From 3dc522e19b92602843cc89cc44f45983c71c1316 Mon Sep 17 00:00:00 2001 From: David Reid Date: Sat, 10 Jul 2021 15:54:52 +1000 Subject: [PATCH] Remove the Speex resampler. --- extras/speex_resampler/README.md | 32 - extras/speex_resampler/ma_speex_resampler.h | 135 -- extras/speex_resampler/thirdparty/arch.h | 219 --- extras/speex_resampler/thirdparty/resample.c | 1239 ----------------- .../speex_resampler/thirdparty/resample_sse.h | 128 -- .../thirdparty/speex_resampler.h | 343 ----- miniaudio.h | 333 +---- tools/audioconverter/audioconverter.c | 53 +- 8 files changed, 10 insertions(+), 2472 deletions(-) delete mode 100644 extras/speex_resampler/README.md delete mode 100644 extras/speex_resampler/ma_speex_resampler.h delete mode 100644 extras/speex_resampler/thirdparty/arch.h delete mode 100644 extras/speex_resampler/thirdparty/resample.c delete mode 100644 extras/speex_resampler/thirdparty/resample_sse.h delete mode 100644 extras/speex_resampler/thirdparty/speex_resampler.h diff --git a/extras/speex_resampler/README.md b/extras/speex_resampler/README.md deleted file mode 100644 index 31555a3e..00000000 --- a/extras/speex_resampler/README.md +++ /dev/null @@ -1,32 +0,0 @@ -This code in the `thirdparty` directory is taken from opus-tools (https://github.com/xiph/opus-tools). Note -that unlike miniaudio, this code is _not_ public domain. The opus-tools license is below: - -``` -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. - -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 COPYRIGHT OWNER -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. -``` - -Note that miniaudio does not use any of this code by default and is strictly opt-in. While miniaudio reproduces -this license text in it's source redistributions (in this file, and in each source file), it does not have any -control over binary distributions. When opting-in to use the Speex resampler you will need to consider this if -you redistribute a binary. diff --git a/extras/speex_resampler/ma_speex_resampler.h b/extras/speex_resampler/ma_speex_resampler.h deleted file mode 100644 index a80b6d32..00000000 --- a/extras/speex_resampler/ma_speex_resampler.h +++ /dev/null @@ -1,135 +0,0 @@ - -#ifndef ma_speex_resampler_h -#define ma_speex_resampler_h - -#define OUTSIDE_SPEEX -#define RANDOM_PREFIX ma_speex -#include "thirdparty/speex_resampler.h" - -#if defined(_MSC_VER) - typedef unsigned __int64 spx_uint64_t; -#else - #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) - #pragma GCC diagnostic push - #pragma GCC diagnostic ignored "-Wlong-long" - #if defined(__clang__) - #pragma GCC diagnostic ignored "-Wc++11-long-long" - #endif - #endif - typedef unsigned long long spx_uint64_t; - #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) - #pragma GCC diagnostic pop - #endif -#endif - -int ma_speex_resampler_get_required_input_frame_count(const SpeexResamplerState* st, spx_uint64_t out_len, spx_uint64_t* in_len); -int ma_speex_resampler_get_expected_output_frame_count(const SpeexResamplerState* st, spx_uint64_t in_len, spx_uint64_t* out_len); - -#endif /* ma_speex_resampler_h */ - -#if defined(MINIAUDIO_SPEEX_RESAMPLER_IMPLEMENTATION) -/* The Speex resampler uses "inline", which is not defined for C89. We need to define it here. */ -#if !defined(__cplusplus) - #if defined(__GNUC__) && !defined(_MSC_VER) - #if defined(__STRICT_ANSI__) - #if !defined(inline) - #define inline __inline__ __attribute__((always_inline)) - #define MA_SPEEX_INLINE_DEFINED - #endif - #endif - #endif - #if defined(_MSC_VER) && _MSC_VER <= 1400 /* 1400 = Visual Studio 2005 */ - #define inline _inline - #define MA_SPEEX_INLINE_DEFINED - #endif -#endif - -#if defined(_MSC_VER) && !defined(__clang__) - #pragma warning(push) - #pragma