diff --git a/PlatformIO/lib/esp_sr/libc_speech_features.a b/PlatformIO/lib/esp_sr/libc_speech_features.a deleted file mode 100644 index 4e665a5..0000000 Binary files a/PlatformIO/lib/esp_sr/libc_speech_features.a and /dev/null differ diff --git a/PlatformIO/lib/esp_sr/libdl_lib.a b/PlatformIO/lib/esp_sr/libdl_lib.a deleted file mode 100644 index 65614cd..0000000 Binary files a/PlatformIO/lib/esp_sr/libdl_lib.a and /dev/null differ diff --git a/PlatformIO/lib/esp_sr/libnn_model_alexa_wn3.a b/PlatformIO/lib/esp_sr/libnn_model_alexa_wn3.a deleted file mode 100644 index 1889573..0000000 Binary files a/PlatformIO/lib/esp_sr/libnn_model_alexa_wn3.a and /dev/null differ diff --git a/PlatformIO/lib/esp_sr/libwakenet.a b/PlatformIO/lib/esp_sr/libwakenet.a deleted file mode 100644 index 4c3acd1..0000000 Binary files a/PlatformIO/lib/esp_sr/libwakenet.a and /dev/null differ diff --git a/PlatformIO/lib/libspeex/arch.h b/PlatformIO/lib/libspeex/arch.h deleted file mode 100644 index 4449751..0000000 --- a/PlatformIO/lib/libspeex/arch.h +++ /dev/null @@ -1,235 +0,0 @@ -/* 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 - -/* 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/speexdsp_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 WORD2INT(x) ((x) < -32767 ? -32768 : ((x) > 32766 ? 32767 : (x))) - -#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 SATURATE32PSHR(x,shift,a) (x) - -#define PSHR(a,shift) (a) -#define SHR(a,shift) (a) -#define SHL(a,shift) (a) -#define SATURATE(x,a) (x) - -#define ADD16(a,b) ((a)+(b)) -#define SUB16(a,b) ((a)-(b)) -#define ADD32(a,b) ((a)+(b)) -#define SUB32(a,b) ((a)-(b)) -#define MULT16_16_16(a,b) ((a)*(b)) -#define MULT16_16(a,b) ((spx_word32_t)(a)*(spx_word32_t)(b)) -#define MAC16_16(c,a,b) ((c)+(spx_word32_t)(a)*(spx_word32_t)(b)) - -#define MULT16_32_Q11(a,b) ((a)*(b)) -#define MULT16_32_Q13(a,b) ((a)*(b)) -#define MULT16_32_Q14(a,b) ((a)*(b)) -#define MULT16_32_Q15(a,b) ((a)*(b)) -#define MULT16_32_P15(a,b) ((a)*(b)) - -#define MAC16_32_Q11(c,a,b) ((c)+(a)*(b)) -#define MAC16_32_Q15(c,a,b) ((c)+(a)*(b)) - -#define MAC16_16_Q11(c,a,b) ((c)+(a)*(b)) -#define MAC16_16_Q13(c,a,b) ((c)+(a)*(b)) -#define MAC16_16_P13(c,a,b) ((c)+(a)*(b)) -#define MULT16_16_Q11_32(a,b) ((a)*(b)) -#define MULT16_16_Q13(a,b) ((a)*(b)) -#define MULT16_16_Q14(a,b) ((a)*(b)) -#define MULT16_16_Q15(a,b) ((a)*(b)) -#define MULT16_16_P15(a,b) ((a)*(b)) -#define MULT16_16_P13(a,b) ((a)*(b)) -#define MULT16_16_P14(a,b) ((a)*(b)) - -#define DIV32_16(a,b) (((spx_word32_t)(a))/(spx_word16_t)(b)) -#define PDIV32_16(a,b) (((spx_word32_t)(a))/(spx_word16_t)(b)) -#define DIV32(a,b) (((spx_word32_t)(a))/(spx_word32_t)(b)) -#define PDIV32(a,b) (((spx_word32_t)(a))/(spx_word32_t)(b)) - -#define WORD2INT(x) ((x) < -32767.5f ? -32768 : \ - ((x) > 32766.5f ? 32767 : (spx_int16_t)floor(.5 + (x)))) -#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/PlatformIO/lib/libspeex/fixed_generic.h b/PlatformIO/lib/libspeex/fixed_generic.h deleted file mode 100644 index 12d27aa..0000000 --- a/PlatformIO/lib/libspeex/fixed_generic.h +++ /dev/null @@ -1,110 +0,0 @@ -/* Copyright (C) 2003 Jean-Marc Valin */ -/** - @file fixed_generic.h - @brief Generic fixed-point operations -*/ -/* - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -*/ - -#ifndef FIXED_GENERIC_H -#define FIXED_GENERIC_H - -#define QCONST16(x,bits) ((spx_word16_t)(.5+(x)*(((spx_word32_t)1)<<(bits)))) -#define QCONST32(x,bits) ((spx_word32_t)(.5+(x)*(((spx_word32_t)1)<<(bits)))) - -#define NEG16(x) (-(x)) -#define NEG32(x) (-(x)) -#define EXTRACT16(x) ((spx_word16_t)(x)) -#define EXTEND32(x) ((spx_word32_t)(x)) -#define SHR16(a,shift) ((a) >> (shift)) -#define SHL16(a,shift) ((a) << (shift)) -#define SHR32(a,shift) ((a) >> (shift)) -#define SHL32(a,shift) ((a) << (shift)) -#define PSHR16(a,shift) (SHR16((a)+((1<<((shift))>>1)),shift)) -#define PSHR32(a,shift) (SHR32((a)+((EXTEND32(1)<<((shift))>>1)),shift)) -#define VSHR32(a, shift) (((shift)>0) ? SHR32(a, shift) : SHL32(a, -(shift))) -#define SATURATE16(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x))) -#define SATURATE32(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x))) - -#define SATURATE32PSHR(x,shift,a) (((x)>=(SHL32(a,shift))) ? (a) : \ - (x)<=-(SHL32(a,shift)) ? -(a) : \ - (PSHR32(x, shift))) - -#define SHR(a,shift) ((a) >> (shift)) -#define SHL(a,shift) ((spx_word32_t)(a) << (shift)) -#define PSHR(a,shift) (SHR((a)+((EXTEND32(1)<<((shift))>>1)),shift)) -#define SATURATE(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x))) - - -#define ADD16(a,b) ((spx_word16_t)((spx_word16_t)(a)+(spx_word16_t)(b))) -#define SUB16(a,b) ((spx_word16_t)(a)-(spx_word16_t)(b)) -#define ADD32(a,b) ((spx_word32_t)(a)+(spx_word32_t)(b)) -#define SUB32(a,b) ((spx_word32_t)(a)-(spx_word32_t)(b)) - - -/* result fits in 16 bits */ -#define