From ae148498649e38105607e9da2262fc276ffe2049 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Jo=C3=A3o=20Valverde?= Date: Thu, 15 Dec 2022 18:54:48 +0000 Subject: [PATCH] Windows: Use SpeexDSP binary package Remove bundled code and use vcpkg binary library instead. --- CMakeLists.txt | 17 +- cmake/modules/FindSpeexDSP.cmake | 40 +- cmakeconfig.h.in | 3 - docbook/release-notes.adoc | 3 + doxygen.cfg.in | 1 - packaging/nsis/CMakeLists.txt | 2 + packaging/wix/CMakeLists.txt | 2 + sharkd_session.c | 6 +- speexdsp/CMakeLists.txt | 15 - speexdsp/README.txt | 2 - speexdsp/arch.h | 237 ------ speexdsp/resample.c | 1211 ------------------------------ speexdsp/speex_resampler.h | 347 --------- speexdsp/stack_alloc.h | 115 --- tools/win-setup.ps1 | 2 + ui/logray/logray_main.cpp | 5 - ui/qt/main.cpp | 5 - ui/qt/rtp_audio_stream.cpp | 4 - ui/qt/utils/rtp_audio_file.h | 4 - 19 files changed, 44 insertions(+), 1977 deletions(-) delete mode 100644 speexdsp/CMakeLists.txt delete mode 100644 speexdsp/README.txt delete mode 100644 speexdsp/arch.h delete mode 100644 speexdsp/resample.c delete mode 100644 speexdsp/speex_resampler.h delete mode 100644 speexdsp/stack_alloc.h diff --git a/CMakeLists.txt b/CMakeLists.txt index 7abb22db86..8f4758ae64 100644 --- a/CMakeLists.txt +++ b/CMakeLists.txt @@ -1327,19 +1327,7 @@ find_package(DOXYGEN) # The SpeexDSP resampler is required iff building wireshark or sharkd. if(BUILD_wireshark OR BUILD_logray OR BUILD_sharkd) - # We don't provide a binary package for SpeexDSP in our repository. - # If using the repository don't bother searching for a system SpeexDSP - # installation and just use the bundled resampler code instead. - if (NOT USE_REPOSITORY) - find_package(SpeexDSP) - endif() - if(SpeexDSP_FOUND) - set(HAVE_SPEEXDSP 1) - else() - add_subdirectory(speexdsp) - set(SPEEXDSP_INCLUDE_DIRS "") - set(SPEEXDSP_LIBRARIES "speexresampler") - endif() + find_package(SpeexDSP REQUIRED) endif() # Generate the distribution tarball. @@ -2171,6 +2159,9 @@ if(USE_REPOSITORY) list (APPEND THIRD_PARTY_DLLS "${BROTLI_DLL_DIR}/${_dll}") endforeach(_dll) endif(BROTLI_FOUND) + if (SPEEXDSP_FOUND) + list (APPEND THIRD_PARTY_DLLS "${SPEEXDSP_DLL_DIR}/${SPEEXDSP_DLL}") + endif() # With libs downloaded to c:/wireshark-win64-libs this currently # (early 2018) expands to about 1900 characters. diff --git a/cmake/modules/FindSpeexDSP.cmake b/cmake/modules/FindSpeexDSP.cmake index 6c111111e5..cd6ada0c42 100644 --- a/cmake/modules/FindSpeexDSP.cmake +++ b/cmake/modules/FindSpeexDSP.cmake @@ -1,18 +1,26 @@ +# Find the system's SpeexDSP includes and library # -# - Find speexdsp libraries -# -# SPEEXDSP_INCLUDE_DIRS - where to find speexdsp headers. -# SPEEXDSP_LIBRARIES - List of libraries when using speexdsp. -# SPEEXDSP_FOUND - True if speexdsp is found. +# SPEEXDSP_INCLUDE_DIRS - where to find SpeexDSP headers +# SPEEXDSP_LIBRARIES - List of libraries when using SpeexDSP +# SPEEXDSP_FOUND - True if SpeexDSP found +# SPEEXDSP_DLL_DIR - (Windows) Path to the SpeexDSP DLL +# SPEEXDSP_DLL - (Windows) Name of the SpeexDSP DLL + +include(FindWSWinLibs) +FindWSWinLibs("speexdsp-.*" "SPEEXDSP_HINTS") + +if(NOT WIN32) + find_package(PkgConfig) + pkg_search_module(PC_SPEEXDSP speexdsp) +endif() -find_package(PkgConfig QUIET) -pkg_search_module(PC_SPEEXDSP QUIET speexdsp) find_path(SPEEXDSP_INCLUDE_DIR NAMES speex/speex_resampler.h HINTS ${PC_SPEEXDSP_INCLUDE_DIRS} + ${SPEEXDSP_HINTS}/include ) find_library(SPEEXDSP_LIBRARY @@ -20,16 +28,28 @@ find_library(SPEEXDSP_LIBRARY speexdsp HINTS ${PC_SPEEXDSP_LIBRARY_DIRS} + ${SPEEXDSP_HINTS}/lib ) include(FindPackageHandleStandardArgs) -find_package_handle_standard_args(SpeexDSP - REQUIRED_VARS SPEEXDSP_LIBRARY SPEEXDSP_INCLUDE_DIR - VERSION_VAR PC_SPEEXDSP_VERSION) +find_package_handle_standard_args(SpeexDSP DEFAULT_MSG SPEEXDSP_LIBRARY SPEEXDSP_INCLUDE_DIR) if(SPEEXDSP_FOUND) set(SPEEXDSP_LIBRARIES ${SPEEXDSP_LIBRARY}) set(SPEEXDSP_INCLUDE_DIRS ${SPEEXDSP_INCLUDE_DIR}) + if(WIN32) + set(SPEEXDSP_DLL_DIR "${SPEEXDSP_HINTS}/bin" + CACHE