler, gpointer sbuf[]) \
+{ \
+ guint i, c, channels = resampler->channels; \
+ gint si = resampler->n_taps / 2; \
+ gint n_taps = resampler->n_taps; \
+ for (c = 0; c < channels; c++) { \
+ type *s = sbuf[c]; \
+ for (i = 0; i < si; i++) \
+ s[i] = -s[n_taps - i]; \
+ } \
+}
+
+MAKE_MIRROR_FUNC (gdouble);
+MAKE_MIRROR_FUNC (gfloat);
+MAKE_MIRROR_FUNC (gint32);
+MAKE_MIRROR_FUNC (gint16);
+
+static void
+calculate_kaiser_params (GstAudioResampler * resampler)
+{
+ gdouble A, B, dw, tr_bw, Fc;
+ gint n;
+ const KaiserQualityMap *q = &kaiser_qualities[DEFAULT_QUALITY];
+
+ /* default cutoff */
+ Fc = q->cutoff;
+ if (resampler->out_rate < resampler->in_rate)
+ Fc *= q->downsample_cutoff_factor;
+
+ Fc = GET_OPT_CUTOFF (resampler->options, Fc);
+ A = GET_OPT_STOP_ATTENUATION (resampler->options, q->stopband_attenuation);
+ tr_bw =
+ GET_OPT_TRANSITION_BANDWIDTH (resampler->options,
+ q->transition_bandwidth);
+
+ GST_LOG ("Fc %f, A %f, tr_bw %f", Fc, A, tr_bw);
+
+ /* calculate Beta */
+ if (A > 50)
+ B = 0.1102 * (A - 8.7);
+ else if (A >= 21)
+ B = 0.5842 * pow (A - 21, 0.4) + 0.07886 * (A - 21);
+ else
+ B = 0.0;
+ /* calculate transition width in radians */
+ dw = 2 * G_PI * (tr_bw);
+ /* order of the filter */
+ n = (A - 8.0) / (2.285 * dw);
+
+ resampler->kaiser_beta = B;
+ resampler->n_taps = n + 1;
+ resampler->cutoff = Fc;
+
+ GST_LOG ("using Beta %f n_taps %d cutoff %f", resampler->kaiser_beta,
+ resampler->n_taps, resampler->cutoff);
+}
+
+static void
+resampler_calculate_taps (GstAudioResampler * resampler)
+{
+ gint bps;
+ gint j;
+ gint n_taps;
+ gint out_rate;
+ gint in_rate;
+
+ switch (resampler->method) {
+ case GST_AUDIO_RESAMPLER_METHOD_NEAREST:
+ resampler->n_taps = 2;
+ break;
+ case GST_AUDIO_RESAMPLER_METHOD_LINEAR:
+ resampler->n_taps = GET_OPT_N_TAPS (resampler->options, 2);
+ break;
+ case GST_AUDIO_RESAMPLER_METHOD_CUBIC:
+ resampler->n_taps = GET_OPT_N_TAPS (resampler->options, 4);
+ resampler->b = GET_OPT_CUBIC_B (resampler->options);
+ resampler->c = GET_OPT_CUBIC_C (resampler->options);;
+ break;
+ case GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL:
+ {
+ const BlackmanQualityMap *q = &blackman_qualities[DEFAULT_QUALITY];
+ resampler->n_taps = GET_OPT_N_TAPS (resampler->options, q->n_taps);
+ resampler->cutoff = GET_OPT_CUTOFF (resampler->options, q->cutoff);
+ break;
+ }
+ case GST_AUDIO_RESAMPLER_METHOD_KAISER:
+ calculate_kaiser_params (resampler);
+ break;
+ }
+
+ in_rate = resampler->in_rate;
+ out_rate = resampler->out_rate;
+
+ if (out_rate < in_rate) {
+ resampler->cutoff = resampler->cutoff * out_rate / in_rate;
+ resampler->n_taps = resampler->n_taps * in_rate / out_rate;
+ }
+ /* only round up for bigger taps, the small taps are used for nearest,
+ * linear and cubic and we want to use less taps for those. */
+ if (resampler->n_taps > 4)
+ resampler->n_taps = GST_ROUND_UP_8 (resampler->n_taps);
+
+ n_taps = resampler->n_taps;
+ bps = resampler->bps;
+
+ GST_LOG ("using n_taps %d cutoff %f", n_taps, resampler->cutoff);
+
+ resampler->taps = g_realloc_n (resampler->taps, out_rate, sizeof (Tap));
+ resampler->coeff = g_realloc_n (resampler->coeff, out_rate, bps * n_taps);
+ resampler->tmpcoeff =
+ g_realloc_n (resampler->tmpcoeff, n_taps, sizeof (gdouble));
+
+ resampler->samp_inc = in_rate / out_rate;
+ resampler->samp_frac = in_rate % out_rate;
+
+ for (j = 0; j < out_rate; j++) {
+ Tap *t = &resampler->taps[j];
+ t->taps = NULL;
+ t->sample_inc = (j + in_rate) / out_rate;
+ t->next_phase = (j + in_rate) % out_rate;
+ }
+
+ switch (resampler->format) {
+ case GST_AUDIO_FORMAT_F64:
+ if (resampler->channels == 2 && n_taps >= 4) {
+ resampler->resample = resample_interleaved_gdouble_2;
+ resampler->deinterleave = deinterleave_copy;
+ } else {
+ resampler->resample = resample_gdouble;
+ resampler->deinterleave = deinterleave_gdouble;
+ }
+ resampler->mirror = mirror_gdouble;
+ break;
+ case GST_AUDIO_FORMAT_F32:
+ resampler->resample = resample_gfloat;
+ resampler->deinterleave = deinterleave_gfloat;
+ resampler->mirror = mirror_gfloat;
+ break;
+ case GST_AUDIO_FORMAT_S32:
+ resampler->resample = resample_gint32;
+ resampler->deinterleave = deinterleave_gint32;
+ resampler->mirror = mirror_gint32;
+ break;
+ case GST_AUDIO_FORMAT_S16:
+ if (resampler->channels == 2 && n_taps >= 4) {
+ resampler->resample = resample_interleaved_gint16_2;
+ resampler->deinterleave = deinterleave_copy;
+ } else {
+ resampler->resample = resample_gint16;
+ resampler->deinterleave = deinterleave_gint16;
+ }
+ resampler->mirror = mirror_gint16;
+ break;
+ default:
+ break;
+ }
+ if (resampler->flags & GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED) {
+ resampler->deinterleave = deinterleave_copy_n;
+ resampler->ostride = 1;
+ } else {
+ resampler->ostride = resampler->channels;
+ }
+}
+
+#define PRINT_TAPS(type,print) \
+G_STMT_START { \
+ type sum = 0.0, *taps; \
+ \
+ if (t->taps == NULL) \
+ make_taps (resampler, t, i); \
+ \
+ taps = t->taps; \
+ for (j = 0; j < n_taps; j++) { \
+ type tap = taps[j]; \
+ fprintf (stderr, "\t%" print " ", tap); \
+ sum += tap; \
+ } \
+ fprintf (stderr, "\t: sum %" print "\n", sum);\
+} G_STMT_END
+
+static void
+resampler_dump (GstAudioResampler * resampler)
+{
+#if 0
+ gint i, n_taps, out_rate;
+ gint64 a;
+
+ out_rate = resampler->out_rate;
+ n_taps = resampler->n_taps;
+
+ fprintf (stderr, "out size %d, max taps %d\n", out_rate, n_taps);
+
+ a = g_get_monotonic_time ();
+
+ for (i = 0; i < out_rate; i++) {
+ gint j;
+ Tap *t = &resampler->taps[i];
+
+ fprintf (stderr, "%u: %d %d\t ", i, t->sample_inc, t->next_phase);
+ switch (resampler->format) {
+ case GST_AUDIO_FORMAT_F64:
+ PRINT_TAPS (gdouble, "f");
+ break;
+ case GST_AUDIO_FORMAT_F32:
+ PRINT_TAPS (gfloat, "f");
+ break;
+ case GST_AUDIO_FORMAT_S32:
+ PRINT_TAPS (gint32, "d");
+ break;
+ case GST_AUDIO_FORMAT_S16:
+ PRINT_TAPS (gint16, "d");
+ break;
+ default:
+ break;
+ }
+ }
+ fprintf (stderr, "time %" G_GUINT64_FORMAT "\n", g_get_monotonic_time () - a);
+#endif
+}
+
+/**
+ * gst_audio_resampler_options_set_quality:
+ * @method: a #GstAudioResamplerMethod
+ * @quality: the quality
+ * @in_rate: the input rate
+ * @out_rate: the output rate
+ * @options: a #GstStructure
+ *
+ * Set the parameters for resampling from @in_rate to @out_rate using @method
+ * for @quality in @options.
+ */
+void
+gst_audio_resampler_options_set_quality (GstAudioResamplerMethod method,
+ guint quality, guint in_rate, guint out_rate, GstStructure * options)
+{
+ g_return_if_fail (options != NULL);
+ g_return_if_fail (quality < 11);
+ g_return_if_fail (in_rate != 0 && out_rate != 0);
+
+ switch (method) {
+ case GST_AUDIO_RESAMPLER_METHOD_NEAREST:
+ break;
+ case GST_AUDIO_RESAMPLER_METHOD_LINEAR:
+ gst_structure_set (options,
+ GST_AUDIO_RESAMPLER_OPT_N_TAPS, G_TYPE_INT, 2, NULL);
+ break;
+ case GST_AUDIO_RESAMPLER_METHOD_CUBIC:
+ gst_structure_set (options,
+ GST_AUDIO_RESAMPLER_OPT_N_TAPS, G_TYPE_INT, 4,
+ GST_AUDIO_RESAMPLER_OPT_CUBIC_B, G_TYPE_DOUBLE, DEFAULT_OPT_CUBIC_B,
+ GST_AUDIO_RESAMPLER_OPT_CUBIC_C, G_TYPE_DOUBLE, DEFAULT_OPT_CUBIC_C,
+ NULL);
+ break;
+ case GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL:
+ {
+ const BlackmanQualityMap *map = &blackman_qualities[quality];
+ gst_structure_set (options,
+ GST_AUDIO_RESAMPLER_OPT_N_TAPS, G_TYPE_INT, map->n_taps,
+ GST_AUDIO_RESAMPLER_OPT_CUTOFF, G_TYPE_DOUBLE, map->cutoff, NULL);
+ break;
+ }
+ case GST_AUDIO_RESAMPLER_METHOD_KAISER:
+ {
+ const KaiserQualityMap *map = &kaiser_qualities[quality];
+ gdouble cutoff;
+
+ cutoff = map->cutoff;
+ if (out_rate < in_rate)
+ cutoff *= map->downsample_cutoff_factor;
+
+ gst_structure_set (options,
+ GST_AUDIO_RESAMPLER_OPT_CUTOFF, G_TYPE_DOUBLE, cutoff,
+ GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION, G_TYPE_DOUBLE,
+ map->stopband_attenuation,
+ GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH, G_TYPE_DOUBLE,
+ map->transition_bandwidth, NULL);
+ break;
+ }
+ }
+}
+
+/**
+ * gst_audio_resampler_new:
+ * @resampler: a #GstAudioResampler
+ * @method: a #GstAudioResamplerMethod
+ * @flags: #GstAudioResamplerFlags
+ * @in_rate: input rate
+ * @out_rate: output rate
+ * @options: extra options
+ *
+ * Make a new resampler.
+ *
+ * Returns: %TRUE on success
+ */
+GstAudioResampler *
+gst_audio_resampler_new (GstAudioResamplerMethod method,
+ GstAudioResamplerFlags flags,
+ GstAudioFormat format, guint channels,
+ guint in_rate, guint out_rate, GstStructure * options)
+{
+ GstAudioResampler *resampler;
+ const GstAudioFormatInfo *info;
+
+ g_return_val_if_fail (in_rate != 0, FALSE);
+ g_return_val_if_fail (out_rate != 0, FALSE);
+
+ resampler = g_slice_new0 (GstAudioResampler);
+ resampler->method = method;
+ resampler->flags = flags;
+ resampler->format = format;
+ resampler->channels = channels;
+
+ info = gst_audio_format_get_info (format);
+ resampler->bps = GST_AUDIO_FORMAT_INFO_WIDTH (info) / 8;
+ resampler->bpf = resampler->bps * channels;
+ resampler->sbuf = g_malloc0 (sizeof (gpointer) * channels);
+
+ GST_DEBUG ("method %d, bps %d, bpf %d", method, resampler->bps,
+ resampler->bpf);
+
+ gst_audio_resampler_update (resampler, in_rate, out_rate, options);
+
+ return resampler;
+}
+
+/**
+ * gst_audio_resampler_update:
+ * @resampler: a #GstAudioResampler
+ * @in_rate: new input rate
+ * @out_rate: new output rate
+ * @options: new options or %NULL
+ *
+ * Update the resampler parameters for @resampler. This function should
+ * not be called concurrently with any other function on @resampler.
+ *
+ * Returns: %TRUE if the new parameters could be set
+ */
+gboolean
+gst_audio_resampler_update (GstAudioResampler * resampler,
+ guint in_rate, guint out_rate, GstStructure * options)
+{
+ gint gcd;
+
+ g_return_val_if_fail (resampler != NULL, FALSE);
+ g_return_val_if_fail (in_rate != 0, FALSE);
+ g_return_val_if_fail (out_rate != 0, FALSE);
+
+ gcd = gst_util_greatest_common_divisor (in_rate, out_rate);
+ in_rate /= gcd;
+ out_rate /= gcd;
+
+ resampler->in_rate = in_rate;
+ resampler->out_rate = out_rate;
+ if (options) {
+ if (resampler->options)
+ gst_structure_free (resampler->options);
+ resampler->options = gst_structure_copy (options);
+ }
+
+ GST_DEBUG ("%u->%u", in_rate, out_rate);
+
+ resampler_calculate_taps (resampler);
+ resampler_dump (resampler);
+
+ resampler->filling = TRUE;
+ resampler->samp_index = 0;
+ resampler->samp_phase = 0;
+ resampler->samples_avail = resampler->n_taps / 2 - 1;
+
+ return TRUE;
+}
+
+/**
+ * gst_audio_resampler_free:
+ * @resampler: a #GstAudioResampler
+ *
+ * Free a previously allocated #GstAudioResampler @resampler.
+ *
+ * Since: 1.6
+ */
+void
+gst_audio_resampler_free (GstAudioResampler * resampler)
+{
+ g_return_if_fail (resampler != NULL);
+
+ g_free (resampler->taps);
+ g_free (resampler->coeff);
+ g_free (resampler->tmpcoeff);
+ g_free (resampler->samples);
+ g_free (resampler->sbuf);
+ if (resampler->options)
+ gst_structure_free (resampler->options);
+ g_slice_free (GstAudioResampler, resampler);
+}
+
+static inline gsize
+calc_out (GstAudioResampler * resampler, gsize in)
+{
+ return ((in * resampler->out_rate -
+ resampler->samp_phase) / resampler->in_rate) + 1;
+}
+
+/**
+ * gst_audio_resampler_get_out_frames:
+ * @resampler: a #GstAudioResampler
+ * @in_frames: number of input frames
+ *
+ * Get the number of output frames that would be currently available when
+ * @in_frames are given to @resampler.
+ *
+ * Returns: The number of frames that would be availabe after giving
+ * @in_frames as input to @resampler.
+ */
+gsize
+gst_audio_resampler_get_out_frames (GstAudioResampler * resampler,
+ gsize in_frames)
+{
+ gsize need, avail;
+
+ g_return_val_if_fail (resampler != NULL, 0);
+
+ need = resampler->n_taps + resampler->samp_index + resampler->skip;
+ avail = resampler->samples_avail + in_frames;
+ if (avail < need)
+ return 0;
+
+ return calc_out (resampler, avail - need);
+}
+
+/**
+ * gst_audio_resampler_get_in_frames:
+ * @resampler: a #GstAudioResampler
+ * @out_frames: number of input frames
+ *
+ * Get the number of input frames that would currently be needed
+ * to produce @out_frames from @resampler.
+ *
+ * Returns: The number of input frames needed for producing
+ * @out_frames of data from @resampler.
+ */
+gsize
+gst_audio_resampler_get_in_frames (GstAudioResampler * resampler,
+ gsize out_frames)
+{
+ gsize in_frames;
+
+ g_return_val_if_fail (resampler != NULL, 0);
+
+ in_frames =
+ (resampler->samp_phase +
+ out_frames * resampler->samp_frac) / resampler->out_rate;
+ in_frames += out_frames * resampler->samp_inc;
+
+ return in_frames;
+}
+
+/**
+ * gst_audio_resampler_get_max_latency:
+ * @resampler: a #GstAudioResampler
+ *
+ * Get the maximum number of input samples that the resampler would
+ * need before producing output.
+ *
+ * Returns: the latency of @resampler as expressed in the number of
+ * frames.
+ */
+gsize
+gst_audio_resampler_get_max_latency (GstAudioResampler * resampler)
+{
+ g_return_val_if_fail (resampler != NULL, 0);
+
+ return resampler->n_taps / 2;
+}
+
+/* make the buffers to hold the (deinterleaved) samples */
+static inline gpointer *
+get_sample_bufs (GstAudioResampler * resampler, gsize need)
+{
+ if (resampler->samples_len < need) {
+ guint c, channels = resampler->channels;
+ GST_LOG ("realloc %d -> %d", (gint) resampler->samples_len, (gint) need);
+ /* FIXME, move history */
+ resampler->samples = g_realloc (resampler->samples, need * resampler->bpf);
+ resampler->samples_len = need;
+ /* set up new pointers */
+ for (c = 0; c < channels; c++)
+ resampler->sbuf[c] =
+ (gint8 *) resampler->samples +
+ (c * resampler->samples_len * resampler->bps);
+ }
+ return resampler->sbuf;
+}
+
+/**
+ * gst_audio_resampler_resample:
+ * @resampler: a #GstAudioResampler
+ * @in: input samples
+ * @in_frames: number of input frames
+ * @out: output samples
+ * @out_frames: maximum output frames
+ * @consumed: number of frames consumed
+ * @produced: number of frames produced
+ *
+ * Perform resampling on @in_frames frames in @in and write at most
+ * @out_frames of frames to @out.
+ *
+ * In case the samples are interleaved, @in and @out must point to an
+ * array with a single element pointing to a block of interleaved samples.
+ *
+ * If non-interleaved samples are used, @in and @out must point to an
+ * array with pointers to memory blocks, one for each channel.
+ *
+ * @in may be %NULL, in which case @in_frames of 0 samples are pushed
+ * into the resampler.
+ *
+ * The number of frames consumed is returned in @consumed and can be
+ * less than @in_frames due to latency of the resampler or because
+ * the number of samples produced equals @out_frames.
+ *
+ * The number of frames produced is returned in @produced.
+ */
+void
+gst_audio_resampler_resample (GstAudioResampler * resampler,
+ gpointer in[], gsize in_frames, gpointer out[], gsize out_frames,
+ gsize * consumed, gsize * produced)
+{
+ gsize samples_avail;
+ gsize out2, need;
+ gpointer *sbuf;
+
+ /* do sample skipping */
+ if (resampler->skip >= in_frames) {
+ /* we need tp skip all input */
+ resampler->skip -= in_frames;
+ *consumed = in_frames;
+ *produced = 0;
+ return;
+ }
+ /* skip the last samples by advancing the sample index */
+ resampler->samp_index += resampler->skip;
+
+ samples_avail = resampler->samples_avail;
+
+ /* make sure we have enough space to copy our samples */
+ sbuf = get_sample_bufs (resampler, in_frames + samples_avail);
+
+ /* copy/deinterleave the samples */
+ resampler->deinterleave (resampler, sbuf, in, in_frames);
+
+ /* update new amount of samples in our buffer */
+ resampler->samples_avail = samples_avail += in_frames;
+
+ need = resampler->n_taps + resampler->samp_index;
+ if (samples_avail < need) {
+ /* not enough samples to start */
+ *consumed = in_frames;
+ *produced = 0;
+ return;
+ }
+
+ if (resampler->filling) {
+ /* if we are filling up our history duplicate the samples to the left */
+ resampler->mirror (resampler, sbuf);
+ resampler->filling = FALSE;
+ }
+
+ /* calculate maximum number of available output samples */
+ out2 = calc_out (resampler, samples_avail - need);
+ out_frames = MIN (out2, out_frames);
+
+ /* resample all channels */
+ resampler->resample (resampler, sbuf, samples_avail, out, out_frames,
+ consumed, produced, TRUE);
+
+ GST_LOG ("in %" G_GSIZE_FORMAT ", used %" G_GSIZE_FORMAT ", consumed %"
+ G_GSIZE_FORMAT ", produced %" G_GSIZE_FORMAT, in_frames, samples_avail,
+ *consumed, *produced);
+
+ /* update pointers */
+ if (*consumed > 0) {
+ gssize left = samples_avail - *consumed;
+ if (left > 0) {
+ /* we consumed part of our samples */
+ resampler->samples_avail = left;
+ } else {
+ /* we consumed all our samples, empty our buffers */
+ resampler->samples_avail = 0;
+ resampler->skip = -left;
+ }
+ /* we always consume everything */
+ *consumed = in_frames;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2015> Wim Taymans <wim.taymans@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_AUDIO_RESAMPLER_H__
+#define __GST_AUDIO_RESAMPLER_H__
+
+#include <gst/gst.h>
+#include <gst/audio/audio.h>
+
+G_BEGIN_DECLS
+
+typedef struct _GstAudioResampler GstAudioResampler;
+
+/**
+ * GST_AUDIO_RESAMPLER_OPT_CUTOFF
+ *
+ * G_TYPE_DOUBLE, Cutoff parameter for the filter. 0.940 is the default.
+ */
+#define GST_AUDIO_RESAMPLER_OPT_CUTOFF "GstAudioResampler.cutoff"
+/**
+ * GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUTATION
+ *
+ * G_TYPE_DOUBLE, stopband attenuation in debibels. The attenutation
+ * after the stopband for the kaiser window. 85 dB is the default.
+ */
+#define GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION "GstAudioResampler.stop-attenutation"
+/**
+ * GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH
+ *
+ * G_TYPE_DOUBLE, transition bandwidth. The width of the
+ * transition band for the kaiser window. 0.087 is the default.
+ */
+#define GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH "GstAudioResampler.transition-bandwidth"
+
+/**
+ * GST_AUDIO_RESAMPLER_OPT_CUBIC_B:
+ *
+ * G_TYPE_DOUBLE, B parameter of the cubic filter.
+ * Values between 0.0 and 2.0 are accepted. 1.0 is the default.
+ *
+ * Below are some values of popular filters:
+ * B C
+ * Hermite 0.0 0.0
+ * Spline 1.0 0.0
+ * Catmull-Rom 0.0 1/2
+ */
+#define GST_AUDIO_RESAMPLER_OPT_CUBIC_B "GstAudioResampler.cubic-b"
+/**
+ * GST_AUDIO_RESAMPLER_OPT_CUBIC_C:
+ *
+ * G_TYPE_DOUBLE, C parameter of the cubic filter.
+ * Values between 0.0 and 2.0 are accepted. 0.0 is the default.
+ *
+ * See #GST_AUDIO_RESAMPLER_OPT_CUBIC_B for some more common values
+ */
+#define GST_AUDIO_RESAMPLER_OPT_CUBIC_C "GstAudioResampler.cubic-c"
+
+/**
+ * GST_AUDIO_RESAMPLER_OPT_N_TAPS:
+ *
+ * G_TYPE_INT: the number of taps to use for the filter.
+ * 0 is the default and selects the taps automatically.
+ */
+#define GST_AUDIO_RESAMPLER_OPT_N_TAPS "GstAudioResampler.n-taps"
+
+/**
+ * GstAudioResamplerFilterMode:
+ * @GST_AUDIO_RESAMPLER_FILTER_MODE_INTERPOLATED: Use interpolated filter tables. This
+ * uses less memory but more CPU and is slightly less accurate.
+ * @GST_AUDIO_RESAMPLER_FILTER_MODE_FULL: Use full filter table. This uses more memory
+ * but less CPU.
+ * @GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO: Automatically choose between interpolated
+ * and full filter tables.
+ *
+ * Select for the filter tables should be set up.
+ */
+typedef enum {
+ GST_AUDIO_RESAMPLER_FILTER_MODE_INTERPOLATED = (0),
+ GST_AUDIO_RESAMPLER_FILTER_MODE_FULL,
+ GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO,
+} GstAudioResamplerFilterMode;
+/**
+ * GST_AUDIO_RESAMPLER_OPT_FILTER_MODE:
+ *
+ * GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE: how the filter tables should be
+ * constructed.
+ * GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is the default.
+ */
+#define GST_AUDIO_RESAMPLER_OPT_FILTER_MODE "GstAudioResampler.filter-mode"
+/**
+ * GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD:
+ *
+ * G_TYPE_UINT: the amount of memory to use for full filter tables before
+ * switching to interpolated filter tables.
+ * 1048576 is the default.
+ */
+#define GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD "GstAudioResampler.filter-mode-threshold"
+
+/**
+ * GstAudioResamplerMethod:
+ * @GST_AUDIO_RESAMPLER_METHOD_NEAREST: Duplicates the samples when
+ * upsampling and drops when downsampling
+ * @GST_AUDIO_RESAMPLER_METHOD_LINEAR: Uses linear interpolation to reconstruct
+ * missing samples and averaging to downsample
+ * @GST_AUDIO_RESAMPLER_METHOD_CUBIC: Uses cubic interpolation
+ * @GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL: Uses Blackman-Nuttall windowed sinc interpolation
+ * @GST_AUDIO_RESAMPLER_METHOD_KAISER: Uses Kaiser windowed sinc interpolation
+ *
+ * Different subsampling and upsampling methods
+ *
+ * Since: 1.6
+ */
+typedef enum {
+ GST_AUDIO_RESAMPLER_METHOD_NEAREST,
+ GST_AUDIO_RESAMPLER_METHOD_LINEAR,
+ GST_AUDIO_RESAMPLER_METHOD_CUBIC,
+ GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL,
+ GST_AUDIO_RESAMPLER_METHOD_KAISER
+} GstAudioResamplerMethod;
+
+/**
+ * GstAudioResamplerFlags:
+ * @GST_AUDIO_RESAMPLER_FLAG_NONE: no flags
+ * @GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED: samples are non-interleaved. an array
+ * of blocks of samples, one for each channel, should be passed to the resample
+ * function.
+ * @GST_AUDIO_RESAMPLER_FLAG_VARIABLE_RATE: allow arbitrary sample rate changes.
+ *
+ * Different resampler flags.
+ */
+typedef enum {
+ GST_AUDIO_RESAMPLER_FLAG_NONE = (0),
+ GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED = (1 << 0),
+ GST_AUDIO_RESAMPLER_FLAG_VARIABLE_RATE = (1 << 1),
+} GstAudioResamplerFlags;
+
+#define GST_AUDIO_RESAMPLER_QUALITY_MIN 0
+#define GST_AUDIO_RESAMPLER_QUALITY_MAX 10
+#define GST_AUDIO_RESAMPLER_QUALITY_DEFAULT 4
+
+void gst_audio_resampler_options_set_quality (GstAudioResamplerMethod method,
+ guint quality,
+ guint in_rate, guint out_rate,
+ GstStructure *options);
+
+GstAudioResampler * gst_audio_resampler_new (GstAudioResamplerMethod method,
+ GstAudioResamplerFlags flags,
+ GstAudioFormat format, guint channels,
+ guint in_rate, guint out_rate,
+ GstStructure *options);
+void gst_audio_resampler_free (GstAudioResampler *resampler);
+
+
+gboolean gst_audio_resampler_update (GstAudioResampler *resampler,
+ guint in_rate, guint out_rate,
+ GstStructure *options);
+
+gsize gst_audio_resampler_get_out_frames (GstAudioResampler *resampler,
+ gsize in_frames);
+gsize gst_audio_resampler_get_in_frames (GstAudioResampler *resampler,
+ gsize out_frames);
+
+gsize gst_audio_resampler_get_max_latency (GstAudioResampler *resampler);
+
+void gst_audio_resampler_resample (GstAudioResampler * resampler,
+ gpointer in[], gsize in_frames,
+ gpointer out[], gsize out_frames,
+ gsize *produced, gsize *consumed);
+
+G_END_DECLS
+
+#endif /* __GST_AUDIO_RESAMPLER_H__ */
#include <gst/audio/audio-info.h>
#include <gst/audio/audio-quantize.h>
#include <gst/audio/audio-converter.h>
+#include <gst/audio/audio-resampler.h>
G_BEGIN_DECLS
--- /dev/null
+/* Copyright(C) 1996 Takuya OOURA
+
+You may use, copy, modify this code for any purpose and
+without fee.