MULT16_16_16(a,b) ((((spx_word16_t)(a))*((spx_word16_t)(b)))) - -/* (spx_word32_t)(spx_word16_t) gives TI compiler a hint that it's 16x16->32 multiply */ -#define MULT16_16(a,b) (((spx_word32_t)(spx_word16_t)(a))*((spx_word32_t)(spx_word16_t)(b))) - -#define MAC16_16(c,a,b) (ADD32((c),MULT16_16((a),(b)))) -#define MULT16_32_Q12(a,b) ADD32(MULT16_16((a),SHR((b),12)), SHR(MULT16_16((a),((b)&0x00000fff)),12)) -#define MULT16_32_Q13(a,b) ADD32(MULT16_16((a),SHR((b),13)), SHR(MULT16_16((a),((b)&0x00001fff)),13)) -#define MULT16_32_Q14(a,b) ADD32(MULT16_16((a),SHR((b),14)), SHR(MULT16_16((a),((b)&0x00003fff)),14)) - -#define MULT16_32_Q11(a,b) ADD32(MULT16_16((a),SHR((b),11)), SHR(MULT16_16((a),((b)&0x000007ff)),11)) -#define MAC16_32_Q11(c,a,b) ADD32(c,ADD32(MULT16_16((a),SHR((b),11)), SHR(MULT16_16((a),((b)&0x000007ff)),11))) - -#define MULT16_32_P15(a,b) ADD32(MULT16_16((a),SHR((b),15)), PSHR(MULT16_16((a),((b)&0x00007fff)),15)) -#define MULT16_32_Q15(a,b) ADD32(MULT16_16((a),SHR((b),15)), SHR(MULT16_16((a),((b)&0x00007fff)),15)) -#define MAC16_32_Q15(c,a,b) ADD32(c,ADD32(MULT16_16((a),SHR((b),15)), SHR(MULT16_16((a),((b)&0x00007fff)),15))) - - -#define MAC16_16_Q11(c,a,b) (ADD32((c),SHR(MULT16_16((a),(b)),11))) -#define MAC16_16_Q13(c,a,b) (ADD32((c),SHR(MULT16_16((a),(b)),13))) -#define MAC16_16_P13(c,a,b) (ADD32((c),SHR(ADD32(4096,MULT16_16((a),(b))),13))) - -#define MULT16_16_Q11_32(a,b) (SHR(MULT16_16((a),(b)),11)) -#define MULT16_16_Q13(a,b) (SHR(MULT16_16((a),(b)),13)) -#define MULT16_16_Q14(a,b) (SHR(MULT16_16((a),(b)),14)) -#define MULT16_16_Q15(a,b) (SHR(MULT16_16((a),(b)),15)) - -#define MULT16_16_P13(a,b) (SHR(ADD32(4096,MULT16_16((a),(b))),13)) -#define MULT16_16_P14(a,b) (SHR(ADD32(8192,MULT16_16((a),(b))),14)) -#define MULT16_16_P15(a,b) (SHR(ADD32(16384,MULT16_16((a),(b))),15)) - -#define MUL_16_32_R15(a,bh,bl) ADD32(MULT16_16((a),(bh)), SHR(MULT16_16((a),(bl)),15)) - -#define DIV32_16(a,b) ((spx_word16_t)(((spx_word32_t)(a))/((spx_word16_t)(b)))) -#define PDIV32_16(a,b) ((spx_word16_t)(((spx_word32_t)(a)+((spx_word16_t)(b)>>1))/((spx_word16_t)(b)))) -#define DIV32(a,b) (((spx_word32_t)(a))/((spx_word32_t)(b))) -#define PDIV32(a,b) (((spx_word32_t)(a)+((spx_word16_t)(b)>>1))/((spx_word32_t)(b))) - -#endif diff --git a/PlatformIO/lib/libspeex/resample.c b/PlatformIO/lib/libspeex/resample.c deleted file mode 100644 index 15d58df..0000000 --- a/PlatformIO/lib/libspeex/resample.c +++ /dev/null @@ -1,1239 +0,0 @@ -/* Copyright (C) 2007-2008 Jean-Marc Valin - Copyright (C) 2008 Thorvald Natvig - - File: resample.c - Arbitrary resampling code - - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions are - met: - - 1. Redistributions of source code must retain the above copyright notice, - this list of conditions and the following disclaimer. - - 2. Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - 3. The name of the author may not be used to endorse or promote products - derived from this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR - IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES - OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE - DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, - INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES - (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR - SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) - HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, - STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN - ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE - POSSIBILITY OF SUCH DAMAGE. -*/ - -/* - The design goals of this code are: - - Very fast algorithm - - SIMD-friendly algorithm - - Low memory requirement - - Good *perceptual* quality (and not best SNR) - - Warning: This resampler is relatively new. Although I think I got rid of - all the major bugs and I don't expect the API to change anymore, there - may be something I've missed. So use with caution. - - This algorithm is based on this original resampling algorithm: - Smith, Julius O. Digital Audio Resampling Home Page - Center for Computer Research in Music and Acoustics (CCRMA), - Stanford University, 2007. - Web published at https://ccrma.stanford.edu/~jos/resample/. - - There is one main difference, though. This resampler uses cubic - interpolation instead of linear interpolation in the above paper. This - makes the table much smaller and makes it possible to compute that table - on a per-stream basis. In turn, being able to tweak the table for each - stream makes it possible to both reduce complexity on simple ratios - (e.g. 2/3), and get rid of the rounding operations in the inner loop. - The latter both reduces CPU time and makes the algorithm more SIMD-friendly. -*/ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#ifdef OUTSIDE_SPEEX -#include -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);} -#ifndef EXPORT -#define