PATH "Path to SpeexDSP DLL" + ) + file(GLOB _speexdsp_dll RELATIVE "${SPEEXDSP_DLL_DIR}" + "${SPEEXDSP_DLL_DIR}/libspeexdsp.dll" + ) + set(SPEEXDSP_DLL ${_speexdsp_dll} + # We're storing filenames only. Should we use STRING instead? + CACHE FILEPATH "SpeexDSP DLL file name" + ) + mark_as_advanced(SPEEXDSP_DLL_DIR SPEEXDSP_DLL) + endif() else() set(SPEEXDSP_LIBRARIES) set(SPEEXDSP_INCLUDE_DIRS) diff --git a/cmakeconfig.h.in b/cmakeconfig.h.in index 9fbedea831..1b60bc23ad 100644 --- a/cmakeconfig.h.in +++ b/cmakeconfig.h.in @@ -235,9 +235,6 @@ /* Define to 1 if you have the opus library. */ #cmakedefine HAVE_OPUS 1 -/* Define to 1 if you have the speexdsp library. */ -#cmakedefine HAVE_SPEEXDSP 1 - /* Define to 1 if you have the lixbml2 library. */ #cmakedefine HAVE_LIBXML2 1 diff --git a/docbook/release-notes.adoc b/docbook/release-notes.adoc index 7730d5203a..7d3cf60f15 100644 --- a/docbook/release-notes.adoc +++ b/docbook/release-notes.adoc @@ -25,6 +25,9 @@ A new display filter feature for filtering raw bytes has been added. Display filter autocomplete is smarter about not suggesting invalid syntax. +The Windows build has a new SpeexDSP external dependency (https://www.speex.org). +The speex code that was previously bundled has been removed. + Many other improvements have been made. See the “New and Updated Features” section below for more details. diff --git a/doxygen.cfg.in b/doxygen.cfg.in index 8b73ab5137..4175d09647 100644 --- a/doxygen.cfg.in +++ b/doxygen.cfg.in @@ -837,7 +837,6 @@ INPUT = @DOXYGEN_INPUT_DIRECTORY@/wireshark.dox \ @DOXYGEN_INPUT_DIRECTORY@/extcap \ @DOXYGEN_INPUT_DIRECTORY@/plugins \ @DOXYGEN_INPUT_DIRECTORY@/randpkt_core \ - @DOXYGEN_INPUT_DIRECTORY@/speexdsp \ @DOXYGEN_INPUT_DIRECTORY@/ui \ @DOXYGEN_INPUT_DIRECTORY@/wiretap \ @DOXYGEN_INPUT_DIRECTORY@/writecap \ diff --git a/packaging/nsis/CMakeLists.txt b/packaging/nsis/CMakeLists.txt index 8d4b61513b..a6b2060ff6 100644 --- a/packaging/nsis/CMakeLists.txt +++ b/packaging/nsis/CMakeLists.txt @@ -177,6 +177,7 @@ if (BUILD_wireshark) ${LZ4_DLL} ${NGHTTP2_DLL} ${SBC_DLL} ${SMI_DLL} ${SNAPPY_DLL} ${SPANDSP_DLL} ${BCG729_DLL} ${LIBXML2_DLLS} ${WINSPARKLE_DLL} ${ZLIB_DLL} ${BROTLI_DLLS} ${ZSTD_DLL} ${ILBC_DLL} ${OPUS_DLL} + ${SPEEXDSP_DLL} # Needed for mmdbresolve ${MAXMINDDB_DLL} ) @@ -215,6 +216,7 @@ if (BUILD_logray) ${LZ4_DLL} ${NGHTTP2_DLL} ${SBC_DLL} ${SMI_DLL} ${SNAPPY_DLL} ${SPANDSP_DLL} ${BCG729_DLL} ${LIBXML2_DLLS} ${WINSPARKLE_DLL} ${ZLIB_DLL} ${BROTLI_DLLS} ${ZSTD_DLL} ${ILBC_DLL} ${OPUS_DLL} + ${SPEEXDSP_DLL} # Needed for mmdbresolve ${MAXMINDDB_DLL} ) diff --git a/packaging/wix/CMakeLists.txt b/packaging/wix/CMakeLists.txt index a0925e9e71..6ce90ef2fd 100644 --- a/packaging/wix/CMakeLists.txt +++ b/packaging/wix/CMakeLists.txt @@ -138,6 +138,7 @@ foreach(_dll ${CARES_DLL} ${PCRE2_DLL} ${GCRYPT_DLLS} ${LZ4_DLL} ${NGHTTP2_DLL} ${SBC_DLL} ${SMI_DLL} ${SNAPPY_DLL} ${SPANDSP_DLL} ${BCG729_DLL} ${LIBXML2_DLLS} ${WINSPARKLE_DLL} ${ZLIB_DLL} ${BROTLI_DLLS} ${ZSTD_DLL} ${ILBC_DLL} ${OPUS_DLL} + ${SPEEXDSP_DLL} # Required for mmdbresolve ${MAXMINDDB_DLL} ) @@ -177,6 +178,7 @@ foreach(_dll ${CARES_DLL} ${PCRE2_DLL} ${GCRYPT_DLLS} ${LZ4_DLL} ${NGHTTP2_DLL} ${SBC_DLL} ${SMI_DLL} ${SNAPPY_DLL} ${SPANDSP_DLL} ${BCG729_DLL} ${LIBXML2_DLLS} ${WINSPARKLE_DLL} ${ZLIB_DLL} ${BROTLI_DLLS} ${ZSTD_DLL} ${ILBC_DLL} ${OPUS_DLL} + ${SPEEXDSP_DLL} # mmdbresolve ${MAXMINDDB_DLL} ) diff --git a/sharkd_session.c b/sharkd_session.c index b34ab30f84..3f5414aee1 100644 --- a/sharkd_session.c +++ b/sharkd_session.c @@ -61,11 +61,7 @@ #include #include #include -#ifdef HAVE_SPEEXDSP -# include -#else -# include "speexdsp/speex_resampler.h" -#endif /* HAVE_SPEEXDSP */ +#include #include diff --git a/speexdsp/CMakeLists.txt b/speexdsp/CMakeLists.txt deleted file mode 100644 index 1667dfec28..0000000000 --- a/speexdsp/CMakeLists.txt +++ /dev/null @@ -1,15 +0,0 @@ -# CMakeLists.txt -# -# Wireshark - Network traffic analyzer -# By Gerald Combs -# Copyright 1998 Gerald Combs -# -# SPDX-License-Identifier: GPL-2.0-or-later -# - -add_library(speexresampler STATIC resample.c) - -set_target_properties(speexresampler PROPERTIES - LINK_FLAGS "${WS_LINK_FLAGS}" - FOLDER "Libs" -) diff --git a/speexdsp/README.txt b/speexdsp/README.txt deleted file mode 100644 index ee8cd66e13..0000000000 --- a/speexdsp/README.txt +++ /dev/null @@ -1,2 +0,0 @@ -Copied from http://git.xiph.org/speexdsp.git c470e2e89a6ca75b507437467692cd684b71a526 -and modified to compile in the Wireshark source tree. diff --git a/speexdsp/arch.h b/speexdsp/arch.h deleted file mode 100644 index eaad6b9260..0000000000 --- a/speexdsp/arch.h +++ /dev/null @@ -1,237 +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 - -#ifndef ABS /* already defined by glib.h */ -#define ABS(x) ((x) < 0 ? (-(x)) : (x)) /**< Absolute integer value. */ -#endif -#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) ((spx_int16_t)((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) ((spx_word16_t)(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) ((spx_word32_t)((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) ((spx_int16_t)((x) < -32767.5f ? -32768 : ((x) > 32766.5f ? 32767 : 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/speexdsp/resample.c b/speexdsp/resample.c deleted file mode 100644 index 32197fe5f6..0000000000 --- a/speexdsp/resample.c +++ /dev/null @@ -1,1211 +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 http://www-ccrma.stanford.edu/~jos/resample/. - - There is one main difference, though. This resampler uses cubic - interpolation instead of linear interpolation in the above paper. This - makes the table much smaller and makes it possible to compute that table - on a per-stream basis. In turn, being able to tweak the table for each - stream makes it possible to both reduce complexity on simple ratios - (e.g. 2/3), and get rid of the rounding operations in the inner loop. - The latter both reduces CPU time and makes the algorithm more SIMD-friendly. -*/ - -#include - -#define OUTSIDE_SPEEX 1 -#define FLOATING_POINT 1 - -#ifdef OUTSIDE_SPEEX -#include -static void *speex_alloc (size_t size) {return g_malloc0(size);} -static void *speex_realloc (void *ptr, size_t size) {return g_realloc(ptr, size);} -static void speex_free (void *ptr) {g_free(ptr);} -#include "speex_resampler.h" -#include "arch.h" -#else /* OUTSIDE_SPEEX */ - -#include "speex/speex_resampler.h" -#include "arch.h" -#include "os_support.h" -#endif /* OUTSIDE_SPEEX */ - -#include "stack_alloc.h" -#include -#include - -#include "ws_attributes.h" - -#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 - -#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 = {kaiser12_table, 64}; -#define KAISER12 (&_KAISER12) -/*static struct FuncDef _KAISER12 = {kaiser12_table, 32}; -#define KAISER12 (&_KAISER12)*/ -static const struct FuncDef _KAISER10 = {kaiser10_table, 32}; -#define KAISER10 (&_KAISER10) -static const struct FuncDef _KAISER8 = {kaiser8_table, 32}; -#define KAISER8 (&_KAISER8) -static const struct FuncDef _KAISER6 = {kaiser6_table, 32}; -#define KAISER6 (&_KAISER6) - -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 - -DIAG_OFF_CLANG(self-assign) /* SATURATE32PSHR */ - -#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 (spx_word16_t)(cutoff*sin(M_PI*xx)/(M_PI*xx) * compute_func((float)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] = (spx_word16_t)(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 _U_, 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; - - 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 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; - /* FIXME: divide the numerator and denominator by a certain amount if they're too large */ - st->filt_len = st->filt_len*st->num_rate / st->den_rate; - /* Round 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; - } - - /* Choose the resampling type that requires the least amount of memory */ -#ifdef RESAMPLE_FULL_SINC_TABLE - use_direct = 1; - if (INT_MAX/sizeof(spx_word16_t)/st->den_rate < st->filt_len) - goto fail; -#else - use_direct = st->filt_len*st->den_rate <= st->filt_len*st->oversample+8 - && INT_MAX/(spx_uint32_t)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/(spx_uint32_t)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_uint32_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/(spx_uint32_t)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) -{ - spx_uint32_t i; - 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)); - 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 */ - st->last_sample = (spx_int32_t*)speex_alloc(nb_channels*sizeof(spx_int32_t)); - st->magic_samples = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(spx_uint32_t)); - st->samp_frac_num = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(spx_uint32_t)); - for (i=0;ilast_sample[i] = 0; - st->magic_samples[i] = 0; - st->samp_frac_num[i] = 0; - } - - speex_resampler_set_quality(st, quality); - speex_resampler_set_rate_frac(st, ratio_num, ratio_den, in_rate, out_rate); - - 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; -} - -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 -{ - spx_uint32_t 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 -{ - spx_uint32_t j; - const int istride_save = st->in_stride; - const int ostride_save = st->out_stride; - spx_uint32_t ilen = *in_len; - spx_uint32_t olen = *out_len; - spx_word16_t *x = st->mem + channel_index * st->mem_alloc_size; - const spx_uint32_t xlen = st->mem_alloc_size - (st->filt_len - 1); -#ifdef VAR_ARRAYS - const unsigned int ylen = (olen < FIXED_STACK_ALLOC) ? olen : FIXED_STACK_ALLOC; - VARDECL(spx_word16_t *ystack); - ALLOC(ystack, ylen, spx_word16_t); -#else - const unsigned int ylen = FIXED_STACK_ALLOC; - spx_word16_t ystack[FIXED_STACK_ALLOC]; -#endif - - st->out_stride = 1; - - while (ilen && olen) { - spx_word16_t *y = ystack; - spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen; - spx_uint32_t ochunk = (olen > ylen) ? ylen : olen; - spx_uint32_t omagic = 0; - - if (st->magic_samples[channel_index]) { - omagic = speex_resampler_magic(st, channel_index, &y, ochunk); - ochunk -= omagic; - olen -= omagic; - } - if (! st->magic_samples[channel_index]) { - if (in) { - for(j=0;jfilt_len-1]=WORD2INT(in[j*istride_save]); -#else - x[j+st->filt_len-1]=in[j*istride_save]; -#endif - } else { - for(j=0;jfilt_len-1]=0; - } - - speex_resampler_process_native(st, channel_index, &ichunk, y, &ochunk); - } else { - ichunk = 0; - ochunk = 0; - } - - for (j=0;jout_stride = ostride_save; - *in_len -= ilen; - *out_len -= olen; - - return 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; -} - -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; - /* FIXME: This is terribly inefficient, but who cares (at least for now)? */ - for (fact=2;fact<=IMIN(st->num_rate, st->den_rate);fact++) - { - while ((st->num_rate % fact == 0) && (st->den_rate % fact == 0)) - { - st->num_rate /= fact; - st->den_rate /= fact; - } - } - - if (old_den > 0) - { - for (i=0;inb_channels;i++) - { - st->samp_frac_num[i]=st->samp_frac_num[i]*st->den_rate/old_den; - /* Safety net */ - if (st->samp_frac_num[i] >= st->den_rate) - st->samp_frac_num[i] = st->den_rate-1; - } - } - - if (st->initialised) - 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/speexdsp/speex_resampler.h b/speexdsp/speex_resampler.h deleted file mode 100644 index 284e7ec8ed..0000000000 --- a/speexdsp/speex_resampler.h +++ /dev/null @@ -1,347 +0,0 @@ -/** @file - - 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 - -#define OUTSIDE_SPEEX 1 -#define RANDOM_PREFIX ws_codec -#include -#include -#define EXPORT - -#ifdef OUTSIDE_SPEEX - -/********* WARNING: MENTAL SANITY ENDS HERE *************/ - -/* If the resampler is defined outside of Speex, we change the symbol names so that - there won't be any clash if linking with Speex later on. */ - -/* #define RANDOM_PREFIX your software name here */ -#ifndef RANDOM_PREFIX -#error "Please define RANDOM_PREFIX (above) to something specific to your project to prevent symbol name clashes" -#endif - -#define CAT_PREFIX2(a,b) a ## b -#define CAT_PREFIX(a,b) CAT_PREFIX2(a, b) - -#define speex_resampler_init CAT_PREFIX(RANDOM_PREFIX,_resampler_init) -#define speex_resampler_init_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_init_frac) -#define speex_resampler_destroy CAT_PREFIX(RANDOM_PREFIX,_resampler_destroy) -#define speex_resampler_process_float CAT_PREFIX(RANDOM_PREFIX,_resampler_process_float) -#define speex_resampler_process_int CAT_PREFIX(RANDOM_PREFIX,_resampler_process_int) -#define speex_resampler_process_interleaved_float CAT_PREFIX(RANDOM_PREFIX,_resampler_process_interleaved_float) -#define speex_resampler_process_interleaved_int CAT_PREFIX(RANDOM_PREFIX,_resampler_process_interleaved_int) -#define speex_resampler_set_rate CAT_PREFIX(RANDOM_PREFIX,_resampler_set_rate) -#define speex_resampler_get_rate CAT_PREFIX(RANDOM_PREFIX,_resampler_get_rate) -#define speex_resampler_set_rate_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_set_rate_frac) -#define speex_resampler_get_ratio CAT_PREFIX(RANDOM_PREFIX,_resampler_get_ratio) -#define speex_resampler_set_quality CAT_PREFIX(RANDOM_PREFIX,_resampler_set_quality) -#define speex_resampler_get_quality CAT_PREFIX(RANDOM_PREFIX,_resampler_get_quality) -#define speex_resampler_set_input_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_set_input_stride) -#define speex_resampler_get_input_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_get_input_stride) -#define speex_resampler_set_output_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_set_output_stride) -#define speex_resampler_get_output_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_get_output_stride) -#define speex_resampler_get_input_latency CAT_PREFIX(RANDOM_PREFIX,_resampler_get_input_latency) -#define speex_resampler_get_output_latency CAT_PREFIX(RANDOM_PREFIX,_resampler_get_output_latency) -#define speex_resampler_skip_zeros CAT_PREFIX(RANDOM_PREFIX,_resampler_skip_zeros) -#define speex_resampler_reset_mem CAT_PREFIX(RANDOM_PREFIX,_resampler_reset_mem) -#define speex_resampler_strerror CAT_PREFIX(RANDOM_PREFIX,_resampler_strerror) - -#define spx_int16_t short -#define spx_int32_t int -#define spx_uint16_t unsigned short -#define spx_uint32_t unsigned int - -#else /* OUTSIDE_SPEEX */ - -#include "speexdsp_types.h" - -#endif /* OUTSIDE_SPEEX */ - -#ifdef __cplusplus -extern "C" { -#endif - -#define SPEEX_RESAMPLER_QUALITY_MAX 10 -#define SPEEX_RESAMPLER_QUALITY_MIN 0 -#define SPEEX_RESAMPLER_QUALITY_DEFAULT 4 -#define SPEEX_RESAMPLER_QUALITY_VOIP 3 -#define SPEEX_RESAMPLER_QUALITY_DESKTOP 5 - -enum { - RESAMPLER_ERR_SUCCESS = 0, - RESAMPLER_ERR_ALLOC_FAILED = 1, - RESAMPLER_ERR_BAD_STATE = 2, - RESAMPLER_ERR_INVALID_ARG = 3, - RESAMPLER_ERR_PTR_OVERLAP = 4, - - RESAMPLER_ERR_MAX_ERROR -}; - -struct SpeexResamplerState_; -typedef struct SpeexResamplerState_ SpeexResamplerState; - -/** Create a new resampler with integer input and output rates. - * @param nb_channels Number of channels to be processed - * @param in_rate Input sampling rate (integer number of Hz). - * @param out_rate Output sampling rate (integer number of Hz). - * @param quality Resampling quality between 0 and 10, where 0 has poor quality - * and 10 has very high quality. - * @return Newly created resampler state - * @retval NULL Error: not enough memory - */ -SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels, - spx_uint32_t in_rate, - spx_uint32_t out_rate, - int quality, - int *err); - -/** Create a new resampler with fractional input/output rates. The sampling - * rate ratio is an arbitrary rational number with both the numerator and - * denominator being 32-bit integers. - * @param nb_channels Number of channels to be processed - * @param ratio_num Numerator of the sampling rate ratio - * @param ratio_den Denominator of the sampling rate ratio - * @param in_rate Input sampling rate rounded to the nearest integer (in Hz). - * @param out_rate Output sampling rate rounded to the nearest integer (in Hz). - * @param quality Resampling quality between 0 and 