+
+Package home: http://www.kurims.kyoto-u.ac.jp/~ooura/bessel.html
+*/
+
+/* Bessel I_0(x) function in double precision */
+
+#include <math.h>
+
+static double
+dbesi0 (double x)
+{
+ int k;
+ double w, t, y;
+ static double a[65] = {
+ 8.5246820682016865877e-11, 2.5966600546497407288e-9,
+ 7.9689994568640180274e-8, 1.9906710409667748239e-6,
+ 4.0312469446528002532e-5, 6.4499871606224265421e-4,
+ 0.0079012345761930579108, 0.071111111109207045212,
+ 0.444444444444724909, 1.7777777777777532045,
+ 4.0000000000000011182, 3.99999999999999998,
+ 1.0000000000000000001,
+ 1.1520919130377195927e-10, 2.2287613013610985225e-9,
+ 8.1903951930694585113e-8, 1.9821560631611544984e-6,
+ 4.0335461940910133184e-5, 6.4495330974432203401e-4,
+ 0.0079013012611467520626, 0.071111038160875566622,
+ 0.44444450319062699316, 1.7777777439146450067,
+ 4.0000000132337935071, 3.9999999968569015366,
+ 1.0000000003426703174,
+ 1.5476870780515238488e-10, 1.2685004214732975355e-9,
+ 9.2776861851114223267e-8, 1.9063070109379044378e-6,
+ 4.0698004389917945832e-5, 6.4370447244298070713e-4,
+ 0.0079044749458444976958, 0.071105052411749363882,
+ 0.44445280640924755082, 1.7777694934432109713,
+ 4.0000055808824003386, 3.9999977081165740932,
+ 1.0000004333949319118,
+ 2.0675200625006793075e-10, -6.1689554705125681442e-10,
+ 1.2436765915401571654e-7, 1.5830429403520613423e-6,
+ 4.2947227560776583326e-5, 6.3249861665073441312e-4,
+ 0.0079454472840953930811, 0.070994327785661860575,
+ 0.44467219586283000332, 1.7774588182255374745,
+ 4.0003038986252717972, 3.9998233869142057195,
+ 1.0000472932961288324,
+ 2.7475684794982708655e-10, -3.8991472076521332023e-9,
+ 1.9730170483976049388e-7, 5.9651531561967674521e-7,
+ 5.1992971474748995357e-5, 5.7327338675433770752e-4,
+ 0.0082293143836530412024, 0.069990934858728039037,
+ 0.44726764292723985087, 1.7726685170014087784,
+ 4.0062907863712704432, 3.9952750700487845355,
+ 1.0016354346654179322
+ };
+ static double b[70] = {
+ 6.7852367144945531383e-8, 4.6266061382821826854e-7,
+ 6.9703135812354071774e-6, 7.6637663462953234134e-5,
+ 7.9113515222612691636e-4, 0.0073401204731103808981,
+ 0.060677114958668837046, 0.43994941411651569622,
+ 2.7420017097661750609, 14.289661921740860534,
+ 59.820609640320710779, 188.78998681199150629,
+ 399.8731367825601118, 427.56411572180478514,
+ 1.8042097874891098754e-7, 1.2277164312044637357e-6,
+ 1.8484393221474274861e-5, 2.0293995900091309208e-4,
+ 0.0020918539850246207459, 0.019375315654033949297,
+ 0.15985869016767185908, 1.1565260527420641724,
+ 7.1896341224206072113, 37.354773811947484532,
+ 155.80993164266268457, 489.5211371158540918,
+ 1030.9147225169564806, 1093.5883545113746958,
+ 4.8017305613187493564e-7, 3.261317843912380074e-6,
+ 4.9073137508166159639e-5, 5.3806506676487583755e-4,
+ 0.0055387918291051866561, 0.051223717488786549025,
+ 0.42190298621367914765, 3.0463625987357355872,
+ 18.895299447327733204, 97.915189029455461554,
+ 407.13940115493494659, 1274.3088990480582632,
+ 2670.9883037012547506, 2815.7166284662544712,
+ 1.2789926338424623394e-6, 8.6718263067604918916e-6,
+ 1.3041508821299929489e-4, 0.001428224737372747892,
+ 0.014684070635768789378, 0.13561403190404185755,
+ 1.1152592585977393953, 8.0387088559465389038,
+ 49.761318895895479206, 257.2684232313529138,
+ 1066.8543146269566231, 3328.3874581009636362,
+ 6948.8586598121634874, 7288.4893398212481055,
+ 3.409350368197032893e-6, 2.3079025203103376076e-5,
+ 3.4691373283901830239e-4, 0.003794994977222908545,
+ 0.038974209677945602145, 0.3594948380414878371,
+ 2.9522878893539528226, 21.246564609514287056,
+ 131.28727387146173141, 677.38107093296675421,
+ 2802.3724744545046518, 8718.5731420798254081,
+ 18141.348781638832286, 18948.925349296308859
+ };
+ static double c[45] = {
+ 2.5568678676452702768e-15, 3.0393953792305924324e-14,
+ 6.3343751991094840009e-13, 1.5041298011833009649e-11,
+ 4.4569436918556541414e-10, 1.746393051427167951e-8,
+ 1.0059224011079852317e-6, 1.0729838945088577089e-4,
+ 0.05150322693642527738,
+ 5.2527963991711562216e-15, 7.202118481421005641e-15,
+ 7.2561421229904797156e-13, 1.482312146673104251e-11,
+ 4.4602670450376245434e-10, 1.7463600061788679671e-8,
+ 1.005922609132234756e-6, 1.0729838937545111487e-4,
+ 0.051503226936437300716,
+ 1.3365917359358069908e-14, -1.2932643065888544835e-13,
+ 1.7450199447905602915e-12, 1.0419051209056979788e-11,
+ 4.58047881980598326e-10, 1.7442405450073548966e-8,
+ 1.0059461453281292278e-6, 1.0729837434500161228e-4,
+ 0.051503226940658446941,
+ 5.3771611477352308649e-14, -1.1396193006413731702e-12,
+ 1.2858641335221653409e-11, -5.9802086004570057703e-11,
+ 7.3666894305929510222e-10, 1.6731837150730356448e-8,
+ 1.0070831435812128922e-6, 1.0729733111203704813e-4,
+ 0.051503227360726294675,
+ 3.7819492084858931093e-14, -4.8600496888588034879e-13,
+ 1.6898350504817224909e-12, 4.5884624327524255865e-11,
+ 1.2521615963377513729e-10, 1.8959658437754727957e-8,
+ 1.0020716710561353622e-6, 1.073037119856927559e-4,
+ 0.05150322383300230775
+ };
+
+ w = fabs (x);
+ if (w < 8.5) {
+ t = w * w * 0.0625;
+ k = 13 * ((int) t);
+ y = (((((((((((a[k] * t + a[k + 1]) * t +
+ a[k + 2]) * t + a[k + 3]) * t +
+ a[k + 4]) * t + a[k + 5]) * t + a[k +
+ 6]) * t + a[k + 7]) * t + a[k + 8]) * t + a[k +
+ 9]) * t + a[k + 10]) * t + a[k + 11]) * t + a[k + 12];
+ } else if (w < 12.5) {
+ k = (int) w;
+ t = w - k;
+ k = 14 * (k - 8);
+ y = ((((((((((((b[k] * t + b[k + 1]) * t + b[k + 2]) * t + b[k + 3]) * t +
+ b[k + 4]) * t + b[k + 5]) * t + b[k +
+ 6]) * t + b[k + 7]) * t + b[k + 8]) * t +
+ b[k + 9]) * t + b[k + 10]) * t + b[k + 11]) * t + b[k +
+ 12]) * t + b[k + 13];
+ } else {
+ t = 60 / w;
+ k = 9 * ((int) t);
+ y = ((((((((c[k] * t + c[k + 1]) * t +
+ c[k + 2]) * t + c[k + 3]) * t + c[k + 4]) * t +
+ c[k + 5]) * t + c[k + 6]) * t + c[k + 7]) * t +
+ c[k + 8]) * sqrt (t) * exp (w);
+ }
+ return y;
+}
endif
libgstaudioresample_la_SOURCES = \
- gstaudioresample.c \
- speex_resampler_int.c \
- speex_resampler_float.c \
- speex_resampler_double.c
+ gstaudioresample.c
+
+nodist_libgstaudioresample_la_SOURCES = $(BUILT_SOURCES)
libgstaudioresample_la_CFLAGS = \
$(GST_PLUGINS_BASE_CFLAGS) \
libgstaudioresample_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = \
- arch.h \
- fixed_arm4.h \
- fixed_arm5e.h \
- fixed_bfin.h \
- fixed_debug.h \
- fixed_generic.h \
- gstaudioresample.h \
- resample.c \
- resample_sse.h \
- resample_neon.h \
- speex_resampler.h \
- speex_resampler_wrapper.h
+ gstaudioresample.h
+++ /dev/null
-/* Copyright (C) 2003 Jean-Marc Valin */
-/**
- @file arch.h
- @brief Various architecture definitions Speex
-*/
-/*
- Redistribution and use in source and binary forms, with or without
- modification, are permitted provided that the following conditions
- are met:
-
- - Redistributions of source code must retain the above copyright
- notice, this list of conditions and the following disclaimer.
-
- - Redistributions in binary form must reproduce the above copyright
- notice, this list of conditions and the following disclaimer in the
- documentation and/or other materials provided with the distribution.
-
- - Neither the name of the Xiph.org Foundation nor the names of its
- contributors may be used to endorse or promote products derived from
- this software without specific prior written permission.
-
- THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
- ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
- LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
- A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
- CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
- EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
- PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
- LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
- NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
- SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#ifndef ARCH_H
-#define ARCH_H
-
-#ifndef SPEEX_VERSION
-#define SPEEX_MAJOR_VERSION 1 /**< Major Speex version. */
-#define SPEEX_MINOR_VERSION 1 /**< Minor Speex version. */
-#define SPEEX_MICRO_VERSION 15 /**< Micro Speex version. */
-#define SPEEX_EXTRA_VERSION "" /**< Extra Speex version. */
-#define SPEEX_VERSION "speex-1.2beta3" /**< Speex version string. */
-#endif
-
-/* A couple test to catch stupid option combinations */
-#ifdef FIXED_POINT
-
-#ifdef FLOATING_POINT
-#error You cannot compile as floating point and fixed point at the same time
-#endif
-#ifdef _USE_SSE
-#error SSE is only for floating-point
-#endif
-#if ((defined (ARM4_ASM)||defined (ARM4_ASM)) && defined(BFIN_ASM)) || (defined (ARM4_ASM)&&defined(ARM5E_ASM))
-#error Make up your mind. What CPU do you have?
-#endif
-#ifdef VORBIS_PSYCHO
-#error Vorbis-psy model currently not implemented in fixed-point
-#endif
-
-#else
-
-#ifndef FLOATING_POINT
-#error You now need to define either FIXED_POINT or FLOATING_POINT
-#endif
-#if defined (ARM4_ASM) || defined(ARM5E_ASM) || defined(BFIN_ASM)
-#error I suppose you can have a [ARM4/ARM5E/Blackfin] that has float instructions?
-#endif
-#ifdef FIXED_POINT_DEBUG
-#error "Don't you think enabling fixed-point is a good thing to do if you want to debug that?"
-#endif
-
-
-#endif
-
-#ifndef OUTSIDE_SPEEX
-#include "../include/speex/speex_types.h"
-#endif
-
-#ifndef ABS
-#define ABS(x) ((x) < 0 ? (-(x)) : (x)) /**< Absolute integer value. */
-#endif
-
-#define ABS16(x) ((x) < 0 ? (-(x)) : (x)) /**< Absolute 16-bit value. */
-#define MIN16(a,b) ((a) < (b) ? (a) : (b)) /**< Maximum 16-bit value. */
-#define MAX16(a,b) ((a) > (b) ? (a) : (b)) /**< Maximum 16-bit value. */
-#define ABS32(x) ((x) < 0 ? (-(x)) : (x)) /**< Absolute 32-bit value. */
-#define MIN32(a,b) ((a) < (b) ? (a) : (b)) /**< Maximum 32-bit value. */
-#define MAX32(a,b) ((a) > (b) ? (a) : (b)) /**< Maximum 32-bit value. */
-
-#ifdef FIXED_POINT
-
-typedef spx_int16_t spx_word16_t;
-typedef spx_int32_t spx_word32_t;
-typedef spx_word32_t spx_mem_t;
-typedef spx_word16_t spx_coef_t;
-typedef spx_word16_t spx_lsp_t;
-typedef spx_word32_t spx_sig_t;
-
-#define Q15ONE 32767
-
-#define LPC_SCALING 8192
-#define SIG_SCALING 16384
-#define LSP_SCALING 8192.
-#define GAMMA_SCALING 32768.
-#define GAIN_SCALING 64
-#define GAIN_SCALING_1 0.015625
-
-#define LPC_SHIFT 13
-#define LSP_SHIFT 13
-#define SIG_SHIFT 14
-#define GAIN_SHIFT 6
-
-#define VERY_SMALL 0
-#define VERY_LARGE32 ((spx_word32_t)2147483647)
-#define VERY_LARGE16 ((spx_word16_t)32767)
-#define Q15_ONE ((spx_word16_t)32767)
-
-
-#ifdef FIXED_DEBUG
-#include "fixed_debug.h"
-#else
-
-#include "fixed_generic.h"
-
-#ifdef ARM5E_ASM
-#include "fixed_arm5e.h"
-#elif defined (ARM4_ASM)
-#include "fixed_arm4.h"
-#elif defined (BFIN_ASM)
-#include "fixed_bfin.h"
-#endif
-
-#endif
-
-
-#else
-
-#ifdef DOUBLE_PRECISION
-typedef double spx_mem_t;
-typedef double spx_coef_t;
-typedef double spx_lsp_t;
-typedef double spx_sig_t;
-typedef double spx_word16_t;
-typedef double spx_word32_t;
-
-#define Q15ONE 1.0
-#define LPC_SCALING 1.
-#define SIG_SCALING 1.
-#define LSP_SCALING 1.
-#define GAMMA_SCALING 1.
-#define GAIN_SCALING 1.
-#define GAIN_SCALING_1 1.
-
-
-#define VERY_SMALL 1e-20
-#define VERY_LARGE32 1e20
-#define VERY_LARGE16 1e20
-#define Q15_ONE ((spx_word16_t)1.)
-#else /* !DOUBLE_PRECISION */
-typedef float spx_mem_t;
-typedef float spx_coef_t;
-typedef float spx_lsp_t;
-typedef float spx_sig_t;
-typedef float spx_word16_t;
-typedef float spx_word32_t;
-
-#define Q15ONE 1.0f
-#define LPC_SCALING 1.f
-#define SIG_SCALING 1.f
-#define LSP_SCALING 1.f
-#define GAMMA_SCALING 1.f
-#define GAIN_SCALING 1.f
-#define GAIN_SCALING_1 1.f
-
-
-#define VERY_SMALL 1e-15f
-#define VERY_LARGE32 1e15f
-#define VERY_LARGE16 1e15f
-#define Q15_ONE ((spx_word16_t)1.f)
-#endif /* DOUBLE_PRECISION */
-
-#define QCONST16(x,bits) (x)
-#define QCONST32(x,bits) (x)
-
-#define NEG16(x) (-(x))
-#define NEG32(x) (-(x))
-#define EXTRACT16(x) (x)
-#define EXTEND32(x) (x)
-#define SHR16(a,shift) (a)
-#define SHL16(a,shift) (a)
-#define SHR32(a,shift) (a)
-#define SHL32(a,shift) (a)
-#define PSHR16(a,shift) (a)
-#define PSHR32(a,shift) (a)
-#define VSHR32(a,shift) (a)
-#define SATURATE16(x,a) (x)
-#define SATURATE32(x,a) (x)
-#define SATURATE32PSHR(x,shift,a) (x)
-
-#define PSHR(a,shift) (a)
-#define SHR(a,shift) (a)
-#define SHL(a,shift) (a)
-#define SATURATE(x,a) (x)
-
-#define ADD16(a,b) ((a)+(b))
-#define SUB16(a,b) ((a)-(b))
-#define ADD32(a,b) ((a)+(b))
-#define SUB32(a,b) ((a)-(b))
-#define MULT16_16_16(a,b) ((a)*(b))
-#define MULT16_16(a,b) ((spx_word32_t)(a)*(spx_word32_t)(b))
-#define MAC16_16(c,a,b) ((c)+(spx_word32_t)(a)*(spx_word32_t)(b))
-
-#define MULT16_32_Q11(a,b) ((a)*(b))
-#define MULT16_32_Q13(a,b) ((a)*(b))
-#define MULT16_32_Q14(a,b) ((a)*(b))
-#define MULT16_32_Q15(a,b) ((a)*(b))
-#define MULT16_32_P15(a,b) ((a)*(b))
-
-#define MAC16_32_Q11(c,a,b) ((c)+(a)*(b))
-#define MAC16_32_Q15(c,a,b) ((c)+(a)*(b))
-
-#define MAC16_16_Q11(c,a,b) ((c)+(a)*(b))
-#define MAC16_16_Q13(c,a,b) ((c)+(a)*(b))
-#define MAC16_16_P13(c,a,b) ((c)+(a)*(b))
-#define MULT16_16_Q11_32(a,b) ((a)*(b))
-#define MULT16_16_Q13(a,b) ((a)*(b))
-#define MULT16_16_Q14(a,b) ((a)*(b))
-#define MULT16_16_Q15(a,b) ((a)*(b))
-#define MULT16_16_P15(a,b) ((a)*(b))
-#define MULT16_16_P13(a,b) ((a)*(b))
-#define MULT16_16_P14(a,b) ((a)*(b))
-
-#define DIV32_16(a,b) (((spx_word32_t)(a))/(spx_word16_t)(b))
-#define PDIV32_16(a,b) (((spx_word32_t)(a))/(spx_word16_t)(b))
-#define DIV32(a,b) (((spx_word32_t)(a))/(spx_word32_t)(b))
-#define PDIV32(a,b) (((spx_word32_t)(a))/(spx_word32_t)(b))
-
-
-#endif
-
-
-#if defined (CONFIG_TI_C54X) || defined (CONFIG_TI_C55X)
-
-/* 2 on TI C5x DSP */
-#define BYTES_PER_CHAR 2
-#define BITS_PER_CHAR 16
-#define LOG2_BITS_PER_CHAR 4
-
-#else
-
-#define BYTES_PER_CHAR 1
-#define BITS_PER_CHAR 8
-#define LOG2_BITS_PER_CHAR 3
-
-#endif
-
-
-
-#ifdef FIXED_DEBUG
-extern long long spx_mips;
-#endif
-
-
-#endif
+++ /dev/null
-/* Copyright (C) 2004 Jean-Marc Valin */
-/**
- @file fixed_arm4.h
- @brief ARM4 fixed-point operations
-*/
-/*
- Redistribution and use in source and binary forms, with or without
- modification, are permitted provided that the following conditions
- are met:
-
- - Redistributions of source code must retain the above copyright
- notice, this list of conditions and the following disclaimer.
-
- - Redistributions in binary form must reproduce the above copyright
- notice, this list of conditions and the following disclaimer in the
- documentation and/or other materials provided with the distribution.
-
- - Neither the name of the Xiph.org Foundation nor the names of its
- contributors may be used to endorse or promote products derived from
- this software without specific prior written permission.
-
- THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
- ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
- LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
- A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
- CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
- EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
- PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
- LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
- NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
- SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#ifndef FIXED_ARM4_H
-#define FIXED_ARM4_H
-
-#undef MULT16_32_Q14
-static inline spx_word32_t MULT16_32_Q14(spx_word16_t x, spx_word32_t y) {
- int res;
- int dummy;
- asm (
- "smull %0,%1,%2,%3 \n\t"
- "mov %0, %0, lsr #14 \n\t"
- "add %0, %0, %1, lsl #18 \n\t"
- : "=&r"(res), "=&r" (dummy)
- : "r"(y),"r"((int)x));
- return(res);
-}
-
-#undef MULT16_32_Q15
-static inline spx_word32_t MULT16_32_Q15(spx_word16_t x, spx_word32_t y) {
- int res;
- int dummy;
- asm (
- "smull %0,%1,%2,%3 \n\t"
- "mov %0, %0, lsr #15 \n\t"
- "add %0, %0, %1, lsl #17 \n\t"
- : "=&r"(res), "=&r" (dummy)
- : "r"(y),"r"((int)x));
- return(res);
-}
-
-#undef DIV32_16
-static inline short DIV32_16(int a, int b)
-{
- int res=0;
- int dead1, dead2, dead3, dead4, dead5;
- __asm__ __volatile__ (
- "\teor %5, %0, %1\n"
- "\tmovs %4, %0\n"
- "\trsbmi %0, %0, #0 \n"
- "\tmovs %4, %1\n"
- "\trsbmi %1, %1, #0 \n"
- "\tmov %4, #1\n"
-
- "\tsubs %3, %0, %1, asl #14 \n"
- "\tmovpl %0, %3 \n"
- "\torrpl %2, %2, %4, asl #14 \n"
-
- "\tsubs %3, %0, %1, asl #13 \n"
- "\tmovpl %0, %3 \n"
- "\torrpl %2, %2, %4, asl #13 \n"
-
- "\tsubs %3, %0, %1, asl #12 \n"
- "\tmovpl %0, %3 \n"
- "\torrpl %2, %2, %4, asl #12 \n"
-
- "\tsubs %3, %0, %1, asl #11 \n"
- "\tmovpl %0, %3 \n"
- "\torrpl %2, %2, %4, asl #11 \n"
-
- "\tsubs %3, %0, %1, asl #10 \n"
- "\tmovpl %0, %3 \n"
- "\torrpl %2, %2, %4, asl #10 \n"
-
- "\tsubs %3, %0, %1, asl #9 \n"
- "\tmovpl %0, %3 \n"
- "\torrpl %2, %2, %4, asl #9 \n"
-
- "\tsubs %3, %0, %1, asl #8 \n"
- "\tmovpl %0, %3 \n"
- "\torrpl %2, %2, %4, asl #8 \n"
-
- "\tsubs %3, %0, %1, asl #7 \n"
- "\tmovpl %0, %3 \n"
- "\torrpl %2, %2, %4, asl #7 \n"
-
- "\tsubs %3, %0, %1, asl #6 \n"
- "\tmovpl %0, %3 \n"
- "\torrpl %2, %2, %4, asl #6 \n"
-
- "\tsubs %3, %0, %1, asl #5 \n"
- "\tmovpl %0, %3 \n"
- "\torrpl %2, %2, %4, asl #5 \n"
-
- "\tsubs %3, %0, %1, asl #4 \n"
- "\tmovpl %0, %3 \n"
- "\torrpl %2, %2, %4, asl #4 \n"
-
- "\tsubs %3, %0, %1, asl #3 \n"
- "\tmovpl %0, %3 \n"
- "\torrpl %2, %2, %4, asl #3 \n"
-
- "\tsubs %3, %0, %1, asl #2 \n"
- "\tmovpl %0, %3 \n"
- "\torrpl %2, %2, %4, asl #2 \n"
-
- "\tsubs %3, %0, %1, asl #1 \n"
- "\tmovpl %0, %3 \n"
- "\torrpl %2, %2, %4, asl #1 \n"
-
- "\tsubs %3, %0, %1 \n"
- "\tmovpl %0, %3 \n"
- "\torrpl %2, %2, %4 \n"
-
- "\tmovs %5, %5, lsr #31 \n"
- "\trsbne %2, %2, #0 \n"
- : "=r" (dead1), "=r" (dead2), "=r" (res),
- "=r" (dead3), "=r" (dead4), "=r" (dead5)
- : "0" (a), "1" (b), "2" (res)
- : "cc"
- );
- return res;
-}
-
-
-#endif
+++ /dev/null
-/* Copyright (C) 2003 Jean-Marc Valin */
-/**
- @file fixed_arm5e.h
- @brief ARM-tuned fixed-point operations
-*/
-/*
- Redistribution and use in source and binary forms, with or without
- modification, are permitted provided that the following conditions
- are met:
-
- - Redistributions of source code must retain the above copyright
- notice, this list of conditions and the following disclaimer.
-
- - Redistributions in binary form must reproduce the above copyright
- notice, this list of conditions and the following disclaimer in the
- documentation and/or other materials provided with the distribution.
-
- - Neither the name of the Xiph.org Foundation nor the names of its
- contributors may be used to endorse or promote products derived from
- this software without specific prior written permission.
-
- THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
- ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
- LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
- A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
- CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
- EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
- PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
- LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
- NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
- SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#ifndef FIXED_ARM5E_H
-#define FIXED_ARM5E_H
-
-#undef MULT16_16
-static inline spx_word32_t MULT16_16(spx_word16_t x, spx_word16_t y) {
- int res;
- asm ("smulbb %0,%1,%2;\n"
- : "=&r"(res)
- : "%r"(x),"r"(y));
- return(res);
-}
-
-#undef MAC16_16
-static inline spx_word32_t MAC16_16(spx_word32_t a, spx_word16_t x, spx_word32_t y) {
- int res;
- asm ("smlabb %0,%1,%2,%3;\n"
- : "=&r"(res)
- : "%r"(x),"r"(y),"r"(a));
- return(res);
-}
-
-#undef MULT16_32_Q15
-static inline spx_word32_t MULT16_32_Q15(spx_word16_t x, spx_word32_t y) {
- int res;
- asm ("smulwb %0,%1,%2;\n"
- : "=&r"(res)
- : "%r"(y<<1),"r"(x));
- return(res);
-}
-
-#undef MAC16_32_Q15
-static inline spx_word32_t MAC16_32_Q15(spx_word32_t a, spx_word16_t x, spx_word32_t y) {
- int res;
- asm ("smlawb %0,%1,%2,%3;\n"
- : "=&r"(res)
- : "%r"(y<<1),"r"(x),"r"(a));
- return(res);
-}
-
-#undef MULT16_32_Q11
-static inline spx_word32_t MULT16_32_Q11(spx_word16_t x, spx_word32_t y) {
- int res;
- asm ("smulwb %0,%1,%2;\n"
- : "=&r"(res)
- : "%r"(y<<5),"r"(x));
- return(res);
-}
-
-#undef MAC16_32_Q11
-static inline spx_word32_t MAC16_32_Q11(spx_word32_t a, spx_word16_t x, spx_word32_t y) {
- int res;
- asm ("smlawb %0,%1,%2,%3;\n"
- : "=&r"(res)
- : "%r"(y<<5),"r"(x),"r"(a));
- return(res);
-}
-
-#undef DIV32_16
-static inline short DIV32_16(int a, int b)
-{
- int res=0;
- int dead1, dead2, dead3, dead4, dead5;
- __asm__ __volatile__ (
- "\teor %5, %0, %1\n"
- "\tmovs %4, %0\n"
- "\trsbmi %0, %0, #0 \n"
- "\tmovs %4, %1\n"
- "\trsbmi %1, %1, #0 \n"
- "\tmov %4, #1\n"
-
- "\tsubs %3, %0, %1, asl #14 \n"
- "\torrpl %2, %2, %4, asl #14 \n"
- "\tmovpl %0, %3 \n"
-
- "\tsubs %3, %0, %1, asl #13 \n"
- "\torrpl %2, %2, %4, asl #13 \n"
- "\tmovpl %0, %3 \n"
-
- "\tsubs %3, %0, %1, asl #12 \n"
- "\torrpl %2, %2, %4, asl #12 \n"
- "\tmovpl %0, %3 \n"
-
- "\tsubs %3, %0, %1, asl #11 \n"
- "\torrpl %2, %2, %4, asl #11 \n"
- "\tmovpl %0, %3 \n"
-
- "\tsubs %3, %0, %1, asl #10 \n"
- "\torrpl %2, %2, %4, asl #10 \n"
- "\tmovpl %0, %3 \n"
-
- "\tsubs %3, %0, %1, asl #9 \n"
- "\torrpl %2, %2, %4, asl #9 \n"
- "\tmovpl %0, %3 \n"
-
- "\tsubs %3, %0, %1, asl #8 \n"
- "\torrpl %2, %2, %4, asl #8 \n"
- "\tmovpl %0, %3 \n"
-
- "\tsubs %3, %0, %1, asl #7 \n"
- "\torrpl %2, %2, %4, asl #7 \n"
- "\tmovpl %0, %3 \n"
-
- "\tsubs %3, %0, %1, asl #6 \n"
- "\torrpl %2, %2, %4, asl #6 \n"
- "\tmovpl %0, %3 \n"
-
- "\tsubs %3, %0, %1, asl #5 \n"
- "\torrpl %2, %2, %4, asl #5 \n"
- "\tmovpl %0, %3 \n"
-
- "\tsubs %3, %0, %1, asl #4 \n"
- "\torrpl %2, %2, %4, asl #4 \n"
- "\tmovpl %0, %3 \n"
-
- "\tsubs %3, %0, %1, asl #3 \n"
- "\torrpl %2, %2, %4, asl #3 \n"
- "\tmovpl %0, %3 \n"
-
- "\tsubs %3, %0, %1, asl #2 \n"
- "\torrpl %2, %2, %4, asl #2 \n"
- "\tmovpl %0, %3 \n"
-
- "\tsubs %3, %0, %1, asl #1 \n"
- "\torrpl %2, %2, %4, asl #1 \n"
- "\tmovpl %0, %3 \n"
-
- "\tsubs %3, %0, %1 \n"
- "\torrpl %2, %2, %4 \n"
- "\tmovpl %0, %3 \n"
-
- "\tmovs %5, %5, lsr #31 \n"
- "\trsbne %2, %2, #0 \n"
- : "=r" (dead1), "=r" (dead2), "=r" (res),
- "=r" (dead3), "=r" (dead4), "=r" (dead5)
- : "0" (a), "1" (b), "2" (res)
- : "memory", "cc"
- );
- return res;
-}
-
-
-
-
-#endif
+++ /dev/null
-/* Copyright (C) 2005 Analog Devices
- Author: Jean-Marc Valin */
-/**
- @file fixed_bfin.h
- @brief Blackfin fixed-point operations
-*/
-/*
- Redistribution and use in source and binary forms, with or without
- modification, are permitted provided that the following conditions
- are met:
-
- - Redistributions of source code must retain the above copyright
- notice, this list of conditions and the following disclaimer.
