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-rw-r--r--docs/plugins/inspect/plugin-speexresample.xml34
-rw-r--r--gst/speexresample/Makefile.am31
-rw-r--r--gst/speexresample/README80
-rw-r--r--gst/speexresample/arch.h241
-rw-r--r--gst/speexresample/fixed_generic.h106
-rw-r--r--gst/speexresample/gstspeexresample.c991
-rw-r--r--gst/speexresample/gstspeexresample.h82
-rw-r--r--gst/speexresample/resample.c1372
-rw-r--r--gst/speexresample/speex_resampler.h342
-rw-r--r--gst/speexresample/speex_resampler_float.c24
-rw-r--r--gst/speexresample/speex_resampler_int.c24
-rw-r--r--gst/speexresample/speex_resampler_wrapper.h86
12 files changed, 0 insertions, 3413 deletions
diff --git a/docs/plugins/inspect/plugin-speexresample.xml b/docs/plugins/inspect/plugin-speexresample.xml
deleted file mode 100644
index 40127c49..00000000
--- a/docs/plugins/inspect/plugin-speexresample.xml
+++ /dev/null
@@ -1,34 +0,0 @@
-<plugin>
- <name>speexresample</name>
- <description>Resamples audio</description>
- <filename>../../gst/speexresample/.libs/libgstspeexresample.so</filename>
- <basename>libgstspeexresample.so</basename>
- <version>0.10.8</version>
- <license>LGPL</license>
- <source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
- <origin>Unknown package origin</origin>
- <elements>
- <element>
- <name>speexresample</name>
- <longname>Audio resampler</longname>
- <class>Filter/Converter/Audio</class>
- <description>Resamples audio</description>
- <author>Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</author>
- <pads>
- <caps>
- <name>src</name>
- <direction>source</direction>
- <presence>always</presence>
- <details>audio/x-raw-float, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], endianness=(int)1234, width=(int)32; audio/x-raw-int, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], endianness=(int)1234, width=(int)16, depth=(int)16, signed=(boolean)true</details>
- </caps>
- <caps>
- <name>sink</name>
- <direction>sink</direction>
- <presence>always</presence>
- <details>audio/x-raw-float, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], endianness=(int)1234, width=(int)32; audio/x-raw-int, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], endianness=(int)1234, width=(int)16, depth=(int)16, signed=(boolean)true</details>
- </caps>
- </pads>
- </element>
- </elements>
-</plugin> \ No newline at end of file
diff --git a/gst/speexresample/Makefile.am b/gst/speexresample/Makefile.am
deleted file mode 100644
index 3e0c4924..00000000
--- a/gst/speexresample/Makefile.am
+++ /dev/null
@@ -1,31 +0,0 @@
-plugin_LTLIBRARIES = libgstspeexresample.la
-
-libgstspeexresample_la_SOURCES = \
- gstspeexresample.c \
- speex_resampler_int.c \
- speex_resampler_float.c
-
-libgstspeexresample_la_CFLAGS = \
- $(GST_PLUGINS_BASE_CFLAGS) \
- $(GST_BASE_CFLAGS) \
- $(GST_CFLAGS)
-
-libgstspeexresample_la_LIBADD = \
- $(GST_PLUGINS_BASE_LIBS) \
- $(GST_BASE_LIBS) \
- $(GST_LIBS) \
- -lgstaudio-$(GST_MAJORMINOR) \
- $(LIBM)
-
-libgstspeexresample_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
-
-noinst_HEADERS = \
- arch.h \
- fixed_generic.h \
- gstspeexresample.h \
- speex_resampler.h \
- speex_resampler_wrapper.h
-
-EXTRA_DIST = \
- resample.c
-
diff --git a/gst/speexresample/README b/gst/speexresample/README
deleted file mode 100644
index 68d8c290..00000000
--- a/gst/speexresample/README
+++ /dev/null
@@ -1,80 +0,0 @@
-resample.c
-arch.h
-fixed_generic.h
-speex_resampler.h
-
-are taken from http://svn.xiph.org/trunk/speex/ revision 14232.