10, where 0 has poor quality - * and 10 has very high quality. - * @return Newly created resampler state - * @retval NULL Error: not enough memory - */ -SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels, - spx_uint32_t ratio_num, - spx_uint32_t ratio_den, - spx_uint32_t in_rate, - spx_uint32_t out_rate, - int quality, - int *err); - -/** Destroy a resampler state. - * @param st Resampler state - */ -void speex_resampler_destroy(SpeexResamplerState *st); - -/** Resample a float array. The input and output buffers must *not* overlap. - * @param st Resampler state - * @param channel_index Index of the channel to process for the multi-channel - * base (0 otherwise) - * @param in Input buffer - * @param in_len Number of input samples in the input buffer. Returns the - * number of samples processed - * @param out Output buffer - * @param out_len Size of the output buffer. Returns the number of samples written - */ -int speex_resampler_process_float(SpeexResamplerState *st, - spx_uint32_t channel_index, - const float *in, - spx_uint32_t *in_len, - float *out, - spx_uint32_t *out_len); - -/** Resample an int array. The input and output buffers must *not* overlap. - * @param st Resampler state - * @param channel_index Index of the channel to process for the multi-channel - * base (0 otherwise) - * @param in Input buffer - * @param in_len Number of input samples in the input buffer. Returns the number - * of samples processed - * @param out Output buffer - * @param out_len Size of the output buffer. Returns the number of samples written - */ -int speex_resampler_process_int(SpeexResamplerState *st, - spx_uint32_t channel_index, - const spx_int16_t *in, - spx_uint32_t *in_len, - spx_int16_t *out, - spx_uint32_t *out_len); - -/** Resample an interleaved float array. The input and output buffers must *not* overlap. - * @param st Resampler state - * @param in Input buffer - * @param in_len Number of input samples in the input buffer. Returns the number - * of samples processed. This is all per-channel. - * @param out Output buffer - * @param out_len Size of the output buffer. Returns the number of samples written. - * This is all per-channel. - */ -int speex_resampler_process_interleaved_float(SpeexResamplerState *st, - const float *in, - spx_uint32_t *in_len, - float *out, - spx_uint32_t *out_len); - -/** Resample an interleaved int array. The input and output buffers must *not* overlap. - * @param st Resampler state - * @param in Input buffer - * @param in_len Number of input samples in the input buffer. Returns the number - * of samples processed. This is all per-channel. - * @param out Output buffer - * @param out_len Size of the output buffer. Returns the number of samples written. - * This is all per-channel. - */ -int speex_resampler_process_interleaved_int(SpeexResamplerState *st, - const spx_int16_t *in, - spx_uint32_t *in_len, - spx_int16_t *out, - spx_uint32_t *out_len); - -/** Set (change) the input/output sampling rates (integer value). - * @param st Resampler state - * @param in_rate Input sampling rate (integer number of Hz). - * @param out_rate Output sampling rate (integer number of Hz). - */ -int speex_resampler_set_rate(SpeexResamplerState *st, - spx_uint32_t in_rate, - spx_uint32_t out_rate); - -/** Get the current input/output sampling rates (integer value). - * @param st Resampler state - * @param in_rate Input sampling rate (integer number of Hz) copied. - * @param out_rate Output sampling rate (integer number of Hz) copied. - */ -void speex_resampler_get_rate(SpeexResamplerState *st, - spx_uint32_t *in_rate, - spx_uint32_t *out_rate); - -/** Set (change) the input/output sampling rates and resampling ratio - * (fractional values in Hz supported). - * @param st Resampler state - * @param ratio_num Numerator of the sampling rate ratio - * @param ratio_den Denominator of the sampling rate ratio - * @param in_rate Input sampling rate rounded to the nearest integer (in Hz). - * @param out_rate Output sampling rate rounded to the nearest integer (in