diff options
Diffstat (limited to 'pph')
| -rw-r--r-- | pph/Makefile.am | 19 | ||||
| -rw-r--r-- | pph/rate_speexrate.c | 162 | ||||
| -rw-r--r-- | pph/resample.c | 933 | ||||
| -rw-r--r-- | pph/speex_resampler.h | 319 |
4 files changed, 1433 insertions, 0 deletions
diff --git a/pph/Makefile.am b/pph/Makefile.am new file mode 100644 index 0000000..d695ef6 --- /dev/null +++ b/pph/Makefile.am @@ -0,0 +1,19 @@ +asound_module_rate_speexrate_LTLIBRARIES = libasound_module_rate_speexrate.la + +asound_module_rate_speexratedir = $(libdir)/alsa-lib + +AM_CFLAGS = -DOUTSIDE_SPEEX -Wall -g @ALSA_CFLAGS@ +AM_LDFLAGS = -module -avoid-version -export-dynamic + +libasound_module_rate_speexrate_la_SOURCES = rate_speexrate.c resample.c +libasound_module_rate_speexrate_la_LIBADD = @ALSA_LIBS@ + +install-exec-hook: + rm -f $(DESTDIR)$(libdir)/alsa-lib/libasound_module_rate_speexrate_*.so + $(LN_S) libasound_module_rate_speexrate.so $(DESTDIR)$(libdir)/alsa-lib/libasound_module_rate_speexrate_best.so + $(LN_S) libasound_module_rate_speexrate.so $(DESTDIR)$(libdir)/alsa-lib/libasound_module_rate_speexrate_medium.so + +uninstall-hook: + rm -f $(DESTDIR)$(libdir)/alsa-lib/libasound_module_rate_speexrate_*.so + +noinst_HEADERS = speex_resampler.h diff --git a/pph/rate_speexrate.c b/pph/rate_speexrate.c new file mode 100644 index 0000000..07679b4 --- /dev/null +++ b/pph/rate_speexrate.c @@ -0,0 +1,162 @@ +/* Rate converter plugin using Public Parrot Hack + Copyright (C) 2007 Jean-Marc Valin + + 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. +*/ + +#include <stdio.h> +#include <samplerate.h> +#include <alsa/asoundlib.h> +#include <alsa/pcm_rate.h> + +#include "speex_resampler.h" + +struct rate_src { + int quality; + unsigned int channels; + SpeexResamplerState *st; +}; + +static snd_pcm_uframes_t input_frames(void *obj, snd_pcm_uframes_t frames) +{ + int num, den; + struct rate_src *rate = obj; + if (frames == 0) + return 0; + speex_resampler_get_ratio(rate->st, &num, &den); + return (snd_pcm_uframes_t)((frames*num+(den>>1))/den); +} + +static snd_pcm_uframes_t output_frames(void *obj, snd_pcm_uframes_t frames) +{ + int num, den; + struct rate_src *rate = obj; + if (frames == 0) + return 0; + speex_resampler_get_ratio(rate->st, &num, &den); + return (snd_pcm_uframes_t)((frames*den+(num>>1))/num); +} + +static void pcm_src_free(void *obj) +{ + struct rate_src *rate = obj; + if (rate->st) + { + speex_resampler_destroy(rate->st); + rate->st = NULL; + } +} + +static int pcm_src_init(void *obj, snd_pcm_rate_info_t *info) +{ + struct rate_src *rate = obj; + + if (! rate->st || rate->channels != info->channels) { + if (rate->st) + speex_resampler_destroy(rate->st); + rate->channels = info->channels; + rate->st = speex_resampler_init_frac(rate->channels, info->in.period_size, info->out.period_size, info->in.rate, info->out.rate, rate->quality); + if (! rate->st) + return -EINVAL; + } + + return 0; +} + +static int pcm_src_adjust_pitch(void *obj, snd_pcm_rate_info_t *info) +{ + struct rate_src *rate = obj; + speex_resampler_set_rate_frac(rate->st, info->in.period_size, info->out.period_size, info->in.rate, info->out.rate); + return 0; +} + +static void pcm_src_reset(void *obj) +{ + struct rate_src *rate = obj; + speex_resampler_reset_mem(rate->st); +} + +static void pcm_src_convert_s16(void *obj, int16_t *dst, unsigned int dst_frames, + const int16_t *src, unsigned int src_frames) +{ + struct rate_src *rate = obj; + speex_resampler_process_interleaved_int(rate->st, src, &src_frames, dst, &dst_frames); +} + +static void pcm_src_close(void *obj) +{ + free(obj); +} + +static snd_pcm_rate_ops_t pcm_src_ops = { + .close = pcm_src_close, + .init = pcm_src_init, + .free = pcm_src_free, + .reset = pcm_src_reset, + .adjust_pitch = pcm_src_adjust_pitch, + .convert_s16 = pcm_src_convert_s16, + .input_frames = input_frames, + .output_frames = output_frames, +}; + +static int pcm_src_open(unsigned int version, void **objp, + snd_pcm_rate_ops_t *ops, int quality) +{ + struct rate_src *rate; + + if (version != SND_PCM_RATE_PLUGIN_VERSION) { + fprintf(stderr, "Invalid rate plugin version %x\n", version); + return -EINVAL; + } + + rate = calloc(1, sizeof(*rate)); + if (! rate) + return -ENOMEM; + rate->quality = quality; + + *objp = rate; + *ops = pcm_src_ops; + return 0; +} + +int SND_PCM_RATE_PLUGIN_ENTRY(speexrate) (unsigned int version, void **objp, + snd_pcm_rate_ops_t *ops) +{ + return pcm_src_open(version, objp, ops, 3); +} + +int SND_PCM_RATE_PLUGIN_ENTRY(speexrate_best) (unsigned int version, void **objp, + snd_pcm_rate_ops_t *ops) +{ + return pcm_src_open(version, objp, ops, 10); +} + +int SND_PCM_RATE_PLUGIN_ENTRY(speexrate_medium) (unsigned int version, void **objp, + snd_pcm_rate_ops_t *ops) +{ + return pcm_src_open(version, objp, ops, 5); +} diff --git a/pph/resample.c b/pph/resample.c new file mode 100644 index 0000000..8255b83 --- /dev/null +++ b/pph/resample.c @@ -0,0 +1,933 @@ +/* Copyright (C) 2007 Jean-Marc Valin + + 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) + + The code is working, but it's in a very early stage, so it may have + artifacts, noise or subliminal messages from satan. Also, the API + isn't stable and I can actually promise that I *will* change the API + some time in the future. + +TODO list: + - Variable calculation resolution depending on quality setting + - Single vs double in float mode + - 16-bit vs 32-bit (sinc only) in fixed-point mode + - Make sure the filter update works even when changing params + after only a few samples procesed +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#ifdef OUTSIDE_SPEEX +#include <stdlib.h> +void *speex_alloc (int size) {return calloc(size,1);} +void *speex_realloc (void *ptr, int size) {return realloc(ptr, size);} +void speex_free (void *ptr) {free(ptr);} +#include "speex_resampler.h" +#else +#include "speex/speex_resampler.h" +#include "misc.h" +#endif + +#include <math.h> + +#ifndef M_PI +#define M_PI 3.14159263 +#endif + +#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 float double*/ +#define FILTER_SIZE 64 +#define OVERSAMPLE 8 + +#define IMAX(a,b) ((a) > (b) ? (a) : (b)) + + +typedef int (*resampler_basic_func)(SpeexResamplerState *, int , const spx_word16_t *, int *, spx_word16_t *, int *); + +struct SpeexResamplerState_ { + int in_rate; + int out_rate; + int num_rate; + int den_rate; + + int quality; + int nb_channels; + int filt_len; + int mem_alloc_size; + int int_advance; + int frac_advance; + float cutoff; + int oversample; + int initialised; + int started; + + /* These are per-channel */ + int *last_sample; + int *samp_frac_num; + int *magic_samples; + + spx_word16_t *mem; + spx_word16_t *sinc_table; + int sinc_table_length; + resampler_basic_func resampler_ptr; + + int in_stride; + int out_stride; +} ; + +static 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 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 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 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 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 { + 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). +*/ +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, 