diff --git a/libspeex/arch.h b/libspeex/arch.h new file mode 100644 index 0000000000..d38c36ce7c --- /dev/null +++ b/libspeex/arch.h @@ -0,0 +1,239 @@ +/* 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 "speex/speex_types.h" +#endif + +#define ABS(x) ((x) < 0 ? (-(x)) : (x)) /**< Absolute integer value. */ +#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 + +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) + +#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 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 diff --git a/libspeex/config.h b/libspeex/config.h new file mode 100644 index 0000000000..abd35f0914 --- /dev/null +++ b/libspeex/config.h @@ -0,0 +1,20 @@ +// Microsoft version of 'inline' +#define inline __inline + +// Visual Studio support alloca(), but it always align variables to 16-bit +// boundary, while SSE need 128-bit alignment. So we disable alloca() when +// SSE is enabled. +#ifndef _USE_SSE +# define USE_ALLOCA +#endif + +/* Default to floating point */ +#ifndef FIXED_POINT +# define FLOATING_POINT +# define USE_SMALLFT +#else +# define USE_KISS_FFT +#endif + +/* We don't support visibility on Win32 */ +#define EXPORT diff --git a/libspeex/os_support.h b/libspeex/os_support.h new file mode 100644 index 0000000000..6b74b0c22f --- /dev/null +++ b/libspeex/os_support.h @@ -0,0 +1,169 @@ +/* Copyright (C) 2007 Jean-Marc Valin + + File: os_support.h + This is the (tiny) OS abstraction layer. Aside from math.h, this is the + only place where system headers are allowed. + + 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 OS_SUPPORT_H +#define OS_SUPPORT_H + +#include +#include +#include + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#ifdef OS_SUPPORT_CUSTOM +#include "os_support_custom.h" +#endif + +/** Speex wrapper for calloc. To do your own dynamic allocation, all you need to do is replace this function, speex_realloc and speex_free + NOTE: speex_alloc needs to CLEAR THE MEMORY */ +#ifndef OVERRIDE_SPEEX_ALLOC +static inline void *speex_alloc (int size) +{ + /* WARNING: this is not equivalent to malloc(). If you want to use malloc() + or your own allocator, YOU NEED TO CLEAR THE MEMORY ALLOCATED. Otherwise + you will experience strange bugs */ + return calloc(size,1); +} +#endif + +/** Same as speex_alloc, except that the area is only needed inside a Speex call (might cause problem with wideband though) */ +#ifndef OVERRIDE_SPEEX_ALLOC_SCRATCH +static inline void *speex_alloc_scratch (int size) +{ + /* Scratch space doesn't need to be cleared */ + return calloc(size,1); +} +#endif + +/** Speex wrapper for realloc. To do your own dynamic allocation, all you need to do is replace this function, speex_alloc and speex_free */ +#ifndef OVERRIDE_SPEEX_REALLOC +static inline void *speex_realloc (void *ptr, int size) +{ + return realloc(ptr, size); +} +#endif + +/** Speex wrapper for calloc. To do your own dynamic allocation, all you need to do is replace this function, speex_realloc and speex_alloc */ +#ifndef OVERRIDE_SPEEX_FREE +static inline void speex_free (void *ptr) +{ + free(ptr); +} +#endif + +/** Same as speex_free, except that the area is only needed inside a Speex call (might cause problem with wideband though) */ +#ifndef OVERRIDE_SPEEX_FREE_SCRATCH +static inline void speex_free_scratch (void *ptr) +{ + free(ptr); +} +#endif + +/** Copy n bytes of memory from src to dst. The 0* term provides compile-time type checking */ +#ifndef OVERRIDE_SPEEX_COPY +#define SPEEX_COPY(dst, src, n) (memcpy((dst), (src), (n)*sizeof(*(dst)) + 0*((dst)-(src)) )) +#endif + +/** Copy n bytes of memory from src to dst, allowing overlapping regions. The 0* term + provides compile-time type checking */ +#ifndef OVERRIDE_SPEEX_MOVE +#define SPEEX_MOVE(dst, src, n) (memmove((dst), (src), (n)*sizeof(*(dst)) + 0*((dst)-(src)) )) +#endif + +/** Set n bytes of memory to value of c, starting at address s */ +#ifndef OVERRIDE_SPEEX_MEMSET +#define SPEEX_MEMSET(dst, c, n) (memset((dst), (c), (n)*sizeof(*(dst)))) +#endif + + +#ifndef OVERRIDE_SPEEX_FATAL +static inline void _speex_fatal(const char *str, const char *file, int line) +{ + fprintf (stderr, "Fatal (internal) error in %s, line %d: %s\n", file, line, str); + exit(1); +} +#endif + +#ifndef OVERRIDE_SPEEX_WARNING +static inline void speex_warning(const char *str) +{ +#ifndef DISABLE_WARNINGS + fprintf (stderr, "warning: %s\n", str); +#endif +} +#endif + +#ifndef OVERRIDE_SPEEX_WARNING_INT +static inline void speex_warning_int(const char *str, int val) +{ +#ifndef DISABLE_WARNINGS + fprintf (stderr, "warning: %s %d\n", str, val); +#endif +} +#endif + +#ifndef OVERRIDE_SPEEX_NOTIFY +static inline void speex_notify(const char *str) +{ +#ifndef DISABLE_NOTIFICATIONS + fprintf (stderr, "notification: %s\n", str); +#endif +} +#endif + +#ifndef OVERRIDE_SPEEX_PUTC +/** Speex wrapper for putc */ +static inline void _speex_putc(int ch, void *file) +{ + FILE *f = (FILE *)file; + fprintf(f, "%c", ch); +} +#endif + +#define speex_fatal(str) _speex_fatal(str, __FILE__, __LINE__); +#define speex_assert(cond) {if (!