Hz). - */ -int speex_resampler_set_rate_frac(SpeexResamplerState *st, - spx_uint32_t ratio_num, - spx_uint32_t ratio_den, - spx_uint32_t in_rate, - spx_uint32_t out_rate); - -/** Get the current resampling ratio. This will be reduced to the least - * common denominator. - * @param st Resampler state - * @param ratio_num Numerator of the sampling rate ratio copied - * @param ratio_den Denominator of the sampling rate ratio copied - */ -void speex_resampler_get_ratio(SpeexResamplerState *st, - spx_uint32_t *ratio_num, - spx_uint32_t *ratio_den); - -/** Set (change) the conversion quality. - * @param st Resampler state - * @param quality Resampling quality between 0 and 10, where 0 has poor - * quality and 10 has very high quality. - */ -int speex_resampler_set_quality(SpeexResamplerState *st, - int quality); - -/** Get the conversion quality. - * @param st Resampler state - * @param quality Resampling quality between 0 and 10, where 0 has poor - * quality and 10 has very high quality. - */ -void speex_resampler_get_quality(SpeexResamplerState *st, - int *quality); - -/** Set (change) the input stride. - * @param st Resampler state - * @param stride Input stride - */ -void speex_resampler_set_input_stride(SpeexResamplerState *st, - spx_uint32_t stride); - -/** Get the input stride. - * @param st Resampler state - * @param stride Input stride copied - */ -void speex_resampler_get_input_stride(SpeexResamplerState *st, - spx_uint32_t *stride); - -/** Set (change) the output stride. - * @param st Resampler state - * @param stride Output stride - */ -void speex_resampler_set_output_stride(SpeexResamplerState *st, - spx_uint32_t stride); - -/** Get the output stride. - * @param st Resampler state copied - * @param stride Output stride - */ -void speex_resampler_get_output_stride(SpeexResamplerState *st, - spx_uint32_t *stride); - -/** Get the latency 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/speexdsp/stack_alloc.h b/speexdsp/stack_alloc.h deleted file mode 100644 index a446065825..0000000000 --- a/speexdsp/stack_alloc.h +++ /dev/null @@ -1,115 +0,0 @@ -/* Copyright (C) 2002 Jean-Marc Valin */ -/** - @file stack_alloc.h - @brief Temporary memory allocation on stack -*/ -/* - Redistribution and use in source and binary forms, with or without - modification, are permitted provided that the following conditions - are met: - - - Redistributions of source code must retain the above copyright - notice, this list of conditions and the following disclaimer. - - - Redistributions in binary form must reproduce the above copyright - notice, this list of conditions and the following disclaimer in the - documentation and/or other materials provided with the distribution. - - - Neither the name of the Xiph.org Foundation nor the names of its - contributors may be used to endorse or promote products derived from - this software without specific prior written permission. - - THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS - ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT - LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR - A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR - CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, - EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, - PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR - PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF - LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING - NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS - SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -*/ - -#ifndef STACK_ALLOC_H -#define STACK_ALLOC_H - -#ifdef USE_ALLOCA -# ifdef _WIN32 -# include -# else -# ifdef HAVE_ALLOCA_H -# include -# else -# include -# endif -# endif -#endif - -/** - * @def ALIGN(stack, size) - * - * Aligns the stack to a 'size' boundary - * - * @param stack Stack - * @param size New size boundary - */ - -/** - * @def PUSH(stack, size, type) - * - * Allocates 'size' elements of type 'type' on the stack - * - * @param stack Stack - * @param size Number