warning(disable:4244) /* conversion from 'x' to 'y', possible loss of data */ - #pragma warning(disable:4018) /* signed/unsigned mismatch */ - #pragma warning(disable:4706) /* assignment within conditional expression */ -#elif defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) - #pragma GCC diagnostic push - #pragma GCC diagnostic ignored "-Wsign-compare" /* comparison between signed and unsigned integer expressions */ -#endif -#include "thirdparty/resample.c" -#if defined(_MSC_VER) && !defined(__clang__) - #pragma warning(pop) -#elif defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) - #pragma GCC diagnostic pop -#endif - -#if defined(MA_SPEEX_INLINE_DEFINED) - #undef inline - #undef MA_SPEEX_INLINE_DEFINED -#endif - -EXPORT int ma_speex_resampler_get_required_input_frame_count(const SpeexResamplerState* st, spx_uint64_t out_len, spx_uint64_t* in_len) -{ - spx_uint64_t count; - - if (st == NULL || in_len == NULL) { - return RESAMPLER_ERR_INVALID_ARG; - } - - *in_len = 0; - - if (out_len == 0) { - return RESAMPLER_ERR_SUCCESS; /* Nothing to do. */ - } - - /* miniaudio only uses interleaved APIs so we can safely just use channel index 0 for the calculations. */ - if (st->nb_channels == 0) { - return RESAMPLER_ERR_BAD_STATE; - } - - count = out_len * st->int_advance; - count += (st->samp_frac_num[0] + (out_len * st->frac_advance)) / st->den_rate; - - *in_len = count; - - return RESAMPLER_ERR_SUCCESS; -} - -EXPORT int ma_speex_resampler_get_expected_output_frame_count(const SpeexResamplerState* st, spx_uint64_t in_len, spx_uint64_t* out_len) -{ - spx_uint64_t count; - spx_uint64_t last_sample; - spx_uint32_t samp_frac_num; - - if (st == NULL || out_len == NULL) { - return RESAMPLER_ERR_INVALID_ARG; - } - - *out_len = 0; - - if (out_len == 0) { - return RESAMPLER_ERR_SUCCESS; /* Nothing to do. */ - } - - /* miniaudio only uses interleaved APIs so we can safely just use channel index 0 for the calculations. */ - if (st->nb_channels == 0) { - return RESAMPLER_ERR_BAD_STATE; - } - - count = 0; - last_sample = st->last_sample[0]; - samp_frac_num = st->samp_frac_num[0]; - - while (!(last_sample >= in_len)) { - count += 1; - - last_sample += st->int_advance; - samp_frac_num += st->frac_advance; - if (samp_frac_num >= st->den_rate) { - samp_frac_num -= st->den_rate; - last_sample += 1; - } - } - - *out_len = count; - - return RESAMPLER_ERR_SUCCESS; -} -#endif diff --git a/extras/speex_resampler/thirdparty/arch.h b/extras/speex_resampler/thirdparty/arch.h deleted file mode 100644 index 225d7276..00000000 --- a/extras/speex_resampler/thirdparty/arch.h +++ /dev/null @@ -1,219 +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 - -#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 - -#else - -#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 - -#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/extras/speex_resampler/thirdparty/resample.c b/extras/speex_resampler/thirdparty/resample.c deleted file mode 100644 index 04338277..00000000 --- a/extras/speex_resampler/thirdparty/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/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 - -#if defined(__SSE__) && !defined(FIXED_POINT) -#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/extras/speex_resampler/thirdparty/resample_sse.h b/extras/speex_resampler/thirdparty/resample_sse.h deleted file mode 100644 index a0c7a204..00000000 --- a/extras/speex_resampler/thirdparty/resample_sse.h +++ /dev/null @@ -1,128 +0,0 @@ -/* Copyright (C) 2007-2008 Jean-Marc Valin - * Copyright (C) 2008 Thorvald Natvig - */ -/** - @file resample_sse.h - @brief Resampler functions (SSE version) -*/ -/* - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -*/ - -#include - -#define OVERRIDE_INNER_PRODUCT_SINGLE -static inline float inner_product_single(const float *a, const float *b, unsigned int len) -{ - int i; - float ret; - __m128 sum = _mm_setzero_ps(); - for (i=0;i -#define OVERRIDE_INNER_PRODUCT_DOUBLE - -static inline double inner_product_double(const float *a, const float *b, unsigned int len) -{ - int i; - double ret; - __m128d sum = _mm_setzero_pd(); - __m128 t; - for (i=0;iresampling.algorithm; converterConfig.resampling.linear.lpfOrder = pDevice->resampling.linear.lpfOrder; - converterConfig.resampling.speex.quality = pDevice->resampling.speex.quality; result = ma_data_converter_init(&converterConfig, &pDevice->capture.converter); if (result != MA_SUCCESS) { @@ -32643,7 +32586,6 @@ static ma_result ma_device__post_init_setup(ma_device* pDevice, ma_device_type d converterConfig.resampling.allowDynamicSampleRate = MA_FALSE; converterConfig.resampling.algorithm = pDevice->resampling.algorithm; converterConfig.resampling.linear.lpfOrder = pDevice->resampling.linear.lpfOrder; - converterConfig.resampling.speex.quality = pDevice->resampling.speex.quality; result = ma_data_converter_init(&converterConfig, &pDevice->playback.converter); if (result != MA_SUCCESS) { @@ -33350,7 +33292,6 @@ MA_API ma_device_config ma_device_config_init(ma_device_type deviceType) /* Resampling defaults. We must never use the Speex backend by default because it uses licensed third party code. */ config.resampling.algorithm = ma_resample_algorithm_linear; config.resampling.linear.lpfOrder = ma_min(MA_DEFAULT_RESAMPLER_LPF_ORDER, MA_MAX_FILTER_ORDER); - config.resampling.speex.quality = 3; return config; } @@ -33427,7 +33368,6 @@ MA_API ma_result ma_device_init(ma_context* pContext, const ma_device_config* pC pDevice->sampleRate = pConfig->sampleRate; pDevice->resampling.algorithm = pConfig->resampling.algorithm; pDevice->resampling.linear.lpfOrder = pConfig->resampling.linear.lpfOrder; - pDevice->resampling.speex.quality = pConfig->resampling.speex.quality; pDevice->capture.shareMode = pConfig->capture.shareMode; pDevice->capture.format = pConfig->capture.format; @@ -39361,24 +39301,6 @@ MA_API ma_uint64 ma_linear_resampler_get_output_latency(const ma_linear_resample } -#if defined(ma_speex_resampler_h) -#define MA_HAS_SPEEX_RESAMPLER - -static ma_result ma_result_from_speex_err(int err) -{ - switch (err) - { - case RESAMPLER_ERR_SUCCESS: return MA_SUCCESS; - case RESAMPLER_ERR_ALLOC_FAILED: return MA_OUT_OF_MEMORY; - case RESAMPLER_ERR_BAD_STATE: return MA_ERROR; - case RESAMPLER_ERR_INVALID_ARG: return MA_INVALID_ARGS; - case RESAMPLER_ERR_PTR_OVERLAP: return MA_INVALID_ARGS; - case RESAMPLER_ERR_OVERFLOW: return MA_ERROR; - default: return MA_ERROR; - } -} -#endif /* ma_speex_resampler_h */ - MA_API ma_resampler_config ma_resampler_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut, ma_resample_algorithm algorithm) { ma_resampler_config config; @@ -39394,9 +39316,6 @@ MA_API ma_resampler_config ma_resampler_config_init(ma_format format, ma_uint32 config.linear.lpfOrder = ma_min(MA_DEFAULT_RESAMPLER_LPF_ORDER, MA_MAX_FILTER_ORDER); config.linear.lpfNyquistFactor = 1; - /* Speex. */ - config.speex.quality = 3; /* Cannot leave this as 0 as that is actually a valid value for Speex resampling quality. */ - return config; } @@ -39435,20 +39354,6 @@ MA_API ma_result ma_resampler_init(const ma_resampler_config* pConfig, ma_resamp } } break; - case ma_resample_algorithm_speex: - { - #if defined(MA_HAS_SPEEX_RESAMPLER) - int speexErr; - pResampler->state.speex.pSpeexResamplerState = speex_resampler_init(pConfig->channels, pConfig->sampleRateIn, pConfig->sampleRateOut, pConfig->speex.quality, &speexErr); - if (pResampler->state.speex.pSpeexResamplerState == NULL) { - return ma_result_from_speex_err(speexErr); - } - #else - /* Speex resampler not available. */ - return MA_NO_BACKEND; - #endif - } break; - default: return MA_INVALID_ARGS; } @@ -39464,12 +39369,6 @@ MA_API void ma_resampler_uninit(ma_resampler* pResampler) if (pResampler->config.algorithm == ma_resample_algorithm_linear) { ma_linear_resampler_uninit(&pResampler->state.linear); } - -#if defined(MA_HAS_SPEEX_RESAMPLER) - if (pResampler->config.algorithm == ma_resample_algorithm_speex) { - speex_resampler_destroy((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState); - } -#endif } static ma_result ma_resampler_process_pcm_frames__read__linear(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut) @@ -39477,76 +39376,6 @@ static ma_result ma_resampler_process_pcm_frames__read__linear(ma_resampler* pRe return ma_linear_resampler_process_pcm_frames(&pResampler->state.linear, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut); } -#if defined(MA_HAS_SPEEX_RESAMPLER) -static ma_result ma_resampler_process_pcm_frames__read__speex(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut) -{ - int speexErr; - ma_uint64 frameCountOut; - ma_uint64 frameCountIn; - ma_uint64 framesProcessedOut; - ma_uint64 framesProcessedIn; - unsigned int framesPerIteration = UINT_MAX; - - MA_ASSERT(pResampler != NULL); - MA_ASSERT(pFramesOut != NULL); - MA_ASSERT(pFrameCountOut != NULL); - MA_ASSERT(pFrameCountIn != NULL); - - /* - Reading from the Speex resampler requires a bit of dancing around for a few reasons. The first thing is that it's frame counts - are in unsigned int's whereas ours is in ma_uint64. We therefore need to run the conversion in a loop. The other, more complicated - problem, is that we need to keep track of the input time, similar to what we do with the linear resampler. The reason we need to - do this is for ma_resampler_get_required_input_frame_count() and ma_resampler_get_expected_output_frame_count(). - */ - frameCountOut = *pFrameCountOut; - frameCountIn = *pFrameCountIn; - framesProcessedOut = 0; - framesProcessedIn = 0; - - while (framesProcessedOut < frameCountOut && framesProcessedIn < frameCountIn) { - unsigned int frameCountInThisIteration; - unsigned int frameCountOutThisIteration; - const void* pFramesInThisIteration; - void* pFramesOutThisIteration; - - frameCountInThisIteration = framesPerIteration; - if ((ma_uint64)frameCountInThisIteration > (frameCountIn - framesProcessedIn)) { - frameCountInThisIteration = (unsigned int)(frameCountIn - framesProcessedIn); - } - - frameCountOutThisIteration = framesPerIteration; - if ((ma_uint64)frameCountOutThisIteration > (frameCountOut - framesProcessedOut)) { - frameCountOutThisIteration = (unsigned int)(frameCountOut - framesProcessedOut); - } - - pFramesInThisIteration = ma_offset_ptr(pFramesIn, framesProcessedIn * ma_get_bytes_per_frame(pResampler->config.format, pResampler->config.channels)); - pFramesOutThisIteration = ma_offset_ptr(pFramesOut, framesProcessedOut * ma_get_bytes_per_frame(pResampler->config.format, pResampler->config.channels)); - - if (pResampler->config.format == ma_format_f32) { - speexErr = speex_resampler_process_interleaved_float((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState, (const float*)pFramesInThisIteration, &frameCountInThisIteration, (float*)pFramesOutThisIteration, &frameCountOutThisIteration); - } else if (pResampler->config.format == ma_format_s16) { - speexErr = speex_resampler_process_interleaved_int((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState, (const spx_int16_t*)pFramesInThisIteration, &frameCountInThisIteration, (spx_int16_t*)pFramesOutThisIteration, &frameCountOutThisIteration); - } else { - /* Format not