(cond)) {speex_fatal("assertion failed: " #cond);}} + +#ifndef RELEASE +static inline void print_vec(float *vec, int len, char *name) +{ + int i; + printf ("%s ", name); + for (i=0;i +static void *speex_alloc (int size) {return calloc(size,1);} +static void *speex_realloc (void *ptr, int size) {return realloc(ptr, size);} +static void speex_free (void *ptr) {free(ptr);} +#include "speex_resampler.h" +#include "arch.h" +#else /* OUTSIDE_SPEEX */ + +#include "speex/speex_resampler.h" +#include "arch.h" +#include "os_support.h" +#endif /* OUTSIDE_SPEEX */ + +#include "stack_alloc.h" +#include + +#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 IMAX(a,b) ((a) > (b) ? (a) : (b)) +#define IMIN(a,b) ((a) < (b) ? (a) : (b)) + +#ifndef NULL +#define NULL 0 +#endif + +#ifdef _USE_SSE +#include "resample_sse.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 + +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; + + /* 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; +} ; + +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). +*/ +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*/ +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 +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, 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]; + +#ifndef OVERRIDE_INNER_PRODUCT_SINGLE + float accum[4] = {0,0,0,0}; + + for(j=0;j= 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; +} + +#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]; + +#ifndef OVERRIDE_INNER_PRODUCT_DOUBLE + double accum[4] = {0,0,0,0}; + + for(j=0;j= 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 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 = PDIV32(SHL32((samp_frac_num*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]; + + +#ifndef OVERRIDE_INTERPOLATE_PRODUCT_SINGLE + spx_word32_t accum[4] = {0,0,0,0}; + + for(j=0;jsinc_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]); +#else + cubic_coef(frac, interp); + sum = interpolate_product_single(iptr, st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample, interp); +#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; +} + +#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 + const spx_word16_t frac = ((float)((samp_frac_num*st->oversample) % st->den_rate))/st->den_rate; +#endif + spx_word16_t interp[4]; + + +#ifndef OVERRIDE_INTERPOLATE_PRODUCT_DOUBLE + double accum[4] = {0,0,0,0}; + + for(j=0;jsinc_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]); +#else + cubic_coef(frac, interp); + sum = interpolate_product_double(iptr, st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample, interp); +#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 */ + if (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;iden_rate;i++) + { + spx_int32_t j; + for (j=0;jfilt_len;j++) + { + st->sinc_table[i*st->filt_len+j] = sinc(st->cutoff,((j-(spx_int32_t)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 { + 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] = 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; + + + /* 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;inb_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;inb_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;jmagic_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;jmem[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 (;jfilt_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;jfilt_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;inb_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;jfilt_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, int *err) +{ + return speex_resampler_init_frac(nb_channels, in_rate, out_rate, in_rate, out_rate, quality, err); +} + +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, int *err) +{ + spx_uint32_t i; + SpeexResamplerState *st; + if (quality > 10 || quality < 0) + { + 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->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 + + /* 