-
-The only changes are:
-
---- speex/libspeex/arch.h 2007-11-21 11:05:46.000000000 +0100
-+++ speexresample/arch.h 2007-11-20 05:41:09.000000000 +0100
-@@ -78,7 +78,9 @@
- #include "speex/speex_types.h"
- #endif
-
-+#ifndef ABS
- #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. */
-
---- speex/include/speex/speex_resampler.h 2007-11-21 11:05:44.000000000 +0100
-+++ speexresample/speex_resampler.h 2007-11-21 11:10:02.000000000 +0100
-@@ -41,6 +41,8 @@
-
- #ifdef OUTSIDE_SPEEX
-
-+#include <glib.h>
-+
- /********* WARNING: MENTAL SANITY ENDS HERE *************/
-
- /* If the resampler is defined outside of Speex, we change the symbol names so that
-@@ -75,10 +77,10 @@
- #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
-+#define spx_int16_t gint16
-+#define spx_int32_t gint32
-+#define spx_uint16_t guint16
-+#define spx_uint32_t guint32
-
- #else /* OUTSIDE_SPEEX */
-
---- speex/libspeex/resample.c 2007-11-25 14:15:38.000000000 +0100
-+++ speexresample/resample.c 2007-11-25 14:15:31.000000000 +0100
-@@ -62,20 +62,23 @@
-
- #ifdef OUTSIDE_SPEEX
- #include <stdlib.h>
--static void *
-+#include <glib.h>
-+
-+static inline void *
- speex_alloc (int size)
- {
-- return calloc (size, 1);
-+ return g_malloc0 (size);
- }
--static void *
-+static inline void *
- speex_realloc (void *ptr, int size)
- {
-- return realloc (ptr, size);
-+ return g_realloc (ptr, size);
- }
--static void
-+
-+static inline void
- speex_free (void *ptr)
- {
-- free (ptr);
-+ g_free (ptr);
- }
-
- #include "speex_resampler.h"
diff --git a/gst/speexresample/arch.h b/gst/speexresample/arch.h
deleted file mode 100644
index 3b341f0a..00000000
--- a/gst/speexresample/arch.h
+++ /dev/null
@@ -1,241 +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
-
-#ifndef SPEEX_VERSION
-#define SPEEX_MAJOR_VERSION 1 /**< Major Speex version. */
-#define SPEEX_MINOR_VERSION 1 /**< Minor Speex version. */
-#define SPEEX_MICRO_VERSION 15 /**< Micro Speex version. */
-#define SPEEX_EXTRA_VERSION "" /**< Extra Speex version. */
-#define SPEEX_VERSION "speex-1.2beta4" /**< Speex version string. */
-#endif
-
-/* A couple test to catch stupid option combinations */
-#ifdef FIXED_POINT
-
-#ifdef FLOATING_POINT
-#error You cannot compile as floating point and fixed point at the same time
-#endif
-#ifdef _USE_SSE
-#error SSE is only for floating-point
-#endif
-#if ((defined (ARM4_ASM)||defined (ARM4_ASM)) && defined(BFIN_ASM)) || (defined (ARM4_ASM)&&defined(ARM5E_ASM))
-#error Make up your mind. What CPU do you have?
-#endif
-#ifdef VORBIS_PSYCHO
-#error Vorbis-psy model currently not implemented in fixed-point
-#endif
-
-#else
-
-#ifndef FLOATING_POINT
-#error You now need to define either FIXED_POINT or FLOATING_POINT
-#endif
-#if defined (ARM4_ASM) || defined(ARM5E_ASM) || defined(BFIN_ASM)
-#error I suppose you can have a [ARM4/ARM5E/Blackfin] that has float instructions?
-#endif
-#ifdef FIXED_POINT_DEBUG
-#error "Don't you think enabling fixed-point is a good thing to do if you want to debug that?"