supported. Should never get here. */ - MA_ASSERT(MA_FALSE); - return MA_INVALID_OPERATION; - } - - if (speexErr != RESAMPLER_ERR_SUCCESS) { - return ma_result_from_speex_err(speexErr); - } - - framesProcessedIn += frameCountInThisIteration; - framesProcessedOut += frameCountOutThisIteration; - } - - *pFrameCountOut = framesProcessedOut; - *pFrameCountIn = framesProcessedIn; - - return MA_SUCCESS; -} -#endif - static ma_result ma_resampler_process_pcm_frames__read(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut) { MA_ASSERT(pResampler != NULL); @@ -39569,15 +39398,6 @@ static ma_result ma_resampler_process_pcm_frames__read(ma_resampler* pResampler, return ma_resampler_process_pcm_frames__read__linear(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut); } - case ma_resample_algorithm_speex: - { - #if defined(MA_HAS_SPEEX_RESAMPLER) - return ma_resampler_process_pcm_frames__read__speex(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut); - #else - break; - #endif - } - default: break; } @@ -39595,81 +39415,6 @@ static ma_result ma_resampler_process_pcm_frames__seek__linear(ma_resampler* pRe return ma_linear_resampler_process_pcm_frames(&pResampler->state.linear, pFramesIn, pFrameCountIn, NULL, pFrameCountOut); } -#if defined(MA_HAS_SPEEX_RESAMPLER) -static ma_result ma_resampler_process_pcm_frames__seek__speex(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, ma_uint64* pFrameCountOut) -{ - /* The generic seek method is implemented in on top of ma_resampler_process_pcm_frames__read() by just processing into a dummy buffer. */ - float devnull[4096]; - ma_uint64 totalOutputFramesToProcess; - ma_uint64 totalOutputFramesProcessed; - ma_uint64 totalInputFramesProcessed; - ma_uint32 bpf; - ma_result result; - - MA_ASSERT(pResampler != NULL); - - totalOutputFramesProcessed = 0; - totalInputFramesProcessed = 0; - bpf = ma_get_bytes_per_frame(pResampler->config.format, pResampler->config.channels); - - if (pFrameCountOut != NULL) { - /* Seek by output frames. */ - totalOutputFramesToProcess = *pFrameCountOut; - } else { - /* Seek by input frames. */ - MA_ASSERT(pFrameCountIn != NULL); - totalOutputFramesToProcess = ma_resampler_get_expected_output_frame_count(pResampler, *pFrameCountIn); - } - - if (pFramesIn != NULL) { - /* Process input data. */ - MA_ASSERT(pFrameCountIn != NULL); - while (totalOutputFramesProcessed < totalOutputFramesToProcess && totalInputFramesProcessed < *pFrameCountIn) { - ma_uint64 inputFramesToProcessThisIteration = (*pFrameCountIn - totalInputFramesProcessed); - ma_uint64 outputFramesToProcessThisIteration = (totalOutputFramesToProcess - totalOutputFramesProcessed); - if (outputFramesToProcessThisIteration > sizeof(devnull) / bpf) { - outputFramesToProcessThisIteration = sizeof(devnull) / bpf; - } - - result = ma_resampler_process_pcm_frames__read(pResampler, ma_offset_ptr(pFramesIn, totalInputFramesProcessed*bpf), &inputFramesToProcessThisIteration, ma_offset_ptr(devnull, totalOutputFramesProcessed*bpf), &outputFramesToProcessThisIteration); - if (result != MA_SUCCESS) { - return result; - } - - totalOutputFramesProcessed += outputFramesToProcessThisIteration; - totalInputFramesProcessed += inputFramesToProcessThisIteration; - } - } else { - /* Don't process input data - just update timing and filter state as if zeroes were passed in. */ - while (totalOutputFramesProcessed < totalOutputFramesToProcess) { - ma_uint64 inputFramesToProcessThisIteration = 16384; - ma_uint64 outputFramesToProcessThisIteration = (totalOutputFramesToProcess - totalOutputFramesProcessed); - if (outputFramesToProcessThisIteration > sizeof(devnull) / bpf) { - outputFramesToProcessThisIteration = sizeof(devnull) / bpf; - } - - result = ma_resampler_process_pcm_frames__read(pResampler, NULL, &inputFramesToProcessThisIteration, ma_offset_ptr(devnull, totalOutputFramesProcessed*bpf), &outputFramesToProcessThisIteration); - if (result != MA_SUCCESS) { - return result; - } - - totalOutputFramesProcessed += outputFramesToProcessThisIteration; - totalInputFramesProcessed += inputFramesToProcessThisIteration; - } - } - - - if (pFrameCountIn != NULL) { - *pFrameCountIn = totalInputFramesProcessed; - } - if (pFrameCountOut != NULL) { - *pFrameCountOut = totalOutputFramesProcessed; - } - - return MA_SUCCESS; -} -#endif - static ma_result ma_resampler_process_pcm_frames__seek(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, ma_uint64* pFrameCountOut) { MA_ASSERT(pResampler != NULL); @@ -39681,15 +39426,6 @@ static ma_result ma_resampler_process_pcm_frames__seek(ma_resampler* pResampler, return ma_resampler_process_pcm_frames__seek__linear(pResampler, pFramesIn, pFrameCountIn, pFrameCountOut); } break; - case ma_resample_algorithm_speex: - { - #if defined(MA_HAS_SPEEX_RESAMPLER) - return ma_resampler_process_pcm_frames__seek__speex(pResampler, pFramesIn, pFrameCountIn, pFrameCountOut); - #else - break; - #endif - }; - default: break; } @@ -39738,15 +39474,6 @@ MA_API ma_result ma_resampler_set_rate(ma_resampler* pResampler, ma_uint32 sampl return ma_linear_resampler_set_rate(&pResampler->state.linear, sampleRateIn, sampleRateOut); } break; - case ma_resample_algorithm_speex: - { - #if defined(MA_HAS_SPEEX_RESAMPLER) - return ma_result_from_speex_err(speex_resampler_set_rate((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState, sampleRateIn, sampleRateOut)); - #else - break; - #endif - }; - default: break; } @@ -39798,21 +39525,6 @@ MA_API ma_uint64 ma_resampler_get_required_input_frame_count(const ma_resampler* return ma_linear_resampler_get_required_input_frame_count(&pResampler->state.linear, outputFrameCount); } - case ma_resample_algorithm_speex: - { - #if defined(MA_HAS_SPEEX_RESAMPLER) - spx_uint64_t count; - int speexErr = ma_speex_resampler_get_required_input_frame_count((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState, outputFrameCount, &count); - if (speexErr != RESAMPLER_ERR_SUCCESS) { - return 0; - } - - return (ma_uint64)count; - #else - break; - #endif - } - default: break; } @@ -39838,21 +39550,6 @@ MA_API ma_uint64 ma_resampler_get_expected_output_frame_count(const ma_resampler return ma_linear_resampler_get_expected_output_frame_count(&pResampler->state.linear, inputFrameCount); } - case ma_resample_algorithm_speex: - { - #if defined(MA_HAS_SPEEX_RESAMPLER) - spx_uint64_t count; - int speexErr = ma_speex_resampler_get_expected_output_frame_count((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState, inputFrameCount, &count); - if (speexErr != RESAMPLER_ERR_SUCCESS) { - return 0; - } - - return (ma_uint64)count; - #else - break; - #endif - } - default: break; } @@ -39874,15 +39571,6 @@ MA_API ma_uint64 ma_resampler_get_input_latency(const ma_resampler* pResampler) return ma_linear_resampler_get_input_latency(&pResampler->state.linear); } - case ma_resample_algorithm_speex: - { - #if defined(MA_HAS_SPEEX_RESAMPLER) - return (ma_uint64)ma_speex_resampler_get_input_latency((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState); - #else - break; - #endif - } - default: break; } @@ -39904,15 +39592,6 @@ MA_API ma_uint64 ma_resampler_get_output_latency(const ma_resampler* pResampler) return ma_linear_resampler_get_output_latency(&pResampler->state.linear); } - case ma_resample_algorithm_speex: - { - #if defined(MA_HAS_SPEEX_RESAMPLER) - return (ma_uint64)ma_speex_resampler_get_output_latency((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState); - #else - break; - #endif - } - default: break; } @@ -40825,9 +40504,6 @@ MA_API ma_data_converter_config ma_data_converter_config_init_default() config.resampling.linear.lpfOrder = 1; config.resampling.linear.lpfNyquistFactor = 1; - /* Speex resampling defaults. */ - config.resampling.speex.quality = 3; - return config; } @@ -40928,7 +40604,6 @@ MA_API ma_result ma_data_converter_init(const ma_data_converter_config* pConfig, resamplerConfig = ma_resampler_config_init(midFormat, resamplerChannels, pConverter->config.sampleRateIn, pConverter->config.sampleRateOut, pConverter->config.resampling.algorithm); resamplerConfig.linear.lpfOrder = pConverter->config.resampling.linear.lpfOrder; resamplerConfig.linear.lpfNyquistFactor = pConverter->config.resampling.linear.lpfNyquistFactor; - resamplerConfig.speex.quality = pConverter->config.resampling.speex.quality; result = ma_resampler_init(&resamplerConfig, &pConverter->resampler); if (result != MA_SUCCESS) { @@ -46793,7 +46468,6 @@ MA_API ma_decoder_config ma_decoder_config_init(ma_format outputFormat, ma_uint3 config.sampleRate = outputSampleRate; config.resampling.algorithm = ma_resample_algorithm_linear; config.resampling.linear.lpfOrder = ma_min(MA_DEFAULT_RESAMPLER_LPF_ORDER, MA_MAX_FILTER_ORDER); - config.resampling.speex.quality = 3; config.encodingFormat = ma_encoding_format_unknown; /* Note that we are intentionally leaving the channel map empty here which will cause the default channel map to be used. */ @@ -46885,7 +46559,6 @@ static ma_result ma_decoder__init_data_converter(ma_decoder* pDecoder, const ma_ converterConfig.resampling.allowDynamicSampleRate = MA_FALSE; /* Never allow dynamic sample rate conversion. Setting this to true will disable passthrough optimizations. */ converterConfig.resampling.algorithm = pConfig->resampling.algorithm; converterConfig.resampling.linear.lpfOrder = pConfig->resampling.linear.lpfOrder; - converterConfig.resampling.speex.quality = pConfig->resampling.speex.quality; return ma_data_converter_init(&converterConfig, &pDecoder->converter); } diff --git a/tools/audioconverter/audioconverter.c b/tools/audioconverter/audioconverter.c index 1ec4a015..be39b470 100644 --- a/tools/audioconverter/audioconverter.c +++ b/tools/audioconverter/audioconverter.c @@ -4,25 +4,12 @@ USAGE: audioconverter [input file] [output file] [format] [channels] [rate] EXAMPLES: audioconverter my_file.flac my_file.wav audioconverter my_file.flac my_file.wav f32 44100 linear --linear-order 8 - audioconverter my_file.flac my_file.wav s16 2 44100 speex --speex-quality 10 -*/ - -/* -Note about Speex resampling. If you decide to enable the Speex resampler with ENABLE_SPEEX, this program will use licensed third party code. If you compile and -redistribute this program you need to include a copy of the license which can be found at https://github.com/xiph/opus-tools/blob/master/COPYING. You can also -find a copy of this text in extras/speex_resampler/README.md in the miniaudio repository. */ #define _CRT_SECURE_NO_WARNINGS /* For stb_vorbis' usage of fopen() instead of fopen_s(). */ #define STB_VORBIS_HEADER_ONLY #include "../../extras/stb_vorbis.c" /* Enables Vorbis decoding. */ -/* Enable Speex resampling, but only if requested on the command line at build time. */ -#if defined(ENABLE_SPEEX) - #define MINIAUDIO_SPEEX_RESAMPLER_IMPLEMENTATION - #include "../../extras/speex_resampler/ma_speex_resampler.h" -#endif - #define MA_NO_DEVICE_IO #define MA_NO_THREADING #define MINIAUDIO_IMPLEMENTATION @@ -44,7 +31,6 @@ void print_usage() printf("\n"); printf("PARAMETERS:\n"); printf(" --linear-order [0..