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;ilast_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; + 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;jmagic_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;imagic_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 +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 +{ + 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 && olen) { + spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen; + spx_uint32_t ochunk = olen; + + if (in) { + for(j=0;jout_stride; + if (in) + in += ichunk * istride; + } + } + *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 && olen) { + 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;jfilt_len-1]=WORD2INT(in[j*istride_save]); +#else + x[j+st->filt_len-1]=in[j*istride_save]; +#endif + } else { + for(j=0;jfilt_len-1]=0; + } + + speex_resampler_process_native(st, channel_index, &ichunk, y, &ochunk); + } else { + ichunk = 0; + ochunk = 0; + } + + for (j=0;jout_stride = ostride_save; + *in_len -= ilen; + *out_len -= olen; + + return RESAMPLER_ERR_SUCCESS; +} + +EXPORT int speex_resampler_process_interleaved_float(SpeexResamplerState *st, const float *in, spx_uint32_t *in_len, float *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;inb_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;inb_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;inb_channels;i++) + { + st->samp_frac_num[i]=st->samp_frac_num[i]*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_skip_zeros(SpeexResamplerState *st) +{ + spx_uint32_t i; + for (i=0;inb_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;inb_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."; + } +} diff --git a/libspeex/speex/speex_resampler.h b/libspeex/speex/speex_resampler.h new file mode 100644 index 0000000000..54eef8d7b8 --- /dev/null +++ b/libspeex/speex/speex_resampler.h @@ -0,0 +1,340 @@ +/* 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_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 short +#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" + +#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 +}; + +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(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 + * 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(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); + +/** 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 + */ +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 + * 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. + */ +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. + * @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 in input samples introduced by the resampler. + * @param st Resampler state + */ +int speex_resampler_get_input_latency(SpeexResamplerState *st); + +/** Get the latency in output samples introduced by the resampler. + * @param st Resampler state + */ +int speex_resampler_get_output_latency(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 diff --git a/libspeex/speex/speex_types.h b/libspeex/speex/speex_types.h new file mode 100644 index 0000000000..852fed801d --- /dev/null +++ b/libspeex/speex/speex_types.h @@ -0,0 +1,126 @@ +/* speex_types.h taken from libogg */ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the Xiph.Org Foundation http://www.xiph.org/ * + * * + ******************************************************************** + + function: #ifdef jail to whip a few platforms into the UNIX ideal. + last mod: $Id: os_types.h 7524 2004-08-11 04:20:36Z conrad $ + + ********************************************************************/ +/** + @file speex_types.h + @brief Speex types +*/ +#ifndef _SPEEX_TYPES_H +#define _SPEEX_TYPES_H + +#if defined(_WIN32) + +# if defined(__CYGWIN__) +# include <_G_config.h> + typedef _G_int32_t spx_int32_t; + typedef _G_uint32_t spx_uint32_t; + typedef _G_int16_t spx_int16_t; + typedef _G_uint16_t spx_uint16_t; +# elif defined(__MINGW32__) + typedef short spx_int16_t; + typedef unsigned short spx_uint16_t; + typedef int spx_int32_t; + typedef unsigned int spx_uint32_t; +# elif defined(__MWERKS__) + typedef int spx_int32_t; + typedef unsigned int spx_uint32_t; + typedef short spx_int16_t; + typedef unsigned short spx_uint16_t; +# else + /* MSVC/Borland */ + typedef __int32 spx_int32_t; + typedef unsigned __int32 spx_uint32_t; + typedef __int16 spx_int16_t; + typedef unsigned __int16 spx_uint16_t; +# endif + +#elif defined(__MACOS__) + +# include + typedef SInt16 spx_int16_t; + typedef UInt16 spx_uint16_t; + typedef SInt32 spx_int32_t; + typedef