of elements - * @param type Type of element - */ - -/** - * @def VARDECL(var) - * - * Declare variable on stack - * - * @param var Variable to declare - */ - -/** - * @def ALLOC(var, size, type) - * - * Allocate 'size' elements of 'type' on stack - * - * @param var Name of variable to allocate - * @param size Number of elements - * @param type Type of element - */ - -#ifdef ENABLE_VALGRIND - -#include - -#define ALIGN(stack, size) ((stack) += ((size) - (long)(stack)) & ((size) - 1)) - -#define PUSH(stack, size, type) (VALGRIND_MAKE_NOACCESS(stack, 1000),ALIGN((stack),sizeof(type)),VALGRIND_MAKE_WRITABLE(stack, ((size)*sizeof(type))),(stack)+=((size)*sizeof(type)),(type*)((stack)-((size)*sizeof(type)))) - -#else - -#define ALIGN(stack, size) ((stack) += ((size) - (long)(stack)) & ((size) - 1)) - -#define PUSH(stack, size, type) (ALIGN((stack),sizeof(type)),(stack)+=((size)*sizeof(type)),(type*)((stack)-((size)*sizeof(type)))) - -#endif - -#if defined(VAR_ARRAYS) -#define VARDECL(var) -#define ALLOC(var, size, type) type var[size] -#elif defined(USE_ALLOCA) -#define VARDECL(var) var -#define ALLOC(var, size, type) var = alloca(sizeof(type)*(size)) -#else -#define VARDECL(var) var -#define ALLOC(var, size, type) var = PUSH(stack, size, type) -#endif - - -#endif diff --git a/tools/win-setup.ps1 b/tools/win-setup.ps1 index 41534b8952..b4d19cb6c3 100644 --- a/tools/win-setup.ps1 +++ b/tools/win-setup.ps1 @@ -92,6 +92,7 @@ $Win64Archives = @{ "sbc/sbc-1.3-1-win64ws.zip" = "08cef6898c421277a6582ef3225d8820f74a037cbd5b6e673a4d8f4593ce80a1"; "snappy/snappy-1.1.9-1-win64ws.zip" = "fa907724be019bcc55d27ebe88257ba8898b5c38b719099b8164ac78600d81cc"; "spandsp/spandsp-0.0.6-2-win64ws.zip" = "2eb8278633037f60f44815ea1606486ab5dcdf3bddc500b20c9fe356856236b2"; + "speexdsp/speexdsp-1.21.1-1-win64ws.zip" = "d36db62e64ffaee38d9f607bef07d3778d8957ad29757f3eba169eb135f1a4e5"; "vcpkg-export/vcpkg-export-20220726-1-win64ws.zip" = "b1eaa8124802532fa8d30789219906f90fb80908844e4458327b3f73995a44b0"; "WinSparkle/WinSparkle-0.5.7.zip" = "56d396ef0c4e8b0589ea74134e484376ca6459d972cd1ab1da6b9624d82e6d04"; "zstd/zstd-1.5.2-1-win64ws.zip" = "d920afe636951cfcf144824d9c075d1f2c13387f4739152fe185fd9c09fc58f2"; @@ -141,6 +142,7 @@ $CleanupItems = @( "sbc-1.3-win??ws" "snappy-1.1.*-win??ws" "spandsp-0.0.6-win??ws" + "speexdsp-*-win??ws" "user-guide" "vcpkg-export-*-win??ws" "zstd-*-win??ws" diff --git a/ui/logray/logray_main.cpp b/ui/logray/logray_main.cpp index 774840593c..f1e67b30b0 100644 --- a/ui/logray/logray_main.cpp +++ b/ui/logray/logray_main.cpp @@ -225,11 +225,6 @@ gather_wireshark_qt_compiled_info(feature_list l) without_feature(l, "AirPcap"); #endif #endif /* _WIN32 */ -#ifdef HAVE_SPEEXDSP - with_feature(l, "SpeexDSP (using system library)"); -#else - with_feature(l, "SpeexDSP (using bundled resampler)"); -#endif #ifdef HAVE_MINIZIP with_feature(l, "Minizip"); diff --git a/ui/qt/main.cpp b/ui/qt/main.cpp index 23da6a4083..68b53ffa7e 100644 --- a/ui/qt/main.cpp +++ b/ui/qt/main.cpp @@ -227,11 +227,6 @@ gather_wireshark_qt_compiled_info(feature_list l) without_feature(l, "AirPcap"); #endif #endif /* _WIN32 */ -#ifdef HAVE_SPEEXDSP - with_feature(l, "SpeexDSP (using system library)"); -#else - with_feature(l, "SpeexDSP (using bundled resampler)"); -#endif #ifdef HAVE_MINIZIP with_feature(l, "Minizip"); diff --git a/ui/qt/rtp_audio_stream.cpp b/ui/qt/rtp_audio_stream.cpp index 4229854c07..6f93c12403 100644 --- a/ui/qt/rtp_audio_stream.cpp +++ b/ui/qt/rtp_audio_stream.cpp @@ -11,11 +11,7 @@ #ifdef QT_MULTIMEDIA_LIB -#ifdef HAVE_SPEEXDSP #include -#else -#include "../../speexdsp/speex_resampler.h" -#endif /* HAVE_SPEEXDSP */ #include #include diff --git a/ui/qt/utils/rtp_audio_file.h b/ui/qt/utils/rtp_audio_file.h index 3024d29287..addc3015d5 100644 --- a/ui/qt/utils/rtp_audio_file.h +++ b/ui/qt/utils/rtp_audio_file.h @@ -13,11 +13,7 @@ #include "config.h" #include -#ifdef HAVE_SPEEXDSP #include -#else -#include "../../speexdsp/speex_resampler.h" -#endif /* HAVE_SPEEXDSP */ #include #include