-#endif
-
-
-#endif
-
-#ifndef OUTSIDE_SPEEX
-#include "speex/speex_types.h"
-#endif
-
-#ifndef ABS
-#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 VERY_SMALL 0
-#define VERY_LARGE32 ((spx_word32_t)2147483647)
-#define VERY_LARGE16 ((spx_word16_t)32767)
-#define Q15_ONE ((spx_word16_t)32767)
-
-
-#ifdef FIXED_DEBUG
-#include "fixed_debug.h"
-#else
-
-#include "fixed_generic.h"
-
-#ifdef ARM5E_ASM
-#include "fixed_arm5e.h"
-#elif defined (ARM4_ASM)
-#include "fixed_arm4.h"
-#elif defined (BFIN_ASM)
-#include "fixed_bfin.h"
-#endif
-
-#endif
-
-
-#else
-
-typedef float spx_mem_t;
-typedef float spx_coef_t;
-typedef float spx_lsp_t;
-typedef float spx_sig_t;
-typedef float spx_word16_t;
-typedef float spx_word32_t;
-
-#define Q15ONE 1.0f
-#define LPC_SCALING 1.f
-#define SIG_SCALING 1.f
-#define LSP_SCALING 1.f
-#define GAMMA_SCALING 1.f
-#define GAIN_SCALING 1.f
-#define GAIN_SCALING_1 1.f
-
-
-#define VERY_SMALL 1e-15f
-#define VERY_LARGE32 1e15f
-#define VERY_LARGE16 1e15f
-#define Q15_ONE ((spx_word16_t)1.f)
-
-#define QCONST16(x,bits) (x)
-#define QCONST32(x,bits) (x)
-
-#define NEG16(x) (-(x))
-#define NEG32(x) (-(x))
-#define EXTRACT16(x) (x)
-#define EXTEND32(x) (x)
-#define SHR16(a,shift) (a)
-#define SHL16(a,shift) (a)
-#define SHR32(a,shift) (a)
-#define SHL32(a,shift) (a)
-#define PSHR16(a,shift) (a)
-#define PSHR32(a,shift) (a)
-#define VSHR32(a,shift) (a)
-#define SATURATE16(x,a) (x)
-#define SATURATE32(x,a) (x)
-
-#define PSHR(a,shift) (a)
-#define SHR(a,shift) (a)
-#define SHL(a,shift) (a)
-#define SATURATE(x,a) (x)
-
-#define ADD16(a,b) ((a)+(b))
-#define SUB16(a,b) ((a)-(b))
-#define ADD32(a,b) ((a)+(b))
-#define SUB32(a,b) ((a)-(b))
-#define MULT16_16_16(a,b) ((a)*(b))
-#define MULT16_16(a,b) ((spx_word32_t)(a)*(spx_word32_t)(b))
-#define MAC16_16(c,a,b) ((c)+(spx_word32_t)(a)*(spx_word32_t)(b))
-
-#define MULT16_32_Q11(a,b) ((a)*(b))
-#define MULT16_32_Q13(a,b) ((a)*(b))
-#define MULT16_32_Q14(a,b) ((a)*(b))
-#define MULT16_32_Q15(a,b) ((a)*(b))
-#define MULT16_32_P15(a,b) ((a)*(b))
-
-#define MAC16_32_Q11(c,a,b) ((c)+(a)*(b))
-#define MAC16_32_Q15(c,a,b) ((c)+(a)*(b))
-
-#define MAC16_16_Q11(c,a,b) ((c)+(a)*(b))
-#define MAC16_16_Q13(c,a,b) ((c)+(a)*(b))
-#define MAC16_16_P13(c,a,b) ((c)+(a)*(b))
-#define MULT16_16_Q11_32(a,b) ((a)*(b))
-#define MULT16_16_Q13(a,b) ((a)*(b))
-#define MULT16_16_Q14(a,b) ((a)*(b))
-#define MULT16_16_Q15(a,b) ((a)*(b))
-#define MULT16_16_P15(a,b) ((a)*(b))
-#define MULT16_16_P13(a,b) ((a)*(b))
-#define MULT16_16_P14(a,b) ((a)*(b))
-
-#define DIV32_16(a,b) (((spx_word32_t)(a))/(spx_word16_t)(b))
-#define PDIV32_16(a,b) (((spx_word32_t)(a))/(spx_word16_t)(b))
-#define DIV32(a,b) (((spx_word32_t)(a))/(spx_word32_t)(b))
-#define PDIV32(a,b) (((spx_word32_t)(a))/(spx_word32_t)(b))
-
-
-#endif
-
-
-#if defined (CONFIG_TI_C54X) || defined (CONFIG_TI_C55X)
-
-/* 2 on TI C5x DSP */
-#define BYTES_PER_CHAR 2
-#define BITS_PER_CHAR 16
-#define LOG2_BITS_PER_CHAR 4
-
-#else
-
-#define BYTES_PER_CHAR 1
-#define BITS_PER_CHAR 8
-#define LOG2_BITS_PER_CHAR 3
-
-#endif
-
-
-