UInt32 spx_uint32_t; + +#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */ + +# include + typedef int16_t spx_int16_t; + typedef u_int16_t spx_uint16_t; + typedef int32_t spx_int32_t; + typedef u_int32_t spx_uint32_t; + +#elif defined(__BEOS__) + + /* Be */ +# include + typedef int16_t spx_int16_t; + typedef u_int16_t spx_uint16_t; + typedef int32_t spx_int32_t; + typedef u_int32_t spx_uint32_t; + +#elif defined (__EMX__) + + /* OS/2 GCC */ + typedef short spx_int16_t; + typedef unsigned short spx_uint16_t; + typedef int spx_int32_t; + typedef unsigned int spx_uint32_t; + +#elif defined (DJGPP) + + /* DJGPP */ + typedef short spx_int16_t; + typedef int spx_int32_t; + typedef unsigned int spx_uint32_t; + +#elif defined(R5900) + + /* PS2 EE */ + typedef int spx_int32_t; + typedef unsigned spx_uint32_t; + typedef short spx_int16_t; + +#elif defined(__SYMBIAN32__) + + /* Symbian GCC */ + typedef signed short spx_int16_t; + typedef unsigned short spx_uint16_t; + typedef signed int spx_int32_t; + typedef unsigned int spx_uint32_t; + +#elif defined(CONFIG_TI_C54X) || defined (CONFIG_TI_C55X) + + typedef short spx_int16_t; + typedef unsigned short spx_uint16_t; + typedef long spx_int32_t; + typedef unsigned long spx_uint32_t; + +#elif defined(CONFIG_TI_C6X) + + typedef short spx_int16_t; + typedef unsigned short spx_uint16_t; + typedef int spx_int32_t; + typedef unsigned int spx_uint32_t; + +#else + +# include + +#endif + +#endif /* _SPEEX_TYPES_H */ diff --git a/libspeex/stack_alloc.h b/libspeex/stack_alloc.h new file mode 100644 index 0000000000..f2a10921c5 --- /dev/null +++ b/libspeex/stack_alloc.h @@ -0,0 +1,115 @@ +/* Copyright (C) 2002 Jean-Marc Valin */ +/** + @file stack_alloc.h + @brief Temporary memory allocation on stack +*/ +/* + 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 STACK_ALLOC_H +#define STACK_ALLOC_H + +#ifdef USE_ALLOCA +# ifdef WIN32 +# include +# else +# ifdef HAVE_ALLOCA_H +# include +# else +# include +# endif +# endif +#endif + +/** + * @def ALIGN(stack, size) + * + * Aligns the stack to a 'size' boundary + * + * @param stack Stack + * @param size New size boundary + */ + +/** + * @def PUSH(stack, size, type) + * + * Allocates 'size' elements of type 'type' on the stack + * + * @param stack Stack + * @param size Number of elements + * @param type Type of element + */ + +/** + * @def VARDECL(var) + * + * Declare variable on stack + * + * @param var Variable to declare + */ + +/** + * @def ALLOC(var, size, type) + * + * Allocate 'size' elements of 'type' on stack + * + * @param var Name of variable to allocate + * @param size Number of elements + * @param type Type of element + */ + +#ifdef ENABLE_VALGRIND + +#include + +#define ALIGN(stack, size) ((stack) += ((size) - (long)(stack)) & ((size) - 1)) + +#define PUSH(stack, size, type) (VALGRIND_MAKE_NOACCESS(stack, 1000),ALIGN((stack),sizeof(type)),VALGRIND_MAKE_WRITABLE(stack, ((size)*sizeof(type))),(stack)+=((size)*sizeof(type)),(type*)((stack)-((size)*sizeof(type)))) + +#else + +#define ALIGN(stack, size) ((stack) += ((size) - (long)(stack)) & ((size) - 1)) + +#define PUSH(stack, size, type) (ALIGN((stack),sizeof(type)),(stack)+=((size)*sizeof(type)),(type*)((stack)-((size)*sizeof(type)))) + +#endif + +#if defined(VAR_ARRAYS) +#define VARDECL(var) +#define ALLOC(var, size, type) type var[size] +#elif defined(USE_ALLOCA) +#define VARDECL(var) var +#define ALLOC(var, size, type) var = _alloca(sizeof(type)*(size)) +#else +#define VARDECL(var) var +#define ALLOC(var, size, type) var = PUSH(stack, size, type) +#endif + + +#endif diff --git a/projects/openrct2.vcxproj b/projects/openrct2.vcxproj index 3077811de3..9af8c93c1f 100644 --- a/projects/openrct2.vcxproj +++ b/projects/openrct2.vcxproj @@ -33,6 +33,7 @@ + @@ -63,6 +64,7 @@ + @@ -80,6 +82,7 @@ + @@ -186,7 +189,7 @@ $(SolutionDir)..\obj\$(Configuration)\ - $(SolutionDir)..\lodepng;$(SolutionDir)..\sdl\include;$(IncludePath) + $(SolutionDir)..\lodepng;$(SolutionDir)..\sdl\include;$(SolutionDir)..\libspeex;$(IncludePath) $(SolutionDir)..\sdl\lib\x86;$(LibraryPath) $(SolutionDir)..\build\$(Configuration)\ $(SolutionDir)..