%d]\n", MA_MAX_FILTER_ORDER); - printf(" --speex-quality [0..10]\n"); } ma_result do_conversion(ma_decoder* pDecoder, ma_encoder* pEncoder) @@ -64,9 +50,9 @@ ma_result do_conversion(ma_decoder* pDecoder, ma_encoder* pEncoder) ma_uint64 framesToReadThisIteration; framesToReadThisIteration = sizeof(pRawData) / ma_get_bytes_per_frame(pDecoder->outputFormat, pDecoder->outputChannels); - framesReadThisIteration = ma_decoder_read_pcm_frames(pDecoder, pRawData, framesToReadThisIteration); - if (framesReadThisIteration == 0) { - break; /* Reached the end. */ + result = ma_decoder_read_pcm_frames(pDecoder, pRawData, framesToReadThisIteration, &framesReadThisIteration); + if (result != MA_SUCCESS) { + break; /* Reached the end, or an error occurred. */ } /* At this point we have the raw data from the decoder. We now just need to write it to the encoder. */ @@ -159,8 +145,6 @@ ma_bool32 try_parse_resample_algorithm(const char* str, ma_resample_algorithm* p /* */ if (strcmp(str, "linear") == 0) { algorithm = ma_resample_algorithm_linear; - } else if (strcmp(str, "speex") == 0) { - algorithm = ma_resample_algorithm_speex; } else { return MA_FALSE; /* Not a valid algorithm */ } @@ -179,13 +163,12 @@ int main(int argc, char** argv) ma_decoder decoder; ma_encoder_config encoderConfig; ma_encoder encoder; - ma_resource_format outputResourceFormat; + ma_encoding_format outputEncodingFormat; ma_format format = ma_format_unknown; ma_uint32 channels = 0; ma_uint32 rate = 0; ma_resample_algorithm resampleAlgorithm; ma_uint32 linearOrder = 8; - ma_uint32 speexQuality = 3; int iarg; const char* pOutputFilePath; @@ -204,12 +187,7 @@ int main(int argc, char** argv) return -1; } - /* Default to Speex if it's enabled. */ -#if defined(ENABLE_SPEEX) - resampleAlgorithm = ma_resample_algorithm_speex; -#else resampleAlgorithm = ma_resample_algorithm_linear; -#endif /* The fourth and fifth arguments can be a format and/or rate specifier. It doesn't matter which order they are in as we can identify them by whether or @@ -230,20 +208,6 @@ int main(int argc, char** argv) continue; } - if (strcmp(argv[iarg], "--speex-quality") == 0) { - iarg += 1; - if (iarg >= argc) { - break; - } - - if (!try_parse_uint32_in_range(argv[iarg], &speexQuality, 0, 10)) { - printf("Expecting a number between 0 and 10 for --speex-quality.\n"); - return -1; - } - - continue; - } - if (try_parse_resample_algorithm(argv[iarg], &resampleAlgorithm)) { continue; } @@ -268,9 +232,6 @@ int main(int argc, char** argv) decoderConfig = ma_decoder_config_init(format, channels, rate); decoderConfig.resampling.algorithm = resampleAlgorithm; decoderConfig.resampling.linear.lpfOrder = linearOrder; -#if defined(ENABLE_SPEEX) - decoderConfig.resampling.speex.quality = speexQuality; -#endif result = ma_decoder_init_file(argv[1], &decoderConfig, &decoder); if (result != MA_SUCCESS) { @@ -296,15 +257,15 @@ int main(int argc, char** argv) pOutputFilePath = argv[2]; - outputResourceFormat = ma_resource_format_wav; /* Wave by default in case we don't know the file extension. */ + outputEncodingFormat = ma_encoding_format_wav; /* Wave by default in case we don't know the file extension. */ if (ma_path_extension_equal(pOutputFilePath, "wav")) { - outputResourceFormat = ma_resource_format_wav; + outputEncodingFormat = ma_encoding_format_wav; } else { printf("Warning: Unknown file extension \"%s\". Encoding as WAV.\n", ma_path_extension(pOutputFilePath)); } /* Initialize the encoder for the output file. */ - encoderConfig = ma_encoder_config_init(ma_resource_format_wav, decoder.outputFormat, decoder.outputChannels, decoder.outputSampleRate); + encoderConfig = ma_encoder_config_init(outputEncodingFormat, decoder.outputFormat, decoder.outputChannels, decoder.outputSampleRate); result = ma_encoder_init_file(pOutputFilePath, &encoderConfig, &encoder); if (result != MA_SUCCESS) { ma_decoder_uninit(&decoder);