-#ifdef FIXED_DEBUG
-long long spx_mips=0;
-#endif
-
-
-#endif
diff --git a/gst/speexresample/fixed_generic.h b/gst/speexresample/fixed_generic.h
deleted file mode 100644
index 2948177c..00000000
--- a/gst/speexresample/fixed_generic.h
+++ /dev/null
@@ -1,106 +0,0 @@
-/* Copyright (C) 2003 Jean-Marc Valin */
-/**
- @file fixed_generic.h
- @brief Generic fixed-point operations
-*/
-/*
- Redistribution and use in source and binary forms, with or without
- modification, are permitted provided that the following conditions
- are met:
-
- - Redistributions of source code must retain the above copyright
- notice, this list of conditions and the following disclaimer.
-
- - Redistributions in binary form must reproduce the above copyright
- notice, this list of conditions and the following disclaimer in the
- documentation and/or other materials provided with the distribution.
-
- - Neither the name of the Xiph.org Foundation nor the names of its
- contributors may be used to endorse or promote products derived from
- this software without specific prior written permission.
-
- THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
- ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
- LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
- A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
- CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
- EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
- PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
- LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
- NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
- SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-*/
-
-#ifndef FIXED_GENERIC_H
-#define FIXED_GENERIC_H
-
-#define QCONST16(x,bits) ((spx_word16_t)(.5+(x)*(((spx_word32_t)1)<<(bits))))
-#define QCONST32(x,bits) ((spx_word32_t)(.5+(x)*(((spx_word32_t)1)<<(bits))))
-
-#define NEG16(x) (-(x))
-#define NEG32(x) (-(x))
-#define EXTRACT16(x) ((spx_word16_t)(x))
-#define EXTEND32(x) ((spx_word32_t)(x))
-#define SHR16(a,shift) ((a) >> (shift))
-#define SHL16(a,shift) ((a) << (shift))
-#define SHR32(a,shift) ((a) >> (shift))
-#define SHL32(a,shift) ((a) << (shift))
-#define PSHR16(a,shift) (SHR16((a)+((1<<((shift))>>1)),shift))
-#define PSHR32(a,shift) (SHR32((a)+((1<<((shift))>>1)),shift))
-#define VSHR32(a, shift) (((shift)>0) ? SHR32(a, shift) : SHL32(a, -(shift)))
-#define SATURATE16(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x)))
-#define SATURATE32(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x)))
-
-#define SHR(a,shift) ((a) >> (shift))
-#define SHL(a,shift) ((spx_word32_t)(a) << (shift))
-#define PSHR(a,shift) (SHR((a)+((1<<((shift))>>1)),shift))
-#define SATURATE(x,a) (((x)>(a) ? (a) : (x)<-(a) ? -(a) : (x)))
-
-
-#define ADD16(a,b) ((spx_word16_t)((spx_word16_t)(a)+(spx_word16_t)(b)))
-#define SUB16(a,b) ((spx_word16_t)(a)-(spx_word16_t)(b))
-#define ADD32(a,b) ((spx_word32_t)(a)+(spx_word32_t)(b))
-#define SUB32(a,b) ((spx_word32_t)(a)-(spx_word32_t)(b))
-
-
-/* result fits in 16 bits */