EXPORT -#endif -#include "speex_resampler.h" -#include "arch.h" -#else /* OUTSIDE_SPEEX */ - -#include "speex_resampler.h" -#include "arch.h" -#include "os_support.h" -#endif /* OUTSIDE_SPEEX */ - -#include -#include - -#ifndef M_PI -#define M_PI 3.14159265358979323846 -#endif - -#define IMAX(a,b) ((a) > (b) ? (a) : (b)) -#define IMIN(a,b) ((a) < (b) ? (a) : (b)) - -#ifndef NULL -#define NULL 0 -#endif - -#ifndef UINT32_MAX -#define UINT32_MAX 4294967295U -#endif - -#ifdef USE_SSE -#include "resample_sse.h" -#endif - -#ifdef USE_NEON -#include "resample_neon.h" -#endif - -/* Numer of elements to allocate on the stack */ -#ifdef VAR_ARRAYS -#define FIXED_STACK_ALLOC 8192 -#else -#define FIXED_STACK_ALLOC 1024 -#endif - -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 const double kaiser12_table[68] = { - 0.99859849, 1.00000000, 0.99859849, 0.99440475, 0.98745105, 0.97779076, - 0.96549770, 0.95066529, 0.93340547, 0.91384741, 0.89213598, 0.86843014, - 0.84290116, 0.81573067, 0.78710866, 0.75723148, 0.72629970, 0.69451601, - 0.66208321, 0.62920216, 0.59606986, 0.56287762, 0.52980938, 0.49704014, - 0.46473455, 0.43304576, 0.40211431, 0.37206735, 0.34301800, 0.31506490, - 0.28829195, 0.26276832, 0.23854851, 0.21567274, 0.19416736, 0.17404546, - 0.15530766, 0.13794294, 0.12192957, 0.10723616, 0.09382272, 0.08164178, - 0.07063950, 0.06075685, 0.05193064, 0.04409466, 0.03718069, 0.03111947, - 0.02584161, 0.02127838, 0.01736250, 0.01402878, 0.01121463, 0.00886058, - 0.00691064, 0.00531256, 0.00401805, 0.00298291, 0.00216702, 0.00153438, - 0.00105297, 0.00069463, 0.00043489, 0.00025272, 0.00013031, 0.0000527734, - 0.00001000, 0.00000000}; -/* -static const double kaiser12_table[36] = { - 0.99440475, 1.00000000, 0.99440475, 0.97779076, 0.95066529, 0.91384741, - 0.86843014, 0.81573067, 0.75723148, 0.69451601, 0.62920216, 0.56287762, - 0.49704014, 0.43304576, 0.37206735, 0.31506490, 0.26276832, 0.21567274, - 0.17404546, 0.13794294, 0.10723616, 0.08164178, 0.06075685, 0.04409466, - 0.03111947, 0.02127838, 0.01402878, 0.00886058, 0.00531256, 0.00298291, - 0.00153438, 0.00069463, 0.00025272, 0.0000527734, 0.00000500, 0.00000000}; -*/ -static const double kaiser10_table[36] = { - 0.99537781, 1.00000000, 0.99537781, 0.98162644, 0.95908712, 0.92831446, - 0.89005583, 0.84522401, 0.79486424, 0.74011713, 0.68217934, 0.62226347, - 0.56155915, 0.50119680, 0.44221549, 0.38553619, 0.33194107, 0.28205962, - 0.23636152, 0.19515633, 0.15859932, 0.12670280, 0.09935205, 0.07632451, - 0.05731132, 0.04193980, 0.02979584, 0.02044510, 0.01345224, 0.00839739, - 0.00488951, 0.00257636, 0.00115101, 0.00035515, 0.00000000, 0.00000000}; - -static const double kaiser8_table[36] = { - 0.99635258, 1.00000000, 0.99635258, 0.98548012, 0.96759014, 0.94302200, - 0.91223751, 0.87580811, 0.83439927, 0.78875245, 0.73966538, 0.68797126, - 0.63451750, 0.58014482, 0.52566725, 0.47185369, 0.41941150, 0.36897272, - 0.32108304, 0.27619388, 0.23465776, 0.19672670, 0.16255380, 0.13219758, - 0.10562887, 0.08273982, 0.06335451, 0.04724088, 0.03412321, 0.02369490, - 0.01563093, 0.00959968, 0.00527363, 0.00233883, 0.00050000, 0.00000000}; - -static const double kaiser6_table[36] = { - 0.99733006, 1.00000000, 0.99733006, 0.98935595, 0.97618418, 0.95799003, - 0.93501423, 0.90755855, 0.87598009, 0.84068475, 0.80211977, 0.76076565, - 0.71712752, 0.67172623, 0.62508937, 0.57774224, 0.53019925, 0.48295561, - 0.43647969, 0.39120616, 0.34752997, 0.30580127, 0.26632152, 0.22934058, - 0.19505503, 0.16360756, 0.13508755, 0.10953262, 0.08693120, 0.06722600, - 0.05031820, 0.03607231, 0.02432151, 0.01487334, 0.00752000, 0.00000000}; - -struct FuncDef { - const double *table; - int oversample; -}; - -static const struct FuncDef kaiser12_funcdef = {kaiser12_table, 64}; -#define KAISER12 (&kaiser12_funcdef) -static const struct FuncDef kaiser10_funcdef = {kaiser10_table, 32}; -#define KAISER10 (&kaiser10_funcdef) -static const struct FuncDef kaiser8_funcdef = {kaiser8_table, 32}; -#define KAISER8 (&kaiser8_funcdef) -static const struct FuncDef kaiser6_funcdef = {kaiser6_table, 32}; -#define KAISER6 (&kaiser6_funcdef) - -struct QualityMapping { - int base_length; - int oversample; - float downsample_bandwidth; - float upsample_bandwidth; - const 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, const 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, const 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, const 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; - - while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len)) - { - const spx_word16_t *sinct = & sinc_table[samp_frac_num*N]; - const spx_word16_t *iptr = & in[last_sample]; - -#ifndef OVERRIDE_INNER_PRODUCT_SINGLE - int j; - sum = 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; - - while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len)) - { - const