-
- - Redistributions in binary form must reproduce the above copyright
- notice, this list of conditions and the following disclaimer in the
- documentation and/or other materials provided with the distribution.
-
- - Neither the name of the Xiph.org Foundation nor the names of its
- contributors may be used to endorse or promote products derived from
- this software without specific prior written permission.
-
- THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
- ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
- LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
- A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
- CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
- EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
- PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
- LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
- NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
- SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#ifndef FIXED_BFIN_H
-#define FIXED_BFIN_H
-
-#include "bfin.h"
-
-#undef PDIV32_16
-static inline spx_word16_t PDIV32_16(spx_word32_t a, spx_word16_t b)
-{
- spx_word32_t res, bb;
- bb = b;
- a += b>>1;
- __asm__ (
- "P0 = 15;\n\t"
- "R0 = %1;\n\t"
- "R1 = %2;\n\t"
- //"R0 = R0 + R1;\n\t"
- "R0 <<= 1;\n\t"
- "DIVS (R0, R1);\n\t"
- "LOOP divide%= LC0 = P0;\n\t"
- "LOOP_BEGIN divide%=;\n\t"
- "DIVQ (R0, R1);\n\t"
- "LOOP_END divide%=;\n\t"
- "R0 = R0.L;\n\t"
- "%0 = R0;\n\t"
- : "=m" (res)
- : "m" (a), "m" (bb)
- : "P0", "R0", "R1", "ASTAT" BFIN_HWLOOP0_REGS);
- return res;
-}
-
-#undef DIV32_16
-static inline spx_word16_t DIV32_16(spx_word32_t a, spx_word16_t b)
-{
- spx_word32_t res, bb;
- bb = b;
- /* Make the roundinf consistent with the C version
- (do we need to do that?)*/
- if (a<0)
- a += (b-1);
- __asm__ (
- "P0 = 15;\n\t"
- "R0 = %1;\n\t"
- "R1 = %2;\n\t"
- "R0 <<= 1;\n\t"
- "DIVS (R0, R1);\n\t"
- "LOOP divide%= LC0 = P0;\n\t"
- "LOOP_BEGIN divide%=;\n\t"
- "DIVQ (R0, R1);\n\t"
- "LOOP_END divide%=;\n\t"
- "R0 = R0.L;\n\t"
- "%0 = R0;\n\t"
- : "=m" (res)
- : "m" (a), "m" (bb)
- : "P0", "R0", "R1", "ASTAT" BFIN_HWLOOP0_REGS);
- return res;
-}
-
-#undef MAX16
-static inline spx_word16_t MAX16(spx_word16_t a, spx_word16_t b)
-{
- spx_word32_t res;
- __asm__ (
- "%1 = %1.L (X);\n\t"
- "%2 = %2.L (X);\n\t"
- "%0 = MAX(%1,%2);"
- : "=d" (res)
- : "%d" (a), "d" (b)
- : "ASTAT"
- );
- return res;
-}
-
-#undef MULT16_32_Q15
-static inline spx_word32_t MULT16_32_Q15(spx_word16_t a, spx_word32_t b)
-{
- spx_word32_t res;
- __asm__
- (
- "A1 = %2.L*%1.L (M);\n\t"
- "A1 = A1 >>> 15;\n\t"
- "%0 = (A1 += %2.L*%1.H) ;\n\t"
- : "=&W" (res), "=&d" (b)
- : "d" (a), "1" (b)
- : "A1", "ASTAT"
- );
- return res;
-}
-
-#undef MAC16_32_Q15
-static inline spx_word32_t MAC16_32_Q15(spx_word32_t c, spx_word16_t a, spx_word32_t b)
-{
- spx_word32_t res;
- __asm__
- (
- "A1 = %2.L*%1.L (M);\n\t"
- "A1 = A1 >>> 15;\n\t"
- "%0 = (A1 += %2.L*%1.H);\n\t"
- "%0 = %0 + %4;\n\t"
- : "=&W" (res), "=&d" (b)
- : "d" (a), "1" (b), "d" (c)
- : "A1", "ASTAT"
- );
- return res;
-}
-
-#undef MULT16_32_Q14
-static inline spx_word32_t MULT16_32_Q14(spx_word16_t a, spx_word32_t b)
-{
- spx_word32_t res;
- __asm__
- (
- "%2 <<= 1;\n\t"
- "A1 = %1.L*%2.L (M);\n\t"
- "A1 = A1 >>> 15;\n\t"
- "%0 = (A1 += %1.L*%2.H);\n\t"
- : "=W" (res), "=d" (a), "=d" (b)
- : "1" (a), "2" (b)
- : "A1", "ASTAT"
- );
- return res;
-}
-
-#undef MAC16_32_Q14
-static inline spx_word32_t MAC16_32_Q14(spx_word32_t c, spx_word16_t a, spx_word32_t b)
-{
- spx_word32_t res;
- __asm__
- (
- "%1 <<= 1;\n\t"
- "A1 = %2.L*%1.L (M);\n\t"
- "A1 = A1 >>> 15;\n\t"
- "%0 = (A1 += %2.L*%1.H);\n\t"
- "%0 = %0 + %4;\n\t"
- : "=&W" (res), "=&d" (b)
- : "d" (a), "1" (b), "d" (c)
- : "A1", "ASTAT"
- );
- return res;
-}
-
-#endif
+++ /dev/null
-/* Copyright (C) 2003 Jean-Marc Valin */
-/**
- @file fixed_debug.h
- @brief Fixed-point operations with debugging
-*/
-/*
- Redistribution and use in source and binary forms, with or without
- modification, are permitted provided that the following conditions
- are met:
-
- - Redistributions of source code must retain the above copyright
- notice, this list of conditions and the following disclaimer.
-
- - Redistributions in binary form must reproduce the above copyright
- notice, this list of conditions and the following disclaimer in the
- documentation and/or other materials provided with the distribution.
-
- - Neither the name of the Xiph.org Foundation nor the names of its
- contributors may be used to endorse or promote products derived from
- this software without specific prior written permission.
-
- THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
- ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
- LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
- A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
- CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
- EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
- PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
- LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
- NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
- SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#ifndef FIXED_DEBUG_H
-#define FIXED_DEBUG_H
-
-#include <stdio.h>
-
-extern long long spx_mips;
-#define MIPS_INC spx_mips++,
-
-#define QCONST16(x,bits) ((spx_word16_t)(.5+(x)*(((spx_word32_t)1)<<(bits))))
-#define QCONST32(x,bits) ((spx_word32_t)(.5+(x)*(((spx_word32_t)1)<<(bits))))
-
-
-#define VERIFY_SHORT(x) ((x)<=32767&&(x)>=-32768)
-#define VERIFY_INT(x) ((x)<=2147483647LL&&(x)>=-2147483648LL)
-
-static inline short NEG16(int x)
-{
- int res;
- if (!VERIFY_SHORT(x))
- {
- fprintf (stderr, "NEG16: input is not short: %d\n", (int)x);
- }
- res = -x;
- if (!VERIFY_SHORT(res))
- fprintf (stderr, "NEG16: output is not short: %d\n", (int)res);
- spx_mips++;
- return res;
-}
-static inline int NEG32(long long x)
-{
- long long res;
- if (!VERIFY_INT(x))
- {
- fprintf (stderr, "NEG16: input is not int: %d\n", (int)x);
- }
- res = -x;
- if (!VERIFY_INT(res))
- fprintf (stderr, "NEG16: output is not int: %d\n", (int)res);
- spx_mips++;
- return res;
-}
-
-#define EXTRACT16(x) _EXTRACT16(x, __FILE__, __LINE__)
-static inline short _EXTRACT16(int x, char *file, int line)
-{
- int res;
- if (!VERIFY_SHORT(x))
- {
- fprintf (stderr, "EXTRACT16: input is not short: %d in %s: line %d\n", x, file, line);
- }
- res = x;
- spx_mips++;
- return res;
-}
-
-#define EXTEND32(x) _EXTEND32(x, __FILE__, __LINE__)
-static inline int _EXTEND32(int x, char *file, int line)
-{
- int res;
- if (!VERIFY_SHORT(x))
- {
- fprintf (stderr, "EXTEND32: input is not short: %d in %s: line %d\n", x, file, line);
- }
- res = x;
- spx_mips++;
- return res;
-}
-
-#define SHR16(a, shift) _SHR16(a, shift, __FILE__, __LINE__)
-static inline short _SHR16(int a, int shift, char *file, int line)
-{
- int res;
- if (!VERIFY_SHORT(a) || !VERIFY_SHORT(shift))
- {
- fprintf (stderr, "SHR16: inputs are not short: %d >> %d in %s: line %d\n", a, shift, file, line);
- }
- res = a>>shift;
- if (!VERIFY_SHORT(res))
- fprintf (stderr, "SHR16: output is not short: %d in %s: line %d\n", res, file, line);
- spx_mips++;
- return res;
-}
-#define SHL16(a, shift) _SHL16(a, shift, __FILE__, __LINE__)
-static inline short _SHL16(int a, int shift, char *file, int line)
-{
- int res;
- if (!VERIFY_SHORT(a) || !VERIFY_SHORT(shift))
- {
- fprintf (stderr, "SHL16: inputs are not short: %d %d in %s: line %d\n", a, shift, file, line);
- }
- res = a<<shift;
- if (!VERIFY_SHORT(res))
- fprintf (stderr, "SHL16: output is not short: %d in %s: line %d\n", res, file, line);
- spx_mips++;
- return res;
-}
-
-static inline int SHR32(long long a, int shift)
-{
- long long res;
- if (!VERIFY_INT(a) || !VERIFY_SHORT(shift))
- {
- fprintf (stderr, "SHR32: inputs are not int: %d %d\n", (int)a, shift);
- }
- res = a>>shift;
- if (!VERIFY_INT(res))
- {
- fprintf (stderr, "SHR32: output is not int: %d\n", (int)res);
- }
- spx_mips++;
- return res;
-}
-static inline int SHL32(long long a, int shift)
-{
- long long res;
- if (!VERIFY_INT(a) || !VERIFY_SHORT(shift))
- {
- fprintf (stderr, "SHL32: inputs are not int: %d %d\n", (int)a, shift);
- }
- res = a<<shift;
- if (!VERIFY_INT(res))
- {
- fprintf (stderr, "SHL32: output is not int: %d\n", (int)res);
- }
- spx_mips++;
- return res;
-}
-
-#define PSHR16(a,shift) (SHR16(ADD16((a),((1<<((shift))>>1))),shift))
-#define PSHR32(a,shift) (SHR32(ADD32((a),((EXTEND32(1)<<((shift))>>1))),shift))
-#define VSHR32(a, shift) (((shift)>0) ? SHR32(a, shift) : SHL32(a, -(shift)))
-
-#define SATURATE16(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x)))
-#define SATURATE32(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x)))
-
-//#define SHR(a,shift) ((a) >> (shift))
-//#define SHL(a,shift) ((a) << (shift))
-
-#define ADD16(a, b) _ADD16(a, b, __FILE__, __LINE__)
-static inline short _ADD16(int a, int b, char *file, int line)
-{
- int res;
- if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b))
- {
- fprintf (stderr, "ADD16: inputs are not short: %d %d in %s: line %d\n", a, b, file, line);
- }
- res = a+b;
- if (!VERIFY_SHORT(res))
- {
- fprintf (stderr, "ADD16: output is not short: %d+%d=%d in %s: line %d\n", a,b,res, file, line);
- }
- spx_mips++;
- return res;
-}
-
-#define SUB16(a, b) _SUB16(a, b, __FILE__, __LINE__)
-static inline short _SUB16(int a, int b, char *file, int line)
-{
- int res;
- if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b))
- {
- fprintf (stderr, "SUB16: inputs are not short: %d %d in %s: line %d\n", a, b, file, line);
- }
- res = a-b;
- if (!VERIFY_SHORT(res))
- fprintf (stderr, "SUB16: output is not short: %d in %s: line %d\n", res, file, line);
- spx_mips++;
- return res;
-}
-
-#define ADD32(a, b) _ADD32(a, b, __FILE__, __LINE__)
-static inline int _ADD32(long long a, long long b, char *file, int line)
-{
- long long res;
- if (!VERIFY_INT(a) || !VERIFY_INT(b))
- {
- fprintf (stderr, "ADD32: inputs are not int: %d %d in %s: line %d\n", (int)a, (int)b, file, line);
- }
- res = a+b;
- if (!VERIFY_INT(res))
- {
- fprintf (stderr, "ADD32: output is not int: %d in %s: line %d\n", (int)res, file, line);
- }
- spx_mips++;
- return res;
-}
-
-static inline int SUB32(long long a, long long b)
-{
- long long res;
- if (!VERIFY_INT(a) || !VERIFY_INT(b))
- {
- fprintf (stderr, "SUB32: inputs are not int: %d %d\n", (int)a, (int)b);
- }
- res = a-b;
- if (!VERIFY_INT(res))
- fprintf (stderr, "SUB32: output is not int: %d\n", (int)res);
- spx_mips++;
- return res;
-}
-
-#define ADD64(a,b) (MIPS_INC(a)+(b))
-
-/* result fits in 16 bits */
-static inline short MULT16_16_16(int a, int b)
-{
- int res;
- if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b))
- {
- fprintf (stderr, "MULT16_16_16: inputs are not short: %d %d\n", a, b);
- }
- res = a*b;
- if (!VERIFY_SHORT(res))
- fprintf (stderr, "MULT16_16_16: output is not short: %d\n", res);
- spx_mips++;
- return res;
-}
-
-#define MULT16_16(a, b) _MULT16_16(a, b, __FILE__, __LINE__)
-static inline int _MULT16_16(int a, int b, char *file, int line)
-{
- long long res;
- if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b))
- {
- fprintf (stderr, "MULT16_16: inputs are not short: %d %d in %s: line %d\n", a, b, file, line);
- }
- res = ((long long)a)*b;
- if (!VERIFY_INT(res))
- fprintf (stderr, "MULT16_16: output is not int: %d in %s: line %d\n", (int)res, file, line);
- spx_mips++;
- return res;
-}
-
-#define MAC16_16(c,a,b) (spx_mips--,ADD32((c),MULT16_16((a),(b))))
-#define MAC16_16_Q11(c,a,b) (EXTRACT16(ADD16((c),EXTRACT16(SHR32(MULT16_16((a),(b)),11)))))
-#define MAC16_16_Q13(c,a,b) (EXTRACT16(ADD16((c),EXTRACT16(SHR32(MULT16_16((a),(b)),13)))))
-#define MAC16_16_P13(c,a,b) (EXTRACT16(ADD32((c),SHR32(ADD32(4096,MULT16_16((a),(b))),13))))
-
-
-#define MULT16_32_QX(a, b, Q) _MULT16_32_QX(a, b, Q, __FILE__, __LINE__)
-static inline int _MULT16_32_QX(int a, long long b, int Q, char *file, int line)
-{
- long long res;
- if (!VERIFY_SHORT(a) || !VERIFY_INT(b))
- {
- fprintf (stderr, "MULT16_32_Q%d: inputs are not short+int: %d %d in %s: line %d\n", Q, (int)a, (int)b, file, line);
- }
- if (ABS32(b)>=(EXTEND32(1)<<(15+Q)))
- fprintf (stderr, "MULT16_32_Q%d: second operand too large: %d %d in %s: line %d\n", Q, (int)a, (int)b, file, line);
- res = (((long long)a)*(long long)b) >> Q;
- if (!VERIFY_INT(res))
- fprintf (stderr, "MULT16_32_Q%d: output is not int: %d*%d=%d in %s: line %d\n", Q, (int)a, (int)b,(int)res, file, line);
- spx_mips+=5;
- return res;
-}
-
-static inline int MULT16_32_PX(int a, long long b, int Q)
-{
- long long res;
- if (!VERIFY_SHORT(a) || !VERIFY_INT(b))
- {
- fprintf (stderr, "MULT16_32_P%d: inputs are not short+int: %d %d\n", Q, (int)a, (int)b);
- }
- if (ABS32(b)>=(EXTEND32(1)<<(15+Q)))
- fprintf (stderr, "MULT16_32_Q%d: second operand too large: %d %d\n", Q, (int)a, (int)b);
- res = ((((long long)a)*(long long)b) + ((EXTEND32(1)<<Q)>>1))>> Q;
- if (!VERIFY_INT(res))
- fprintf (stderr, "MULT16_32_P%d: output is not int: %d*%d=%d\n", Q, (int)a, (int)b,(int)res);
- spx_mips+=5;
- return res;
-}
-
-
-#define MULT16_32_Q11(a,b) MULT16_32_QX(a,b,11)
-#define MAC16_32_Q11(c,a,b) ADD32((c),MULT16_32_Q11((a),(b)))
-#define MULT16_32_Q12(a,b) MULT16_32_QX(a,b,12)
-#define MULT16_32_Q13(a,b) MULT16_32_QX(a,b,13)
-#define MULT16_32_Q14(a,b) MULT16_32_QX(a,b,14)
-#define MULT16_32_Q15(a,b) MULT16_32_QX(a,b,15)
-#define MULT16_32_P15(a,b) MULT16_32_PX(a,b,15)
-#define MAC16_32_Q15(c,a,b) ADD32((c),MULT16_32_Q15((a),(b)))
-
-static inline int SATURATE(int a, int b)
-{
- if (a>b)
- a=b;
- if (a<-b)
- a = -b;
- return a;
-}
-
-static inline int MULT16_16_Q11_32(int a, int b)
-{
- long long res;
- if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b))
- {
- fprintf (stderr, "MULT16_16_Q11: inputs are not short: %d %d\n", a, b);
- }
- res = ((long long)a)*b;
- res >>= 11;
- if (!VERIFY_INT(res))
- fprintf (stderr, "MULT16_16_Q11: output is not short: %d*%d=%d\n", (int)a, (int)b, (int)res);
- spx_mips+=3;
- return res;
-}
-static inline short MULT16_16_Q13(int a, int b)
-{
- long long res;
- if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b))
- {
- fprintf (stderr, "MULT16_16_Q13: inputs are not short: %d %d\n", a, b);
- }
- res = ((long long)a)*b;
- res >>= 13;
- if (!VERIFY_SHORT(res))
- fprintf (stderr, "MULT16_16_Q13: output is not short: %d*%d=%d\n", a, b, (int)res);
- spx_mips+=3;
- return res;
-}
-static inline short MULT16_16_Q14(int a, int b)
-{
- long long res;
- if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b))
- {
- fprintf (stderr, "MULT16_16_Q14: inputs are not short: %d %d\n", a, b);
- }
- res = ((long long)a)*b;
- res >>= 14;
- if (!VERIFY_SHORT(res))
- fprintf (stderr, "MULT16_16_Q14: output is not short: %d\n", (int)res);
- spx_mips+=3;
- return res;
-}
-static inline short MULT16_16_Q15(int a, int b)
-{
- long long res;
- if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b))
- {
- fprintf (stderr, "MULT16_16_Q15: inputs are not short: %d %d\n", a, b);
- }
- res = ((long long)a)*b;
- res >>= 15;
- if (!VERIFY_SHORT(res))
- {
- fprintf (stderr, "MULT16_16_Q15: output is not short: %d\n", (int)res);
- }
- spx_mips+=3;
- return res;
-}
-
-static inline short MULT16_16_P13(int a, int b)
-{
- long long res;
- if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b))
- {
- fprintf (stderr, "MULT16_16_P13: inputs are not short: %d %d\n", a, b);
- }
- res = ((long long)a)*b;
- res += 4096;
- if (!VERIFY_INT(res))
- fprintf (stderr, "MULT16_16_P13: overflow: %d*%d=%d\n", a, b, (int)res);
- res >>= 13;
- if (!VERIFY_SHORT(res))
- fprintf (stderr, "MULT16_16_P13: output is not short: %d*%d=%d\n", a, b, (int)res);
- spx_mips+=4;
- return res;
-}
-static inline short MULT16_16_P14(int a, int b)
-{
- long long res;
- if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b))
- {
- fprintf (stderr, "MULT16_16_P14: inputs are not short: %d %d\n", a, b);
- }
- res = ((long long)a)*b;
- res += 8192;
- if (!VERIFY_INT(res))
- fprintf (stderr, "MULT16_16_P14: overflow: %d*%d=%d\n", a, b, (int)res);
- res >>= 14;
- if (!VERIFY_SHORT(res))
- fprintf (stderr, "MULT16_16_P14: output is not short: %d*%d=%d\n", a, b, (int)res);
- spx_mips+=4;
- return res;
-}
-static inline short MULT16_16_P15(int a, int b)
-{
- long long res;
- if (!VERIFY_SHORT(a) || !VERIFY_SHORT(b))
- {
- fprintf (stderr, "MULT16_16_P15: inputs are not short: %d %d\n", a, b);
- }
- res = ((long long)a)*b;
- res += 16384;
- if (!VERIFY_INT(res))
- fprintf (stderr, "MULT16_16_P15: overflow: %d*%d=%d\n", a, b, (int)res);
- res >>= 15;
- if (!VERIFY_SHORT(res))
- fprintf (stderr, "MULT16_16_P15: output is not short: %d*%d=%d\n", a, b, (int)res);
- spx_mips+=4;
- return res;
-}
-
-#define DIV32_16(a, b) _DIV32_16(a, b, __FILE__, __LINE__)
-
-static inline int _DIV32_16(long long a, long long b, char *file, int line)
-{
- long long res;
- if (b==0)
- {
- fprintf(stderr, "DIV32_16: divide by zero: %d/%d in %s: line %d\n", (int)a, (int)b, file, line);
- return 0;
- }
- if (!VERIFY_INT(a) || !VERIFY_SHORT(b))
- {
- fprintf (stderr, "DIV32_16: inputs are not int/short: %d %d in %s: line %d\n", (int)a, (int)b, file, line);
- }
- res = a/b;
- if (!VERIFY_SHORT(res))
- {
- fprintf (stderr, "DIV32_16: output is not short: %d / %d = %d in %s: line %d\n", (int)a,(int)b,(int)res, file, line);
- if (res>32767)
- res = 32767;
- if (res<-32768)
- res = -32768;
- }
- spx_mips+=20;
- return res;
-}
-
-#define DIV32(a, b) _DIV32(a, b, __FILE__, __LINE__)
-static inline int _DIV32(long long a, long long b, char *file, int line)
-{
- long long res;
- if (b==0)
- {
- fprintf(stderr, "DIV32: divide by zero: %d/%d in %s: line %d\n", (int)a, (int)b, file, line);
- return 0;
- }
-
- if (!VERIFY_INT(a) || !VERIFY_INT(b))
- {
- fprintf (stderr, "DIV32: inputs are not int/short: %d %d in %s: line %d\n", (int)a, (int)b, file, line);
- }
- res = a/b;
- if (!VERIFY_INT(res))
- fprintf (stderr, "DIV32: output is not int: %d in %s: line %d\n", (int)res, file, line);
- spx_mips+=36;
- return res;
-}
-#define PDIV32(a,b) DIV32(ADD32((a),(b)>>1),b)
-#define PDIV32_16(a,b) DIV32_16(ADD32((a),(b)>>1),b)
-
-#endif
+++ /dev/null
-/* Copyright (C) 2003 Jean-Marc Valin */
-/**
- @file fixed_generic.h
- @brief Generic fixed-point operations
-*/
-/*
- Redistribution and use in source and binary forms, with or without
- modification, are permitted provided that the following conditions
- are met:
-
- - Redistributions of source code must retain the above copyright
- notice, this list of conditions and the following disclaimer.
-
- - Redistributions in binary form must reproduce the above copyright
- notice, this list of conditions and the following disclaimer in the
- documentation and/or other materials provided with the distribution.
-
- - Neither the name of the Xiph.org Foundation nor the names of its
- contributors may be used to endorse or promote products derived from
- this software without specific prior written permission.