\obj\$(Configuration)\ @@ -197,7 +200,7 @@ Disabled true 1Byte - _CRT_SECURE_NO_WARNINGS;%(PreprocessorDefinitions) + _CRT_SECURE_NO_WARNINGS;HAVE_CONFIG_H;_USE_MATH_DEFINES;%(PreprocessorDefinitions) true @@ -218,7 +221,7 @@ false - _CRT_SECURE_NO_WARNINGS;%(PreprocessorDefinitions) + _CRT_SECURE_NO_WARNINGS;HAVE_CONFIG_H;_USE_MATH_DEFINES;%(PreprocessorDefinitions) true diff --git a/projects/openrct2.vcxproj.filters b/projects/openrct2.vcxproj.filters index 59424e14b6..8f175ca65e 100644 --- a/projects/openrct2.vcxproj.filters +++ b/projects/openrct2.vcxproj.filters @@ -162,6 +162,9 @@ Header Files + + Header Files + @@ -386,6 +389,12 @@ Windows + + Source Files + + + Source Files + diff --git a/src/audio.c b/src/audio.c index fe46f41668..b366b5f9f9 100644 --- a/src/audio.c +++ b/src/audio.c @@ -215,14 +215,14 @@ int dsound_create_primary_buffer(int a, int device, int channels, int samples, i } dsdevice = &RCT2_GLOBAL(RCT2_ADDRESS_DSOUND_DEVICES, rct_dsdevice*)[device]; } - memset(&RCT2_GLOBAL(RCT2_ADDRESS_AUDIO_INFO, rct_audio_info), 0, sizeof(rct_audio_info)); - RCT2_GLOBAL(RCT2_ADDRESS_AUDIO_INFO, rct_audio_info).var_0 = 1; - RCT2_GLOBAL(RCT2_ADDRESS_AUDIO_INFO, rct_audio_info).channels = channels; - RCT2_GLOBAL(RCT2_ADDRESS_AUDIO_INFO, rct_audio_info).samples = samples; - RCT2_GLOBAL(RCT2_ADDRESS_AUDIO_INFO, rct_audio_info).var_8 = samples * RCT2_GLOBAL(0x01425B4C, uint16); - RCT2_GLOBAL(RCT2_ADDRESS_AUDIO_INFO, rct_audio_info).bytes = bits * channels / 8; - RCT2_GLOBAL(RCT2_ADDRESS_AUDIO_INFO, rct_audio_info).bits = bits; - RCT2_GLOBAL(RCT2_ADDRESS_AUDIO_INFO, rct_audio_info).var_E = 0; + memset(&RCT2_GLOBAL(RCT2_ADDRESS_AUDIO_INFO, WAVEFORMATEX), 0, sizeof(WAVEFORMATEX)); + RCT2_GLOBAL(RCT2_ADDRESS_AUDIO_INFO, WAVEFORMATEX).wFormatTag = 1; + RCT2_GLOBAL(RCT2_ADDRESS_AUDIO_INFO, WAVEFORMATEX).nChannels = channels; + RCT2_GLOBAL(RCT2_ADDRESS_AUDIO_INFO, WAVEFORMATEX).nSamplesPerSec = samples; + RCT2_GLOBAL(RCT2_ADDRESS_AUDIO_INFO, WAVEFORMATEX).nAvgBytesPerSec = samples * RCT2_GLOBAL(0x01425B4C, uint16); + RCT2_GLOBAL(RCT2_ADDRESS_AUDIO_INFO, WAVEFORMATEX).nBlockAlign = bits * channels / 8; + RCT2_GLOBAL(RCT2_ADDRESS_AUDIO_INFO, WAVEFORMATEX).wBitsPerSample = bits; + RCT2_GLOBAL(RCT2_ADDRESS_AUDIO_INFO, WAVEFORMATEX).cbSize = 0; DSBUFFERDESC bufferdesc; memset(&bufferdesc, 0, sizeof(bufferdesc)); bufferdesc.dwSize = sizeof(bufferdesc); @@ -466,8 +466,8 @@ int sub_4015E7(int channel) } else { sound_channel->var_168 = 1; sound_channel->var_15C = read; - rct_audio_info* audio_info = sound_channel->hmem; - uint16 v = ((audio_info->var_E != 8) - 1) & 0x80; + LPWAVEFORMATEX waveformat = sound_channel->hmem; + uint16 v = ((waveformat->nBlockAlign != 8) - 1) & 0x80; memset(&buf1[read], v, buf1size - r); } } @@ -531,7 +531,7 @@ MMRESULT mmio_open(char* filename, HMMIO* hmmio, HGLOBAL* hmem, LPMMCKINFO mmcki HMMIO hmmio1; MMRESULT result; MMCKINFO mmckinfo1; - rct_audio_info audio_info; + WAVEFORMATEX waveformat; hmemold = hmem; *hmem = 0; @@ -555,8 +555,8 @@ MMRESULT mmio_open(char* filename, HMMIO* hmmio, HGLOBAL* hmem, LPMMCKINFO mmcki result = 57601; goto label20; } - if (mmioRead(hmmio1, (HPSTR)&audio_info, 16) == 16) { - if (audio_info.var_0 == 1) { + if (mmioRead(hmmio1, (HPSTR)&waveformat, 16) == 16) { + if (waveformat.wFormatTag == 1) { //strcpy(audio_info.var_0, "\x01"); hmem = 0; label11: @@ -566,7 +566,7 @@ MMRESULT mmio_open(char* filename, HMMIO* hmmio, HGLOBAL* hmem, LPMMCKINFO mmcki result = 57344; goto label20; } - memcpy(hmemold2, &audio_info, 16); + memcpy(hmemold2, &waveformat, 16); *((uint16*)*hmemold + 8) = (uint16)hmem; if (!(uint16)hmem || mmioRead(hmmio1, (char*)*hmemold + 18, (uint16)hmem) == (uint16)hmem) { result = mmioAscend(hmmio1, &mmckinfo1, 0); diff --git a/src/audio.h b/src/audio.h index c68b059e96..af0b1a9776 100644 --- a/src/audio.h +++ b/src/audio.h @@ -59,16 +59,6 @@ typedef struct rct_sound { struct rct_sound* next; } rct_sound; -typedef struct { - uint16 var_0; - uint16 channels; - uint32 samples; - uint32 var_8; - uint16 bytes; - uint16 bits; - uint16 var_E; -} rct_audio_info; - typedef struct { uint32 var_0; uint32 var_4; diff --git a/src/mixer.cpp b/src/mixer.cpp new file mode 100644 index 0000000000..a7959e34cf --- /dev/null +++ b/src/mixer.cpp @@ -0,0 +1,401 @@ +/***************************************************************************** + * Copyright (c) 2014 Ted John + * OpenRCT2, an open source clone of Roller Coaster Tycoon 2. + * + * This file is part of OpenRCT2. + * + * OpenRCT2 is free software: you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, either version 3 of the License, or + * (at your option) any later version. + + * This program 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 General Public License for more details. + + * You should have received a copy of the GNU General Public License + * along with this program. If not, see . + *****************************************************************************/ + +#include +#include +#include + +extern "C" { +#include "audio.h" +#include "config.h" +} +#include "mixer.h" + +Sample::Sample() +{ + data = 0; + length = 0; + issdlwav = false; +} + +Sample::~Sample() +{ + Unload(); +} + +bool Sample::Load(char* filename) +{ + Unload(); + SDL_RWops* rw = SDL_RWFromFile(filename, "rb"); + if (!rw) { + SDL_RWclose(rw); + return false; + } + SDL_AudioSpec audiospec; + memset(&audiospec, 0, sizeof(audiospec)); + SDL_AudioSpec* spec = SDL_LoadWAV_RW(rw, false, &audiospec, &data, &length); + if (spec != NULL) { + format.freq = spec->freq; + format.format = spec->format; + format.channels = spec->channels; + } else { + issdlwav = true; + } + return true; +} + +bool Sample::LoadCSS1(char* filename, unsigned int offset) +{ + Unload(); + SDL_RWops* rw = SDL_RWFromFile(filename, "rb"); + if (!rw) { + return false; + } + Uint32 numsounds; + SDL_RWread(rw, &numsounds, sizeof(numsounds), 1); + if (offset > numsounds) { + SDL_RWclose(rw); + return false; + } + SDL_RWseek(rw, offset * 4, RW_SEEK_CUR); + Uint32 soundoffset; + SDL_RWread(rw, &soundoffset, sizeof(soundoffset), 1); + SDL_RWseek(rw, soundoffset, RW_SEEK_SET); + Uint32 soundsize; + SDL_RWread(rw, &soundsize, sizeof(soundsize), 1); + length = soundsize; + WAVEFORMATEX waveformat; + SDL_RWread(rw, &waveformat, sizeof(waveformat), 1); + format.freq = waveformat.nSamplesPerSec; + format.format = AUDIO_S16; + format.channels = (Uint8)waveformat.nChannels; + data = new Uint8[length]; + SDL_RWread(rw, data, length, 1); + SDL_RWclose(rw); + return true; +} + +void Sample::Unload() +{ + SDL_LockAudio(); + if (data) { + if (issdlwav) { + SDL_FreeWAV(data); + } else { + delete[] data; + } + data = 0; + } + issdlwav = false; + length = 0; + SDL_UnlockAudio(); +} + +bool Sample::Convert(AudioFormat format) +{ + if(Sample::format.format != format.format || Sample::format.channels != format.channels || Sample::format.freq != format.freq){ + SDL_AudioCVT cvt; + if (SDL_BuildAudioCVT(&cvt, Sample::format.format, Sample::format.channels, Sample::format.freq, format.format, format.channels, format.freq) < 0) { + return false; + } + cvt.len = length; + cvt.buf = new Uint8[cvt.len * cvt.len_mult]; + memcpy(cvt.buf, data, length); + if (SDL_ConvertAudio(&cvt) < 0) { + delete[] cvt.buf; + return false; + } + Unload(); + data = cvt.buf; + length = cvt.len_cvt; + Sample::format = format; + } + return true; +} + +const Uint8* Sample::Data() +{ + return data; +} + +int Sample::Length() +{ + return length; +} + +Stream::Stream() +{ + sourcetype = SOURCE_NONE; +} + +int Stream::GetSome(int offset, const Uint8** data, int length) +{ + int size = length; + switch(sourcetype) { + case SOURCE_SAMPLE: + if (offset >= sample->Length()) { + return 0; + } + if (offset + length > sample->Length()) { + size = sample->Length() - offset; + } + *data = &sample->Data()[offset]; + return size; + break; + } + return 0; +} + +int Stream::Length() +{ + switch(sourcetype) { + case SOURCE_SAMPLE: + return sample->Length(); + break; + } + return 0; +} + +void Stream::SetSource_Sample(Sample& sample) +{ + sourcetype = SOURCE_SAMPLE; + Stream::sample = &sample; +} + +const AudioFormat* Stream::Format() +{ + switch(sourcetype) { + case SOURCE_SAMPLE: + return &sample->format; + break; + } + return 0; +} + +Channel::Channel() +{ + rate = 1; + resampler = 0; + SetVolume(SDL_MIX_MAXVOLUME); +} + +Channel::~Channel() +{ + if (resampler) { + speex_resampler_destroy(resampler); + resampler = 0; + } +} + +void Channel::Play(Stream& stream, int loop = 0) +{ + Channel::stream = &stream; + Channel::loop = loop; + offset = 0; +} + +void Channel::SetRate(double rate) +{ + Channel::rate = rate; + if (Channel::rate < 0.001) { + Channel::rate = 0.001; + } +} + +void Channel::SetVolume(int volume) +{ + Channel::volume = volume; + if (Channel::volume > SDL_MIX_MAXVOLUME) { + Channel::volume = SDL_MIX_MAXVOLUME; + } + if (Channel::volume < 0) { + Channel::volume = 0; + } +} + +void Mixer::Init(char* device) +{ + Close(); + SDL_AudioSpec want, have; + SDL_zero(want); + want.freq = 22050; + want.format = AUDIO_S16; + want.channels = 2; + want.samples = 1024; + want.callback = Callback; + want.userdata = this; + deviceid = SDL_OpenAudioDevice(device, 0, &want, &have, 0); + format.format = have.format; + format.channels = have.channels; + format.freq = have.freq; + char css1filename[260]; + strcpy(css1filename, gGeneral_config.game_path); + strcat(css1filename, "\\Data\\css1.dat"); + for (int i = 0; i < 