spx_word16_t *sinct = & sinc_table[samp_frac_num*N]; - const spx_word16_t *iptr = & in[last_sample]; - -#ifndef OVERRIDE_INNER_PRODUCT_DOUBLE - int j; - 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; - 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 - int j; - 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],SHR32(accum[0], 1)) + MULT16_32_Q15(interp[1],SHR32(accum[1], 1)) + MULT16_32_Q15(interp[2],SHR32(accum[2], 1)) + MULT16_32_Q15(interp[3],SHR32(accum[3], 1)); - sum = SATURATE32PSHR(sum, 15, 32767); -#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++] = sum; - 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; - 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 - int j; - 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 - -/* This resampler is used to produce zero output in situations where memory - for the filter could not be allocated. The expected numbers of input and - output samples are still processed so that callers failing to check error - codes are not surprised, possibly getting into infinite loops. */ -static int resampler_basic_zero(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) -{ - 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; - - (void)in; - while (!(last_sample >= (spx_int32_t)*in_len || out_sample >= (spx_int32_t)*out_len)) - { - out[out_stride * out_sample++] = 0; - 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; -} - -static int multiply_frac(spx_uint32_t *result, spx_uint32_t value, spx_uint32_t num, spx_uint32_t den) -{ - spx_uint32_t major = value / den; - spx_uint32_t remain = value % den; - /* TODO: Could use 64 bits operation to check for overflow. But only guaranteed in C99+ */ - if (remain > UINT32_MAX / num || major > UINT32_MAX / num - || major * num > UINT32_MAX - remain * num / den) - return RESAMPLER_ERR_OVERFLOW; - *result = remain * num / den + major * num; - return RESAMPLER_ERR_SUCCESS; -} - -static int update_filter(SpeexResamplerState *st) -{ - spx_uint32_t old_length = st->filt_len; - spx_uint32_t old_alloc_size = st->mem_alloc_size; - int use_direct; - spx_uint32_t min_sinc_table_length; - spx_uint32_t min_alloc_size; - - st->int_advance = st->num_rate/st->den_rate; - st->frac_advance = st->num_rate%st->den_rate; - 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; - if (multiply_frac(&st->filt_len,st->filt_len,st->num_rate,st->den_rate) != RESAMPLER_ERR_SUCCESS) - goto fail; - /* Round up to make sure we have a multiple of 8 for SSE */ - st->filt_len = ((st->filt_len-1)&(~0x7))+8; - 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; - } - -#ifdef RESAMPLE_FULL_SINC_TABLE - use_direct = 1; - if (INT_MAX/sizeof(spx_word16_t)/st->den_rate < st->filt_len) - goto fail; -#else - /* Choose the resampling type that requires the least amount of memory */ - use_direct = st->filt_len*st->den_rate <= st->filt_len*st->oversample+8 - && INT_MAX/sizeof(spx_word16_t)/st->den_rate >= st->filt_len; -#endif - if (use_direct) - { - min_sinc_table_length = st->filt_len*st->den_rate; - } else { - if ((INT_MAX/sizeof(spx_word16_t)-8)/st->oversample < st->filt_len) - goto fail; - - min_sinc_table_length = st->filt_len*st->oversample+8; - } - if (st->sinc_table_length < min_sinc_table_length) - { - spx_word16_t *sinc_table = (spx_word16_t *)speex_realloc(st->sinc_table,min_sinc_table_length*sizeof(spx_word16_t)); - if (!sinc_table) - goto fail; - - st->sinc_table = sinc_table; - st->sinc_table_length = min_sinc_table_length; - } - if (use_direct) - { - spx_uint32_t i; - 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; - 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);*/ - } - - /* 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. */ - - /* Adding buffer_size to filt_len won't overflow here because filt_len - could be multiplied by sizeof(spx_word16_t) above. */ - min_alloc_size = st->filt_len-1 + st->buffer_size; - if (min_alloc_size > st->mem_alloc_size) - { - spx_word16_t *mem; - if (INT_MAX/sizeof(spx_word16_t)/st->nb_channels < min_alloc_size) - goto fail; - else if (!(mem = (spx_word16_t*)speex_realloc(st->mem, st->nb_channels*min_alloc_size * sizeof(*mem)))) - goto fail; - - st->mem = mem; - st->mem_alloc_size = min_alloc_size; - } - if (!st->started) - { - spx_uint32_t i; - 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_uint32_t i; - /* Increase the filter length */ - /*speex_warning("increase filter size");*/ - for (i=st->nb_channels;i--;) - { - spx_uint32_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-1+st->magic_samples[i];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; - } - } - return RESAMPLER_ERR_SUCCESS; - -fail: - st->resampler_ptr = resampler_basic_zero; - /* st->mem may still contain consumed input samples for the filter. - Restore filt_len so that filt_len - 1 still points to the position after - the last of these samples. */ - st->filt_len = old_length; - return RESAMPLER_ERR_ALLOC_FAILED; -} - -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) -{ - SpeexResamplerState *st; - int filter_err; - - if (nb_channels == 0 || ratio_num == 0 || ratio_den == 0 || quality > 10 || quality < 0) - { - if (err) - *err = RESAMPLER_ERR_INVALID_ARG; - return NULL; - } - st = (SpeexResamplerState *)speex_alloc(sizeof(SpeexResamplerState)); - if (!st) - { - if (err) - *err = RESAMPLER_ERR_ALLOC_FAILED; - return NULL; - } - 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; - - st->buffer_size = 160; - - /* Per channel data */ - if (!