-
- THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
- ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
- LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
- A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
- CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
- EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
- PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
- LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
- NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
- SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#ifndef FIXED_GENERIC_H
-#define FIXED_GENERIC_H
-
-#define QCONST16(x,bits) ((spx_word16_t)(.5+(x)*(((spx_word32_t)1)<<(bits))))
-#define QCONST32(x,bits) ((spx_word32_t)(.5+(x)*(((spx_word32_t)1)<<(bits))))
-
-#define NEG16(x) (-(x))
-#define NEG32(x) (-(x))
-#define EXTRACT16(x) ((spx_word16_t)(x))
-#define EXTEND32(x) ((spx_word32_t)(x))
-#define SHR16(a,shift) ((a) >> (shift))
-#define SHL16(a,shift) ((a) << (shift))
-#define SHR32(a,shift) ((a) >> (shift))
-#define SHL32(a,shift) ((a) << (shift))
-#define PSHR16(a,shift) (SHR16((a)+((1<<((shift))>>1)),shift))
-#define PSHR32(a,shift) (SHR32((a)+((EXTEND32(1)<<((shift))>>1)),shift))
-#define VSHR32(a, shift) (((shift)>0) ? SHR32(a, shift) : SHL32(a, -(shift)))
-#define SATURATE16(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x)))
-#define SATURATE32(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x)))
-
-#define SATURATE32PSHR(x,shift,a) (((x)>=(SHL32(a,shift))) ? (a) : \
- (x)<=-(SHL32(a,shift)) ? -(a) : \
- (PSHR32(x, shift)))
-
-#define SHR(a,shift) ((a) >> (shift))
-#define SHL(a,shift) ((spx_word32_t)(a) << (shift))
-#define PSHR(a,shift) (SHR((a)+((EXTEND32(1)<<((shift))>>1)),shift))
-#define SATURATE(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x)))
-
-
-#define ADD16(a,b) ((spx_word16_t)((spx_word16_t)(a)+(spx_word16_t)(b)))
-#define SUB16(a,b) ((spx_word16_t)(a)-(spx_word16_t)(b))
-#define ADD32(a,b) ((spx_word32_t)(a)+(spx_word32_t)(b))
-#define SUB32(a,b) ((spx_word32_t)(a)-(spx_word32_t)(b))
-
-
-/* result fits in 16 bits */
-#define MULT16_16_16(a,b) ((((spx_word16_t)(a))*((spx_word16_t)(b))))
-
-/* (spx_word32_t)(spx_word16_t) gives TI compiler a hint that it's 16x16->32 multiply */
-#define MULT16_16(a,b) (((spx_word32_t)(spx_word16_t)(a))*((spx_word32_t)(spx_word16_t)(b)))
-
-#define MAC16_16(c,a,b) (ADD32((c),MULT16_16((a),(b))))
-#define MULT16_32_Q12(a,b) ADD32(MULT16_16((a),SHR((b),12)), SHR(MULT16_16((a),((b)&0x00000fff)),12))
-#define MULT16_32_Q13(a,b) ADD32(MULT16_16((a),SHR((b),13)), SHR(MULT16_16((a),((b)&0x00001fff)),13))
-#define MULT16_32_Q14(a,b) ADD32(MULT16_16((a),SHR((b),14)), SHR(MULT16_16((a),((b)&0x00003fff)),14))
-
-#define MULT16_32_Q11(a,b) ADD32(MULT16_16((a),SHR((b),11)), SHR(MULT16_16((a),((b)&0x000007ff)),11))
-#define MAC16_32_Q11(c,a,b) ADD32(c,ADD32(MULT16_16((a),SHR((b),11)), SHR(MULT16_16((a),((b)&0x000007ff)),11)))
-
-#define MULT16_32_P15(a,b) ADD32(MULT16_16((a),SHR((b),15)), PSHR(MULT16_16((a),((b)&0x00007fff)),15))
-#define MULT16_32_Q15(a,b) ADD32(MULT16_16((a),SHR((b),15)), SHR(MULT16_16((a),((b)&0x00007fff)),15))
-#define MAC16_32_Q15(c,a,b) ADD32(c,ADD32(MULT16_16((a),SHR((b),15)), SHR(MULT16_16((a),((b)&0x00007fff)),15)))
-
-
-#define MAC16_16_Q11(c,a,b) (ADD32((c),SHR(MULT16_16((a),(b)),11)))
-#define MAC16_16_Q13(c,a,b) (ADD32((c),SHR(MULT16_16((a),(b)),13)))
-#define MAC16_16_P13(c,a,b) (ADD32((c),SHR(ADD32(4096,MULT16_16((a),(b))),13)))
-
-#define MULT16_16_Q11_32(a,b) (SHR(MULT16_16((a),(b)),11))
-#define MULT16_16_Q13(a,b) (SHR(MULT16_16((a),(b)),13))
-#define MULT16_16_Q14(a,b) (SHR(MULT16_16((a),(b)),14))
-#define MULT16_16_Q15(a,b) (SHR(MULT16_16((a),(b)),15))
-
-#define MULT16_16_P13(a,b) (SHR(ADD32(4096,MULT16_16((a),(b))),13))
-#define MULT16_16_P14(a,b) (SHR(ADD32(8192,MULT16_16((a),(b))),14))
-#define MULT16_16_P15(a,b) (SHR(ADD32(16384,MULT16_16((a),(b))),15))
-
-#define MUL_16_32_R15(a,bh,bl) ADD32(MULT16_16((a),(bh)), SHR(MULT16_16((a),(bl)),15))
-
-#define DIV32_16(a,b) ((spx_word16_t)(((spx_word32_t)(a))/((spx_word16_t)(b))))
-#define PDIV32_16(a,b) ((spx_word16_t)(((spx_word32_t)(a)+((spx_word16_t)(b)>>1))/((spx_word16_t)(b))))
-#define DIV32(a,b) (((spx_word32_t)(a))/((spx_word32_t)(b)))
-#define PDIV32(a,b) (((spx_word32_t)(a)+((spx_word16_t)(b)>>1))/((spx_word32_t)(b)))
-
-#endif
GST_DEBUG_CATEGORY_STATIC (GST_CAT_PERFORMANCE);
#endif
-#define GST_TYPE_SPEEX_RESAMPLER_SINC_FILTER_MODE (speex_resampler_sinc_filter_mode_get_type ())
+#undef USE_SPEEX
+
+#define DEFAULT_QUALITY GST_AUDIO_RESAMPLER_QUALITY_DEFAULT
+#define DEFAULT_SINC_FILTER_MODE GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO
+#define DEFAULT_SINC_FILTER_AUTO_THRESHOLD (1*1048576)
enum
{
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
#define SUPPORTED_CAPS \
- GST_AUDIO_CAPS_MAKE ("{ F32LE, F64LE, S32LE, S24LE, S16LE, S8 }") \
+ GST_AUDIO_CAPS_MAKE ("{ F32LE, F64LE, S32LE, S16LE }") \
", layout = (string) { interleaved, non-interleaved }"
#else
#define SUPPORTED_CAPS \
- GST_AUDIO_CAPS_MAKE ("{ F32BE, F64BE, S32BE, S24BE, S16BE, S8 }") \
+ GST_AUDIO_CAPS_MAKE ("{ F32BE, F64BE, S32BE, S16BE }") \
", layout = (string) { interleaved, non-interleaved }"
#endif
static void gst_audio_resample_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
-static GType speex_resampler_sinc_filter_mode_get_type (void);
-
/* vmethods */
static gboolean gst_audio_resample_get_unit_size (GstBaseTransform * base,
GstCaps * caps, gsize * size);
g_object_class_install_property (gobject_class, PROP_QUALITY,
g_param_spec_int ("quality", "Quality", "Resample quality with 0 being "
"the lowest and 10 being the best",
- SPEEX_RESAMPLER_QUALITY_MIN, SPEEX_RESAMPLER_QUALITY_MAX,
- SPEEX_RESAMPLER_QUALITY_DEFAULT,
+ GST_AUDIO_RESAMPLER_QUALITY_MIN, GST_AUDIO_RESAMPLER_QUALITY_MAX,
+ DEFAULT_QUALITY,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SINC_FILTER_MODE,
g_param_spec_enum ("sinc-filter-mode", "Sinc filter table mode",
"What sinc filter table mode to use",
- GST_TYPE_SPEEX_RESAMPLER_SINC_FILTER_MODE,
- SPEEX_RESAMPLER_SINC_FILTER_DEFAULT,
+ GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE,
+ DEFAULT_SINC_FILTER_MODE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
g_param_spec_uint ("sinc-filter-auto-threshold",
"Sinc filter auto mode threshold",
"Memory usage threshold to use if sinc filter mode is AUTO, given in bytes",
- 0, G_MAXUINT, SPEEX_RESAMPLER_SINC_FILTER_AUTO_THRESHOLD_DEFAULT,
+ 0, G_MAXUINT, DEFAULT_SINC_FILTER_AUTO_THRESHOLD,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_static_pad_template (gstelement_class,
{
GstBaseTransform *trans = GST_BASE_TRANSFORM (resample);
- resample->quality = SPEEX_RESAMPLER_QUALITY_DEFAULT;
- resample->sinc_filter_mode = SPEEX_RESAMPLER_SINC_FILTER_DEFAULT;
- resample->sinc_filter_auto_threshold =
- SPEEX_RESAMPLER_SINC_FILTER_AUTO_THRESHOLD_DEFAULT;
+ resample->method = GST_AUDIO_RESAMPLER_METHOD_KAISER;
+ resample->quality = DEFAULT_QUALITY;
+ resample->sinc_filter_mode = DEFAULT_SINC_FILTER_MODE;
+ resample->sinc_filter_auto_threshold = DEFAULT_SINC_FILTER_AUTO_THRESHOLD;
gst_base_transform_set_gap_aware (trans, TRUE);
gst_pad_set_query_function (trans->srcpad, gst_audio_resample_query);
resample->samples_in = 0;
resample->samples_out = 0;
- resample->tmp_in = NULL;
- resample->tmp_in_size = 0;
- resample->tmp_out = NULL;
- resample->tmp_out_size = 0;
-
return TRUE;
}
{
GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
- if (resample->state) {
- resample->funcs->destroy (resample->state);
- resample->state = NULL;
+ if (resample->resamp) {
+ gst_audio_resampler_free (resample->resamp);
+ resample->resamp = NULL;
}
-
- resample->funcs = NULL;
-
- g_free (resample->tmp_in);
- resample->tmp_in = NULL;
- resample->tmp_in_size = 0;
-
- g_free (resample->tmp_out);
- resample->tmp_out = NULL;
- resample->tmp_out_size = 0;
-
return TRUE;
}
return othercaps;
}
-static const SpeexResampleFuncs *
-gst_audio_resample_get_funcs (gint width, gboolean fp)
+static GstStructure *
+make_options (GstAudioResample * resample, GstAudioInfo * in,
+ GstAudioInfo * out)
{
- const SpeexResampleFuncs *funcs = NULL;
-
- if (gst_audio_resample_use_int && (width == 8 || width == 16) && !fp)
- funcs = &int_funcs;
- else if ((!gst_audio_resample_use_int && (width == 8 || width == 16) && !fp)
- || (width == 32 && fp))
- funcs = &float_funcs;
- else if ((width == 64 && fp) || ((width == 32 || width == 24) && !fp))
- funcs = &double_funcs;
- else
- g_assert_not_reached ();
+ GstStructure *options;
- return funcs;
-}
+ options = gst_structure_new_empty ("resampler-options");
+ gst_audio_resampler_options_set_quality (resample->method,
+ resample->quality, in->rate, out->rate, options);
-static SpeexResamplerState *
-gst_audio_resample_init_state (GstAudioResample * resample, gint width,
- gint channels, gint inrate, gint outrate, gint quality, gboolean fp,
- SpeexResamplerSincFilterMode sinc_filter_mode,
- guint32 sinc_filter_auto_threshold)
-{
- SpeexResamplerState *ret = NULL;
- gint err = RESAMPLER_ERR_SUCCESS;
- const SpeexResampleFuncs *funcs = gst_audio_resample_get_funcs (width, fp);
-
- ret = funcs->init (channels, inrate, outrate, quality,
- sinc_filter_mode, sinc_filter_auto_threshold, &err);
-
- if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS)) {
- GST_ERROR_OBJECT (resample, "Failed to create resampler state: %s",
- funcs->strerror (err));
- return NULL;
- }
+ gst_structure_set (options,
+ GST_AUDIO_RESAMPLER_OPT_FILTER_MODE, GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE,
+ resample->sinc_filter_mode, GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD,
+ G_TYPE_UINT, resample->sinc_filter_auto_threshold, NULL);
- if (sinc_filter_mode == SPEEX_RESAMPLER_SINC_FILTER_AUTO) {
- GST_INFO_OBJECT (resample, "Using the %s sinc filter table",
- funcs->get_sinc_filter_mode (ret) ? "full" : "interpolated");
- }
-
- funcs->skip_zeros (ret);
-
- return ret;
+ return options;
}
static gboolean
-gst_audio_resample_update_state (GstAudioResample * resample, gint width,
- gint channels, gint inrate, gint outrate, gint quality, gboolean fp,
- SpeexResamplerSincFilterMode sinc_filter_mode,
- guint32 sinc_filter_auto_threshold)
+gst_audio_resample_update_state (GstAudioResample * resample, GstAudioInfo * in,
+ GstAudioInfo * out)
{
- gboolean ret = TRUE;
gboolean updated_latency = FALSE;
+ gsize old_latency = -1;
+ GstStructure *options;
- updated_latency = (resample->inrate != inrate
- || quality != resample->quality) && resample->state != NULL;
-
- if (resample->state == NULL) {
- ret = TRUE;
- } else if (resample->channels != channels || fp != resample->fp
- || width != resample->width
- || sinc_filter_mode != resample->sinc_filter_mode
- || sinc_filter_auto_threshold != resample->sinc_filter_auto_threshold) {
- resample->funcs->destroy (resample->state);
- resample->state =
- gst_audio_resample_init_state (resample, width, channels, inrate,
- outrate, quality, fp, sinc_filter_mode, sinc_filter_auto_threshold);
-
- resample->funcs = gst_audio_resample_get_funcs (width, fp);
- ret = (resample->state != NULL);
- } else if (resample->inrate != inrate || resample->outrate != outrate) {
- gint err = RESAMPLER_ERR_SUCCESS;
+ if (resample->resamp == NULL && in == NULL && out == NULL)
+ return TRUE;
- err = resample->funcs->set_rate (resample->state, inrate, outrate);
+ options = make_options (resample, in, out);
- if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS))
- GST_ERROR_OBJECT (resample, "Failed to update rate: %s",
- resample->funcs->strerror (err));
+ if (resample->resamp)
+ old_latency = gst_audio_resampler_get_max_latency (resample->resamp);
- ret = (err == RESAMPLER_ERR_SUCCESS);
- } else if (quality != resample->quality) {
- gint err = RESAMPLER_ERR_SUCCESS;
+ /* if channels and layout changed, destroy existing resampler */
+ if ((in->finfo != resample->in.finfo ||
+ in->channels != resample->in.channels ||
+ in->layout != resample->in.layout) && resample->resamp) {
+ gst_audio_resampler_free (resample->resamp);
+ resample->resamp = NULL;
+ }
+ if (resample->resamp == NULL) {
+ GstAudioResamplerFlags flags = 0;
- err = resample->funcs->set_quality (resample->state, quality);
+ if (in->layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED)
+ flags |= GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED;
- if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS))
- GST_ERROR_OBJECT (resample, "Failed to update quality: %s",
- resample->funcs->strerror (err));
+ resample->resamp = gst_audio_resampler_new (resample->method,
+ flags, in->finfo->format, in->channels, in->rate, out->rate, options);
+ if (resample->resamp == NULL)
+ goto resampler_failed;
+ } else {
+ gboolean ret;
- ret = (err == RESAMPLER_ERR_SUCCESS);
+ ret =
+ gst_audio_resampler_update (resample->resamp, in->rate, out->rate,
+ options);
+ if (!ret)
+ goto update_failed;
}
-
- resample->width = width;
- resample->channels = channels;
- resample->fp = fp;
- resample->quality = quality;
- resample->inrate = inrate;
- resample->outrate = outrate;
- resample->sinc_filter_mode = sinc_filter_mode;
- resample->sinc_filter_auto_threshold = sinc_filter_auto_threshold;
+ if (old_latency != -1)
+ updated_latency =
+ old_latency != gst_audio_resampler_get_max_latency (resample->resamp);
if (updated_latency)
gst_element_post_message (GST_ELEMENT (resample),
gst_message_new_latency (GST_OBJECT (resample)));
- return ret;
+ return TRUE;
+
+ /* ERRORS */
+resampler_failed:
+ {
+ GST_ERROR_OBJECT (resample, "failed to create resampler");
+ return FALSE;
+ }
+update_failed:
+ {
+ GST_ERROR_OBJECT (resample, "failed to update resampler");
+ return FALSE;
+ }
}
static void
gst_audio_resample_reset_state (GstAudioResample * resample)
{
- if (resample->state)
- resample->funcs->reset_mem (resample->state);
}
static gint
gst_audio_resample_set_caps (GstBaseTransform * base, GstCaps * incaps,
GstCaps * outcaps)
{
- gboolean ret;
- gint width, inrate, outrate, channels;
- gboolean fp;
GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
GstAudioInfo in, out;
goto invalid_outcaps;
/* FIXME do some checks */
+ gst_audio_resample_update_state (resample, &in, &out);
- /* take new values */
- width = GST_AUDIO_FORMAT_INFO_WIDTH (in.finfo);
- channels = GST_AUDIO_INFO_CHANNELS (&in);
- inrate = GST_AUDIO_INFO_RATE (&in);
- outrate = GST_AUDIO_INFO_RATE (&out);
- fp = GST_AUDIO_FORMAT_INFO_IS_FLOAT (in.finfo);
-
- ret =
- gst_audio_resample_update_state (resample, width, channels, inrate,
- outrate, resample->quality, fp, resample->sinc_filter_mode,
- resample->sinc_filter_auto_threshold);
-
- if (G_UNLIKELY (!ret))
- return FALSE;
+ resample->in = in;
+ resample->out = out;
return TRUE;
}
}
-#define GST_MAXINT24 (8388607)
-#define GST_MININT24 (-8388608)
-
-#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
-#define GST_READ_UINT24 GST_READ_UINT24_LE
-#define GST_WRITE_UINT24 GST_WRITE_UINT24_LE
-#else
-#define GST_READ_UINT24 GST_READ_UINT24_BE
-#define GST_WRITE_UINT24 GST_WRITE_UINT24_BE
-#endif
-
-static void
-gst_audio_resample_convert_buffer (GstAudioResample * resample,
- const guint8 * in, guint8 * out, guint len, gboolean inverse)
-{
- len *= resample->channels;
-
- if (inverse) {
- if (gst_audio_resample_use_int && resample->width == 8 && !resample->fp) {
- gint8 *o = (gint8 *) out;
- gint16 *i = (gint16 *) in;
- gint32 tmp;
-
- while (len) {
- tmp = *i + (G_MAXINT8 >> 1);
- *o = CLAMP (tmp >> 8, G_MININT8, G_MAXINT8);
- o++;
- i++;
- len--;
- }
- } else if (!gst_audio_resample_use_int && resample->width == 8
- && !resample->fp) {
- gint8 *o = (gint8 *) out;
- gfloat *i = (gfloat *) in;
- gfloat tmp;
-
- while (len) {
- tmp = *i;
- *o = (gint8) CLAMP (tmp * G_MAXINT8 + 0.5, G_MININT8, G_MAXINT8);
- o++;
- i++;
- len--;
- }
- } else if (!gst_audio_resample_use_int && resample->width == 16
- && !resample->fp) {
- gint16 *o = (gint16 *) out;
- gfloat *i = (gfloat *) in;
- gfloat tmp;
-
- while (len) {
- tmp = *i;
- *o = (gint16) CLAMP (tmp * G_MAXINT16 + 0.5, G_MININT16, G_MAXINT16);
- o++;
- i++;
- len--;
- }
- } else if (resample->width == 24 && !resample->fp) {
- guint8 *o = (guint8 *) out;
- gdouble *i = (gdouble *) in;
- gdouble tmp;
-
- while (len) {
- tmp = *i;
- GST_WRITE_UINT24 (o, (gint32) CLAMP (tmp * GST_MAXINT24 + 0.5,
- GST_MININT24, GST_MAXINT24));
- o += 3;
- i++;
- len--;
- }
- } else if (resample->width == 32 && !resample->fp) {
- gint32 *o = (gint32 *) out;
- gdouble *i = (gdouble *) in;
- gdouble tmp;
-
- while (len) {
- tmp = *i;
- *o = (gint32) CLAMP (tmp * G_MAXINT32 + 0.5, G_MININT32, G_MAXINT32);
- o++;
- i++;
- len--;
- }
- } else {
- g_assert_not_reached ();
- }
- } else {
- if (gst_audio_resample_use_int && resample->width == 8 && !resample->fp) {
- gint8 *i = (gint8 *) in;
- gint16 *o = (gint16 *) out;
- gint32 tmp;
-
- while (len) {
- tmp = *i;
- *o = tmp << 8;
- o++;
- i++;
- len--;
- }
- } else if (!gst_audio_resample_use_int && resample->width == 8
- && !resample->fp) {
- gint8 *i = (gint8 *) in;
- gfloat *o = (gfloat *) out;
- gfloat tmp;
-
- while (len) {
- tmp = *i;
- *o = tmp / G_MAXINT8;
- o++;
- i++;
- len--;
- }
- } else if (!gst_audio_resample_use_int && resample->width == 16
- && !resample->fp) {
- gint16 *i = (gint16 *) in;
- gfloat *o = (gfloat *) out;
- gfloat tmp;
-
- while (len) {
- tmp = *i;
- *o = tmp / G_MAXINT16;
- o++;
- i++;
- len--;
- }
- } else if (resample->width == 24 && !resample->fp) {
- guint8 *i = (guint8 *) in;
- gdouble *o = (gdouble *) out;
- gdouble tmp;
- guint32 tmp2;
-
- while (len) {
- tmp2 = GST_READ_UINT24 (i);
- if (tmp2 & 0x00800000)
- tmp2 |= 0xff000000;
- tmp = (gint32) tmp2;
- *o = tmp / GST_MAXINT24;
- o++;
- i += 3;
- len--;
- }
- } else if (resample->width == 32 && !resample->fp) {
- gint32 *i = (gint32 *) in;
- gdouble *o = (gdouble *) out;
- gdouble tmp;
-
- while (len) {
- tmp = *i;
- *o = tmp / G_MAXINT32;
- o++;
- i++;
- len--;
- }
- } else {
- g_assert_not_reached ();
- }
- }
-}
-
-static guint8 *
-gst_audio_resample_workspace_realloc (guint8 ** workspace, guint * size,
- guint new_size)
-{
- guint8 *new;
- if (new_size <= *size)
- /* no need to resize */
- return *workspace;
- new = g_realloc (*workspace, new_size);
- if (!new)
- /* failure (re)allocating memeory */
- return NULL;
- /* success */
- *workspace = new;
- *size = new_size;
- return *workspace;
-}
-
/* Push history_len zeros into the filter, but discard the output. */
static void
gst_audio_resample_dump_drain (GstAudioResample * resample, guint history_len)
{
+#if 0
gint outsize;
guint in_len G_GNUC_UNUSED, in_processed;
guint out_len, out_processed;
guint num, den;
gpointer buf;
- g_assert (resample->state != NULL);
+ g_assert (resample->resamp != NULL);
resample->funcs->get_ratio (resample->state, &num, &den);
g_free (buf);
g_assert (in_len == in_processed);
+#endif
}
static void
GstBuffer *outbuf;
GstFlowReturn res;
gint outsize;
- guint in_len, in_processed;
- guint out_len, out_processed;
- gint err;
- guint num, den;
+ gsize in_processed;
+ gsize out_len, out_processed;
GstMapInfo map;
+ gpointer out[1];
- g_assert (resample->state != NULL);
+ g_assert (resample->resamp != NULL);
/* Don't drain samples if we were reset. */
if (!GST_CLOCK_TIME_IS_VALID (resample->t0))
return;
- resample->funcs->get_ratio (resample->state, &num, &den);
-
- in_len = in_processed = history_len;
- out_len = out_processed =
- gst_util_uint64_scale_int_ceil (history_len, den, num);
- outsize = out_len * resample->channels * (resample->width / 8);
-
+ out_len = gst_audio_resampler_get_out_frames (resample->resamp, history_len);
if (out_len == 0)
return;
+ outsize = out_len * resample->in.bpf;
outbuf = gst_buffer_new_and_alloc (outsize);
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
- if (resample->funcs->width != resample->width) {
- /* need to convert data format; allocate workspace */
- if (!gst_audio_resample_workspace_realloc (&resample->tmp_out,
- &resample->tmp_out_size, (resample->funcs->width / 8) * out_len *
- resample->channels)) {
- GST_ERROR_OBJECT (resample, "failed to allocate workspace");
- return;
- }
-
- /* process */
- err = resample->funcs->process (resample->state, NULL, &in_processed,
- resample->tmp_out, &out_processed);
-
- /* convert output format */
- gst_audio_resample_convert_buffer (resample, resample->tmp_out,
- map.data, out_processed, TRUE);
- } else {
- /* don't need to convert data format; process */
- err = resample->funcs->process (resample->state, NULL, &in_processed,
- map.data, &out_processed);
- }
+ out[0] = map.data;
+ gst_audio_resampler_resample (resample->resamp, NULL, history_len,
+ out, out_len, &in_processed, &out_processed);
/* If we wrote more than allocated something is really wrong now
* and we should better abort immediately */
- g_assert (out_len >= out_processed);
-
- outsize = out_processed * resample->channels * (resample->width / 8);
+ g_assert (out_len == out_processed);
gst_buffer_unmap (outbuf, &map);
- gst_buffer_resize (outbuf, 0, outsize);
-
- if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS)) {
- GST_WARNING_OBJECT (resample, "Failed to process drain: %s",
- resample->funcs->strerror (err));
- gst_buffer_unref (outbuf);
- return;
- }
/* time */
if (GST_CLOCK_TIME_IS_VALID (resample->t0)) {
GST_BUFFER_TIMESTAMP (outbuf) = resample->t0 +
gst_util_uint64_scale_int_round (resample->samples_out, GST_SECOND,
- resample->outrate);
+ resample->out.rate);
GST_BUFFER_DURATION (outbuf) = resample->t0 +
gst_util_uint64_scale_int_round (resample->samples_out + out_processed,
- GST_SECOND, resample->outrate) - GST_BUFFER_TIMESTAMP (outbuf);
+ GST_SECOND, resample->out.rate) - GST_BUFFER_TIMESTAMP (outbuf);
} else {
GST_BUFFER_TIMESTAMP (outbuf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE;
resample->samples_out += out_processed;
resample->samples_in += history_len;
- if (G_UNLIKELY (out_processed == 0 && in_len * den > num)) {
- GST_WARNING_OBJECT (resample, "Failed to get drain, dropping buffer");
- gst_buffer_unref (outbuf);
- return;
- }
-
GST_LOG_OBJECT (resample,
"Pushing drain buffer of %u bytes with timestamp %" GST_TIME_FORMAT
" duration %" GST_TIME_FORMAT " offset %" G_GUINT64_FORMAT " offset_end %"
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
gst_audio_resample_reset_state (resample);
- if (resample->state)
- resample->funcs->skip_zeros (resample->state);
+#if 0
+ if (resample->resamp)
+ resample->funcs->skip_zeros (resample->resamp);
+#endif
resample->num_gap_samples = 0;
resample->num_nongap_samples = 0;
resample->t0 = GST_CLOCK_TIME_NONE;
resample->need_discont = TRUE;
break;
case GST_EVENT_SEGMENT:
- if (resample->state) {
- guint latency = resample->funcs->get_input_latency (resample->state);
+#if 0
+ if (resample->resamp) {
+ guint latency = resample->funcs->get_input_latency (resample->resamp);
gst_audio_resample_push_drain (resample, latency);
}
+#endif
gst_audio_resample_reset_state (resample);
- if (resample->state)
- resample->funcs->skip_zeros (resample->state);
+#if 0
+ if (resample->resamp)
+ resample->funcs->skip_zeros (resample->resamp);
+#endif
resample->num_gap_samples = 0;
resample->num_nongap_samples = 0;
resample->t0 = GST_CLOCK_TIME_NONE;