8, 0.922f, 0.940f, KAISER10}, /* Q5 */ /* 89.1% cutoff (~100 dB stop) 10 */ + { 96, 8, 0.940f, 0.945f, KAISER10}, /* Q6 */ /* 91.5% cutoff (~100 dB stop) 10 */ + {128, 16, 0.950f, 0.950f, KAISER10}, /* Q7 */ /* 93.1% cutoff (~100 dB stop) 10 */ + {160, 16, 0.960f, 0.960f, KAISER10}, /* Q8 */ /* 94.5% cutoff (~100 dB stop) 10 */ + {192, 16, 0.968f, 0.968f, KAISER12}, /* Q9 */ /* 95.5% cutoff (~100 dB stop) 10 */ + {256, 16, 0.975f, 0.975f, KAISER12}, /* Q10 */ /* 96.6% cutoff (~100 dB stop) 10 */ +}; +/*8,24,40,56,80,104,128,160,200,256,320*/ +static double compute_func(float x, struct FuncDef *func) +{ + float y, frac; + 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(M_PI*xx)/(M_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 */ +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-6) + return cutoff; + else if (fabs(x) > .5*N) + return 0; + /*FIXME: Can it really be any slower than this? */ + return cutoff*sin(M_PI*xx)/(M_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 + +static int resampler_basic_direct_single(SpeexResamplerState *st, int channel_index, const spx_word16_t *in, int *in_len, spx_word16_t *out, int *out_len) +{ + int N = st->filt_len; + int out_sample = 0; + spx_word16_t *mem; + int last_sample = st->last_sample[channel_index]; + int samp_frac_num = st->samp_frac_num[channel_index]; + mem = st->mem + channel_index * st->mem_alloc_size; + while (!(last_sample >= *in_len || out_sample >= *out_len)) + { + int j; + spx_word32_t sum=0; + + /* We already have all the filter coefficients pre-computed in the table */ + const spx_word16_t *ptr; + /* Do the memory part */ + for (j=0;last_sample-N+1+j < 0;j++) + { + sum += MULT16_16(mem[last_sample+j],st->sinc_table[samp_frac_num*st->filt_len+j]); + } + + /* Do the new part */ + ptr = in+st->in_stride*(last_sample-N+1+j); + for (;j<N;j++) + { + sum += MULT16_16(*ptr,st->sinc_table[samp_frac_num*st->filt_len+j]); + ptr += st->in_stride; + } + + *out = PSHR32(sum,15); + out += st->out_stride; + out_sample++; + last_sample += st->int_advance; + samp_frac_num += st->frac_advance; + if (samp_frac_num >= st->den_rate) + { + samp_frac_num -= st->den_rate; + last_sample++; + } + } + st->last_sample[channel_index] = last_sample; + st->samp_frac_num[channel_index] = samp_frac_num; + return out_sample; +} + +#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, int channel_index, const spx_word16_t *in, int *in_len, spx_word16_t *out, int *out_len) +{ + int N = st->filt_len; + int out_sample = 0; + spx_word16_t *mem; + int last_sample = st->last_sample[channel_index]; + int samp_frac_num = st->samp_frac_num[channel_index]; + mem = st->mem + channel_index * st->mem_alloc_size; + while (!(last_sample >= *in_len || out_sample >= *out_len)) + { + int j; + double sum=0; + + /* We already have all the filter coefficients pre-computed in the table */ + const spx_word16_t *ptr; + /* Do the memory part */ + for (j=0;last_sample-N+1+j < 0;j++) + { + sum += MULT16_16(mem[last_sample+j],(double)st->sinc_table[samp_frac_num*st->filt_len+j]); + } + + /* Do the new part */ + ptr = in+st->in_stride*(last_sample-N+1+j); + for (;j<N;j++) + { + sum += MULT16_16(*ptr,(double)st->sinc_table[samp_frac_num*st->filt_len+j]); + ptr += st->in_stride; + } + + *out = sum; + out += st->out_stride; + out_sample++; + last_sample += st->int_advance; + samp_frac_num += st->frac_advance; + if (samp_frac_num >= st->den_rate) + { + samp_frac_num -= st->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 int resampler_basic_interpolate_single(SpeexResamplerState *st, int channel_index, const spx_word16_t *in, int *in_len, spx_word16_t *out, int *out_len) +{ + int N = st->filt_len; + int out_sample = 0; + spx_word16_t *mem; + int last_sample = st->last_sample[channel_index]; + int samp_frac_num = st->samp_frac_num[channel_index]; + mem = st->mem + channel_index * st->mem_alloc_size; + while (!