-#define MULT16_16_16(a,b) ((((spx_word16_t)(a))*((spx_word16_t)(b))))
-
-/* (spx_word32_t)(spx_word16_t) gives TI compiler a hint that it's 16x16->32 multiply */
-#define MULT16_16(a,b) (((spx_word32_t)(spx_word16_t)(a))*((spx_word32_t)(spx_word16_t)(b)))
-
-#define MAC16_16(c,a,b) (ADD32((c),MULT16_16((a),(b))))
-#define MULT16_32_Q12(a,b) ADD32(MULT16_16((a),SHR((b),12)), SHR(MULT16_16((a),((b)&0x00000fff)),12))
-#define MULT16_32_Q13(a,b) ADD32(MULT16_16((a),SHR((b),13)), SHR(MULT16_16((a),((b)&0x00001fff)),13))
-#define MULT16_32_Q14(a,b) ADD32(MULT16_16((a),SHR((b),14)), SHR(MULT16_16((a),((b)&0x00003fff)),14))
-
-#define MULT16_32_Q11(a,b) ADD32(MULT16_16((a),SHR((b),11)), SHR(MULT16_16((a),((b)&0x000007ff)),11))
-#define MAC16_32_Q11(c,a,b) ADD32(c,ADD32(MULT16_16((a),SHR((b),11)), SHR(MULT16_16((a),((b)&0x000007ff)),11)))
-
-#define MULT16_32_P15(a,b) ADD32(MULT16_16((a),SHR((b),15)), PSHR(MULT16_16((a),((b)&0x00007fff)),15))
-#define MULT16_32_Q15(a,b) ADD32(MULT16_16((a),SHR((b),15)), SHR(MULT16_16((a),((b)&0x00007fff)),15))
-#define MAC16_32_Q15(c,a,b) ADD32(c,ADD32(MULT16_16((a),SHR((b),15)), SHR(MULT16_16((a),((b)&0x00007fff)),15)))
-
-
-#define MAC16_16_Q11(c,a,b) (ADD32((c),SHR(MULT16_16((a),(b)),11)))
-#define MAC16_16_Q13(c,a,b) (ADD32((c),SHR(MULT16_16((a),(b)),13)))
-#define MAC16_16_P13(c,a,b) (ADD32((c),SHR(ADD32(4096,MULT16_16((a),(b))),13)))
-
-#define MULT16_16_Q11_32(a,b) (SHR(MULT16_16((a),(b)),11))
-#define MULT16_16_Q13(a,b) (SHR(MULT16_16((a),(b)),13))
-#define MULT16_16_Q14(a,b) (SHR(MULT16_16((a),(b)),14))
-#define MULT16_16_Q15(a,b) (SHR(MULT16_16((a),(b)),15))
-
-#define MULT16_16_P13(a,b) (SHR(ADD32(4096,MULT16_16((a),(b))),13))
-#define MULT16_16_P14(a,b) (SHR(ADD32(8192,MULT16_16((a),(b))),14))
-#define MULT16_16_P15(a,b) (SHR(ADD32(16384,MULT16_16((a),(b))),15))
-
-#define MUL_16_32_R15(a,bh,bl) ADD32(MULT16_16((a),(bh)), SHR(MULT16_16((a),(bl)),15))
-
-#define DIV32_16(a,b) ((spx_word16_t)(((spx_word32_t)(a))/((spx_word16_t)(b))))
-#define PDIV32_16(a,b) ((spx_word16_t)(((spx_word32_t)(a)+((spx_word16_t)(b)>>1))/((spx_word16_t)(b))))
-#define DIV32(a,b) (((spx_word32_t)(a))/((spx_word32_t)(b)))
-#define PDIV32(a,b) (((spx_word32_t)(a)+((spx_word16_t)(b)>>1))/((spx_word32_t)(b)))
-
-#endif
diff --git a/gst/speexresample/gstspeexresample.c b/gst/speexresample/gstspeexresample.c
deleted file mode 100644
index 096c6b93..00000000
--- a/gst/speexresample/gstspeexresample.c
+++ /dev/null
@@ -1,991 +0,0 @@
-/* GStreamer
- * Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu>
- * Copyright (C) 2003,2004 David A. Schleef <ds@schleef.org>
- * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-/**
- * SECTION:element-speexresample
- *
- * speexresample resamples raw audio buffers to different sample rates using
- * a configurable windowing function to enhance quality.
- *
- * <refsect2>
- * <title>Example launch line</title>
- * |[
- * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! speexresample ! audio/x-raw-int, rate=8000 ! alsasink
- * ]| Decode an Ogg/Vorbis downsample to 8Khz and play sound through alsa.