resample->need_discont = TRUE;
break;
case GST_EVENT_EOS:
- if (resample->state) {
- guint latency = resample->funcs->get_input_latency (resample->state);
+#if 0
+ if (resample->resamp) {
+ guint latency = resample->funcs->get_input_latency (resample->resamp);
gst_audio_resample_push_drain (resample, latency);
}
+#endif
gst_audio_resample_reset_state (resample);
break;
default:
/* convert the inbound timestamp to an offset. */
offset =
gst_util_uint64_scale_int_round (GST_BUFFER_TIMESTAMP (buf) -
- resample->t0, resample->inrate, GST_SECOND);
+ resample->t0, resample->in.rate, GST_SECOND);
/* many elements generate imperfect streams due to rounding errors, so we
* permit a small error (up to one sample) without triggering a filter
/* allow even up to more samples, since sink is not so strict anyway,
* so give that one a chance to handle this as configured */
delta = ABS ((gint64) (offset - resample->samples_in));
- if (delta <= (resample->inrate >> 5))
+ if (delta <= (resample->in.rate >> 5))
return FALSE;
GST_WARNING_OBJECT (resample,
"encountered timestamp discontinuity of %" G_GUINT64_FORMAT " samples = %"
GST_TIME_FORMAT, delta,
GST_TIME_ARGS (gst_util_uint64_scale_int_round (delta, GST_SECOND,
- resample->inrate)));
+ resample->in.rate)));
return TRUE;
}
gsize outsize;
guint32 in_len, in_processed;
guint32 out_len, out_processed;
- guint filt_len = resample->funcs->get_filt_len (resample->state);
+ guint filt_len = gst_audio_resampler_get_max_latency (resample->resamp) * 2;
gst_buffer_map (inbuf, &in_map, GST_MAP_READ);
gst_buffer_map (outbuf, &out_map, GST_MAP_WRITE);
- in_len = in_map.size / resample->channels;
- out_len = out_map.size / resample->channels;
-
- in_len /= (resample->width / 8);
- out_len /= (resample->width / 8);
+ in_len = in_map.size / resample->in.bpf;
+ out_len = out_map.size / resample->out.bpf;
in_processed = in_len;
out_processed = out_len;
{
guint num, den;
- resample->funcs->get_ratio (resample->state, &num, &den);
+
+ num = resample->in.rate;
+ den = resample->out.rate;
+
if (resample->samples_in + in_len >= filt_len / 2)
out_processed =
gst_util_uint64_scale_int_ceil (resample->samples_in + in_len -
in_processed = in_len;
}
} else { /* not a gap */
-
- gint err;
-
if (resample->num_gap_samples > filt_len) {
/* push in enough zeros to restore the filter to the right offset */
- guint num, den;
- resample->funcs->get_ratio (resample->state, &num, &den);
+ guint num;
+
+ num = resample->in.rate;
+
gst_audio_resample_dump_drain (resample,
(resample->num_gap_samples - filt_len) % num);
}
if (resample->num_nongap_samples > filt_len)
resample->num_nongap_samples = filt_len;
}
+ {
+ /* process */
+ {
+ gsize in_proc, out_proc, out_test;
+ gpointer in[1], out[1];
- if (resample->funcs->width != resample->width) {
- /* need to convert data format for processing; ensure we have enough
- * workspace available */
- if (!gst_audio_resample_workspace_realloc (&resample->tmp_in,
- &resample->tmp_in_size, in_len * resample->channels *
- (resample->funcs->width / 8)) ||
- !gst_audio_resample_workspace_realloc (&resample->tmp_out,
- &resample->tmp_out_size, out_len * resample->channels *
- (resample->funcs->width / 8))) {
- GST_ERROR_OBJECT (resample, "failed to allocate workspace");
- gst_buffer_unmap (inbuf, &in_map);
- gst_buffer_unmap (outbuf, &out_map);
- return GST_FLOW_ERROR;
- }
-
- /* convert input */
- gst_audio_resample_convert_buffer (resample, in_map.data,
- resample->tmp_in, in_len, FALSE);
+ out_test =
+ gst_audio_resampler_get_out_frames (resample->resamp, in_len);
+ out_test = MIN (out_test, out_len);
- /* process */
- err = resample->funcs->process (resample->state,
- resample->tmp_in, &in_processed, resample->tmp_out, &out_processed);
+ in[0] = in_map.data;
+ out[0] = out_map.data;
+ gst_audio_resampler_resample (resample->resamp, in, in_len,
+ out, out_len, &in_proc, &out_proc);
- /* convert output */
- gst_audio_resample_convert_buffer (resample, resample->tmp_out,
- out_map.data, out_processed, TRUE);
- } else {
- /* no format conversion required; process */
- err = resample->funcs->process (resample->state,
- in_map.data, &in_processed, out_map.data, &out_processed);
- }
+ in_processed = in_proc;
+ out_processed = out_proc;
- if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS)) {
- GST_ERROR_OBJECT (resample, "Failed to convert data: %s",
- resample->funcs->strerror (err));
- gst_buffer_unmap (inbuf, &in_map);
- gst_buffer_unmap (outbuf, &out_map);
- return GST_FLOW_ERROR;
+ //g_printerr ("in %d, test %d, %d, real %d (%d)\n", (gint) in_len, (gint) out_test, (gint) out_len, (gint) out_proc, (gint) (out_test - out_proc));
+ g_assert (out_test == out_proc);
+ }
}
}
if (GST_CLOCK_TIME_IS_VALID (resample->t0)) {
GST_BUFFER_TIMESTAMP (outbuf) = resample->t0 +
gst_util_uint64_scale_int_round (resample->samples_out, GST_SECOND,
- resample->outrate);
+ resample->out.rate);
GST_BUFFER_DURATION (outbuf) = resample->t0 +
gst_util_uint64_scale_int_round (resample->samples_out + out_processed,
- GST_SECOND, resample->outrate) - GST_BUFFER_TIMESTAMP (outbuf);
+ GST_SECOND, resample->out.rate) - GST_BUFFER_TIMESTAMP (outbuf);
} else {
GST_BUFFER_TIMESTAMP (outbuf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE;
gst_buffer_unmap (inbuf, &in_map);
gst_buffer_unmap (outbuf, &out_map);
- outsize = out_processed * resample->channels * (resample->width / 8);
+ outsize = out_processed * resample->in.bpf;
gst_buffer_resize (outbuf, 0, outsize);
GST_LOG_OBJECT (resample,
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf));
- return GST_FLOW_OK;
+ if (outsize == 0)
+ return GST_BASE_TRANSFORM_FLOW_DROPPED;
+ else
+ return GST_FLOW_OK;
}
static GstFlowReturn
GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
GstFlowReturn ret;
- if (resample->state == NULL) {
- if (G_UNLIKELY (!(resample->state =
- gst_audio_resample_init_state (resample, resample->width,
- resample->channels, resample->inrate, resample->outrate,
- resample->quality, resample->fp, resample->sinc_filter_mode,
- resample->sinc_filter_auto_threshold))))
- return GST_FLOW_ERROR;
-
- resample->funcs =
- gst_audio_resample_get_funcs (resample->width, resample->fp);
- }
-
GST_LOG_OBJECT (resample, "transforming buffer of %" G_GSIZE_FORMAT " bytes,"
" ts %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
/* handle discontinuity */
if (G_UNLIKELY (resample->need_discont)) {
+#if 0
resample->funcs->skip_zeros (resample->state);
+#endif
resample->num_gap_samples = 0;
resample->num_nongap_samples = 0;
/* reset */
resample->in_offset0 = GST_BUFFER_OFFSET (inbuf);
resample->out_offset0 =
gst_util_uint64_scale_int_round (resample->in_offset0,
- resample->outrate, resample->inrate);
+ resample->out.rate, resample->in.rate);
} else {
GST_DEBUG_OBJECT (resample, "... but new offset is invalid");
resample->in_offset0 = GST_BUFFER_OFFSET_NONE;
if (base->segment.format == GST_FORMAT_TIME) {
input =
- gst_audio_buffer_clip (input, &base->segment, resample->inrate,
- resample->channels * resample->width);
+ gst_audio_buffer_clip (input, &base->segment, resample->in.rate,
+ resample->in.bpf);
if (!input)
return GST_FLOW_OK;
GstClockTime min, max;
gboolean live;
guint64 latency;
- gint rate = resample->inrate;
+ gint rate = resample->in.rate;
gint resampler_latency;
+#if 0
if (resample->state)
resampler_latency =
resample->funcs->get_input_latency (resample->state);
else
+#endif
resampler_latency = 0;
if (gst_base_transform_is_passthrough (trans))
const GValue * value, GParamSpec * pspec)
{
GstAudioResample *resample;
- gint quality;
resample = GST_AUDIO_RESAMPLE (object);
switch (prop_id) {
case PROP_QUALITY:
/* FIXME locking! */
- quality = g_value_get_int (value);
- GST_DEBUG_OBJECT (resample, "new quality %d", quality);
-
- gst_audio_resample_update_state (resample, resample->width,
- resample->channels, resample->inrate, resample->outrate,
- quality, resample->fp, resample->sinc_filter_mode,
- resample->sinc_filter_auto_threshold);
+ resample->quality = g_value_get_int (value);
+ GST_DEBUG_OBJECT (resample, "new quality %d", resample->quality);
+ gst_audio_resample_update_state (resample, NULL, NULL);
break;
- case PROP_SINC_FILTER_MODE:{
+ case PROP_SINC_FILTER_MODE:
/* FIXME locking! */
- SpeexResamplerSincFilterMode sinc_filter_mode = g_value_get_enum (value);
-
- gst_audio_resample_update_state (resample, resample->width,
- resample->channels, resample->inrate, resample->outrate,
- resample->quality, resample->fp, sinc_filter_mode,
- resample->sinc_filter_auto_threshold);
-
+ resample->sinc_filter_mode = g_value_get_enum (value);
+ gst_audio_resample_update_state (resample, NULL, NULL);
break;
- }
- case PROP_SINC_FILTER_AUTO_THRESHOLD:{
+ case PROP_SINC_FILTER_AUTO_THRESHOLD:
/* FIXME locking! */
- guint32 sinc_filter_auto_threshold = g_value_get_uint (value);
-
- gst_audio_resample_update_state (resample, resample->width,
- resample->channels, resample->inrate, resample->outrate,
- resample->quality, resample->fp, resample->sinc_filter_mode,
- sinc_filter_auto_threshold);
-
+ resample->sinc_filter_auto_threshold = g_value_get_uint (value);
+ gst_audio_resample_update_state (resample, NULL, NULL);
break;
- }
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
-static GType
-speex_resampler_sinc_filter_mode_get_type (void)
-{
- static GType speex_resampler_sinc_filter_mode_type = 0;
-
- if (!speex_resampler_sinc_filter_mode_type) {
- static const GEnumValue sinc_filter_modes[] = {
- {SPEEX_RESAMPLER_SINC_FILTER_INTERPOLATED, "Use interpolated sinc table",
- "interpolated"},
- {SPEEX_RESAMPLER_SINC_FILTER_FULL, "Use full sinc table", "full"},
- {SPEEX_RESAMPLER_SINC_FILTER_AUTO,
- "Use full table if table size below threshold", "auto"},
- {0, NULL, NULL},
- };
-
- speex_resampler_sinc_filter_mode_type =
- g_enum_register_static ("SpeexResamplerSincFilterMode",
- sinc_filter_modes);
- }
-
- return speex_resampler_sinc_filter_mode_type;
-}
-
/* FIXME: should have a benchmark fallback for the case where orc is disabled */
#if defined(AUDIORESAMPLE_FORMAT_AUTO) && !defined(DISABLE_ORC)
#define BENCHMARK_SIZE 512
static gboolean
-_benchmark_int_float (SpeexResamplerState * st)
+_benchmark_int_float (GstAudioResampler * st)
{
gint16 in[BENCHMARK_SIZE] = { 0, }, G_GNUC_UNUSED out[BENCHMARK_SIZE / 2];
gfloat in_tmp[BENCHMARK_SIZE], out_tmp[BENCHMARK_SIZE / 2];
gint i;
guint32 inlen = BENCHMARK_SIZE, outlen = BENCHMARK_SIZE / 2;
+ gpointer inp[1], outp[1];
+ gsize produced, consumed;
for (i = 0; i < BENCHMARK_SIZE; i++) {
gfloat tmp = in[i];
in_tmp[i] = tmp / G_MAXINT16;
}
- resample_float_resampler_process_interleaved_float (st,
- (const guint8 *) in_tmp, &inlen, (guint8 *) out_tmp, &outlen);
+ inp[0] = in_tmp;
+ outp[0] = out_tmp;
+
+ gst_audio_resampler_resample (st,
+ inp, inlen, outp, outlen, &produced, &consumed);
if (outlen == 0) {
GST_ERROR ("Failed to use float resampler");
}
static gboolean
-_benchmark_int_int (SpeexResamplerState * st)
+_benchmark_int_int (GstAudioResampler * st)
{
gint16 in[BENCHMARK_SIZE] = { 0, }, out[BENCHMARK_SIZE / 2];
guint32 inlen = BENCHMARK_SIZE, outlen = BENCHMARK_SIZE / 2;
+ gpointer inp[1], outp[1];
+ gsize produced, consumed;
+
+ inp[0] = in;
+ outp[0] = out;
- resample_int_resampler_process_interleaved_int (st, (const guint8 *) in,
- &inlen, (guint8 *) out, &outlen);
+ gst_audio_resampler_resample (st, inp, inlen, outp, outlen, &produced,
+ &consumed);
if (outlen == 0) {
GST_ERROR ("Failed to use int resampler");
{
OrcProfile a, b;
gdouble av, bv;
- SpeexResamplerState *sta, *stb;
+ GstAudioResampler *sta, *stb;
int i;
orc_profile_init (&a);
orc_profile_init (&b);
- sta = resample_float_resampler_init (1, 48000, 24000, 4,
- SPEEX_RESAMPLER_SINC_FILTER_INTERPOLATED,
- SPEEX_RESAMPLER_SINC_FILTER_AUTO_THRESHOLD_DEFAULT, NULL);
+ sta = gst_audio_resampler_new (GST_AUDIO_RESAMPLER_METHOD_KAISER,
+ 0, GST_AUDIO_FORMAT_F32LE, 1, 48000, 24000, NULL);
if (sta == NULL) {
GST_ERROR ("Failed to create float resampler state");
return FALSE;
}
- stb = resample_int_resampler_init (1, 48000, 24000, 4,
- SPEEX_RESAMPLER_SINC_FILTER_INTERPOLATED,
- SPEEX_RESAMPLER_SINC_FILTER_AUTO_THRESHOLD_DEFAULT, NULL);
+ stb = gst_audio_resampler_new (GST_AUDIO_RESAMPLER_METHOD_KAISER,
+ 0, GST_AUDIO_FORMAT_S32LE, 1, 48000, 24000, NULL);
if (stb == NULL) {
- resample_float_resampler_destroy (sta);
+ gst_audio_resampler_free (sta);
GST_ERROR ("Failed to create int resampler state");
return FALSE;
}
/* Remember benchmark result in global variable */
gst_audio_resample_use_int = (av > bv);
- resample_float_resampler_destroy (sta);
- resample_int_resampler_destroy (stb);
+ gst_audio_resampler_free (sta);
+ gst_audio_resampler_free (stb);
if (av > bv)
GST_INFO ("Using integer resampler if appropriate: %lf < %lf", bv, av);
return TRUE;
error:
- resample_float_resampler_destroy (sta);
- resample_int_resampler_destroy (stb);
+ gst_audio_resampler_free (sta);
+ gst_audio_resampler_free (stb);
return FALSE;
}
#include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
-#include "speex_resampler_wrapper.h"
-
G_BEGIN_DECLS
#define GST_TYPE_AUDIO_RESAMPLE \
guint64 out_offset0;
guint64 samples_in;
guint64 samples_out;
-
+
guint64 num_gap_samples;
guint64 num_nongap_samples;
/* properties */
+ GstAudioResamplerMethod method;
gint quality;
-
- /* state */
- gboolean fp;
- gint width;
- gint channels;
- gint inrate;
- gint outrate;
-
- SpeexResamplerSincFilterMode sinc_filter_mode;
+ GstAudioResamplerFilterMode sinc_filter_mode;
guint32 sinc_filter_auto_threshold;
- guint8 *tmp_in;
- guint tmp_in_size;
-
- guint8 *tmp_out;
- guint tmp_out_size;
-
- SpeexResamplerState *state;
- const SpeexResampleFuncs *funcs;
+ /* state */
+ GstAudioInfo in;
+ GstAudioInfo out;
+ GstAudioResampler *resamp;
};
struct _GstAudioResampleClass {
+++ /dev/null
-/* Copyright (C) 2007-2008 Jean-Marc Valin
- Copyright (C) 2008 Thorvald Natvig
-
- File: resample.c
- Arbitrary resampling code
-
- Redistribution and use in source and binary forms, with or without
- modification, are permitted provided that the following conditions are
- met:
-
- 1. Redistributions of source code must retain the above copyright notice,
- this list of conditions and the following disclaimer.
-
- 2. Redistributions in binary form must reproduce the above copyright
- notice, this list of conditions and the following disclaimer in the
- documentation and/or other materials provided with the distribution.
-
- 3. The name of the author may not be used to endorse or promote products
- derived from this software without specific prior written permission.
-
- THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
- IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
- OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
- DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
- INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
- (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
- SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
- HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
- STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
- ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
- POSSIBILITY OF SUCH DAMAGE.
-*/
-
-/*
- The design goals of this code are:
- - Very fast algorithm
- - SIMD-friendly algorithm
- - Low memory requirement
- - Good *perceptual* quality (and not best SNR)
-
- Warning: This resampler is relatively new. Although I think I got rid of
- all the major bugs and I don't expect the API to change anymore, there
- may be something I've missed. So use with caution.
-
- This algorithm is based on this original resampling algorithm:
- Smith, Julius O. Digital Audio Resampling Home Page
- Center for Computer Research in Music and Acoustics (CCRMA),
- Stanford University, 2007.
- Web published at http://www-ccrma.stanford.edu/~jos/resample/.
-
- There is one main difference, though. This resampler uses cubic
- interpolation instead of linear interpolation in the above paper. This
- makes the table much smaller and makes it possible to compute that table
- on a per-stream basis. In turn, being able to tweak the table for each
- stream makes it possible to both reduce complexity on simple ratios
- (e.g. 2/3), and get rid of the rounding operations in the inner loop.
- The latter both reduces CPU time and makes the algorithm more SIMD-friendly.
-*/
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#ifdef OUTSIDE_SPEEX
-#include <stdlib.h>
-
-#ifdef HAVE_STRING_H
-#include <string.h>
-#endif
-
-#include <glib.h>
-
-#ifdef HAVE_ORC
-#include <orc/orc.h>
-#endif
-
-#define EXPORT G_GNUC_INTERNAL
-
-#ifdef _USE_SSE
-#if !defined(__SSE__) || !defined(HAVE_XMMINTRIN_H)
-#undef _USE_SSE
-#endif
-#endif
-
-#ifdef _USE_SSE2
-#if !defined(__SSE2__) || !defined(HAVE_EMMINTRIN_H)
-#undef _USE_SSE2
-#endif
-#endif
-
-#ifdef _USE_NEON
-#ifndef HAVE_ARM_NEON
-#undef _USE_NEON
-#endif
-#endif
-
-static inline void *
-speex_alloc (int size)
-{
- return g_malloc0 (size);
-}
-
-static inline void *
-speex_realloc (void *ptr, int size)
-{
- return g_realloc (ptr, size);
-}
-
-static inline void
-speex_free (void *ptr)
-{
- g_free (ptr);
-}
-
-#include "speex_resampler.h"
-#include "arch.h"
-#else /* OUTSIDE_SPEEX */
-
-#include "../include/speex/speex_resampler.h"
-#include "arch.h"
-#include "os_support.h"
-#endif /* OUTSIDE_SPEEX */
-
-#include <math.h>
-
-#ifdef FIXED_POINT
-#define WORD2INT(x) ((x) < -32767 ? -32768 : ((x) > 32766 ? 32767 : (x)))
-#else
-#define WORD2INT(x) ((x) < -32767.5f ? -32768 : ((x) > 32766.5f ? 32767 : floor(.5+(x))))
-#endif
-
-#define IMAX(a,b) ((a) > (b) ? (a) : (b))
-#define IMIN(a,b) ((a) < (b) ? (a) : (b))
-
-#ifndef NULL
-#define NULL 0
-#endif
-
-#if defined _USE_SSE || defined _USE_SSE2
-#include "resample_sse.h"
-#endif
-
-#ifdef _USE_NEON
-#include "resample_neon.h"
-#endif
-
-/* Numer of elements to allocate on the stack */
-#ifdef VAR_ARRAYS
-#define FIXED_STACK_ALLOC 8192
-#else
-#define FIXED_STACK_ALLOC 1024
-#endif
-
-/* Allow selecting SSE or not when compiled with SSE support */
-#ifdef _USE_SSE
-#define SSE_FALLBACK(macro) \
- if (st->use_sse) goto sse_##macro##_sse; {
-#define SSE_IMPLEMENTATION(macro) \
- goto sse_##macro##_end; } sse_##macro##_sse: {
-#define SSE_END(macro) sse_##macro##_end:; }
-#else
-#define SSE_FALLBACK(macro)
-#endif
-
-#ifdef _USE_SSE2
-#define SSE2_FALLBACK(macro) \
- if (st->use_sse2) goto sse2_##macro##_sse2; {
-#define SSE2_IMPLEMENTATION(macro) \
- goto sse2_##macro##_end; } sse2_##macro##_sse2: {
-#define SSE2_END(macro) sse2_##macro##_end:; }
-#else
-#define SSE2_FALLBACK(macro)
-#endif
-
-#ifdef _USE_NEON
-#define NEON_FALLBACK(macro) \
- if (st->use_neon) goto neon_##macro##_neon; {
-#define NEON_IMPLEMENTATION(macro) \
- goto neon_##macro##_end; } neon_##macro##_neon: {
-#define NEON_END(macro) neon_##macro##_end:; }
-#else
-#define NEON_FALLBACK(macro)
-#endif
-
-
-typedef int (*resampler_basic_func) (SpeexResamplerState *, spx_uint32_t,
- const spx_word16_t *, spx_uint32_t *, spx_word16_t *, spx_uint32_t *);
-
-struct SpeexResamplerState_
-{
- spx_uint32_t in_rate;
- spx_uint32_t out_rate;
- spx_uint32_t num_rate;
- spx_uint32_t den_rate;
-
- int quality;
- spx_uint32_t nb_channels;
- spx_uint32_t filt_len;
- spx_uint32_t mem_alloc_size;
- spx_uint32_t buffer_size;
- int int_advance;
- int frac_advance;
- float cutoff;
- spx_uint32_t oversample;
- int initialised;
- int started;
- int use_full_sinc_table;
-
- /* These are per-channel */
- spx_int32_t *last_sample;
- spx_uint32_t *samp_frac_num;
- spx_uint32_t *magic_samples;
-
- spx_word16_t *mem;
- spx_word16_t *sinc_table;
- spx_uint32_t sinc_table_length;
- resampler_basic_func resampler_ptr;
-
- int in_stride;
- int out_stride;
-
- int use_sse:1;
- int use_sse2:1;
- int use_neon:1;
-};
-
-static const double kaiser12_table[68] = {
- 0.99859849, 1.00000000, 0.99859849, 0.99440475, 0.98745105, 0.97779076,
- 0.96549770, 0.95066529, 0.93340547, 0.91384741, 0.89213598, 0.86843014,
- 0.84290116, 0.81573067, 0.78710866, 0.75723148, 0.72629970, 0.69451601,
- 0.66208321, 0.62920216, 0.59606986, 0.56287762, 0.52980938, 0.49704014,
- 0.46473455, 0.43304576, 0.40211431, 0.37206735, 0.34301800, 0.31506490,
- 0.28829195, 0.26276832, 0.23854851, 0.21567274, 0.19416736, 0.17404546,
- 0.15530766, 0.13794294, 0.12192957, 0.10723616, 0.09382272, 0.08164178,
- 0.07063950, 0.06075685, 0.05193064, 0.04409466, 0.03718069, 0.03111947,
- 0.02584161, 0.02127838, 0.01736250, 0.01402878, 0.01121463, 0.00886058,
- 0.00691064, 0.00531256, 0.00401805, 0.00298291, 0.00216702, 0.00153438,
- 0.00105297, 0.00069463, 0.00043489, 0.00025272, 0.00013031, 0.0000527734,
- 0.00001000, 0.00000000
-};
-
-/*
-static const double kaiser12_table[36] = {
- 0.99440475, 1.00000000, 0.99440475, 0.97779076, 0.95066529, 0.91384741,
- 0.86843014, 0.81573067, 0.75723148, 0.69451601, 0.62920216, 0.56287762,
- 0.49704014, 0.43304576, 0.37206735, 0.31506490, 0.26276832, 0.21567274,
- 0.17404546, 0.13794294, 0.10723616, 0.08164178, 0.06075685, 0.04409466,
- 0.03111947, 0.02127838, 0.01402878, 0.00886058, 0.00531256, 0.00298291,
- 0.00153438, 0.00069463, 0.00025272, 0.0000527734, 0.00000500, 0.00000000};
-*/
-static const double kaiser10_table[36] = {
- 0.99537781, 1.00000000, 0.99537781, 0.98162644, 0.95908712, 0.92831446,
- 0.89005583, 0.84522401, 0.79486424, 0.74011713, 0.68217934, 0.62226347,
- 0.56155915, 0.50119680, 0.44221549, 0.38553619, 0.33194107, 0.28205962,
- 0.23636152, 0.19515633, 0.15859932, 0.12670280, 0.09935205, 0.07632451,
- 0.05731132, 0.04193980, 0.02979584, 0.02044510, 0.01345224, 0.00839739,
- 0.00488951, 0.00257636, 0.00115101, 0.00035515, 0.00000000, 0.00000000
-};
-
-static const double kaiser8_table[36] = {
- 0.99635258, 1.00000000, 0.99635258, 0.98548012, 0.96759014, 0.94302200,
- 0.91223751, 0.87580811, 0.83439927, 0.78875245, 0.73966538, 0.68797126,
- 0.63451750, 0.58014482, 0.52566725, 0.47185369, 0.41941150, 0.36897272,
- 0.32108304, 0.27619388, 0.23465776, 0.19672670, 0.16255380, 0.13219758,
- 0.10562887, 0.08273982, 0.06335451, 0.04724088, 0.03412321, 0.02369490,
- 0.01563093, 0.00959968, 0.00527363, 0.00233883, 0.00050000, 0.00000000
-};
-
-static const double kaiser6_table[36] = {
- 0.99733006, 1.00000000, 0.99733006, 0.98935595, 0.97618418, 0.95799003,
- 0.93501423, 0.90755855, 0.87598009, 0.84068475, 0.80211977, 0.76076565,
- 0.71712752, 0.67172623, 0.62508937, 0.57774224, 0.53019925, 0.48295561,
- 0.43647969, 0.39120616, 0.34752997, 0.30580127, 0.26632152, 0.22934058,
- 0.19505503, 0.16360756, 0.13508755, 0.10953262, 0.08693120, 0.06722600,
- 0.05031820, 0.03607231, 0.02432151, 0.01487334, 0.00752000, 0.00000000
-};
-
-struct FuncDef
-{
- const double *table;
- int oversample;
-};
-
-static struct FuncDef _KAISER12 = { kaiser12_table, 64 };
-
-#define KAISER12 (&_KAISER12)
-/*static struct FuncDef _KAISER12 = {kaiser12_table, 32};
-#define KAISER12 (&_KAISER12)*/
-static struct FuncDef _KAISER10 = { kaiser10_table, 32 };
-
-#define KAISER10 (&_KAISER10)
-static struct FuncDef _KAISER8 = { kaiser8_table, 32 };
-
-#define KAISER8 (&_KAISER8)
-static struct FuncDef _KAISER6 = { kaiser6_table, 32 };
-
-#define KAISER6 (&_KAISER6)
-
-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).