(last_sample >= *in_len || out_sample >= *out_len)) + { + int j; + spx_word32_t sum=0; + + /* We need to interpolate the sinc filter */ + spx_word32_t accum[4] = {0.f,0.f, 0.f, 0.f}; + spx_word16_t interp[4]; + const spx_word16_t *ptr; + int offset; + spx_word16_t frac; + offset = samp_frac_num*st->oversample/st->den_rate; +#ifdef FIXED_POINT + frac = PDIV32(SHL32((samp_frac_num*st->oversample) % st->den_rate,15),st->den_rate); +#else + frac = ((float)((samp_frac_num*st->oversample) % st->den_rate))/st->den_rate; +#endif + /* This code is written like this to make it easy to optimise with SIMD. + For most DSPs, it would be best to split the loops in two because most DSPs + have only two accumulators */ + for (j=0;last_sample-N+1+j < 0;j++) + { + spx_word16_t curr_mem = mem[last_sample+j]; + accum[0] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset-2]); + accum[1] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset-1]); + accum[2] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset]); + accum[3] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset+1]); + } + ptr = in+st->in_stride*(last_sample-N+1+j); + /* Do the new part */ + for (;j<N;j++) + { + spx_word16_t curr_in = *ptr; + ptr += st->in_stride; + 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]); + + *out = PSHR32(sum,15); + out += st->out_stride; + out_sample++; + last_sample += st->int_advance; + samp_frac_num += st->frac_advance; + if (samp_frac_num >= st->den_rate) + { + samp_frac_num -= st->den_rate; + last_sample++; + } + } + st->last_sample[channel_index] = last_sample; + st->samp_frac_num[channel_index] = samp_frac_num; + return out_sample; +} + +#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, int channel_index, const spx_word16_t *in, int *in_len, spx_word16_t *out, int *out_len) +{ + int N = st->filt_len; + int out_sample = 0; + spx_word16_t *mem; + int last_sample = st->last_sample[channel_index]; + int samp_frac_num = st->samp_frac_num[channel_index]; + mem = st->mem + channel_index * st->mem_alloc_size; + while (!(last_sample >= *in_len || out_sample >= *out_len)) + { + int j; + spx_word32_t sum=0; + + /* We need to interpolate the sinc filter */ + double accum[4] = {0.f,0.f, 0.f, 0.f}; + float interp[4]; + const spx_word16_t *ptr; + float alpha = ((float)samp_frac_num)/st->den_rate; + int offset = samp_frac_num*st->oversample/st->den_rate; + float frac = alpha*st->oversample - offset; + /* This code is written like this to make it easy to optimise with SIMD. + For most DSPs, it would be best to split the loops in two because most DSPs + have only two accumulators */ + for (j=0;last_sample-N+1+j < 0;j++) + { + double curr_mem = mem[last_sample+j]; + accum[0] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset-2]); + accum[1] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset-1]); + accum[2] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset]); + accum[3] += MULT16_16(curr_mem,st->sinc_table[4+(j+1)*st->oversample-offset+1]); + } + ptr = in+st->in_stride*(last_sample-N+1+j); + /* Do the new part */ + for (;j<N;j++) + { + double curr_in = *ptr; + ptr += st->in_stride; + 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 = interp[0]*accum[0] + interp[1]*accum[1] + interp[2]*accum[2] + interp[3]*accum[3]; + + *out = PSHR32(sum,15); + out += st->out_stride; + out_sample++; + last_sample += st->int_advance; + samp_frac_num += st->frac_advance; + if (samp_frac_num >= st->den_rate) + { + samp_frac_num -= st->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) +{ + int i; + int 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); + } else { + /* up-sampling */ + st->cutoff = quality_map[st->quality].upsample_bandwidth; + } + + /* Choose the resampling type that requires the least amount of memory */ + if (st->den_rate <= st->oversample) + { + 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++) + { + int j; + for (j=0;j<st->filt_len;j++) + { + st->sinc_table[i*st->filt_len+j] = sinc(st->cutoff,((j-st->filt_len/2+1)-((float)i)/st->den_rate), st->filt_len, quality_map[st->quality].window_func); + } + } +#ifdef FIXED_POINT + st->resampler_ptr = resampler_basic_direct_single; +#else + if (st->quality>8) + st->resampler_ptr = resampler_basic_direct_double; + else + st->resampler_ptr = resampler_basic_direct_single; +#endif + /*fprintf (stderr, "resampler uses direct sinc table and normalised cutoff %f\n", cutoff);*/ + } else { + 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<st->oversample*st->filt_len+4;i++) + st->sinc_table[i+4] = sinc(st->cutoff,(i/(float)st->oversample - st->filt_len/2), st->filt_len, quality_map[st->quality].window_func); +#ifdef FIXED_POINT + st->resampler_ptr = resampler_basic_interpolate_single; +#else + if (st->quality>8) + st->resampler_ptr = resampler_basic_interpolate_double; + else + st->resampler_ptr = resampler_basic_interpolate_single; +#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; + + if (!st->mem) + { + st->mem = (spx_word16_t*)speex_alloc(st->nb_channels*(st->filt_len-1) * sizeof(spx_word16_t)); + for (i=0;i<st->nb_channels*(st->filt_len-1);i++) + st->mem[i] = 0; + st->mem_alloc_size = st->filt_len-1; + /*speex_warning("init filter");*/ + } else if (!st->started) + { + st->mem = (spx_word16_t*)speex_realloc(st->mem, st->nb_channels*(st->filt_len-1) * sizeof(spx_word16_t)); + for (i=0;i<st->nb_channels*(st->filt_len-1);i++) + st->mem[i] = 0; + st->mem_alloc_size = st->filt_len-1; + /*speex_warning("reinit filter");*/ + } else if (st->filt_len > old_length) + { + /* Increase the filter length */ + /*speex_warning("increase filter size");*/ + int old_alloc_size = st->mem_alloc_size; + if (st->filt_len-1 > st->mem_alloc_size) + { + st->mem = (spx_word16_t*)speex_realloc(st->mem, st->nb_channels*(st->filt_len-1) * sizeof(spx_word16_t)); + st->mem_alloc_size = st->filt_len-1; + } + for (i=0;i<st->nb_channels;i++) + { + int j; + /* Copy data going backward */ + for (j=0;j<old_length-1;j++) + st->mem[i*st->mem_alloc_size+(st->filt_len-2-j)] = st->mem[i*old_alloc_size+(old_length-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 - old_length)/2; + } + } else if (st->filt_len < old_length) + { + /* Reduce filter length, this a bit tricky */ + /*speex_warning("decrease filter size (unimplemented)");*/ + /* Adjust last_sample (which will likely end up negative) */ + /*st->last_sample += (st->filt_len - old_length)/2;*/ + for (i=0;i<st->nb_channels;i++) + { + int j; + st->magic_samples[i] = (old_length - st->filt_len)/2; + /* Copy data going backward */ + 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]]; + } + } + +} + +SpeexResamplerState *speex_resampler_init(int nb_channels, int in_rate, int out_rate, int quality) +{ + return speex_resampler_init_frac(nb_channels, in_rate, out_rate, in_rate, out_rate, quality); +} + +SpeexResamplerState *speex_resampler_init_frac(int