63; i++) { + css1samples[i].LoadCSS1(css1filename, i); + css1samples[i].Convert(format); // convert to audio output format, saves some cpu usage but requires a bit more memory, optional + css1streams[i].SetSource_Sample(css1samples[i]); + } + SDL_PauseAudioDevice(deviceid, 0); +} + +void Mixer::Close() +{ + SDL_CloseAudioDevice(deviceid); +} + +void SDLCALL Mixer::Callback(void* arg, Uint8* stream, int length) +{ + Mixer* mixer = (Mixer*)arg; + memset(stream, 0, length); + for (int i = 0; i < 10; i++) { + mixer->MixChannel(mixer->channels[i], stream, length); + } +} + +void Mixer::MixChannel(Channel& channel, Uint8* data, int length) +{ + if (channel.stream) { + if (!channel.resampler) { + channel.resampler = speex_resampler_init(format.channels, format.freq, format.freq, 0, 0); + } + AudioFormat channelformat = *channel.stream->Format(); + int loaded = 0; + SDL_AudioCVT cvt; + cvt.len_ratio = 1; + do { + int samplesize = format.channels * format.BytesPerSample(); + int samples = length / samplesize; + int samplesloaded = loaded / samplesize; + int samplestoread = (int)ceil((samples - samplesloaded) * channel.rate); + int lengthloaded = 0; + if (channel.offset < channel.stream->Length()) { + bool mustconvert = false; + if (MustConvert(*channel.stream)) { + if (SDL_BuildAudioCVT(&cvt, channelformat.format, channelformat.channels, channelformat.freq, Mixer::format.format, Mixer::format.channels, Mixer::format.freq) == -1) { + break; + } + mustconvert = true; + } + const Uint8* datastream; + int readfromstream = (channel.stream->GetSome(channel.offset, &datastream, (int)(((samplestoread) * samplesize) / cvt.len_ratio)) / channelformat.BytesPerSample()) * channelformat.BytesPerSample(); + if (readfromstream == 0) { + break; + } + int volume = channel.volume; + if (mustconvert) { + Uint8* dataconverted; + if (Convert(cvt, datastream, readfromstream, &dataconverted)) { + if (channel.rate != 1 && format.format == AUDIO_S16) { + spx_uint32_t in_len = (spx_uint32_t)(ceil((double)cvt.len_cvt / samplesize)); + Uint8* out = new Uint8[length + 200]; // needs some extra, otherwise resampler sometimes doesn't process all the input samples + spx_uint32_t out_len = samples + 20; + speex_resampler_set_rate(channel.resampler, format.freq, (int)(format.freq * (1 / channel.rate))); + speex_resampler_process_interleaved_int(channel.resampler, (const spx_int16_t*)dataconverted, &in_len, (spx_int16_t*)out, &out_len); + int mixlength = (out_len * samplesize); + if (loaded + mixlength > length) { // check for overflow + mixlength = length - loaded; + } + lengthloaded = (out_len * samplesize); + SDL_MixAudioFormat(&data[loaded], out, format.format, mixlength, volume); + delete[] out; + } else { + lengthloaded = (cvt.len_cvt / samplesize) * samplesize; + int mixlength = lengthloaded; + if (loaded + cvt.len_cvt > length) { + mixlength = length - loaded; + } + SDL_MixAudioFormat(&data[loaded], dataconverted, format.format, mixlength, volume); + } + delete[] dataconverted; + } + } else { + if (channel.rate != 1 && format.format == AUDIO_S16) { + spx_uint32_t in_len = (spx_uint32_t)(ceil((double)readfromstream / samplesize)); + Uint8* out = new Uint8[length + 200]; + spx_uint32_t out_len = samples + 20; + speex_resampler_set_rate(channel.resampler, format.freq, (int)(format.freq * (1 / channel.rate))); + speex_resampler_process_interleaved_int(channel.resampler, (const spx_int16_t*)datastream, &in_len, (spx_int16_t*)out, &out_len); + int mixlength = (out_len * samplesize); + if (loaded + mixlength > length) { + mixlength = length - loaded; + } + lengthloaded = (out_len * samplesize); + SDL_MixAudioFormat(&data[loaded], out, format.format, mixlength, volume); + delete[] out; + } else { + lengthloaded = readfromstream; + int mixlength = lengthloaded; + if (loaded + readfromstream > length) { + mixlength = length - loaded; + } + SDL_MixAudioFormat(&data[loaded], datastream, format.format, mixlength, volume); + } + } + + channel.offset += readfromstream; + + } + + loaded += lengthloaded; + + if (channel.loop != 0 && channel.offset >= channel.stream->Length()) { + if (channel.loop != -1) { + channel.loop--; + } + channel.offset = 0; + } + } while(loaded < length || (loaded < length && channel.loop != 0 && channel.offset == 0)); + } +} + +bool Mixer::MustConvert(Stream& stream) +{ + const AudioFormat* streamformat = stream.Format(); + if (!streamformat) { + return false; + } + if (streamformat->format != format.format || streamformat->channels != format.channels || streamformat->freq != format.freq) { + return true; + } + return false; +} + +bool