-*/
-static const struct QualityMapping quality_map[11] = {
- {8, 4, 0.830f, 0.860f, KAISER6}, /* Q0 */
- {16, 4, 0.850f, 0.880f, KAISER6}, /* Q1 */
- {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, 16, 0.922f, 0.940f, KAISER10}, /* Q5 *//* 89.1% cutoff (~100 dB stop) 10 */
- {96, 16, 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, 32, 0.968f, 0.968f, KAISER12}, /* Q9 *//* 95.5% cutoff (~100 dB stop) 10 */
- {256, 32, 0.975f, 0.975f, KAISER12}, /* Q10 *//* 96.6% cutoff (~100 dB stop) 10 */
-};
-
-/*8,24,40,56,80,104,128,160,200,256,320*/
-#ifdef DOUBLE_PRECISION
-static double
-compute_func (double x, struct FuncDef *func)
-{
- double y, frac;
-#else
-static double
-compute_func (float x, struct FuncDef *func)
-{
- float y, frac;
-#endif
- double interp[4];
- int ind;
- y = x * func->oversample;
- ind = (int) floor (y);
- frac = (y - ind);
- /* CSE with handle the repeated powers */
- interp[3] = -0.1666666667 * frac + 0.1666666667 * (frac * frac * frac);
- interp[2] = frac + 0.5 * (frac * frac) - 0.5 * (frac * frac * frac);
- /*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac; */
- interp[0] =
- -0.3333333333 * frac + 0.5 * (frac * frac) -
- 0.1666666667 * (frac * frac * frac);
- /* Just to make sure we don't have rounding problems */
- interp[1] = 1.f - interp[3] - interp[2] - interp[0];
-
- /*sum = frac*accum[1] + (1-frac)*accum[2]; */
- return interp[0] * func->table[ind] + interp[1] * func->table[ind + 1] +
- interp[2] * func->table[ind + 2] + interp[3] * func->table[ind + 3];
-}
-
-#if 0
-#include <stdio.h>
-int
-main (int argc, char **argv)
-{
- int i;
- for (i = 0; i < 256; i++) {
- printf ("%f\n", compute_func (i / 256., KAISER12));
- }
- return 0;
-}
-#endif
-
-#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, struct FuncDef *window_func)
-{
- /*fprintf (stderr, "%f ", x); */
- float xx = x * cutoff;
- if (fabs (x) < 1e-6f)
- return WORD2INT (32768. * cutoff);
- else if (fabs (x) > .5f * N)
- return 0;
- /*FIXME: Can it really be any slower than this? */
- return WORD2INT (32768. * cutoff * sin (G_PI * xx) / (G_PI * xx) *
- compute_func (fabs (2. * x / N), window_func));
-}
-#else
-/* The slow way of computing a sinc for the table. Should improve that some day */
-#ifdef DOUBLE_PRECISION
-static spx_word16_t
-sinc (double cutoff, double x, int N, struct FuncDef *window_func)
-{
- /*fprintf (stderr, "%f ", x); */
- double xx = x * cutoff;
-#else
-static spx_word16_t
-sinc (float cutoff, float x, int N, struct FuncDef *window_func)
-{
- /*fprintf (stderr, "%f ", x); */
- float xx = x * cutoff;
-#endif
- if (fabs (x) < 1e-6)
- return cutoff;
- else if (fabs (x) > .5 * N)
- return 0;
- /*FIXME: Can it really be any slower than this? */
- return cutoff * sin (G_PI * xx) / (G_PI * xx) * compute_func (fabs (2. * x /
- N), window_func);
-}
-#endif
-
-#ifdef FIXED_POINT
-static void
-cubic_coef (spx_word16_t x, spx_word16_t interp[4])
-{
- /* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation
- but I know it's MMSE-optimal on a sinc */
- spx_word16_t x2, x3;
- x2 = MULT16_16_P15 (x, x);
- x3 = MULT16_16_P15 (x, x2);
- interp[0] =
- PSHR32 (MULT16_16 (QCONST16 (-0.16667f, 15),
- x) + MULT16_16 (QCONST16 (0.16667f, 15), x3), 15);
- interp[1] =
- EXTRACT16 (EXTEND32 (x) + SHR32 (SUB32 (EXTEND32 (x2), EXTEND32 (x3)),
- 1));
- interp[3] =
- PSHR32 (MULT16_16 (QCONST16 (-0.33333f, 15),
- x) + MULT16_16 (QCONST16 (.5f, 15),
- x2) - MULT16_16 (QCONST16 (0.16667f, 15), x3), 15);
- /* Just to make sure we don't have rounding problems */
- interp[2] = Q15_ONE - interp[0] - interp[1] - interp[3];
- if (interp[2] < 32767)
- interp[2] += 1;
-}
-#else
-static void
-cubic_coef (spx_word16_t frac, spx_word16_t interp[4])
-{
- /* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation
- but I know it's MMSE-optimal on a sinc */
- interp[0] = -0.16667f * frac + 0.16667f * frac * frac * frac;
- interp[1] = frac + 0.5f * frac * frac - 0.5f * frac * frac * frac;
- /*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac; */
- interp[3] =
- -0.33333f * frac + 0.5f * frac * frac - 0.16667f * frac * frac * frac;
- /* Just to make sure we don't have rounding problems */
- interp[2] = 1. - interp[0] - interp[1] - interp[3];
-}
-#endif
-
-#ifndef DOUBLE_PRECISION
-static int
-resampler_basic_direct_single (SpeexResamplerState * st,
- spx_uint32_t channel_index, const spx_word16_t * in, spx_uint32_t * in_len,
- spx_word16_t * out, spx_uint32_t * out_len)
-{
- const int N = st->filt_len;
- int out_sample = 0;
- int last_sample = st->last_sample[channel_index];
- spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
- const spx_word16_t *sinc_table = st->sinc_table;
- const int out_stride = st->out_stride;
- const int int_advance = st->int_advance;
- const int frac_advance = st->frac_advance;
- const spx_uint32_t den_rate = st->den_rate;
- spx_word32_t sum;
- int j;
-
- while (!(last_sample >= (spx_int32_t) * in_len
- || out_sample >= (spx_int32_t) * out_len)) {
- const spx_word16_t *sinc = &sinc_table[samp_frac_num * N];
- const spx_word16_t *iptr = &in[last_sample];
-
- SSE_FALLBACK (INNER_PRODUCT_SINGLE)
- NEON_FALLBACK (INNER_PRODUCT_SINGLE)
- sum = 0;
- for (j = 0; j < N; j++)
- sum += MULT16_16 (sinc[j], iptr[j]);
-
-/* This code is slower on most DSPs which have only 2 accumulators.
- Plus this forces truncation to 32 bits and you lose the HW guard bits.
- I think we can trust the compiler and let it vectorize and/or unroll itself.
- spx_word32_t accum[4] = {0,0,0,0};
- for(j=0;j<N;j+=4) {
- accum[0] += MULT16_16(sinc[j], iptr[j]);
- accum[1] += MULT16_16(sinc[j+1], iptr[j+1]);
- accum[2] += MULT16_16(sinc[j+2], iptr[j+2]);
- accum[3] += MULT16_16(sinc[j+3], iptr[j+3]);
- }
- sum = accum[0] + accum[1] + accum[2] + accum[3];
-*/
-#if defined(OVERRIDE_INNER_PRODUCT_SINGLE) && defined(_USE_NEON)
- NEON_IMPLEMENTATION (INNER_PRODUCT_SINGLE)
- sum = inner_product_single (sinc, iptr, N);
- NEON_END (INNER_PRODUCT_SINGLE)
-#elif defined(OVERRIDE_INNER_PRODUCT_SINGLE) && defined(_USE_SSE)
- SSE_IMPLEMENTATION (INNER_PRODUCT_SINGLE)
- sum = inner_product_single (sinc, iptr, N);
- SSE_END (INNER_PRODUCT_SINGLE)
-#endif
- out[out_stride * out_sample++] = SATURATE32PSHR (sum, 15, 32767);
- last_sample += int_advance;
- samp_frac_num += frac_advance;
- if (samp_frac_num >= den_rate) {
- samp_frac_num -= den_rate;
- last_sample++;
- }
- }
-
- st->last_sample[channel_index] = last_sample;
- st->samp_frac_num[channel_index] = samp_frac_num;
- return out_sample;
-}
-#endif
-
-#ifdef FIXED_POINT
-#else
-/* This is the same as the previous function, except with a double-precision accumulator */
-static int
-resampler_basic_direct_double (SpeexResamplerState * st,
- spx_uint32_t channel_index, const spx_word16_t * in, spx_uint32_t * in_len,
- spx_word16_t * out, spx_uint32_t * out_len)
-{
- const int N = st->filt_len;
- int out_sample = 0;
- int last_sample = st->last_sample[channel_index];
- spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
- const spx_word16_t *sinc_table = st->sinc_table;
- const int out_stride = st->out_stride;
- const int int_advance = st->int_advance;
- const int frac_advance = st->frac_advance;
- const spx_uint32_t den_rate = st->den_rate;
- double sum;
- int j;
-
- while (!(last_sample >= (spx_int32_t) * in_len
- || out_sample >= (spx_int32_t) * out_len)) {
- const spx_word16_t *sinc = &sinc_table[samp_frac_num * N];
- const spx_word16_t *iptr = &in[last_sample];
-
- SSE2_FALLBACK (INNER_PRODUCT_DOUBLE)
- double accum[4] = { 0, 0, 0, 0 };
-
- for (j = 0; j < N; j += 4) {
- accum[0] += sinc[j] * iptr[j];
- accum[1] += sinc[j + 1] * iptr[j + 1];
- accum[2] += sinc[j + 2] * iptr[j + 2];
- accum[3] += sinc[j + 3] * iptr[j + 3];
- }
- sum = accum[0] + accum[1] + accum[2] + accum[3];
-#if defined(OVERRIDE_INNER_PRODUCT_DOUBLE) && defined(_USE_SSE2)
- SSE2_IMPLEMENTATION (INNER_PRODUCT_DOUBLE)
- sum = inner_product_double (sinc, iptr, N);
- SSE2_END (INNER_PRODUCT_DOUBLE)
-#endif
- out[out_stride * out_sample++] = PSHR32 (sum, 15);
- last_sample += int_advance;
- samp_frac_num += frac_advance;
- if (samp_frac_num >= den_rate) {
- samp_frac_num -= den_rate;
- last_sample++;
- }
- }
-
- st->last_sample[channel_index] = last_sample;
- st->samp_frac_num[channel_index] = samp_frac_num;
- return out_sample;
-}
-#endif
-
-#ifndef DOUBLE_PRECISION
-static int
-resampler_basic_interpolate_single (SpeexResamplerState * st,
- spx_uint32_t channel_index, const spx_word16_t * in, spx_uint32_t * in_len,
- spx_word16_t * out, spx_uint32_t * out_len)
-{
- const int N = st->filt_len;
- int out_sample = 0;
- int last_sample = st->last_sample[channel_index];
- spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
- const int out_stride = st->out_stride;
- const int int_advance = st->int_advance;
- const int frac_advance = st->frac_advance;
- const spx_uint32_t den_rate = st->den_rate;
- int j;
- spx_word32_t sum;
-
- while (!(last_sample >= (spx_int32_t) * in_len
- || out_sample >= (spx_int32_t) * out_len)) {
- const spx_word16_t *iptr = &in[last_sample];
-
- const int offset = samp_frac_num * st->oversample / st->den_rate;
-#ifdef FIXED_POINT
- const spx_word16_t frac =
- ((((gint64) samp_frac_num * (gint64) st->oversample) % st->den_rate)
- << 15) / st->den_rate;
-#else
- const spx_word16_t frac =
- ((float) ((samp_frac_num * st->oversample) % st->den_rate)) /
- st->den_rate;
-#endif
- spx_word16_t interp[4];
-
-
- SSE_FALLBACK (INTERPOLATE_PRODUCT_SINGLE)
- spx_word32_t accum[4] = { 0, 0, 0, 0 };
-
- for (j = 0; j < N; j++) {
- const spx_word16_t curr_in = iptr[j];
- accum[0] +=
- MULT16_16 (curr_in,
- st->sinc_table[4 + (j + 1) * st->oversample - offset - 2]);
- accum[1] +=
- MULT16_16 (curr_in,
- st->sinc_table[4 + (j + 1) * st->oversample - offset - 1]);
- accum[2] +=
- MULT16_16 (curr_in,
- st->sinc_table[4 + (j + 1) * st->oversample - offset]);
- accum[3] +=
- MULT16_16 (curr_in,
- st->sinc_table[4 + (j + 1) * st->oversample - offset + 1]);
- }
-
- cubic_coef (frac, interp);
- sum =
- MULT16_32_Q15 (interp[0], SHR32 (accum[0],
- 1)) + MULT16_32_Q15 (interp[1], SHR32 (accum[1],
- 1)) + MULT16_32_Q15 (interp[2], SHR32 (accum[2],
- 1)) + MULT16_32_Q15 (interp[3], SHR32 (accum[3], 1));
-#if defined(OVERRIDE_INTERPOLATE_PRODUCT_SINGLE) && defined(_USE_SSE)
- SSE_IMPLEMENTATION (INTERPOLATE_PRODUCT_SINGLE)
- cubic_coef (frac, interp);
- sum =
- interpolate_product_single (iptr,
- st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample,
- interp);
- SSE_END (INTERPOLATE_PRODUCT_SINGLE)
-#endif
- out[out_stride * out_sample++] = SATURATE32PSHR (sum, 14, 32767);
- last_sample += int_advance;
- samp_frac_num += frac_advance;
- if (samp_frac_num >= den_rate) {
- samp_frac_num -= den_rate;
- last_sample++;
- }
- }
-
- st->last_sample[channel_index] = last_sample;
- st->samp_frac_num[channel_index] = samp_frac_num;
- return out_sample;
-}
-#endif
-
-#ifdef FIXED_POINT
-#else
-/* This is the same as the previous function, except with a double-precision accumulator */
-static int
-resampler_basic_interpolate_double (SpeexResamplerState * st,
- spx_uint32_t channel_index, const spx_word16_t * in, spx_uint32_t * in_len,
- spx_word16_t * out, spx_uint32_t * out_len)
-{
- const int N = st->filt_len;
- int out_sample = 0;
- int last_sample = st->last_sample[channel_index];
- spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
- const int out_stride = st->out_stride;
- const int int_advance = st->int_advance;
- const int frac_advance = st->frac_advance;
- const spx_uint32_t den_rate = st->den_rate;
- int j;
- spx_word32_t sum;
-
- while (!(last_sample >= (spx_int32_t) * in_len
- || out_sample >= (spx_int32_t) * out_len)) {
- const spx_word16_t *iptr = &in[last_sample];
-
- const int offset = samp_frac_num * st->oversample / st->den_rate;
-#ifdef FIXED_POINT
- const spx_word16_t frac =
- PDIV32 (SHL32 ((samp_frac_num * st->oversample) % st->den_rate, 15),
- st->den_rate);
-#else
-#ifdef DOUBLE_PRECISION
- const spx_word16_t frac =
- ((double) ((samp_frac_num * st->oversample) % st->den_rate)) /
- st->den_rate;
-#else
- const spx_word16_t frac =
- ((float) ((samp_frac_num * st->oversample) % st->den_rate)) /
- st->den_rate;
-#endif
-#endif
- spx_word16_t interp[4];
-
-
- SSE2_FALLBACK (INTERPOLATE_PRODUCT_DOUBLE)
- double accum[4] = { 0, 0, 0, 0 };
-
- for (j = 0; j < N; j++) {
- const double curr_in = iptr[j];
- accum[0] +=
- MULT16_16 (curr_in,
- st->sinc_table[4 + (j + 1) * st->oversample - offset - 2]);
- accum[1] +=
- MULT16_16 (curr_in,
- st->sinc_table[4 + (j + 1) * st->oversample - offset - 1]);
- accum[2] +=
- MULT16_16 (curr_in,
- st->sinc_table[4 + (j + 1) * st->oversample - offset]);
- accum[3] +=
- MULT16_16 (curr_in,
- st->sinc_table[4 + (j + 1) * st->oversample - offset + 1]);
- }
-
- cubic_coef (frac, interp);
- sum =
- MULT16_32_Q15 (interp[0], accum[0]) + MULT16_32_Q15 (interp[1],
- accum[1]) + MULT16_32_Q15 (interp[2],
- accum[2]) + MULT16_32_Q15 (interp[3], accum[3]);
-#if defined(OVERRIDE_INTERPOLATE_PRODUCT_DOUBLE) && defined(_USE_SSE2)
- SSE2_IMPLEMENTATION (INTERPOLATE_PRODUCT_DOUBLE)
- cubic_coef (frac, interp);
- sum =
- interpolate_product_double (iptr,
- st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample,
- interp);
- SSE2_END (INTERPOLATE_PRODUCT_DOUBLE)
-#endif
- out[out_stride * out_sample++] = PSHR32 (sum, 15);
- last_sample += int_advance;
- samp_frac_num += frac_advance;
- if (samp_frac_num >= den_rate) {
- samp_frac_num -= den_rate;
- last_sample++;
- }
- }
-
- st->last_sample[channel_index] = last_sample;
- st->samp_frac_num[channel_index] = samp_frac_num;
- return out_sample;
-}
-#endif
-
-static void
-update_filter (SpeexResamplerState * st)
-{
- spx_uint32_t old_length;
-
- old_length = st->filt_len;
- st->oversample = quality_map[st->quality].oversample;
- st->filt_len = quality_map[st->quality].base_length;
-
- if (st->num_rate > st->den_rate) {
- /* down-sampling */
- st->cutoff =
- quality_map[st->quality].downsample_bandwidth * st->den_rate /
- st->num_rate;
- /* FIXME: divide the numerator and denominator by a certain amount if they're too large */
- st->filt_len = st->filt_len * st->num_rate / st->den_rate;
- /* Round down to make sure we have a multiple of 4 */
- st->filt_len &= (~0x3);
- if (2 * st->den_rate < st->num_rate)
- st->oversample >>= 1;
- if (4 * st->den_rate < st->num_rate)
- st->oversample >>= 1;
- if (8 * st->den_rate < st->num_rate)
- st->oversample >>= 1;
- if (16 * st->den_rate < st->num_rate)
- st->oversample >>= 1;
- if (st->oversample < 1)
- st->oversample = 1;
- } else {
- /* up-sampling */
- st->cutoff = quality_map[st->quality].upsample_bandwidth;
- }
-
- /* Choose the resampling type that requires the least amount of memory */
- /* Or if the full sinc table is explicitely requested, use that */
- if (st->use_full_sinc_table || (st->den_rate <= st->oversample)) {
- spx_uint32_t i;
- if (!st->sinc_table)
- st->sinc_table =
- (spx_word16_t *) speex_alloc (st->filt_len * st->den_rate *
- sizeof (spx_word16_t));
- else if (st->sinc_table_length < st->filt_len * st->den_rate) {
- st->sinc_table =
- (spx_word16_t *) speex_realloc (st->sinc_table,
- st->filt_len * st->den_rate * sizeof (spx_word16_t));
- st->sinc_table_length = st->filt_len * st->den_rate;
- }
- for (i = 0; i < st->den_rate; i++) {
- spx_int32_t j;
- for (j = 0; j < st->filt_len; j++) {
- st->sinc_table[i * st->filt_len + j] =
- sinc (st->cutoff, ((j - (spx_int32_t) st->filt_len / 2 + 1) -
-#ifdef DOUBLE_PRECISION
- ((double) i) / st->den_rate), st->filt_len,
-#else
- ((float) i) / st->den_rate), st->filt_len,
-#endif
- quality_map[st->quality].window_func);
- }
- }
-#ifdef FIXED_POINT
- st->resampler_ptr = resampler_basic_direct_single;
-#else
-#ifdef DOUBLE_PRECISION
- st->resampler_ptr = resampler_basic_direct_double;
-#else
- if (st->quality > 8)
- st->resampler_ptr = resampler_basic_direct_double;
- else
- st->resampler_ptr = resampler_basic_direct_single;
-#endif
-#endif
- /*fprintf (stderr, "resampler uses direct sinc table and normalised cutoff %f\n", cutoff); */
- } else {
- spx_int32_t i;
- if (!st->sinc_table)
- st->sinc_table =
- (spx_word16_t *) speex_alloc ((st->filt_len * st->oversample +
- 8) * sizeof (spx_word16_t));
- else if (st->sinc_table_length < st->filt_len * st->oversample + 8) {
- st->sinc_table =
- (spx_word16_t *) speex_realloc (st->sinc_table,
- (st->filt_len * st->oversample + 8) * sizeof (spx_word16_t));
- st->sinc_table_length = st->filt_len * st->oversample + 8;
- }
- for (i = -4; i < (spx_int32_t) (st->oversample * st->filt_len + 4); i++)
- st->sinc_table[i + 4] =
-#ifdef DOUBLE_PRECISION
- sinc (st->cutoff, (i / (double) st->oversample - st->filt_len / 2),
-#else
- sinc (st->cutoff, (i / (float) st->oversample - st->filt_len / 2),
-#endif
- st->filt_len, quality_map[st->quality].window_func);
-#ifdef FIXED_POINT
- st->resampler_ptr = resampler_basic_interpolate_single;
-#else
-#ifdef DOUBLE_PRECISION
- st->resampler_ptr = resampler_basic_interpolate_double;
-#else
- if (st->quality > 8)
- st->resampler_ptr = resampler_basic_interpolate_double;
- else
- st->resampler_ptr = resampler_basic_interpolate_single;
-#endif
-#endif
- /*fprintf (stderr, "resampler uses interpolated sinc table and normalised cutoff %f\n", cutoff); */
- }
- st->int_advance = st->num_rate / st->den_rate;
- st->frac_advance = st->num_rate % st->den_rate;
-
-
- /* Here's the place where we update the filter memory to take into account
- the change in filter length. It's probably the messiest part of the code
- due to handling of lots of corner cases. */
- if (!st->mem) {
- spx_uint32_t i;
- st->mem_alloc_size = st->filt_len - 1 + st->buffer_size;
- st->mem =
- (spx_word16_t *) speex_alloc (st->nb_channels * st->mem_alloc_size *
- sizeof (spx_word16_t));
- for (i = 0; i < st->nb_channels * st->mem_alloc_size; i++)
- st->mem[i] = 0;
- /*speex_warning("init filter"); */
- } else if (!st->started) {
- spx_uint32_t i;
- st->mem_alloc_size = st->filt_len - 1 + st->buffer_size;
- st->mem =
- (spx_word16_t *) speex_realloc (st->mem,
- st->nb_channels * st->mem_alloc_size * sizeof (spx_word16_t));
- for (i = 0; i < st->nb_channels * st->mem_alloc_size; i++)
- st->mem[i] = 0;
- /*speex_warning("reinit filter"); */
- } else if (st->filt_len > old_length) {
- spx_int32_t i;
- /* Increase the filter length */
- /*speex_warning("increase filter size"); */
- int old_alloc_size = st->mem_alloc_size;
- if ((st->filt_len - 1 + st->buffer_size) > st->mem_alloc_size) {
- st->mem_alloc_size = st->filt_len - 1 + st->buffer_size;
- st->mem =
- (spx_word16_t *) speex_realloc (st->mem,
- st->nb_channels * st->mem_alloc_size * sizeof (spx_word16_t));
- }
- for (i = st->nb_channels - 1; i >= 0; i--) {
- spx_int32_t j;
- spx_uint32_t olen = old_length;
- /*if (st->magic_samples[i]) */
- {
- /* Try and remove the magic samples as if nothing had happened */
-
- /* FIXME: This is wrong but for now we need it to avoid going over the array bounds */
- olen = old_length + 2 * st->magic_samples[i];
- for (j = old_length - 2 + st->magic_samples[i]; j >= 0; j--)
- st->mem[i * st->mem_alloc_size + j + st->magic_samples[i]] =
- st->mem[i * old_alloc_size + j];
- for (j = 0; j < st->magic_samples[i]; j++)
- st->mem[i * st->mem_alloc_size + j] = 0;
- st->magic_samples[i] = 0;
- }
- if (st->filt_len > olen) {
- /* If the new filter length is still bigger than the "augmented" length */
- /* Copy data going backward */
- for (j = 0; j < olen - 1; j++)
- st->mem[i * st->mem_alloc_size + (st->filt_len - 2 - j)] =
- st->mem[i * st->mem_alloc_size + (olen - 2 - j)];
- /* Then put zeros for lack of anything better */
- for (; j < st->filt_len - 1; j++)
- st->mem[i * st->mem_alloc_size + (st->filt_len - 2 - j)] = 0;
- /* Adjust last_sample */
- st->last_sample[i] += (st->filt_len - olen) / 2;
- } else {
- /* Put back some of the magic! */
- st->magic_samples[i] = (olen - st->filt_len) / 2;
- for (j = 0; j < st->filt_len - 1 + st->magic_samples[i]; j++)
- st->mem[i * st->mem_alloc_size + j] =
- st->mem[i * st->mem_alloc_size + j + st->magic_samples[i]];
- }
- }
- } else if (st->filt_len < old_length) {
- spx_uint32_t i;
- /* Reduce filter length, this a bit tricky. We need to store some of the memory as "magic"
- samples so they can be used directly as input the next time(s) */
- for (i = 0; i < st->nb_channels; i++) {
- spx_uint32_t j;
- spx_uint32_t old_magic = st->magic_samples[i];
- st->magic_samples[i] = (old_length - st->filt_len) / 2;
- /* We must copy some of the memory that's no longer used */
- /* Copy data going backward */
- for (j = 0; j < st->filt_len - 1 + st->magic_samples[i] + old_magic; j++)
- st->mem[i * st->mem_alloc_size + j] =
- st->mem[i * st->mem_alloc_size + j + st->magic_samples[i]];
- st->magic_samples[i] += old_magic;
- }
- }
-
-}
-
-EXPORT SpeexResamplerState *
-speex_resampler_init (spx_uint32_t nb_channels, spx_uint32_t in_rate,
- spx_uint32_t out_rate, int quality,
- SpeexResamplerSincFilterMode sinc_filter_mode,
- spx_uint32_t sinc_filter_auto_threshold, int *err)
-{
- return speex_resampler_init_frac (nb_channels, in_rate, out_rate, in_rate,
- out_rate, quality, sinc_filter_mode, sinc_filter_auto_threshold, err);
-}
-
-#if defined HAVE_ORC && !defined DISABLE_ORC
-static void
-check_insn_set (SpeexResamplerState * st, const char *name)
-{
- if (!name)
- return;
-#ifdef _USE_SSE
- if (!strcmp (name, "sse"))
- st->use_sse = 1;
-#endif
-#ifdef _USE_SSE2
- if (!strcmp (name, "sse2"))
- st->use_sse = st->use_sse2 = 1;
-#endif
-#ifdef _USE_NEON
- if (!strcmp (name, "neon"))
- st->use_neon = 1;
-#endif
-}
-#endif
-
-EXPORT SpeexResamplerState *
-speex_resampler_init_frac (spx_uint32_t nb_channels, spx_uint32_t ratio_num,
- spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate,
- int quality, SpeexResamplerSincFilterMode sinc_filter_mode,
- spx_uint32_t sinc_filter_auto_threshold, int *err)
-{
- spx_uint32_t i;
- SpeexResamplerState *st;
- int use_full_sinc_table = 0;
- if (quality > 10 || quality < 0) {
- if (err)
- *err = RESAMPLER_ERR_INVALID_ARG;
- return NULL;
- }
- if (ratio_den == 0) {
- if (err)
- *err = RESAMPLER_ERR_INVALID_ARG;
- return NULL;
- }
- switch (sinc_filter_mode) {
- case RESAMPLER_SINC_FILTER_INTERPOLATED:
- use_full_sinc_table = 0;
- break;
- case RESAMPLER_SINC_FILTER_FULL:
- use_full_sinc_table = 1;
- break;
- case RESAMPLER_SINC_FILTER_AUTO:
- /* Handled below */
- break;
- default:
- if (err)
- *err = RESAMPLER_ERR_INVALID_ARG;
- return NULL;
- }
-
- st = (SpeexResamplerState *) speex_alloc (sizeof (SpeexResamplerState));
- st->initialised = 0;
- st->started = 0;
- st->in_rate = 0;
- st->out_rate = 0;
- st->num_rate = 0;
- st->den_rate = 0;
- st->quality = -1;
- st->sinc_table_length = 0;
- st->mem_alloc_size = 0;
- st->filt_len = 0;
- st->mem = 0;
- st->resampler_ptr = 0;
- st->use_full_sinc_table = use_full_sinc_table;
-
- st->cutoff = 1.f;
- st->nb_channels = nb_channels;
- st->in_stride = 1;
- st->out_stride = 1;
-
-#ifdef FIXED_POINT
- st->buffer_size = 160;
-#else
- st->buffer_size = 160;
-#endif
-
- st->use_sse = st->use_sse2 = 0;
- st->use_neon = 0;
-#if defined HAVE_ORC && !defined DISABLE_ORC
- orc_init ();
- {
- OrcTarget *target = orc_target_get_default ();
- if (target) {
- unsigned int flags = orc_target_get_default_flags (target);
- check_insn_set (st, orc_target_get_name (target));
- for (i = 0; i < 32; ++i) {
- if (flags & (1U << i)) {
- check_insn_set (st, orc_target_get_flag_name (target, i));
- }
- }
- }
- }
-#endif
-
- /* Per channel data */
- st->last_sample = (spx_int32_t *) speex_alloc (nb_channels * sizeof (int));
- st->magic_samples = (spx_uint32_t *) speex_alloc (nb_channels * sizeof (int));
- st->samp_frac_num = (spx_uint32_t *) speex_alloc (nb_channels * sizeof (int));
- for (i = 0; i < nb_channels; i++) {
- st->last_sample[i] = 0;
- st->magic_samples[i] = 0;
- st->samp_frac_num[i] = 0;
- }
-
- speex_resampler_set_quality (st, quality);
- speex_resampler_set_rate_frac (st, ratio_num, ratio_den, in_rate, out_rate);
-
- if (sinc_filter_mode == RESAMPLER_SINC_FILTER_AUTO) {
- /*
- Estimate how big the filter table would become if the full mode were to be used
- calculations used correspond to the ones in update_filter()
- if the size is bigger than the threshold, use interpolated sinc instead
- */
- spx_uint32_t base_filter_length = st->filt_len =
- quality_map[st->quality].base_length;
- spx_uint32_t filter_table_size =
- base_filter_length * st->den_rate * sizeof (spx_uint16_t);