- * To create the Ogg/Vorbis file refer to the documentation of vorbisenc.
- * </refsect2>
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <string.h>
-#include <math.h>
-
-#include "gstspeexresample.h"
-#include <gst/audio/audio.h>
-#include <gst/base/gstbasetransform.h>
-
-GST_DEBUG_CATEGORY (speex_resample_debug);
-#define GST_CAT_DEFAULT speex_resample_debug
-
-enum
-{
- PROP_0,
- PROP_QUALITY
-};
-
-#define SUPPORTED_CAPS \
-GST_STATIC_CAPS ( \
- "audio/x-raw-float, " \
- "rate = (int) [ 1, MAX ], " \
- "channels = (int) [ 1, MAX ], " \
- "endianness = (int) BYTE_ORDER, " \
- "width = (int) 32; " \
- "audio/x-raw-int, " \
- "rate = (int) [ 1, MAX ], " \
- "channels = (int) [ 1, MAX ], " \
- "endianness = (int) BYTE_ORDER, " \
- "width = (int) 16, " \
- "depth = (int) 16, " \
- "signed = (boolean) true" \
-)
-
-static GstStaticPadTemplate gst_speex_resample_sink_template =
-GST_STATIC_PAD_TEMPLATE ("sink",
- GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS);
-
-static GstStaticPadTemplate gst_speex_resample_src_template =
-GST_STATIC_PAD_TEMPLATE ("src",
- GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
-
-static void gst_speex_resample_set_property (GObject * object,
- guint prop_id, const GValue * value, GParamSpec * pspec);
-static void gst_speex_resample_get_property (GObject * object,
- guint prop_id, GValue * value, GParamSpec * pspec);
-
-/* vmethods */
-static gboolean gst_speex_resample_get_unit_size (GstBaseTransform * base,
- GstCaps * caps, guint * size);
-static GstCaps *gst_speex_resample_transform_caps (GstBaseTransform * base,
- GstPadDirection direction, GstCaps * caps);
-static gboolean gst_speex_resample_transform_size (GstBaseTransform * trans,
- GstPadDirection direction, GstCaps * incaps, guint insize,
- GstCaps * outcaps, guint * outsize);
-static gboolean gst_speex_resample_set_caps (GstBaseTransform * base,
- GstCaps * incaps, GstCaps * outcaps);
-static GstFlowReturn gst_speex_resample_transform (GstBaseTransform * base,
- GstBuffer * inbuf, GstBuffer * outbuf);
-static gboolean gst_speex_resample_event (GstBaseTransform * base,
- GstEvent * event);
-static gboolean gst_speex_resample_start (GstBaseTransform * base);
-static gboolean gst_speex_resample_stop (GstBaseTransform * base);
-static gboolean gst_speex_resample_query (GstPad * pad, GstQuery * query);
-static const GstQueryType *gst_speex_resample_query_type (GstPad * pad);
-
-#define DEBUG_INIT(bla) \
- GST_DEBUG_CATEGORY_INIT (speex_resample_debug, "speex_resample", 0, "audio resampling element");
-
-GST_BOILERPLATE_FULL (GstSpeexResample, gst_speex_resample, GstBaseTransform,
- GST_TYPE_BASE_TRANSFORM, DEBUG_INIT);
-
-static void
-gst_speex_resample_base_init (gpointer g_class)
-{
- GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
-
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_speex_resample_src_template));
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_speex_resample_sink_template));
-
- gst_element_class_set_details_simple (gstelement_class, "Audio resampler",
- "Filter/Converter/Audio", "Resamples audio",
- "Sebastian Dröge <slomo@circular-chaos.org>");
-}
-
-static void
-gst_speex_resample_class_init (GstSpeexResampleClass * klass)
-{
- GObjectClass *gobject_class = (GObjectClass *) klass;
-
- gobject_class->set_property = gst_speex_resample_set_property;
- gobject_class->get_property = gst_speex_resample_get_property;
-
- g_object_class_install_property (gobject_class, PROP_QUALITY,