(st->last_sample = (spx_int32_t*)speex_alloc(nb_channels*sizeof(spx_int32_t)))) - goto fail; - if (!(st->magic_samples = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(spx_uint32_t)))) - goto fail; - if (!(st->samp_frac_num = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(spx_uint32_t)))) - goto fail; - - speex_resampler_set_quality(st, quality); - speex_resampler_set_rate_frac(st, ratio_num, ratio_den, in_rate, out_rate); - - filter_err = update_filter(st); - if (filter_err == RESAMPLER_ERR_SUCCESS) - { - st->initialised = 1; - } else { - speex_resampler_destroy(st); - st = NULL; - } - if (err) - *err = filter_err; - - return st; - -fail: - if (err) - *err = RESAMPLER_ERR_ALLOC_FAILED; - speex_resampler_destroy(st); - return NULL; -} - -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 st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : 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; - spx_word16_t ystack[ylen]; -#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 st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : 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_out_len = *out_len; - spx_uint32_t bak_in_len = *in_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_out_len; - *in_len = bak_in_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 st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : 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_out_len = *out_len; - spx_uint32_t bak_in_len = *in_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_out_len; - *in_len = bak_in_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 st->resampler_ptr == resampler_basic_zero ? RESAMPLER_ERR_ALLOC_FAILED : 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; -} - -static inline spx_uint32_t compute_gcd(spx_uint32_t a, spx_uint32_t b) -{ - while (b != 0) - { - spx_uint32_t temp = a; - - a = b; - b = temp % b; - } - return a; -} - -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 (ratio_num == 0 || ratio_den == 0) - return RESAMPLER_ERR_INVALID_ARG; - - 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; - - fact = compute_gcd(st->num_rate, st->den_rate); - - st->num_rate /= fact; - st->den_rate /= fact; - - if (old_den > 0) - { - for (i=0;inb_channels;i++) - { - if (multiply_frac(&st->samp_frac_num[i],st->samp_frac_num[i],st->den_rate,old_den) != RESAMPLER_ERR_SUCCESS) - return RESAMPLER_ERR_OVERFLOW; - /* Safety net */ - if (st->samp_frac_num[i] >= st->den_rate) - st->samp_frac_num[i] = st->den_rate-1; - } - } - - if (st->initialised) - return 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) - return 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;i++) - { - st->last_sample[i] = 0; - st->magic_samples[i] = 0; - st->samp_frac_num[i] = 0; - } - 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/PlatformIO/lib/libspeex/speex_resampler.h b/PlatformIO/lib/libspeex/speex_resampler.h deleted file mode 100644 index c521ff1..0000000 --- a/PlatformIO/lib/libspeex/speex_resampler.h +++ /dev/null @@ -1,294 +0,0 @@ -/* 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 - -#include "speexdsp_types.h" - -#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_OVERFLOW = 5, - - 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 introduced by the resampler measured in input samples. - * @param st Resampler state - */ -int speex_resampler_get_input_latency(SpeexResamplerState *st); - -/** Get the latency introduced by the resampler measured in output samples. - * @param st Resampler state - */ -int speex_resampler_get_output_latency(SpeexResamplerState *st); - -/** 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/PlatformIO/lib/libspeex/speexdsp_types.h b/PlatformIO/lib/libspeex/speexdsp_types.h deleted file mode 100644 index a70a74b..0000000 --- a/PlatformIO/lib/libspeex/speexdsp_types.h +++ /dev/null @@ -1,30 +0,0 @@ -/* speexdsp_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 speexdsp_types.h - @brief Speex types -*/ -#ifndef _SPEEX_TYPES_H -#define _SPEEX_TYPES_H - -#define spx_int16_t short -#define spx_int32_t int -#define spx_uint16_t unsigned short -#define spx_uint32_t unsigned int - -#endif /* _SPEEX_TYPES_H */ diff --git a/PlatformIO/platformio.ini b/PlatformIO/platformio.ini index 18f0182..6d8e076 100644 --- a/PlatformIO/platformio.ini +++ b/PlatformIO/platformio.ini @@ -12,8 +12,7 @@ default_envs = esp32dev [common] build_flags = - '-DFIXED_POINT=1' - '-DOUTSIDE_SPEEX=1' +; '-DFIXED_POINT=1' [env] extra_scripts = pre:load_settings.py diff --git a/PlatformIO/src/Satellite.cpp b/PlatformIO/src/Satellite.cpp index ba08406..5b22d8b 100644 --- a/PlatformIO/src/Satellite.cpp +++ b/PlatformIO/src/Satellite.cpp @@ -103,6 +103,9 @@ v7.8 - Added animations during audio playback, every device has those animation defaulted to not supported - Implemented for M5 Atom echo and Matrix Voice + v7.8.1 + - Matrix Voice now supports mono and stereo + - Supported samplerates: 8000, 16000, 22050 and 44100 16 bit. 44100S is a bit too slow * ************************************************************************ */ diff --git a/PlatformIO/src/StateMachine.hpp b/PlatformIO/src/StateMachine.hpp index 767ef27..94dddf8 100644 --- a/PlatformIO/src/StateMachine.hpp +++ b/PlatformIO/src/StateMachine.hpp @@ -326,13 +326,10 @@ class WifiDisconnected : public StateMachine if (i2sHandle == NULL) { Serial.println("Creating I2Stask"); xTaskCreatePinnedToCore(I2Stask, "I2Stask", 30000, NULL, 3, &i2sHandle, 1); - // give this task a suffciently high priority to push data in time to DMA } else { Serial.println("We already have a I2Stask"); } Serial.println("Enter WifiDisconnected"); - Serial.printf("Total heap: %d\r\n", ESP.getHeapSize()); - Serial.printf("Free heap: %d\r\n", ESP.getFreeHeap()); #if NETWORK_TYPE == NETWORK_ETHERNET WiFi.onEvent(WiFiEvent); @@ -469,7 +466,7 @@ void push_i2s_data(const uint8_t *const payload, size_t len) { if (xEventGroupGetBits(audioGroup) != PLAY) { - Serial.println("Send PlayBytesEvent"); + publishDebug("Send PlayBytesEvent"); send_event(PlayBytesEvent()); } vTaskDelay(pdMS_TO_TICKS(50)); @@ -493,10 +490,10 @@ void handle_playBytes(const std::string& topicstr, uint8_t *payload, size_t len, bitDepth = Message.BitsPerSample; offset = Message.DataStart; - Serial.printf("Samplerate: %d, Channels: %d, Format: %d, Bits per Sample: %d, Start: %d\r\n", sampleRate, numChannels, (int)Message.Format, bitDepth, offset); + char message[100]; + snprintf(message, 100, "Samplerate: %d, Channels: %d, Format: %d, Bits per Sample: %d, Start: %d", sampleRate, numChannels, (int)Message.Format, bitDepth, offset); + publishDebug(message); queueDelay = (sampleRate * numChannels * bitDepth) / 1000; - //delay *= 2; - //Serial.printf("Delay %d\n", (int)queueDelay); } push_i2s_data((uint8_t *)&payload[offset], len - offset); @@ -507,7 +504,7 @@ void handle_playBytes(const std::string& topicstr, uint8_t *payload, size_t len, //At the end, make sure to start play in case the buffer is not full yet if (!audioData.isEmpty() && xEventGroupGetBits(audioGroup) != PLAY) { - Serial.println("Send PlayBytesEvent"); + publishDebug("Send PlayBytesEvent"); send_event(PlayBytesEvent()); } @@ -741,7 +738,9 @@ void onMqttMessage(char *topic, char *payload, AsyncMqttClientMessageProperties } else { - Serial.printf("Unhandled partial message received, topic '%s'\r\n", topic); + char message[100]; + snprintf(message, 100, "Unhandled partial message received, topic '%s'", topic); + publishDebug(message); } } } @@ -774,7 +773,9 @@ void I2Stask(void *p) { { if (!audioData.pop(data[i])) { - Serial.printf("Buffer underflow %d %ld\r\n", played + i, message_size); + char message[100]; + snprintf(message, 100, "Buffer underflow %d %ld", played + i, message_size); + publishDebug(message); vTaskDelay(60); bytes_to_write = (i)*2; } @@ -792,7 +793,9 @@ void I2Stask(void *p) { bytes_written = bytes_to_write; } if (bytes_written != bytes_to_write) { - Serial.printf("Bytes to write %d, but bytes written %d\r\n",bytes_to_write,bytes_written); + char message[100]; + snprintf(message, 100, "Bytes to write %d, but bytes written %d", bytes_to_write, bytes_written); + publishDebug(message); } } } @@ -801,8 +804,9 @@ void I2Stask(void *p) { device->muteOutput(true); xSemaphoreGive(wbSemaphore); audioData.clear(); - Serial.println("Done"); - Serial.println("Send StreamAudioEvent"); + + publishDebug("Done"); + publishDebug("Send StreamAudioEvent"); send_event(StreamAudioEvent()); } if (xEventGroupGetBits(audioGroup) == STREAM && !config.mute_input) { diff --git a/PlatformIO/src/devices/MatrixVoice.hpp b/PlatformIO/src/devices/MatrixVoice.hpp index f92bff3..6542ff3 100644 --- a/PlatformIO/src/devices/MatrixVoice.hpp +++ b/PlatformIO/src/devices/MatrixVoice.hpp @@ -9,9 +9,6 @@ #include "voice_memory_map.h" #include "wishbone_bus.h" #include -extern "C" { - #include "speex_resampler.h" -} // This is used to be able to change brightness, while keeping the colors appear // the same Called gamma correction, check this @@ -36,9 +33,6 @@ const uint8_t PROGMEM gamma8[] = { 215, 218, 220, 223, 225, 228, 231, 233, 236, 239, 241, 244, 247, 249, 252, 255}; -int