- st->use_full_sinc_table =
- (filter_table_size > sinc_filter_auto_threshold) ? 0 : 1;
- }
-
- update_filter (st);
-
- st->initialised = 1;
- if (err)
- *err = RESAMPLER_ERR_SUCCESS;
-
- return st;
-}
-
-EXPORT void
-speex_resampler_destroy (SpeexResamplerState * st)
-{
- speex_free (st->mem);
- speex_free (st->sinc_table);
- speex_free (st->last_sample);
- speex_free (st->magic_samples);
- speex_free (st->samp_frac_num);
- speex_free (st);
-}
-
-static int
-speex_resampler_process_native (SpeexResamplerState * st,
- spx_uint32_t channel_index, spx_uint32_t * in_len, spx_word16_t * out,
- spx_uint32_t * out_len)
-{
- int j = 0;
- const int N = st->filt_len;
- int out_sample = 0;
- spx_word16_t *mem = st->mem + channel_index * st->mem_alloc_size;
- spx_uint32_t ilen;
-
- st->started = 1;
-
- /* Call the right resampler through the function ptr */
- out_sample = st->resampler_ptr (st, channel_index, mem, in_len, out, out_len);
-
- if (st->last_sample[channel_index] < (spx_int32_t) * in_len)
- *in_len = st->last_sample[channel_index];
- *out_len = out_sample;
- st->last_sample[channel_index] -= *in_len;
-
- ilen = *in_len;
-
- for (j = 0; j < N - 1; ++j)
- mem[j] = mem[j + ilen];
-
- return RESAMPLER_ERR_SUCCESS;
-}
-
-static int
-speex_resampler_magic (SpeexResamplerState * st, spx_uint32_t channel_index,
- spx_word16_t ** out, spx_uint32_t out_len)
-{
- spx_uint32_t tmp_in_len = st->magic_samples[channel_index];
- spx_word16_t *mem = st->mem + channel_index * st->mem_alloc_size;
- const int N = st->filt_len;
-
- speex_resampler_process_native (st, channel_index, &tmp_in_len, *out,
- &out_len);
-
- st->magic_samples[channel_index] -= tmp_in_len;
-
- /* If we couldn't process all "magic" input samples, save the rest for next time */
- if (st->magic_samples[channel_index]) {
- spx_uint32_t i;
- for (i = 0; i < st->magic_samples[channel_index]; i++)
- mem[N - 1 + i] = mem[N - 1 + i + tmp_in_len];
- }
- *out += out_len * st->out_stride;
- return out_len;
-}
-
-#ifdef FIXED_POINT
-EXPORT int
-speex_resampler_process_int (SpeexResamplerState * st,
- spx_uint32_t channel_index, const spx_int16_t * in, spx_uint32_t * in_len,
- spx_int16_t * out, spx_uint32_t * out_len)
-#else
-#ifdef DOUBLE_PRECISION
-EXPORT int
-speex_resampler_process_float (SpeexResamplerState * st,
- spx_uint32_t channel_index, const double *in, spx_uint32_t * in_len,
- double *out, spx_uint32_t * out_len)
-#else
-EXPORT int
-speex_resampler_process_float (SpeexResamplerState * st,
- spx_uint32_t channel_index, const float *in, spx_uint32_t * in_len,
- float *out, spx_uint32_t * out_len)
-#endif
-#endif
-{
- int j;
- spx_uint32_t ilen = *in_len;
- spx_uint32_t olen = *out_len;
- spx_word16_t *x = st->mem + channel_index * st->mem_alloc_size;
- const int filt_offs = st->filt_len - 1;
- const spx_uint32_t xlen = st->mem_alloc_size - filt_offs;
- const int istride = st->in_stride;
-
- if (st->magic_samples[channel_index])
- olen -= speex_resampler_magic (st, channel_index, &out, olen);
- if (!st->magic_samples[channel_index]) {
- while (ilen) {
- spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen;
- spx_uint32_t ochunk = olen;
-
- if (in) {
- for (j = 0; j < ichunk; ++j)
- x[j + filt_offs] = in[j * istride];
- } else {
- for (j = 0; j < ichunk; ++j)
- x[j + filt_offs] = 0;
- }
- speex_resampler_process_native (st, channel_index, &ichunk, out, &ochunk);
- ilen -= ichunk;
- olen -= ochunk;
- out += ochunk * st->out_stride;
- if (in)
- in += ichunk * istride;
- if (olen == 0 && ichunk == 0)
- break;
- }
- }
- *in_len -= ilen;
- *out_len -= olen;
- return RESAMPLER_ERR_SUCCESS;
-}
-
-#ifdef FIXED_POINT
-EXPORT int
-speex_resampler_process_float (SpeexResamplerState * st,
- spx_uint32_t channel_index, const float *in, spx_uint32_t * in_len,
- float *out, spx_uint32_t * out_len)
-#else
-EXPORT int
-speex_resampler_process_int (SpeexResamplerState * st,
- spx_uint32_t channel_index, const spx_int16_t * in, spx_uint32_t * in_len,
- spx_int16_t * out, spx_uint32_t * out_len)
-#endif
-{
- int j;
- const int istride_save = st->in_stride;
- const int ostride_save = st->out_stride;
- spx_uint32_t ilen = *in_len;
- spx_uint32_t olen = *out_len;
- spx_word16_t *x = st->mem + channel_index * st->mem_alloc_size;
- const spx_uint32_t xlen = st->mem_alloc_size - (st->filt_len - 1);
-#ifdef VAR_ARRAYS
- const unsigned int ylen =
- (olen < FIXED_STACK_ALLOC) ? olen : FIXED_STACK_ALLOC;
- VARDECL (spx_word16_t * ystack);
- ALLOC (ystack, ylen, spx_word16_t);
-#else
- const unsigned int ylen = FIXED_STACK_ALLOC;
- spx_word16_t ystack[FIXED_STACK_ALLOC];
-#endif
-
- st->out_stride = 1;
-
- while (ilen) {
- spx_word16_t *y = ystack;
- spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen;
- spx_uint32_t ochunk = (olen > ylen) ? ylen : olen;
- spx_uint32_t omagic = 0;
-
- if (st->magic_samples[channel_index]) {
- omagic = speex_resampler_magic (st, channel_index, &y, ochunk);
- ochunk -= omagic;
- olen -= omagic;
- }
- if (!st->magic_samples[channel_index]) {
- if (in) {
- for (j = 0; j < ichunk; ++j)
-#ifdef FIXED_POINT
- x[j + st->filt_len - 1] = WORD2INT (in[j * istride_save]);
-#else
- x[j + st->filt_len - 1] = in[j * istride_save];
-#endif
- } else {
- for (j = 0; j < ichunk; ++j)
- x[j + st->filt_len - 1] = 0;
- }
-
- speex_resampler_process_native (st, channel_index, &ichunk, y, &ochunk);
- } else {
- ichunk = 0;
- ochunk = 0;
- }
-
- for (j = 0; j < ochunk + omagic; ++j)
-#ifdef FIXED_POINT
- out[j * ostride_save] = ystack[j];
-#else
- out[j * ostride_save] = WORD2INT (ystack[j]);
-#endif
-
- ilen -= ichunk;
- olen -= ochunk;
- out += (ochunk + omagic) * ostride_save;
- if (in)
- in += ichunk * istride_save;
- if (olen == 0 && ichunk == 0)
- break;
- }
- st->out_stride = ostride_save;
- *in_len -= ilen;
- *out_len -= olen;
-
- return RESAMPLER_ERR_SUCCESS;
-}
-
-#ifdef DOUBLE_PRECISION
-EXPORT int
-speex_resampler_process_interleaved_float (SpeexResamplerState * st,
- const double *in, spx_uint32_t * in_len, double *out,
- spx_uint32_t * out_len)
-#else
-EXPORT int
-speex_resampler_process_interleaved_float (SpeexResamplerState * st,
- const float *in, spx_uint32_t * in_len, float *out, spx_uint32_t * out_len)
-#endif
-{
- spx_uint32_t i;
- int istride_save, ostride_save;
- spx_uint32_t bak_len = *out_len;
- istride_save = st->in_stride;
- ostride_save = st->out_stride;
- st->in_stride = st->out_stride = st->nb_channels;
- for (i = 0; i < st->nb_channels; i++) {
- *out_len = bak_len;
- if (in != NULL)
- speex_resampler_process_float (st, i, in + i, in_len, out + i, out_len);
- else
- speex_resampler_process_float (st, i, NULL, in_len, out + i, out_len);
- }
- st->in_stride = istride_save;
- st->out_stride = ostride_save;
- return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT int
-speex_resampler_process_interleaved_int (SpeexResamplerState * st,
- const spx_int16_t * in, spx_uint32_t * in_len, spx_int16_t * out,
- spx_uint32_t * out_len)
-{
- spx_uint32_t i;
- int istride_save, ostride_save;
- spx_uint32_t bak_len = *out_len;
- istride_save = st->in_stride;
- ostride_save = st->out_stride;
- st->in_stride = st->out_stride = st->nb_channels;
- for (i = 0; i < st->nb_channels; i++) {
- *out_len = bak_len;
- if (in != NULL)
- speex_resampler_process_int (st, i, in + i, in_len, out + i, out_len);
- else
- speex_resampler_process_int (st, i, NULL, in_len, out + i, out_len);
- }
- st->in_stride = istride_save;
- st->out_stride = ostride_save;
- return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT int
-speex_resampler_set_rate (SpeexResamplerState * st, spx_uint32_t in_rate,
- spx_uint32_t out_rate)
-{
- return speex_resampler_set_rate_frac (st, in_rate, out_rate, in_rate,
- out_rate);
-}
-
-EXPORT void
-speex_resampler_get_rate (SpeexResamplerState * st, spx_uint32_t * in_rate,
- spx_uint32_t * out_rate)
-{
- *in_rate = st->in_rate;
- *out_rate = st->out_rate;
-}
-
-EXPORT int
-speex_resampler_set_rate_frac (SpeexResamplerState * st, spx_uint32_t ratio_num,
- spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate)
-{
- spx_uint32_t fact;
- spx_uint32_t old_den;
- spx_uint32_t i;
- if (st->in_rate == in_rate && st->out_rate == out_rate
- && st->num_rate == ratio_num && st->den_rate == ratio_den)
- return RESAMPLER_ERR_SUCCESS;
-
- old_den = st->den_rate;
- st->in_rate = in_rate;
- st->out_rate = out_rate;
- st->num_rate = ratio_num;
- st->den_rate = ratio_den;
- /* FIXME: This is terribly inefficient, but who cares (at least for now)? */
- for (fact = 2; fact <= IMIN (st->num_rate, st->den_rate); fact++) {
- while ((st->num_rate % fact == 0) && (st->den_rate % fact == 0)) {
- st->num_rate /= fact;
- st->den_rate /= fact;
- }
- }
-
- if (old_den > 0) {
- for (i = 0; i < st->nb_channels; i++) {
- st->samp_frac_num[i] =
- (gint64) st->samp_frac_num[i] * (gint64) st->den_rate / old_den;
- /* Safety net */
- if (st->samp_frac_num[i] >= st->den_rate)
- st->samp_frac_num[i] = st->den_rate - 1;
- }
- }
-
- if (st->initialised)
- update_filter (st);
- return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT void
-speex_resampler_get_ratio (SpeexResamplerState * st, spx_uint32_t * ratio_num,
- spx_uint32_t * ratio_den)
-{
- *ratio_num = st->num_rate;
- *ratio_den = st->den_rate;
-}
-
-EXPORT int
-speex_resampler_set_quality (SpeexResamplerState * st, int quality)
-{
- if (quality > 10 || quality < 0)
- return RESAMPLER_ERR_INVALID_ARG;
- if (st->quality == quality)
- return RESAMPLER_ERR_SUCCESS;
- st->quality = quality;
- if (st->initialised)
- update_filter (st);
- return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT void
-speex_resampler_get_quality (SpeexResamplerState * st, int *quality)
-{
- *quality = st->quality;
-}
-
-EXPORT void
-speex_resampler_set_input_stride (SpeexResamplerState * st, spx_uint32_t stride)
-{
- st->in_stride = stride;
-}
-
-EXPORT void
-speex_resampler_get_input_stride (SpeexResamplerState * st,
- spx_uint32_t * stride)
-{
- *stride = st->in_stride;
-}
-
-EXPORT void
-speex_resampler_set_output_stride (SpeexResamplerState * st,
- spx_uint32_t stride)
-{
- st->out_stride = stride;
-}
-
-EXPORT void
-speex_resampler_get_output_stride (SpeexResamplerState * st,
- spx_uint32_t * stride)
-{
- *stride = st->out_stride;
-}
-
-EXPORT int
-speex_resampler_get_input_latency (SpeexResamplerState * st)
-{
- return st->filt_len / 2;
-}
-
-EXPORT int
-speex_resampler_get_output_latency (SpeexResamplerState * st)
-{
- return ((st->filt_len / 2) * st->den_rate +
- (st->num_rate >> 1)) / st->num_rate;
-}
-
-EXPORT int
-speex_resampler_get_filt_len (SpeexResamplerState * st)
-{
- return st->filt_len;
-}
-
-EXPORT int
-speex_resampler_get_sinc_filter_mode (SpeexResamplerState * st)
-{
- return st->use_full_sinc_table;
-}
-
-EXPORT int
-speex_resampler_skip_zeros (SpeexResamplerState * st)
-{
- spx_uint32_t i;
- for (i = 0; i < st->nb_channels; i++)
- st->last_sample[i] = st->filt_len / 2;
- return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT int
-speex_resampler_reset_mem (SpeexResamplerState * st)
-{
- spx_uint32_t i;
- for (i = 0; i < st->nb_channels * (st->filt_len - 1); i++)
- st->mem[i] = 0;
- return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT const char *
-speex_resampler_strerror (int err)
-{
- switch (err) {
- case RESAMPLER_ERR_SUCCESS:
- return "Success.";
- case RESAMPLER_ERR_ALLOC_FAILED:
- return "Memory allocation failed.";
- case RESAMPLER_ERR_BAD_STATE:
- return "Bad resampler state.";
- case RESAMPLER_ERR_INVALID_ARG:
- return "Invalid argument.";
- case RESAMPLER_ERR_PTR_OVERLAP:
- return "Input and output buffers overlap.";
- default:
- return "Unknown error. Bad error code or strange version mismatch.";
- }
-}
+++ /dev/null
-/* Copyright (C) 2007-2008 Jean-Marc Valin
- * Copyright (C) 2008 Thorvald Natvig
- * Copyright (C) 2011 Texas Instruments
- * author Jyri Sarha
- */
-/**
- @file resample_neon.h
- @brief Resampler functions (NEON version)
-*/
-/*
- Redistribution and use in source and binary forms, with or without
- modification, are permitted provided that the following conditions
- are met:
-
- - Redistributions of source code must retain the above copyright
- notice, this list of conditions and the following disclaimer.
-
- - Redistributions in binary form must reproduce the above copyright
- notice, this list of conditions and the following disclaimer in the
- documentation and/or other materials provided with the distribution.
-
- - Neither the name of the Xiph.org Foundation nor the names of its
- contributors may be used to endorse or promote products derived from
- this software without specific prior written permission.
-
- THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
- ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
- LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
- A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
- CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
- EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
- PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
- LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
- NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
- SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#include <arm_neon.h>
-
-#ifdef FIXED_POINT
-#ifdef __thumb2__
-static inline int32_t saturate_32bit_to_16bit(int32_t a) {
- int32_t ret;
- asm ("ssat %[ret], #16, %[a]"
- : [ret] "=&r" (ret)
- : [a] "r" (a)
- : );
- return ret;
-}
-#else
-static inline int32_t saturate_32bit_to_16bit(int32_t a) {
- int32_t ret;
- asm ("vmov.s32 d0[0], %[a]\n"
- "vqmovn.s32 d0, q0\n"
- "vmov.s16 %[ret], d0[0]\n"
- : [ret] "=&r" (ret)
- : [a] "r" (a)
- : "q0");
- return ret;
-}
-#endif
-#undef WORD2INT
-#define WORD2INT(x) (saturate_32bit_to_16bit(x))
-
-#define OVERRIDE_INNER_PRODUCT_SINGLE
-/* Only works when len % 4 == 0 */
-static inline int32_t inner_product_single(const int16_t *a, const int16_t *b, unsigned int len)
-{
- int32_t ret;
- uint32_t remainder = len % 16;
- len = len - remainder;
-
- asm volatile (" cmp %[len], #0\n"
- " bne 1f\n"
- " vld1.16 {d16}, [%[b]]!\n"
- " vld1.16 {d20}, [%[a]]!\n"
- " subs %[remainder], %[remainder], #4\n"
- " vmull.s16 q0, d16, d20\n"
- " beq 5f\n"
- " b 4f\n"
- "1:"
- " vld1.16 {d16, d17, d18, d19}, [%[b]]!\n"
- " vld1.16 {d20, d21, d22, d23}, [%[a]]!\n"
- " subs %[len], %[len], #16\n"
- " vmull.s16 q0, d16, d20\n"
- " vmlal.s16 q0, d17, d21\n"
- " vmlal.s16 q0, d18, d22\n"
- " vmlal.s16 q0, d19, d23\n"
- " beq 3f\n"
- "2:"
- " vld1.16 {d16, d17, d18, d19}, [%[b]]!\n"
- " vld1.16 {d20, d21, d22, d23}, [%[a]]!\n"
- " subs %[len], %[len], #16\n"
- " vmlal.s16 q0, d16, d20\n"
- " vmlal.s16 q0, d17, d21\n"
- " vmlal.s16 q0, d18, d22\n"
- " vmlal.s16 q0, d19, d23\n"
- " bne 2b\n"
- "3:"
- " cmp %[remainder], #0\n"
- " beq 5f\n"
- "4:"
- " vld1.16 {d16}, [%[b]]!\n"
- " vld1.16 {d20}, [%[a]]!\n"
- " subs %[remainder], %[remainder], #4\n"
- " vmlal.s16 q0, d16, d20\n"
- " bne 4b\n"
- "5:"
- " vaddl.s32 q0, d0, d1\n"
- " vadd.s64 d0, d0, d1\n"
- " vqmovn.s64 d0, q0\n"
- " vqrshrn.s32 d0, q0, #15\n"
- " vmov.s16 %[ret], d0[0]\n"
- : [ret] "=&r" (ret), [a] "+r" (a), [b] "+r" (b),
- [len] "+r" (len), [remainder] "+r" (remainder)
- :
- : "cc", "q0",
- "d16", "d17", "d18", "d19",
- "d20", "d21", "d22", "d23");
-
- return ret;
-}
-#elif defined(FLOATING_POINT)
-
-static inline int32_t saturate_float_to_16bit(float a) {
- int32_t ret;
- asm ("vmov.f32 d0[0], %[a]\n"
- "vcvt.s32.f32 d0, d0, #15\n"
- "vqrshrn.s32 d0, q0, #15\n"
- "vmov.s16 %[ret], d0[0]\n"
- : [ret] "=&r" (ret)
- : [a] "r" (a)
- : "q0");
- return ret;
-}
-#undef WORD2INT
-#define WORD2INT(x) (saturate_float_to_16bit(x))
-
-#define OVERRIDE_INNER_PRODUCT_SINGLE
-/* Only works when len % 4 == 0 */
-static inline float inner_product_single(const float *a, const float *b, unsigned int len)
-{
- float ret;
- uint32_t remainder = len % 16;
- len = len - remainder;
-
- asm volatile (" cmp %[len], #0\n"
- " bne 1f\n"
- " vld1.32 {q4}, [%[b]]!\n"
- " vld1.32 {q8}, [%[a]]!\n"
- " subs %[remainder], %[remainder], #4\n"
- " vmul.f32 q0, q4, q8\n"
- " bne 4f\n"
- " b 5f\n"
- "1:"
- " vld1.32 {q4, q5}, [%[b]]!\n"
- " vld1.32 {q8, q9}, [%[a]]!\n"
- " vld1.32 {q6, q7}, [%[b]]!\n"
- " vld1.32 {q10, q11}, [%[a]]!\n"
- " subs %[len], %[len], #16\n"
- " vmul.f32 q0, q4, q8\n"
- " vmul.f32 q1, q5, q9\n"
- " vmul.f32 q2, q6, q10\n"
- " vmul.f32 q3, q7, q11\n"
- " beq 3f\n"
- "2:"
- " vld1.32 {q4, q5}, [%[b]]!\n"
- " vld1.32 {q8, q9}, [%[a]]!\n"
- " vld1.32 {q6, q7}, [%[b]]!\n"
- " vld1.32 {q10, q11}, [%[a]]!\n"
- " subs %[len], %[len], #16\n"
- " vmla.f32 q0, q4, q8\n"
- " vmla.f32 q1, q5, q9\n"
- " vmla.f32 q2, q6, q10\n"
- " vmla.f32 q3, q7, q11\n"
- " bne 2b\n"
- "3:"
- " vadd.f32 q4, q0, q1\n"
- " vadd.f32 q5, q2, q3\n"
- " cmp %[remainder], #0\n"
- " vadd.f32 q0, q4, q5\n"
- " beq 5f\n"
- "4:"
- " vld1.32 {q6}, [%[b]]!\n"
- " vld1.32 {q10}, [%[a]]!\n"
- " subs %[remainder], %[remainder], #4\n"
- " vmla.f32 q0, q6, q10\n"
- " bne 4b\n"
- "5:"
- " vadd.f32 d0, d0, d1\n"
- " vpadd.f32 d0, d0, d0\n"
- " vmov.f32 %[ret], d0[0]\n"
- : [ret] "=&r" (ret), [a] "+r" (a), [b] "+r" (b),
- [len] "+l" (len), [remainder] "+l" (remainder)
- :
- : "cc", "q0", "q1", "q2", "q3", "q4", "q5", "q6", "q7", "q8",
- "q9", "q10", "q11");
- return ret;
-}
-#endif
-
+++ /dev/null
-/* Copyright (C) 2007-2008 Jean-Marc Valin
- * Copyright (C) 2008 Thorvald Natvig
- */
-/**
- @file resample_sse.h
- @brief Resampler functions (SSE version)
-*/
-/*
- Redistribution and use in source and binary forms, with or without
- modification, are permitted provided that the following conditions
- are met:
-
- - Redistributions of source code must retain the above copyright
- notice, this list of conditions and the following disclaimer.
-
- - Redistributions in binary form must reproduce the above copyright
- notice, this list of conditions and the following disclaimer in the
- documentation and/or other materials provided with the distribution.
-
- - Neither the name of the Xiph.org Foundation nor the names of its
- contributors may be used to endorse or promote products derived from
- this software without specific prior written permission.
-
- THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
- ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
- LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
- A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
- CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
- EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
- PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
- LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
- NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
- SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#ifdef HAVE_XMMINTRIN_H
-#include <xmmintrin.h>
-#endif
-
-#define OVERRIDE_INNER_PRODUCT_SINGLE
-static inline float inner_product_single(const float *a, const float *b, unsigned int len)
-{
- int i = 0;
- float ret = 0;
- __m128 sum = _mm_setzero_ps();
-
- if (len > 7)
- {
- for (;i<len-7;i+=8)
- {
- sum = _mm_add_ps(sum, _mm_mul_ps(_mm_loadu_ps(a+i), _mm_loadu_ps(b+i)));
- sum = _mm_add_ps(sum, _mm_mul_ps(_mm_loadu_ps(a+i+4), _mm_loadu_ps(b+i+4)));
- }
- sum = _mm_add_ps(sum, _mm_movehl_ps(sum, sum));
- sum = _mm_add_ss(sum, _mm_shuffle_ps(sum, sum, 0x55));
- _mm_store_ss(&ret, sum);
- }
-
- for (; i < len; i++)
- ret += a[i] * b[i];
-
- return ret;
-}
-
-#define OVERRIDE_INTERPOLATE_PRODUCT_SINGLE
-static inline float interpolate_product_single(const float *a, const float *b, unsigned int len, const spx_uint32_t oversample, float *frac) {
- int i = 0;
- float ret = 0;
- __m128 sum = _mm_setzero_ps();
- __m128 f = _mm_loadu_ps(frac);
-
- if (len > 1)
- {
- for(;i<len-1;i+=2)
- {
- sum = _mm_add_ps(sum, _mm_mul_ps(_mm_load1_ps(a+i), _mm_loadu_ps(b+i*oversample)));
- sum = _mm_add_ps(sum, _mm_mul_ps(_mm_load1_ps(a+i+1), _mm_loadu_ps(b+(i+1)*oversample)));
- }
-
- sum = _mm_mul_ps(f, sum);
- sum = _mm_add_ps(sum, _mm_movehl_ps(sum, sum));
- sum = _mm_add_ss(sum, _mm_shuffle_ps(sum, sum, 0x55));
- _mm_store_ss(&ret, sum);
- }
-
- if (i == len-1)
- ret += a[i] * (frac[0]*b[i*oversample] + frac[1]*b[i*oversample + 1] + frac[2]*b[i*oversample + 2] + frac[3]*b[i*oversample + 3]);
-
- return ret;
-}
-
-#ifdef _USE_SSE2
-#ifdef HAVE_EMMINTRIN_H
-#include <emmintrin.h>
-#endif
-#define OVERRIDE_INNER_PRODUCT_DOUBLE
-
-#ifdef DOUBLE_PRECISION
-static inline double inner_product_double(const double *a, const double *b, unsigned int len)
-{
- int i = 0;
- double ret = 0;
- __m128d sum = _mm_setzero_pd();
-
- if (len > 3)
- {
- for (;i<len-3;i+=4)
- {
- sum = _mm_add_pd(sum, _mm_mul_pd(_mm_loadu_pd(a+i), _mm_loadu_pd(b+i)));
- sum = _mm_add_pd(sum, _mm_mul_pd(_mm_loadu_pd(a+i+2), _mm_loadu_pd(b+i+2)));
- }
- sum = _mm_add_sd(sum, _mm_unpackhi_pd(sum, sum));
- _mm_store_sd(&ret, sum);
- }
-
- for (; i < len; i++)
- ret += a[i] * b[i];
-
- return ret;
-}
-#else
-static inline double inner_product_double(const float *a, const float *b, unsigned int len)
-{
- int i = 0;
- double ret = 0;
- __m128d sum = _mm_setzero_pd();
- __m128 t;
-
- if (len > 7)
- {
- for (;i<len-7;i+=8)
- {
- t = _mm_mul_ps(_mm_loadu_ps(a+i), _mm_loadu_ps(b+i));
- sum = _mm_add_pd(sum, _mm_cvtps_pd(t));
- sum = _mm_add_pd(sum, _mm_cvtps_pd(_mm_movehl_ps(t, t)));
-
- t = _mm_mul_ps(_mm_loadu_ps(a+i+4), _mm_loadu_ps(b+i+4));
- sum = _mm_add_pd(sum, _mm_cvtps_pd(t));
- sum = _mm_add_pd(sum, _mm_cvtps_pd(_mm_movehl_ps(t, t)));
- }
- sum = _mm_add_sd(sum, _mm_unpackhi_pd(sum, sum));
- _mm_store_sd(&ret, sum);
- }
-
- for (; i < len; i++)
- ret += a[i] * b[i];
-
- return ret;
-}
-#endif
-
-
-#define OVERRIDE_INTERPOLATE_PRODUCT_DOUBLE
-
-#ifdef DOUBLE_PRECISION
-static inline double interpolate_product_double(const double *a, const double *b, unsigned int len, const spx_uint32_t oversample, double *frac) {
- int i = 0;
- double ret = 0;
- __m128d sum;
- __m128d sum1 = _mm_setzero_pd();
- __m128d sum2 = _mm_setzero_pd();
- __m128d f1 = _mm_loadu_pd(frac);
- __m128d f2 = _mm_loadu_pd(frac+2);
- __m128d t;
-
- if (len > 1)
- {
- for(;i<len-1;i+=2)
- {
- t = _mm_load1_pd(a+i);
- sum1 = _mm_add_pd(sum1, _mm_mul_pd(t, _mm_loadu_pd(b+i*oversample)));
- sum2 = _mm_add_pd(sum2, _mm_mul_pd(t, _mm_loadu_pd(b+i*oversample+2)));
-
- t = _mm_load1_pd(a+i+1);
- sum1 = _mm_add_pd(sum1, _mm_mul_pd(t, _mm_loadu_pd(b+(i+1)*oversample)));
- sum2 = _mm_add_pd(sum2, _mm_mul_pd(t, _mm_loadu_pd(b+(i+1)*oversample+2)));
- }
- sum1 = _mm_mul_pd(f1, sum1);
- sum2 = _mm_mul_pd(f2, sum2);
- sum = _mm_add_pd(sum1, sum2);
- sum = _mm_add_sd(sum, _mm_unpackhi_pd(sum, sum));
- _mm_store_sd(&ret, sum);
- }
-
- if (i == len-1)
- ret += a[i] * (frac[0]*b[i*oversample] + frac[1]*b[i*oversample + 1] + frac[2]*b[i*oversample + 2] + frac[3]*b[i*oversample + 3]);
-
- return ret;
-}
-#else
-static inline double interpolate_product_double(const float *a, const float *b, unsigned int len, const spx_uint32_t oversample, float *frac) {
- int i = 0;
- double ret = 0;
- __m128d sum;
- __m128d sum1 = _mm_setzero_pd();
- __m128d sum2 = _mm_setzero_pd();
- __m128 f = _mm_loadu_ps(frac);
- __m128d f1 = _mm_cvtps_pd(f);
- __m128d f2 = _mm_cvtps_pd(_mm_movehl_ps(f,f));
- __m128 t;
-
- if (len > 1)
- {
- for(;i<len-1;i+=2)
- {
- t = _mm_mul_ps(_mm_load1_ps(a+i), _mm_loadu_ps(b+i*oversample));
- sum1 = _mm_add_pd(sum1, _mm_cvtps_pd(t));
- sum2 = _mm_add_pd(sum2, _mm_cvtps_pd(_mm_movehl_ps(t, t)));
-
- t = _mm_mul_ps(_mm_load1_ps(a+i+1), _mm_loadu_ps(b+(i+1)*oversample));
- sum1 = _mm_add_pd(sum1, _mm_cvtps_pd(t));
- sum2 = _mm_add_pd(sum2, _mm_cvtps_pd(_mm_movehl_ps(t, t)));
- }
- sum1 = _mm_mul_pd(f1, sum1);
- sum2 = _mm_mul_pd(f2, sum2);
- sum = _mm_add_pd(sum1, sum2);
- sum = _mm_add_sd(sum, _mm_unpackhi_pd(sum, sum));
- _mm_store_sd(&ret, sum);
- }
-
- if (i == len-1)
- ret += a[i] * (frac[0]*b[i*oversample] + frac[1]*b[i*oversample + 1] + frac[2]*b[i*oversample + 2] + frac[3]*b[i*oversample + 3]);
-
- return ret;
-}
-#endif
-
-#endif
+++ /dev/null
-/* Copyright (C) 2007 Jean-Marc Valin
-
- File: speex_resampler.h
- Resampling code
-
- The design goals of this code are:
- - Very fast algorithm
- - Low memory requirement
- - Good *perceptual* quality (and not best SNR)
-
- Redistribution and use in source and binary forms, with or without
- modification, are permitted provided that the following conditions are
- met:
-
- 1. Redistributions of source code must retain the above copyright notice,
- this list of conditions and the following disclaimer.