nb_channels, int ratio_num, int ratio_den, int in_rate, int out_rate, int quality) +{ + int i; + SpeexResamplerState *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->cutoff = 1.f; + st->nb_channels = nb_channels; + st->in_stride = 1; + st->out_stride = 1; + + /* Per channel data */ + st->last_sample = (int*)speex_alloc(nb_channels*sizeof(int)); + st->magic_samples = (int*)speex_alloc(nb_channels*sizeof(int)); + st->samp_frac_num = (int*)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); + + + update_filter(st); + + st->initialised = 1; + return st; +} + +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 void speex_resampler_process_native(SpeexResamplerState *st, int channel_index, const spx_word16_t *in, int *in_len, spx_word16_t *out, int *out_len) +{ + int j=0; + int N = st->filt_len; + int out_sample = 0; + spx_word16_t *mem; + int tmp_out_len = 0; + mem = st->mem + channel_index * st->mem_alloc_size; + st->started = 1; + + /* Handle the case where we have samples left from a reduction in filter length */ + if (st->magic_samples[channel_index]) + { + int tmp_in_len; + int tmp_magic; + tmp_in_len = st->magic_samples[channel_index]; + tmp_out_len = *out_len; + /* FIXME: Need to handle the case where the out array is too small */ + /* magic_samples needs to be set to zero to avoid infinite recursion */ + tmp_magic = st->magic_samples[channel_index]; + st->magic_samples[channel_index] = 0; + speex_resampler_process_native(st, channel_index, mem+N-1, &tmp_in_len, out, &tmp_out_len); + /*speex_warning_int("extra samples:", tmp_out_len);*/ + /* If we couldn't process all "magic" input samples, save the rest for next time */ + if (tmp_in_len < tmp_magic) + { + int i; + st->magic_samples[channel_index] = tmp_magic-tmp_in_len; + for (i=0;i<st->magic_samples[channel_index];i++) + mem[N-1+i]=mem[N-1+i+tmp_in_len]; + } + out += tmp_out_len; + } + + /* Call the right resampler through the function ptr */ + out_sample = st->resampler_ptr(st, channel_index, in, in_len, out, out_len); + + if (st->last_sample[channel_index] < *in_len) + *in_len = st->last_sample[channel_index]; + *out_len = out_sample+tmp_out_len; + st->last_sample[channel_index] -= *in_len; + + for (j=0;j<N-1-*in_len;j++) + mem[j] = mem[j+*in_len]; + for (;j<N-1;j++) + mem[j] = in[st->in_stride*(j+*in_len-N+1)]; + +} + +#ifdef FIXED_POINT +void speex_resampler_process_float(SpeexResamplerState *st, int channel_index, const float *in, int *in_len, float *out, int *out_len) +{ + int i; + int istride_save, ostride_save; + spx_word16_t x[*in_len]; + spx_word16_t y[*out_len]; + istride_save = st->in_stride; + ostride_save = st->out_stride; + for (i=0;i<*in_len;i++) + x[i] = WORD2INT(in[i*st->in_stride]); + st->in_stride = st->out_stride = 1; + speex_resampler_process_native(st, channel_index, x, in_len, y, out_len); + st->in_stride = istride_save; + st->out_stride = ostride_save; + for (i=0;i<*out_len;i++) + out[i*st->out_stride] = y[i]; +} +void speex_resampler_process_int(SpeexResamplerState *st, int channel_index, const spx_int16_t *in, int *in_len, spx_int16_t *out, int *out_len) +{ + speex_resampler_process_native(st, channel_index, in, in_len, out, out_len); +} +#else +void speex_resampler_process_float(SpeexResamplerState *st, int channel_index, const float *in, int *in_len, float *out, int *out_len) +{ + speex_resampler_process_native(st, channel_index, in, in_len, out, out_len); +} +void speex_resampler_process_int(SpeexResamplerState *st, int channel_index, const spx_int16_t *in, int *in_len, spx_int16_t *out, int *out_len) +{ + int i; + int