diff options
Diffstat (limited to 'src/pulsecore/speex')
| -rw-r--r-- | src/pulsecore/speex/arch.h | 62 | ||||
| -rw-r--r-- | src/pulsecore/speex/fixed_generic.h | 8 | ||||
| -rw-r--r-- | src/pulsecore/speex/resample.c | 123 | ||||
| -rw-r--r-- | src/pulsecore/speex/speex_resampler.h | 116 |
4 files changed, 180 insertions, 129 deletions
diff --git a/src/pulsecore/speex/arch.h b/src/pulsecore/speex/arch.h index e2d731ac..9987c8fb 100644 --- a/src/pulsecore/speex/arch.h +++ b/src/pulsecore/speex/arch.h @@ -7,18 +7,18 @@ 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 @@ -35,6 +35,45 @@ #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 "speex/speex_types.h" #endif @@ -68,6 +107,7 @@ typedef spx_word32_t spx_sig_t; #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) @@ -111,9 +151,6 @@ typedef float spx_word32_t; #define GAIN_SCALING 1.f #define GAIN_SCALING_1 1.f -#define LPC_SHIFT 0 -#define LSP_SHIFT 0 -#define SIG_SHIFT 0 #define VERY_SMALL 1e-15f #define VERY_LARGE32 1e15f @@ -182,11 +219,11 @@ typedef float spx_word32_t; #if defined (CONFIG_TI_C54X) || defined (CONFIG_TI_C55X) /* 2 on TI C5x DSP */ -#define BYTES_PER_CHAR 2 +#define BYTES_PER_CHAR 2 #define BITS_PER_CHAR 16 #define LOG2_BITS_PER_CHAR 4 -#else +#else #define BYTES_PER_CHAR 1 #define BITS_PER_CHAR 8 @@ -194,4 +231,11 @@ typedef float spx_word32_t; #endif + + +#ifdef FIXED_DEBUG +long long spx_mips=0; +#endif + + #endif diff --git a/src/pulsecore/speex/fixed_generic.h b/src/pulsecore/speex/fixed_generic.h index 2948177c..547e22c7 100644 --- a/src/pulsecore/speex/fixed_generic.h +++ b/src/pulsecore/speex/fixed_generic.h @@ -7,18 +7,18 @@ 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 diff --git a/src/pulsecore/speex/resample.c b/src/pulsecore/speex/resample.c index 1cc4d490..1e592002 100644 --- a/src/pulsecore/speex/resample.c +++ b/src/pulsecore/speex/resample.c @@ -1,5 +1,5 @@ /* Copyright (C) 2007 Jean-Marc Valin - + File: resample.c Arbitrary resampling code @@ -37,17 +37,23 @@ - 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 + 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 @@ -62,9 +68,10 @@ static void speex_free (void *ptr) {free(ptr);} #include "speex_resampler.h" #include "arch.h" #else /* OUTSIDE_SPEEX */ - + #include "speex/speex_resampler.h" -#include "misc.h" +#include "arch.h" +#include "os_support.h" #endif /* OUTSIDE_SPEEX */ #include <math.h> @@ -74,11 +81,11 @@ static void speex_free (void *ptr) {free(ptr);} #endif #ifdef FIXED_POINT -#define WORD2INT(x) ((x) < -32767 ? -32768 : ((x) > 32766 ? 32767 : (x))) +#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)))) +#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 @@ -97,7 +104,7 @@ struct SpeexResamplerState_ { 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; @@ -108,17 +115,17 @@ struct SpeexResamplerState_ { spx_uint32_t oversample; int initialised; int started; - + /* 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; } ; @@ -160,7 +167,7 @@ static double kaiser8_table[36] = { 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, @@ -173,7 +180,7 @@ struct FuncDef { double *table; int oversample; }; - + static struct FuncDef _KAISER12 = {kaiser12_table, 64}; #define KAISER12 (&_KAISER12) /*static struct FuncDef _KAISER12 = {kaiser12_table, 32}; @@ -195,7 +202,7 @@ struct QualityMapping { /* This table maps conversion quality to internal parameters. There are two - reasons that explain why the up-sampling bandwidth is larger than the + 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) @@ -221,7 +228,7 @@ static double compute_func(float x, struct FuncDef *func) { float y, frac; double interp[4]; - int ind; + int ind; y = x*func->oversample; ind = (int)floor(y); frac = (y-ind); @@ -232,7 +239,7 @@ static double compute_func(float x, struct FuncDef *func) 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]; } @@ -320,7 +327,7 @@ static int resampler_basic_direct_single(SpeexResamplerState *st, spx_uint32_t c { 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 */ @@ -328,7 +335,7 @@ static int resampler_basic_direct_single(SpeexResamplerState *st, spx_uint32_t c { 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++) @@ -336,7 +343,7 @@ static int resampler_basic_direct_single(SpeexResamplerState *st, spx_uint32_t c 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++; @@ -368,7 +375,7 @@ static int resampler_basic_direct_double(SpeexResamplerState *st, spx_uint32_t c { 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 */ @@ -376,7 +383,7 @@ static int resampler_basic_direct_double(SpeexResamplerState *st, spx_uint32_t c { 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++) @@ -384,7 +391,7 @@ static int resampler_basic_direct_double(SpeexResamplerState *st, spx_uint32_t c 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++; @@ -414,7 +421,7 @@ static int resampler_basic_interpolate_single(SpeexResamplerState *st, spx_uint3 { 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]; @@ -428,7 +435,7 @@ static int resampler_basic_interpolate_single(SpeexResamplerState *st, spx_uint3 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 + 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++) { @@ -451,7 +458,7 @@ static int resampler_basic_interpolate_single(SpeexResamplerState *st, spx_uint3 } 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++; @@ -483,7 +490,7 @@ static int resampler_basic_interpolate_double(SpeexResamplerState *st, spx_uint3 { 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]; @@ -492,7 +499,7 @@ static int resampler_basic_interpolate_double(SpeexResamplerState *st, spx_uint3 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 + 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++) { @@ -515,7 +522,7 @@ static int resampler_basic_interpolate_double(SpeexResamplerState *st, spx_uint3 } 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++; @@ -536,11 +543,11 @@ static int resampler_basic_interpolate_double(SpeexResamplerState *st, spx_uint3 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 */ @@ -616,7 +623,7 @@ static void update_filter(SpeexResamplerState *st) 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. */ @@ -654,7 +661,7 @@ static void update_filter(SpeexResamplerState *st) /*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--) @@ -729,12 +736,12 @@ SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels, spx_uin 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 = (spx_int32_t*)speex_alloc(nb_channels*sizeof(int)); st->magic_samples = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(int)); @@ -749,9 +756,9 @@ SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels, spx_uin 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; if (err) *err = RESAMPLER_ERR_SUCCESS; @@ -780,14 +787,14 @@ static int speex_resampler_process_native(SpeexResamplerState *st, spx_uint32_t spx_uint32_t 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 istride_save; spx_uint32_t tmp_in_len; spx_uint32_t tmp_magic; - + istride_save = st->in_stride; tmp_in_len = st->magic_samples[channel_index]; tmp_out_len = *out_len; @@ -809,20 +816,20 @@ static int speex_resampler_process_native(SpeexResamplerState *st, spx_uint32_t out += tmp_out_len*st->out_stride; *out_len -= 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] < (spx_int32_t)*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-(spx_int32_t)*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)]; - + return RESAMPLER_ERR_SUCCESS; } @@ -879,7 +886,7 @@ int speex_resampler_process_float(SpeexResamplerState *st, spx_uint32_t channel_ olen -= ochunk; } *in_len -= ilen; - *out_len -= olen; + *out_len -= olen; #endif return RESAMPLER_ERR_SUCCESS; } @@ -942,7 +949,7 @@ int speex_resampler_process_int(SpeexResamplerState *st, spx_uint32_t channel_in olen -= ochunk; } *in_len -= ilen; - *out_len -= olen; + *out_len -= olen; #endif return RESAMPLER_ERR_SUCCESS; } @@ -966,7 +973,7 @@ int speex_resampler_process_interleaved_float(SpeexResamplerState *st, const flo return RESAMPLER_ERR_SUCCESS; } - + 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; @@ -1003,7 +1010,7 @@ int speex_resampler_set_rate_frac(SpeexResamplerState *st, spx_uint32_t ratio_nu 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; @@ -1018,7 +1025,7 @@ int speex_resampler_set_rate_frac(SpeexResamplerState *st, spx_uint32_t ratio_nu st->den_rate /= fact; } } - + if (old_den > 0) { for (i=0;i<st->nb_channels;i++) @@ -1029,7 +1036,7 @@ int speex_resampler_set_rate_frac(SpeexResamplerState *st, spx_uint32_t ratio_nu st->samp_frac_num[i] = st->den_rate-1; } } - + if (st->initialised) update_filter(st); return RESAMPLER_ERR_SUCCESS; diff --git a/src/pulsecore/speex/speex_resampler.h b/src/pulsecore/speex/speex_resampler.h index c44fbcd0..8629eeb3 100644 --- a/src/pulsecore/speex/speex_resampler.h +++ b/src/pulsecore/speex/speex_resampler.h @@ -1,8 +1,8 @@ /* 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 @@ -43,7 +43,7 @@ /********* WARNING: MENTAL SANITY ENDS HERE *************/ -/* If the resampler is defined outside of Speex, we change the symbol names so that +/* 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 */ @@ -53,7 +53,7 @@ #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) @@ -79,7 +79,7 @@ #define spx_int32_t int #define spx_uint16_t unsigned short #define spx_uint32_t unsigned int - + #else /* OUTSIDE_SPEEX */ #include "speex/speex_types.h" @@ -102,7 +102,7 @@ enum { RESAMPLER_ERR_BAD_STATE = 2, RESAMPLER_ERR_INVALID_ARG = 3, RESAMPLER_ERR_PTR_OVERLAP = 4, - + RESAMPLER_ERR_MAX_ERROR }; @@ -118,14 +118,14 @@ typedef struct SpeexResamplerState_ SpeexResamplerState; * @return Newly created resampler state * @retval NULL Error: not enough memory */ -SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels, - spx_uint32_t in_rate, - spx_uint32_t out_rate, +SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels, + spx_uint32_t in_rate, + spx_uint32_t out_rate, int quality, 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 +/** 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 @@ -137,11 +137,11 @@ SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels, * @return Newly created resampler state * @retval NULL Error: not enough memory */ -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, +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, int *err); @@ -152,24 +152,24 @@ 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 + * @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 + * @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_float(SpeexResamplerState *st, - spx_uint32_t channel_index, - const float *in, - spx_uint32_t *in_len, - float *out, +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); /** 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 + * @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 @@ -177,11 +177,11 @@ int speex_resampler_process_float(SpeexResamplerState *st, * @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, +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. @@ -193,10 +193,10 @@ int speex_resampler_process_int(SpeexResamplerState *st, * @param out_len Size of the output buffer. Returns the number of samples written. * This is all per-channel. */ -int speex_resampler_process_interleaved_float(SpeexResamplerState *st, - const float *in, - spx_uint32_t *in_len, - float *out, +int speex_resampler_process_interleaved_float(SpeexResamplerState *st, + const float *in, + spx_uint32_t *in_len, + float *out, spx_uint32_t *out_len); /** Resample an interleaved int array. The input and output buffers must *not* overlap. @@ -208,10 +208,10 @@ int speex_resampler_process_interleaved_float(SpeexResamplerState *st, * @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, +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). @@ -219,8 +219,8 @@ int speex_resampler_process_interleaved_int(SpeexResamplerState *st, * @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, +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). @@ -228,11 +228,11 @@ int speex_resampler_set_rate(SpeexResamplerState *st, * @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, +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 +/** 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 @@ -240,10 +240,10 @@ void speex_resampler_get_rate(SpeexResamplerState *st, * @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, +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 @@ -252,56 +252,56 @@ int speex_resampler_set_rate_frac(SpeexResamplerState *st, * @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, +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 + * @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 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 + * @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, +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, +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, +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, +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, +void speex_resampler_get_output_stride(SpeexResamplerState *st, spx_uint32_t *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 +/** 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 |