-
- 2. Redistributions in binary form must reproduce the above copyright
- notice, this list of conditions and the following disclaimer in the
- documentation and/or other materials provided with the distribution.
-
- 3. The name of the author may not be used to endorse or promote products
- derived from this software without specific prior written permission.
-
- THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
- IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
- OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
- DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
- INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
- (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
- SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
- HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
- STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
- ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
- POSSIBILITY OF SUCH DAMAGE.
-*/
-
-
-#ifndef SPEEX_RESAMPLER_H
-#define SPEEX_RESAMPLER_H
-
-#ifdef OUTSIDE_SPEEX
-
-/********* WARNING: MENTAL SANITY ENDS HERE *************/
-
-/* If the resampler is defined outside of Speex, we change the symbol names so that
- there won't be any clash if linking with Speex later on. */
-
-/* #define RANDOM_PREFIX your software name here */
-#ifndef RANDOM_PREFIX
-#error "Please define RANDOM_PREFIX (above) to something specific to your project to prevent symbol name clashes"
-#endif
-
-#define CAT_PREFIX2(a,b) a ## b
-#define CAT_PREFIX(a,b) CAT_PREFIX2(a, b)
-
-#define speex_resampler_init CAT_PREFIX(RANDOM_PREFIX,_resampler_init)
-#define speex_resampler_init_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_init_frac)
-#define speex_resampler_destroy CAT_PREFIX(RANDOM_PREFIX,_resampler_destroy)
-#define speex_resampler_process_float CAT_PREFIX(RANDOM_PREFIX,_resampler_process_float)
-#define speex_resampler_process_int CAT_PREFIX(RANDOM_PREFIX,_resampler_process_int)
-#define speex_resampler_process_interleaved_float CAT_PREFIX(RANDOM_PREFIX,_resampler_process_interleaved_float)
-#define speex_resampler_process_interleaved_int CAT_PREFIX(RANDOM_PREFIX,_resampler_process_interleaved_int)
-#define speex_resampler_set_rate CAT_PREFIX(RANDOM_PREFIX,_resampler_set_rate)
-#define speex_resampler_get_rate CAT_PREFIX(RANDOM_PREFIX,_resampler_get_rate)
-#define speex_resampler_set_rate_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_set_rate_frac)
-#define speex_resampler_get_ratio CAT_PREFIX(RANDOM_PREFIX,_resampler_get_ratio)
-#define speex_resampler_set_quality CAT_PREFIX(RANDOM_PREFIX,_resampler_set_quality)
-#define speex_resampler_get_quality CAT_PREFIX(RANDOM_PREFIX,_resampler_get_quality)
-#define speex_resampler_set_input_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_set_input_stride)
-#define speex_resampler_get_input_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_get_input_stride)
-#define speex_resampler_set_output_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_set_output_stride)
-#define speex_resampler_get_output_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_get_output_stride)
-#define speex_resampler_get_input_latency CAT_PREFIX(RANDOM_PREFIX,_resampler_get_input_latency)
-#define speex_resampler_get_output_latency CAT_PREFIX(RANDOM_PREFIX,_resampler_get_output_latency)
-#define speex_resampler_get_filt_len CAT_PREFIX(RANDOM_PREFIX,_resampler_get_filt_len)
-#define speex_resampler_get_sinc_filter_mode CAT_PREFIX(RANDOM_PREFIX,_resampler_get_sinc_filter_mode)
-#define speex_resampler_skip_zeros CAT_PREFIX(RANDOM_PREFIX,_resampler_skip_zeros)
-#define speex_resampler_reset_mem CAT_PREFIX(RANDOM_PREFIX,_resampler_reset_mem)
-#define speex_resampler_strerror CAT_PREFIX(RANDOM_PREFIX,_resampler_strerror)
-
-#define spx_int16_t gint16
-#define spx_int32_t gint32
-#define spx_uint16_t guint16
-#define spx_uint32_t guint32
-
-#else /* OUTSIDE_SPEEX */
-
-#ifdef _BUILD_SPEEX
-# include "speex_types.h"
-#else
-# include <speex/speex_types.h>
-#endif
-
-#endif /* OUTSIDE_SPEEX */
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-#define SPEEX_RESAMPLER_QUALITY_MAX 10
-#define SPEEX_RESAMPLER_QUALITY_MIN 0
-#define SPEEX_RESAMPLER_QUALITY_DEFAULT 4
-#define SPEEX_RESAMPLER_QUALITY_VOIP 3
-#define SPEEX_RESAMPLER_QUALITY_DESKTOP 5
-
-enum {
- RESAMPLER_ERR_SUCCESS = 0,
- RESAMPLER_ERR_ALLOC_FAILED = 1,
- RESAMPLER_ERR_BAD_STATE = 2,
- RESAMPLER_ERR_INVALID_ARG = 3,
- RESAMPLER_ERR_PTR_OVERLAP = 4,
-
- RESAMPLER_ERR_MAX_ERROR
-};
-
-typedef enum {
- RESAMPLER_SINC_FILTER_INTERPOLATED = 0,
- RESAMPLER_SINC_FILTER_FULL = 1,
- RESAMPLER_SINC_FILTER_AUTO = 2
-} SpeexResamplerSincFilterMode;
-
-#define RESAMPLER_SINC_FILTER_DEFAULT RESAMPLER_SINC_FILTER_INTERPOLATED
-#define RESAMPLER_SINC_FILTER_AUTO_THRESHOLD_DEFAULT (1 * 1048576)
-
-struct SpeexResamplerState_;
-typedef struct SpeexResamplerState_ SpeexResamplerState;
-
-/** Create a new resampler with integer input and output rates.
- * @param nb_channels Number of channels to be processed
- * @param in_rate Input sampling rate (integer number of Hz).
- * @param out_rate Output sampling rate (integer number of Hz).
- * @param quality Resampling quality between 0 and 10, where 0 has poor quality
- * and 10 has very high quality.
- * @param sinc_filter_mode Sinc filter table mode to use
- * @param sinc_filter_auto_threshold Threshold to use if sinc filter mode is auto, in bytes
- * @return Newly created resampler state
- * @retval NULL Error: not enough memory
- *
- * If a full filter table would be larger than the auto threshold, and sinc_filter_mode is AUTO,
- * the resample uses the interpolated mode instead
- *
- * @note A full sinc table can significantly improve the resampler's performance, but calculating the table
- * takes longer, as opposed to the interpolated variant
- */
-SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels,
- spx_uint32_t in_rate,
- spx_uint32_t out_rate,
- int quality,
- SpeexResamplerSincFilterMode sinc_filter_mode,
- spx_uint32_t sinc_filter_auto_threshold,
- int *err);
-
-/** Create a new resampler with fractional input/output rates. The sampling
- * rate ratio is an arbitrary rational number with both the numerator and
- * denominator being 32-bit integers.
- * @param nb_channels Number of channels to be processed
- * @param ratio_num Numerator of the sampling rate ratio
- * @param ratio_den Denominator of the sampling rate ratio
- * @param in_rate Input sampling rate rounded to the nearest integer (in Hz).
- * @param out_rate Output sampling rate rounded to the nearest integer (in Hz).
- * @param quality Resampling quality between 0 and 10, where 0 has poor quality
- * and 10 has very high quality.
- * @param sinc_filter_mode Sinc filter table mode to use
- * @param sinc_filter_auto_threshold Threshold to use if sinc filter mode is auto, in bytes
- * @return Newly created resampler state
- * @retval NULL Error: not enough memory
- *
- * If a full filter table would be larger than the auto threshold, and sinc_filter_mode is AUTO,
- * the resample uses the interpolated mode instead
- *
- * @note A full sinc table can significantly improve the resampler's performance, but calculating the table
- * takes longer, as opposed to the interpolated variant
- */
-SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels,
- spx_uint32_t ratio_num,
- spx_uint32_t ratio_den,
- spx_uint32_t in_rate,
- spx_uint32_t out_rate,
- int quality,
- SpeexResamplerSincFilterMode sinc_filter_mode,
- spx_uint32_t sinc_filter_auto_threshold,
- int *err);
-
-/** Destroy a resampler state.
- * @param st Resampler state
- */
-void speex_resampler_destroy(SpeexResamplerState *st);
-
-/** Resample a float array. The input and output buffers must *not* overlap.
- * @param st Resampler state
- * @param channel_index Index of the channel to process for the multi-channel
- * base (0 otherwise)
- * @param in Input buffer
- * @param in_len Number of input samples in the input buffer. Returns the
- * number of samples processed
- * @param out Output buffer
- * @param out_len Size of the output buffer. Returns the number of samples written
- */
-#ifdef DOUBLE_PRECISION
-int speex_resampler_process_float(SpeexResamplerState *st,
- spx_uint32_t channel_index,
- const double *in,
- spx_uint32_t *in_len,
- double *out,
- spx_uint32_t *out_len);
-#else
-int speex_resampler_process_float(SpeexResamplerState *st,
- spx_uint32_t channel_index,
- const float *in,
- spx_uint32_t *in_len,
- float *out,
- spx_uint32_t *out_len);
-#endif
-
-/** Resample an int array. The input and output buffers must *not* overlap.
- * @param st Resampler state
- * @param channel_index Index of the channel to process for the multi-channel
- * base (0 otherwise)
- * @param in Input buffer
- * @param in_len Number of input samples in the input buffer. Returns the number
- * of samples processed
- * @param out Output buffer
- * @param out_len Size of the output buffer. Returns the number of samples written
- */
-int speex_resampler_process_int(SpeexResamplerState *st,
- spx_uint32_t channel_index,
- const spx_int16_t *in,
- spx_uint32_t *in_len,
- spx_int16_t *out,
- spx_uint32_t *out_len);
-
-/** Resample an interleaved float array. The input and output buffers must *not* overlap.
- * @param st Resampler state
- * @param in Input buffer
- * @param in_len Number of input samples in the input buffer. Returns the number
- * of samples processed. This is all per-channel.
- * @param out Output buffer
- * @param out_len Size of the output buffer. Returns the number of samples written.
- * This is all per-channel.
- */
-#ifdef DOUBLE_PRECISION
-int speex_resampler_process_interleaved_float(SpeexResamplerState *st,
- const double *in,
- spx_uint32_t *in_len,
- double *out,
- spx_uint32_t *out_len);
-#else
-int speex_resampler_process_interleaved_float(SpeexResamplerState *st,
- const float *in,
- spx_uint32_t *in_len,
- float *out,
- spx_uint32_t *out_len);
-#endif
-
-/** Resample an interleaved int array. The input and output buffers must *not* overlap.
- * @param st Resampler state
- * @param in Input buffer
- * @param in_len Number of input samples in the input buffer. Returns the number
- * of samples processed. This is all per-channel.
- * @param out Output buffer
- * @param out_len Size of the output buffer. Returns the number of samples written.
- * This is all per-channel.
- */
-int speex_resampler_process_interleaved_int(SpeexResamplerState *st,
- const spx_int16_t *in,
- spx_uint32_t *in_len,
- spx_int16_t *out,
- spx_uint32_t *out_len);
-
-/** Set (change) the input/output sampling rates (integer value).
- * @param st Resampler state
- * @param in_rate Input sampling rate (integer number of Hz).
- * @param out_rate Output sampling rate (integer number of Hz).
- */
-int speex_resampler_set_rate(SpeexResamplerState *st,
- spx_uint32_t in_rate,
- spx_uint32_t out_rate);
-
-/** Get the current input/output sampling rates (integer value).
- * @param st Resampler state
- * @param in_rate Input sampling rate (integer number of Hz) copied.
- * @param out_rate Output sampling rate (integer number of Hz) copied.
- */
-void speex_resampler_get_rate(SpeexResamplerState *st,
- spx_uint32_t *in_rate,
- spx_uint32_t *out_rate);
-
-/** Set (change) the input/output sampling rates and resampling ratio
- * (fractional values in Hz supported).
- * @param st Resampler state
- * @param ratio_num Numerator of the sampling rate ratio
- * @param ratio_den Denominator of the sampling rate ratio
- * @param in_rate Input sampling rate rounded to the nearest integer (in Hz).
- * @param out_rate Output sampling rate rounded to the nearest integer (in Hz).
- */
-int speex_resampler_set_rate_frac(SpeexResamplerState *st,
- spx_uint32_t ratio_num,
- spx_uint32_t ratio_den,
- spx_uint32_t in_rate,
- spx_uint32_t out_rate);
-
-/** Get the current resampling ratio. This will be reduced to the least
- * common denominator.
- * @param st Resampler state
- * @param ratio_num Numerator of the sampling rate ratio copied
- * @param ratio_den Denominator of the sampling rate ratio copied
- */
-void speex_resampler_get_ratio(SpeexResamplerState *st,
- spx_uint32_t *ratio_num,
- spx_uint32_t *ratio_den);
-
-/** Set (change) the conversion quality.
- * @param st Resampler state
- * @param quality Resampling quality between 0 and 10, where 0 has poor
- * quality and 10 has very high quality.
- */
-int speex_resampler_set_quality(SpeexResamplerState *st,
- int quality);
-
-/** Get the conversion quality.
- * @param st Resampler state
- * @param quality Resampling quality between 0 and 10, where 0 has poor
- * quality and 10 has very high quality.
- */
-void speex_resampler_get_quality(SpeexResamplerState *st,
- int *quality);
-
-/** Set (change) the input stride.
- * @param st Resampler state
- * @param stride Input stride
- */
-void speex_resampler_set_input_stride(SpeexResamplerState *st,
- spx_uint32_t stride);
-
-/** Get the input stride.
- * @param st Resampler state
- * @param stride Input stride copied
- */
-void speex_resampler_get_input_stride(SpeexResamplerState *st,
- spx_uint32_t *stride);
-
-/** Set (change) the output stride.
- * @param st Resampler state
- * @param stride Output stride
- */
-void speex_resampler_set_output_stride(SpeexResamplerState *st,
- spx_uint32_t stride);
-
-/** Get the output stride.
- * @param st Resampler state copied
- * @param stride Output stride
- */
-void speex_resampler_get_output_stride(SpeexResamplerState *st,
- spx_uint32_t *stride);
-
-/** Get the latency introduced by the resampler measured in input samples.
- * @param st Resampler state
- */
-int speex_resampler_get_input_latency(SpeexResamplerState *st);
-
-/** Get the latency introduced by the resampler measured in output samples.
- * @param st Resampler state
- */
-int speex_resampler_get_output_latency(SpeexResamplerState *st);
-
-/** Get the length of the filter in input samples.
- * @param st Resampler state
- */
-int speex_resampler_get_filt_len(SpeexResamplerState *st);
-
-/** Returns 1 if the full sinc filter table is used, 0 if the interpolated one is used
- * @param st Resampler state
- * @return Sinc filter mode
- */
-int speex_resampler_get_sinc_filter_mode(SpeexResamplerState *st);
-
-/** Make sure that the first samples to go out of the resamplers don't have
- * leading zeros. This is only useful before starting to use a newly created
- * resampler. It is recommended to use that when resampling an audio file, as
- * it will generate a file with the same length. For real-time processing,
- * it is probably easier not to use this call (so that the output duration
- * is the same for the first frame).
- * @param st Resampler state
- */
-int speex_resampler_skip_zeros(SpeexResamplerState *st);
-
-/** Reset a resampler so a new (unrelated) stream can be processed.
- * @param st Resampler state
- */
-int speex_resampler_reset_mem(SpeexResamplerState *st);
-
-/** Returns the English meaning for an error code
- * @param err Error code
- * @return English string
- */
-const char *speex_resampler_strerror(int err);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2007-2008 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#define _USE_SSE2
-#define FLOATING_POINT
-#define DOUBLE_PRECISION
-#define OUTSIDE_SPEEX
-#define RANDOM_PREFIX resample_double
-
-#include "resample.c"
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2007-2008 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#define _USE_SSE
-#define _USE_SSE2
-#define _USE_NEON
-#define FLOATING_POINT
-#define OUTSIDE_SPEEX
-#define RANDOM_PREFIX resample_float
-
-#include "resample.c"
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2007-2008 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#define FIXED_POINT 1
-#define OUTSIDE_SPEEX 1
-/* disabled, 16-bit integer NEON support seems broken */
-/* #define _USE_NEON */
-#define RANDOM_PREFIX resample_int
-
-#include "resample.c"
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2007-2008 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#ifndef __SPEEX_RESAMPLER_WRAPPER_H__
-#define __SPEEX_RESAMPLER_WRAPPER_H__
-
-#define SPEEX_RESAMPLER_QUALITY_MAX 10
-#define SPEEX_RESAMPLER_QUALITY_MIN 0
-#define SPEEX_RESAMPLER_QUALITY_DEFAULT 4
-#define SPEEX_RESAMPLER_QUALITY_VOIP 3
-#define SPEEX_RESAMPLER_QUALITY_DESKTOP 5
-
-#define SPEEX_RESAMPLER_SINC_FILTER_DEFAULT SPEEX_RESAMPLER_SINC_FILTER_AUTO
-#define SPEEX_RESAMPLER_SINC_FILTER_AUTO_THRESHOLD_DEFAULT (1 * 1048576)
-
-enum
-{
- RESAMPLER_ERR_SUCCESS = 0,
- RESAMPLER_ERR_ALLOC_FAILED = 1,
- RESAMPLER_ERR_BAD_STATE = 2,
- RESAMPLER_ERR_INVALID_ARG = 3,
- RESAMPLER_ERR_PTR_OVERLAP = 4,
-
- RESAMPLER_ERR_MAX_ERROR
-};
-
-typedef enum {
- SPEEX_RESAMPLER_SINC_FILTER_INTERPOLATED = 0,
- SPEEX_RESAMPLER_SINC_FILTER_FULL = 1,
- SPEEX_RESAMPLER_SINC_FILTER_AUTO = 2
-} SpeexResamplerSincFilterMode;
-
-typedef struct SpeexResamplerState_ SpeexResamplerState;
-
-typedef struct {
- SpeexResamplerState *(*init) (guint32 nb_channels,
- guint32 in_rate, guint32 out_rate, gint quality,
- SpeexResamplerSincFilterMode sinc_filter_mode,
- guint32 sinc_filter_auto_threshold, gint * err);
- void (*destroy) (SpeexResamplerState * st);
- int (*process) (SpeexResamplerState *
- st, const guint8 * in, guint32 * in_len, guint8 * out, guint32 * out_len);
- int (*set_rate) (SpeexResamplerState * st,
- guint32 in_rate, guint32 out_rate);
- void (*get_rate) (SpeexResamplerState * st,
- guint32 * in_rate, guint32 * out_rate);
- void (*get_ratio) (SpeexResamplerState * st,
- guint32 * ratio_num, guint32 * ratio_den);
- int (*get_input_latency) (SpeexResamplerState * st);
- int (*get_filt_len) (SpeexResamplerState * st);
- int (*get_sinc_filter_mode) (SpeexResamplerState * st);
- int (*set_quality) (SpeexResamplerState * st, gint quality);
- int (*reset_mem) (SpeexResamplerState * st);
- int (*skip_zeros) (SpeexResamplerState * st);
- const char * (*strerror) (gint err);
- unsigned int width;
-} SpeexResampleFuncs;
-
-SpeexResamplerState *resample_float_resampler_init (guint32 nb_channels,
- guint32 in_rate, guint32 out_rate, gint quality,
- SpeexResamplerSincFilterMode sinc_filter_mode,
- guint32 sinc_filter_auto_threshold, gint * err);
-void resample_float_resampler_destroy (SpeexResamplerState * st);
-int resample_float_resampler_process_interleaved_float (SpeexResamplerState *
- st, const guint8 * in, guint32 * in_len, guint8 * out, guint32 * out_len);
-int resample_float_resampler_set_rate (SpeexResamplerState * st,
- guint32 in_rate, guint32 out_rate);
-void resample_float_resampler_get_rate (SpeexResamplerState * st,
- guint32 * in_rate, guint32 * out_rate);
-void resample_float_resampler_get_ratio (SpeexResamplerState * st,
- guint32 * ratio_num, guint32 * ratio_den);
-int resample_float_resampler_get_input_latency (SpeexResamplerState * st);
-int resample_float_resampler_get_filt_len (SpeexResamplerState * st);
-int resample_float_resampler_get_sinc_filter_mode (SpeexResamplerState * st);
-int resample_float_resampler_set_quality (SpeexResamplerState * st, gint quality);
-int resample_float_resampler_reset_mem (SpeexResamplerState * st);
-int resample_float_resampler_skip_zeros (SpeexResamplerState * st);
-const char * resample_float_resampler_strerror (gint err);
-
-static const SpeexResampleFuncs float_funcs =
-{
- resample_float_resampler_init,
- resample_float_resampler_destroy,
- resample_float_resampler_process_interleaved_float,
- resample_float_resampler_set_rate,
- resample_float_resampler_get_rate,
- resample_float_resampler_get_ratio,
- resample_float_resampler_get_input_latency,
- resample_float_resampler_get_filt_len,
- resample_float_resampler_get_sinc_filter_mode,
- resample_float_resampler_set_quality,
- resample_float_resampler_reset_mem,
- resample_float_resampler_skip_zeros,
- resample_float_resampler_strerror,
- 32
-};
-
-SpeexResamplerState *resample_double_resampler_init (guint32 nb_channels,
- guint32 in_rate, guint32 out_rate, gint quality,
- SpeexResamplerSincFilterMode sinc_filter_mode,
- guint32 sinc_filter_auto_threshold, gint * err);
-void resample_double_resampler_destroy (SpeexResamplerState * st);
-int resample_double_resampler_process_interleaved_float (SpeexResamplerState *
- st, const guint8 * in, guint32 * in_len, guint8 * out, guint32 * out_len);
-int resample_double_resampler_set_rate (SpeexResamplerState * st,
- guint32 in_rate, guint32 out_rate);
-void resample_double_resampler_get_rate (SpeexResamplerState * st,
- guint32 * in_rate, guint32 * out_rate);
-void resample_double_resampler_get_ratio (SpeexResamplerState * st,
- guint32 * ratio_num, guint32 * ratio_den);
-int resample_double_resampler_get_input_latency (SpeexResamplerState * st);
-int resample_double_resampler_get_filt_len (SpeexResamplerState * st);
-int resample_double_resampler_get_sinc_filter_mode (SpeexResamplerState * st);
-int resample_double_resampler_set_quality (SpeexResamplerState * st, gint quality);
-int resample_double_resampler_reset_mem (SpeexResamplerState * st);
-int resample_double_resampler_skip_zeros (SpeexResamplerState * st);
-const char * resample_double_resampler_strerror (gint err);
-
-static const SpeexResampleFuncs double_funcs =
-{
- resample_double_resampler_init,
- resample_double_resampler_destroy,
- resample_double_resampler_process_interleaved_float,
- resample_double_resampler_set_rate,
- resample_double_resampler_get_rate,
- resample_double_resampler_get_ratio,
- resample_double_resampler_get_input_latency,
- resample_double_resampler_get_filt_len,
- resample_double_resampler_get_sinc_filter_mode,
- resample_double_resampler_set_quality,
- resample_double_resampler_reset_mem,
- resample_double_resampler_skip_zeros,
- resample_double_resampler_strerror,
- 64
-};
-
-SpeexResamplerState *resample_int_resampler_init (guint32 nb_channels,
- guint32 in_rate, guint32 out_rate, gint quality,
- SpeexResamplerSincFilterMode sinc_filter_mode,
- guint32 sinc_filter_auto_threshold, gint * err);
-void resample_int_resampler_destroy (SpeexResamplerState * st);
-int resample_int_resampler_process_interleaved_int (SpeexResamplerState *
- st, const guint8 * in, guint32 * in_len, guint8 * out, guint32 * out_len);
-int resample_int_resampler_set_rate (SpeexResamplerState * st,
- guint32 in_rate, guint32 out_rate);
-void resample_int_resampler_get_rate (SpeexResamplerState * st,
- guint32 * in_rate, guint32 * out_rate);
-void resample_int_resampler_get_ratio (SpeexResamplerState * st,
- guint32 * ratio_num, guint32 * ratio_den);
-int resample_int_resampler_get_input_latency (SpeexResamplerState * st);
-int resample_int_resampler_get_filt_len (SpeexResamplerState * st);
-int resample_int_resampler_get_sinc_filter_mode (SpeexResamplerState * st);
-int resample_int_resampler_set_quality (SpeexResamplerState * st, gint quality);
-int resample_int_resampler_reset_mem (SpeexResamplerState * st);
-int resample_int_resampler_skip_zeros (SpeexResamplerState * st);
-const char * resample_int_resampler_strerror (gint err);
-
-static const SpeexResampleFuncs int_funcs =
-{
- resample_int_resampler_init,
- resample_int_resampler_destroy,
- resample_int_resampler_process_interleaved_int,
- resample_int_resampler_set_rate,
- resample_int_resampler_get_rate,
- resample_int_resampler_get_ratio,
- resample_int_resampler_get_input_latency,
- resample_int_resampler_get_filt_len,
- resample_int_resampler_get_sinc_filter_mode,
- resample_int_resampler_set_quality,
- resample_int_resampler_reset_mem,
- resample_int_resampler_skip_zeros,
- resample_int_resampler_strerror,
- 16
-};
-
-#endif /* __SPEEX_RESAMPLER_WRAPPER_H__ */
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