- g_param_spec_int ("quality", "Quality", "Resample quality with 0 being "
- "the lowest and 10 being the best",
- SPEEX_RESAMPLER_QUALITY_MIN, SPEEX_RESAMPLER_QUALITY_MAX,
- SPEEX_RESAMPLER_QUALITY_DEFAULT,
- G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
-
- GST_BASE_TRANSFORM_CLASS (klass)->start =
- GST_DEBUG_FUNCPTR (gst_speex_resample_start);
- GST_BASE_TRANSFORM_CLASS (klass)->stop =
- GST_DEBUG_FUNCPTR (gst_speex_resample_stop);
- GST_BASE_TRANSFORM_CLASS (klass)->transform_size =
- GST_DEBUG_FUNCPTR (gst_speex_resample_transform_size);
- GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size =
- GST_DEBUG_FUNCPTR (gst_speex_resample_get_unit_size);
- GST_BASE_TRANSFORM_CLASS (klass)->transform_caps =
- GST_DEBUG_FUNCPTR (gst_speex_resample_transform_caps);
- GST_BASE_TRANSFORM_CLASS (klass)->set_caps =
- GST_DEBUG_FUNCPTR (gst_speex_resample_set_caps);
- GST_BASE_TRANSFORM_CLASS (klass)->transform =
- GST_DEBUG_FUNCPTR (gst_speex_resample_transform);
- GST_BASE_TRANSFORM_CLASS (klass)->event =
- GST_DEBUG_FUNCPTR (gst_speex_resample_event);
-
- GST_BASE_TRANSFORM_CLASS (klass)->passthrough_on_same_caps = TRUE;
-}
-
-static void
-gst_speex_resample_init (GstSpeexResample * resample,
- GstSpeexResampleClass * klass)
-{
- GstBaseTransform *trans = GST_BASE_TRANSFORM (resample);
-
- resample->quality = SPEEX_RESAMPLER_QUALITY_DEFAULT;
-
- resample->need_discont = FALSE;
-
- gst_pad_set_query_function (trans->srcpad, gst_speex_resample_query);
- gst_pad_set_query_type_function (trans->srcpad,
- gst_speex_resample_query_type);
-}
-
-/* vmethods */
-static gboolean
-gst_speex_resample_start (GstBaseTransform * base)
-{
- GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
-
- resample->ts_offset = -1;
- resample->offset = -1;
- resample->next_ts = -1;
-
- return TRUE;
-}
-
-static gboolean
-gst_speex_resample_stop (GstBaseTransform * base)
-{
- GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
-
- if (resample->state) {
- resample_resampler_destroy (resample->state);
- resample->state = NULL;
- }
-
- gst_caps_replace (&resample->sinkcaps, NULL);
- gst_caps_replace (&resample->srccaps, NULL);
-
- return TRUE;
-}
-
-static gboolean
-gst_speex_resample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
- guint * size)
-{
- gint width, channels;
- GstStructure *structure;
- gboolean ret;
-
- g_return_val_if_fail (size != NULL, FALSE);
-
- /* this works for both float and int */
- structure = gst_caps_get_structure (caps, 0);
- ret = gst_structure_get_int (structure, "width", &width);
- ret &= gst_structure_get_int (structure, "channels", &channels);
- g_return_val_if_fail (ret, FALSE);
-
- *size = width * channels / 8;
-
- return TRUE;
-}
-
-static GstCaps *
-gst_speex_resample_transform_caps (GstBaseTransform * base,
- GstPadDirection direction, GstCaps * caps)
-{
- GstCaps *res;
- GstStructure *structure;
-
- /* transform caps gives one single caps so we can just replace
- * the rate property with our range. */
- res = gst_caps_copy (caps);
- structure = gst_caps_get_structure (res, 0);
- gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
-
- return res;
-}
-
-static SpeexResamplerState *
-gst_speex_resample_init_state (guint channels, guint inrate, guint outrate,
- guint quality, gboolean fp)
-{
- SpeexResamplerState *ret = NULL;
- gint err = RESAMPLER_ERR_SUCCESS;
-
- if (fp)
- ret =
- resample_float_resampler_init (channels, inrate, outrate, quality,