Mixer::Convert(SDL_AudioCVT& cvt, const Uint8* data, int length, Uint8** dataout) +{ + if (length == 0 || cvt.len_mult == 0) { + return false; + } + cvt.len = length; + cvt.buf = (Uint8*)new char[cvt.len * cvt.len_mult]; + memcpy(cvt.buf, data, length); + if (SDL_ConvertAudio(&cvt) < 0) { + delete[] cvt.buf; + return false; + } + *dataout = cvt.buf; + return true; +} + +void Mixer_Init(char* device) +{ + static Mixer mixer; + mixer.Init(device); +} \ No newline at end of file diff --git a/src/mixer.h b/src/mixer.h new file mode 100644 index 0000000000..9b38dce756 --- /dev/null +++ b/src/mixer.h @@ -0,0 +1,127 @@ +/***************************************************************************** + * Copyright (c) 2014 Ted John + * OpenRCT2, an open source clone of Roller Coaster Tycoon 2. + * + * This file is part of OpenRCT2. + * + * OpenRCT2 is free software: you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, either version 3 of the License, or + * (at your option) any later version. + + * This program 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 General Public License for more details. + + * You should have received a copy of the GNU General Public License + * along with this program. If not, see . + *****************************************************************************/ + +#ifndef _MIXER_H_ +#define _MIXER_H_ + +#include "rct2.h" + +#ifdef __cplusplus + +extern "C" { +#include +} + +struct AudioFormat { + int BytesPerSample() const { return (SDL_AUDIO_BITSIZE(format)) / 8; }; + int freq; + SDL_AudioFormat format; + Uint8 channels; +}; + +class Sample +{ +public: + Sample(); + ~Sample(); + bool Load(char* filename); + bool LoadCSS1(char* filename, unsigned int offset); + void Unload(); + bool Convert(AudioFormat format); + const Uint8* Data(); + int Length(); + + friend class Stream; + +private: + AudioFormat format; + Uint8* data; + Uint32 length; + bool issdlwav; +}; + +class Stream +{ +public: + Stream(); + int GetSome(int offset, const Uint8** data, int length); + int Length(); + void SetSource_Sample(Sample& sample); + const AudioFormat* Format(); + + friend class Mixer; + +private: + enum { + SOURCE_NONE = 0, + SOURCE_SAMPLE + } sourcetype; + Sample* sample; +}; + +class Channel +{ +public: + Channel(); + ~Channel(); + void Play(Stream& stream, int loop); + void SetRate(double rate); + void SetVolume(int volume); + + friend class Mixer; + +private: + int loop; + int offset; + double rate; + int volume; + SpeexResamplerState* resampler; + Stream* stream; +}; + +class Mixer +{ +public: + void Init(char* device); + void Close(); + +private: + static void SDLCALL Callback(void* arg, Uint8* data, int length); + void MixChannel(Channel& channel, Uint8* buffer, int length); + bool MustConvert(Stream& stream); + bool Convert(SDL_AudioCVT& cvt, const Uint8* data, int length, Uint8** dataout); + SDL_AudioDeviceID deviceid; + AudioFormat format; + Sample css1samples[63]; + Stream css1streams[63]; + Channel channels[10]; +}; + +extern "C" +{ +#endif + +void Mixer_Init(char* device); + +#ifdef __cplusplus +} +#endif + +#endif \ No newline at end of file diff --git a/src/osinterface.h b/src/osinterface.h index a9480bb0db..ba2f91a7b8 100644 --- a/src/osinterface.h +++ b/src/osinterface.h @@ -74,7 +74,7 @@ typedef struct { char path[260]; uint32 var_20C; uint8 pad_210[0x100]; - char addon[15][0x80]; + char addon[16][0x80]; uint32 addons; //0xB10 } rct2_install_info; diff --git a/src/rct2.c b/src/rct2.c index 29572477a3..0af4c93dea 100644 --- a/src/rct2.c +++ b/src/rct2.c @@ -39,6 +39,7 @@ #include "intro.h" #include "language.h" #include "map.h" +#include "mixer.h" #include "news_item.h" #include "object.h" #include "osinterface.h" @@ -88,6 +89,7 @@ __declspec(dllexport) int StartOpenRCT(HINSTANCE hInstance, HINSTANCE hPrevInsta config_init(); language_open(gGeneral_config.language); rct2_init(); + Mixer_Init(NULL); rct2_loop(); osinterface_free(); exit(0); diff --git a/src/window_options.c b/src/window_options.c index e1bfc20c90..4df489aa0f 100644 --- a/src/window_options.c +++ b/src/window_options.c @@ -33,6 +33,7 @@ #include "config.h" #include "gfx.h" #include "language.h" +#include "mixer.h" #include "osinterface.h" #include "sprites.h" #include "string_ids.h" @@ -503,6 +504,9 @@ static void window_options_dropdown() switch (widgetIndex) { case WIDX_SOUND_DROPDOWN: audio_init2(dropdownIndex); + if (dropdownIndex < gAudioDeviceCount) { + Mixer_Init(gAudioDevices[dropdownIndex].name); + } /*#ifdef _MSC_VER __asm movzx ax, dropdownIndex #else