err; -SpeexResamplerState *resampler = speex_resampler_init(1, 16000, 16000, 0, &err); - class MatrixVoice : public Device { public: @@ -74,19 +68,25 @@ class MatrixVoice : public Device void interleave(const int16_t * in_L, const int16_t * in_R, int16_t * out, const size_t num_samples); bool FIFOFlush(); void updateColors(int colors, bool usePulse); + void SetPCMSamplingFrequency(uint16_t PCM_constant); uint16_t GetFIFOStatus(); - uint32_t PCM_sampling_frequency = 16000; int fifoSize = 4096; int sampleRate, bitDepth, numChannels; int brightness, pulse = 15; float sample_time = 1.0 / 16000; uint32_t spiLength = 1024; int sleep = int(spiLength * sample_time * 1000); - int count = 0; int position = 0; long currentMillis, startMillis; bool ledsOn = true; bool directionDown = false; + + std::map FrequencyMap = { + {8000, 975}, + {16000, 492 }, + {22050, 355 }, + {44100, 177 } + }; }; MatrixVoice::MatrixVoice() @@ -219,30 +219,31 @@ void MatrixVoice::ampOutput(int output) { wb.SpiWrite(matrix_hal::kConfBaseAddress+11,(const uint8_t *)(&output), sizeof(uint16_t)); }; +void MatrixVoice::SetPCMSamplingFrequency(uint16_t PCM_constant) { + wb.SpiWrite(matrix_hal::kConfBaseAddress+9, (const uint8_t *)(&PCM_constant), sizeof(uint16_t)); +} + void MatrixVoice::setWriteMode(int sampleRate, int bitDepth, int numChannels) { MatrixVoice::sampleRate = sampleRate; MatrixVoice::bitDepth = bitDepth; MatrixVoice::numChannels = numChannels; FIFOFlush(); - speex_resampler_set_rate(resampler,sampleRate,16000); - speex_resampler_skip_zeros(resampler); - switch (sampleRate) - { - case 22050: - writeSize = 1411; - break; - case 44100: - writeSize = 2823; - break; - default: - writeSize = 1024; - break; + if (sampleRate == 8000 || sampleRate == 16000 || sampleRate == 22050 || sampleRate == 44100 ) { + if (sampleRate == 44100 && numChannels == 2) { + //Strange issue with 44100 stereo. When using 177 that output is very bad + //When using 220, output is ok but a tad to slow. + SetPCMSamplingFrequency(220); + } else { + SetPCMSamplingFrequency(FrequencyMap[sampleRate]); + } } + + sample_time = 1.0 / sampleRate; + sleep = int(spiLength * sample_time * 1000); + writeSize = spiLength; if (numChannels == 1) { writeSize = writeSize / sizeof(uint16_t); } - count = 0; - spiLength = 1024; }; bool MatrixVoice::readAudio(uint8_t *data, size_t size) { @@ -265,174 +266,24 @@ void MatrixVoice::writeAudio(uint8_t *data, size_t inputLength, size_t *bytes_wr int16_t mono[monoLength]; int16_t stereo[inputLength]; for (int i = 0; i < inputLength; i += 2) { - mono[i/2] = ((data[i] & 0xff) | (data[i + 1] << 8)); + mono[i/2] = ((data[i] & 0xff) | (data[i + 1] << 8)); } interleave(mono, mono, stereo, monoLength); for (int i = 0; i < inputLength; i++) { - output[i] = stereo[i]; + output[i] = stereo[i]; } } else { for (int i = 0; i < inputLength; i += 2) { - output[i/2] = ((data[i] & 0xff) | (data[i + 1] << 8)); - } - } - - //At this point, we always have a stereo audio message, at whatever rate. - //The length of the bytes is dependant on the input sampleRate. - //1024 with 16000Hz, 2822 for 441000. (44100/16000) * 1024 = 2822 - //We might need to resample of the inputRate is higher than 16000 - uint16_t spiSamples[spiLength]; - if (sampleRate > 16000) { - int16_t resampled[spiLength]; - //Serial.printf("before resample %d, %d\r\n", spiLength, outputLength); - speex_resampler_process_interleaved_int(resampler, output, &outputLength, resampled, &spiLength); - //Serial.printf("after resample %d, %d\r\n", spiLength, outputLength); - for (int i = 0; i < spiLength; i++) { - spiSamples[i] = resampled[i]; - } - } else { - for (int i = 0; i < spiLength; i++) { - spiSamples[i] = output[i]; + output[i/2] = ((data[i] & 0xff) | (data[i + 1] << 8)); } } - if (GetFIFOStatus() > fifoSize * 3 / 4) { std::this_thread::sleep_for(std::chrono::milliseconds(sleep)); } - Serial.printf("Write %d, %d, %d, %d\r\n", spiLength, sizeof(spiSamples), sleep, count); - wb.SpiWrite(matrix_hal::kDACBaseAddress, (const uint8_t *)spiSamples, spiLength); - count++; - - - // if (numChannels == 1) { - // // int16_t mono[size / sizeof(int16_t)]; - // // int16_t stereo[size]; - // // for (int i = 0; i < size; i += 2) { - // // mono[i/2] = ((data[i] & 0xff) | (data[i + 1] << 8)); - // // } - // // interleave(mono, mono, stereo, size / sizeof(int16_t)); - // // if (fifo_status > fifoSize * 3 / 4) { - // // int sleep = int(size * sizeof(int16_t) * sample_time * 1000); - // // std::this_thread::sleep_for(std::chrono::milliseconds(sleep)); - // // } - // // wb.SpiWrite(matrix_hal::kDACBaseAddress, (const uint8_t *)stereo, size * sizeof(int16_t)); - // } else { - // if (sampleRate == 44100) { - // uint32_t in_len = size / sizeof(int16_t); - // uint32_t out_len = 1024 / sizeof(int16_t); - // int16_t input[in_len]; - // int16_t output[out_len]; - // for (int i = 0; i < size; i += 2) { - // input[i/2] = ((data[i] & 0xff) | (data[i + 1] << 8)); - // } - // speex_resampler_process_interleaved_int(resampler, input, &in_len, output, &out_len); - // if (fifo_status > fifoSize * 3 / 4) { - // int sleep = int( ( out_len / sizeof(int16_t)) * sample_time * 1000); - // std::this_thread::sleep_for(std::chrono::milliseconds(sleep)); - // } - // wb.SpiWrite(matrix_hal::kDACBaseAddress, (const uint8_t *)output, out_len * sizeof(int16_t)); - // } else { - // if (fifo_status > fifoSize * 3 / 4) { - // int sleep = int(size * sample_time * 1000); - // std::this_thread::sleep_for(std::chrono::milliseconds(sleep)); - // } - // wb.SpiWrite(matrix_hal::kDACBaseAddress, (const uint8_t *)data, size); - // } - //} + wb.SpiWrite(matrix_hal::kDACBaseAddress, (const uint8_t *)output, outputLength); } -// void MatrixVoice::writeAudio(uint8_t *data, size_t size, size_t *bytes_written) { -// *bytes_written = size; -// float sample_time = 1.0 / 16000; -// uint16_t fifo_status = GetFIFOStatus(); -// uint32_t in_len = size / sizeof(int16_t); -// uint32_t out_len = size / sizeof(int16_t); -// int16_t input[in_len]; -// int16_t output[out_len]; -// int sleep = int(out_len * sizeof(int16_t) * sample_time * 1000); -// for (int i = 0; i < size; i += 2) { -// input[i/2] = ((data[i] & 0xff) | (data[i + 1] << 8)); -// } - -// if (numChannels == 1) { -// int16_t mono[size / sizeof(int16_t)]; -// for (int i = 0; i < size; i += 2) { -// mono[i/2] = ((data[i] & 0xff) | (data[i + 1] << 8)); -// } -// interleave(mono, mono, output, size / sizeof(int16_t)); -// } else { -// // uint32_t in_len; -// // uint32_t out_len; -// // in_len = size; -// // out_len = 512; -// // int16_t output[out_len]; -// // int16_t input[in_len]; -// // //Convert 8 bit to 16 bit -// // for (int i = 0; i < size; i += 2) { -// // input[i/2] = ((data[i] & 0xff) | (data[i + 1] << 8)); -// // } -// // speex_resampler_process_interleaved_int(resampler, input, &in_len, output, &out_len); - -// // if (fifo_status > fifoSize * 3 / 4) { -// // int sleep = int(size * sample_time * 1000); -// // std::this_thread::sleep_for(std::chrono::milliseconds(sleep)); -// // } -// // wb.SpiWrite(matrix_hal::kDACBaseAddress, (const uint8_t *)data, size); -// } - -// if (fifo_status > fifoSize * 3 / 4) { -// std::this_thread::sleep_for(std::chrono::milliseconds(sleep)); -// } -// // wb.SpiWrite(matrix_hal::kDACBaseAddress, (const uint8_t *)output, out_len * sizeof(int16_t)); -// } - - -// void MatrixVoice::writeAudioStereo1600OK(uint8_t *data, size_t size, size_t *bytes_written) { -// *bytes_written = size; -// float sample_time = 1.0 / 16000; -// int sleep = int(size * sample_time * 1000); -// uint16_t fifo_status = GetFIFOStatus(); - -// if (fifo_status > fifoSize * 3 / 4) { -// std::this_thread::sleep_for(std::chrono::milliseconds(sleep)); -// } -// wb.SpiWrite(matrix_hal::kDACBaseAddress, (const uint8_t *)data, size); -// } - -// void MatrixVoice::writeAudio(uint8_t *data, size_t size, size_t *bytes_written) { -// *bytes_written = size; -// uint32_t in_len; -// uint32_t out_len; -// in_len = size / sizeof(int16_t); -// out_len = size * (float)(16000 / sampleRate); -// int16_t output[out_len]; -// int16_t input[in_len]; -// //Convert 8 bit to 16 bit -// for (int i = 0; i < size; i += 2) { -// input[i/2] = ((data[i] & 0xff) | (data[i + 1] << 8)); -// } - -// if (MatrixVoice::numChannels == 2) { -// speex_resampler_process_interleaved_int(resampler, input, &in_len, output, &out_len); - -// //play it! -// playBytes(output, out_len); -// } else { -// speex_resampler_process_int(resampler, 0, input, &in_len, output, &out_len); -// int16_t stereo[out_len * sizeof(int16_t)]; -// int16_t mono[out_len]; - -// for (int i = 0; i < out_len; i++) { -// mono[i] = output[i]; -// } -// MatrixVoice::interleave(mono, mono, stereo, out_len); - -// //play it! -// MatrixVoice::playBytes(stereo, out_len * sizeof(int16_t)); -// } -// }; - void MatrixVoice::interleave(const int16_t * in_L, const int16_t * in_R, int16_t * out, const size_t num_samples) { for (size_t i = 0; i < num_samples; ++i) diff --git a/README.md b/README.md index 777efd3..370f27d 100644 --- a/README.md +++ b/README.md @@ -69,8 +69,7 @@ Restart the device by publishing {"passwordhash":"yourpasswordhash"} to SITEID/r ## Known issues -- Audio playback with sample rate higher than 44100 can lead to jitter due to network. Recommended is to use a samplerate of 16000 or 22050 -- Audio playback with matrix voice is not good, code needs to resample to 44100. 16000 Mono or Stereo should be fine +- Audio playback with sample rate higher than 44100 can lead to jitter due to network. Recommended is to use a samplerate of 16000 or 22050. 44100 stereo plays a bit too slow on the Matrix Voice due to unknown issue - Some settings (like the colors if update via MQTT) do not survive a reboot yet # Adding devices