istride_save, ostride_save; + spx_word16_t x[*in_len]; + spx_word16_t y[*out_len]; + istride_save = st->in_stride; + ostride_save = st->out_stride; + for (i=0;i<*in_len;i++) + x[i] = in[i*st->in_stride]; + st->in_stride = st->out_stride = 1; + speex_resampler_process_native(st, channel_index, x, in_len, y, out_len); + st->in_stride = istride_save; + st->out_stride = ostride_save; + for (i=0;i<*out_len;i++) + out[i*st->out_stride] = WORD2INT(y[i]); +} +#endif + +void speex_resampler_process_interleaved_float(SpeexResamplerState *st, const float *in, int *in_len, float *out, int *out_len) +{ + int i; + int istride_save, ostride_save; + 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++) + { + speex_resampler_process_float(st, i, in+i, in_len, out+i, out_len); + } + st->in_stride = istride_save; + st->out_stride = ostride_save; +} + +void speex_resampler_process_interleaved_int(SpeexResamplerState *st, const spx_int16_t *in, int *in_len, spx_int16_t *out, int *out_len) +{ + int i; + int istride_save, ostride_save; + 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++) + { + speex_resampler_process_int(st, i, in+i, in_len, out+i, out_len); + } + st->in_stride = istride_save; + st->out_stride = ostride_save; +} + +void speex_resampler_set_rate(SpeexResamplerState *st, int in_rate, int out_rate) +{ + speex_resampler_set_rate_frac(st, in_rate, out_rate, in_rate, out_rate); +} + +void speex_resampler_get_rate(SpeexResamplerState *st, int *in_rate, int *out_rate) +{ + *in_rate = st->in_rate; + *out_rate = st->out_rate; +} + +void speex_resampler_set_rate_frac(SpeexResamplerState *st, int ratio_num, int ratio_den, int in_rate, int out_rate) +{ + int fact; + if (st->in_rate == in_rate && st->out_rate == out_rate && st->num_rate == ratio_num && st->den_rate == ratio_den) + return; + + 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<=sqrt(IMAX(in_rate, out_rate));fact++) + { + while ((st->num_rate % fact == 0) && (st->den_rate % fact == 0)) + { + st->num_rate /= fact; + st->den_rate /= fact; + } + } + + if (st->initialised) + update_filter(st); +} + +void speex_resampler_get_ratio(SpeexResamplerState *st, int *ratio_num, int *ratio_den) +{ + *ratio_num = st->num_rate; + *ratio_den = st->den_rate; +} + +void speex_resampler_set_quality(SpeexResamplerState *st, int quality) +{ + if (quality < 0) + quality = 0; + if (quality > 10) + quality = 10; + if (st->quality == quality) + return; + st->quality = quality; + if (st->initialised) + update_filter(st); +} + +void speex_resampler_get_quality(SpeexResamplerState *st, int *quality) +{ + *quality = st->quality; +} + +void speex_resampler_set_input_stride(SpeexResamplerState *st, int stride) +{ + st->in_stride = stride; +} + +void speex_resampler_get_input_stride(SpeexResamplerState *st, int *stride) +{ + *stride = st->in_stride; +} + +void speex_resampler_set_output_stride(SpeexResamplerState *st, int stride) +{ + st->out_stride = stride; +} + +void speex_resampler_get_output_stride(SpeexResamplerState *st, int *stride) +{ + *stride = st->out_stride; +} + +void speex_resampler_skip_zeros(SpeexResamplerState *st) +{ + int i; + for (i=0;i<st->nb_channels;i++) + st->last_sample[i] = st->filt_len/2; +} + +void speex_resampler_reset_mem(SpeexResamplerState *st) +{ + int i; + for (i=0;i<st->nb_channels*(st->filt_len-1);i++) + st->mem[i] = 0; +} + diff --git a/pph/speex_resampler.h b/pph/speex_resampler.h new file mode 100644 index 0000000..8ba6790 --- /dev/null +++ b/pph/speex_resampler.h @@ -0,0 +1,319 @@ +/* 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 ALSA_PUBLIC_PARROT_HACK_PLUGIN +#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_resample_set_output_stride CAT_PREFIX(RANDOM_PREFIX,_resample_set_output_stride) +#define speex_resample_get_output_stride CAT_PREFIX(RANDOM_PREFIX,_resample_get_output_stride) +#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 spx_int16_t short + +#ifdef FIXED_POINT +#define spx_word16_t short +#define spx_word32_t int + +#else /* FIXED_POINT */ + +#define spx_word16_t float +#define spx_word32_t float +#define MULT16_16(a,b) ((a)*(b)) +#define MULT16_32_Q15(a,b) ((a)*(b)) +#define PSHR32(a,b) (a) +#endif /* FIXED_POINT */ + +#else /* OUTSIDE_SPEEX */ + +#include "speex/speex_types.h" + +#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 + +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. + * @return Newly created resampler state + * @retval NULL Error: not enough memory + */ +SpeexResamplerState *speex_resampler_init(int nb_channels, + int in_rate, + int out_rate, + int quality); + +/** 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. + * @return Newly created resampler state + * @retval NULL Error: not enough memory + */ +SpeexResamplerState *speex_resampler_init_frac(int nb_channels, + int ratio_num, + int ratio_den, + int in_rate, + int out_rate, + int quality); + +/** 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 + */ +void speex_resampler_process_float(SpeexResamplerState *st, + int channel_index, + const float *in, + int *in_len, + float *out, + int *out_len); + +/** 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 + */ +void speex_resampler_process_int(SpeexResamplerState *st, + int channel_index, + const spx_int16_t *in, + int *in_len, + spx_int16_t *out, + int *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. + */ +void speex_resampler_process_interleaved_float(SpeexResamplerState *st, + const float *in, + int *in_len, + float *out, + int *out_len); + +/** 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. + */ +void speex_resampler_process_interleaved_int(SpeexResamplerState *st, + const spx_int16_t *in, + int *in_len, + spx_int16_t *out, + int *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). + */ +void speex_resampler_set_rate(SpeexResamplerState *st, + int in_rate, + int 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, + int *in_rate, + int *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). + */ +void speex_resampler_set_rate_frac(SpeexResamplerState *st, + int ratio_num, + int ratio_den, + int in_rate, + int 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, + int *ratio_num, + int *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. + */ +void 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, + int stride); + +/** Get the input stride. + * @param st Resampler state + * @param stride Input stride copied + */ +void speex_resampler_get_input_stride(SpeexResamplerState *st, + int *stride); + +/** Set (change) the output stride. + * @param st Resampler state + * @param stride Output stride + */ +void speex_resample_set_output_stride(SpeexResamplerState *st, + int stride); + +/** Get the output stride. + * @param st Resampler state copied + * @param stride Output stride + */ +void speex_resample_get_output_stride(SpeexResamplerState *st, + int *stride); + +/** 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 + */ +void speex_resampler_skip_zeros(SpeexResamplerState *st); + +/** Reset a resampler so a new (unrelated) stream can be processed. + * @param st Resampler state + */ +void speex_resampler_reset_mem(SpeexResamplerState *st); + +#ifdef __cplusplus +} +#endif + +#endif |