- &err);
- else
- ret =
- resample_int_resampler_init (channels, inrate, outrate, quality, &err);
-
- if (err != RESAMPLER_ERR_SUCCESS) {
- GST_ERROR ("Failed to create resampler state: %s",
- resample_resampler_strerror (err));
- return NULL;
- }
-
- if (fp)
- resample_float_resampler_skip_zeros (ret);
- else
- resample_int_resampler_skip_zeros (ret);
-
- return ret;
-}
-
-static gboolean
-gst_speex_resample_update_state (GstSpeexResample * resample, gint channels,
- gint inrate, gint outrate, gint quality, gboolean fp)
-{
- gboolean ret = TRUE;
- gboolean updated_latency = FALSE;
-
- updated_latency = (resample->inrate != inrate
- || quality != resample->quality) && resample->state != NULL;
-
- if (resample->state == NULL) {
- ret = TRUE;
- } else if (resample->channels != channels || fp != resample->fp) {
- resample_resampler_destroy (resample->state);
- resample->state =
- gst_speex_resample_init_state (channels, inrate, outrate, quality, fp);
-
- ret = (resample->state != NULL);
- } else if (resample->inrate != inrate || resample->outrate != outrate) {
- gint err = RESAMPLER_ERR_SUCCESS;
-
- if (fp)
- err =
- resample_float_resampler_set_rate (resample->state, inrate, outrate);
- else
- err = resample_int_resampler_set_rate (resample->state, inrate, outrate);
-
- if (err != RESAMPLER_ERR_SUCCESS)
- GST_ERROR ("Failed to update rate: %s",
- resample_resampler_strerror (err));
-
- ret = (err == RESAMPLER_ERR_SUCCESS);
- } else if (quality != resample->quality) {
- gint err = RESAMPLER_ERR_SUCCESS;
-
- if (fp)
- err = resample_float_resampler_set_quality (resample->state, quality);
- else
- err = resample_int_resampler_set_quality (resample->state, quality);
-
- if (err != RESAMPLER_ERR_SUCCESS)
- GST_ERROR ("Failed to update quality: %s",
- resample_resampler_strerror (err));
-
- ret = (err == RESAMPLER_ERR_SUCCESS);
- }
-
- resample->channels = channels;
- resample->fp = fp;
- resample->quality = quality;
- resample->inrate = inrate;
- resample->outrate = outrate;
-
- if (updated_latency)
- gst_element_post_message (GST_ELEMENT (resample),
- gst_message_new_latency (GST_OBJECT (resample)));
-
- return ret;
-}
-
-static void
-gst_speex_resample_reset_state (GstSpeexResample * resample)
-{
- if (resample->state && resample->fp)
- resample_float_resampler_reset_mem (resample->state);
- else if (resample->state && !resample->fp)
- resample_int_resampler_reset_mem (resample->state);
-}
-
-static gboolean
-gst_speex_resample_parse_caps (GstCaps * incaps,
- GstCaps * outcaps, gint * channels, gint * inrate, gint * outrate,
- gboolean * fp)
-{
- GstStructure *structure;
- gboolean ret;
- gint myinrate, myoutrate, mychannels;
- gboolean myfp;
-
- GST_DEBUG ("incaps %" GST_PTR_FORMAT ", outcaps %"
- GST_PTR_FORMAT, incaps, outcaps);
-
- structure = gst_caps_get_structure (incaps, 0);
-
- if (g_str_equal (gst_structure_get_name (structure), "audio/x-raw-float"))
- myfp = TRUE;
- else
- myfp = FALSE;
-
- ret = gst_structure_get_int (structure, "rate", &myinrate);
- ret &= gst_structure_get_int (structure, "channels", &mychannels);
- if (!ret)
- goto no_in_rate_channels;
-
- structure = gst_caps_get_structure (outcaps, 0);
- ret = gst_structure_get_int (structure, "rate", &myoutrate);
- if (!ret)
- goto no_out_rate;
-
- if (channels)
- *channels = mychannels;
- if (inrate)
- *inrate = myinrate;
- if (outrate)
- *outrate = myoutrate;
-
- if (fp)
- *fp = myfp;
-
- return TRUE;
-
- /* ERRORS */
-no_in_rate_channels:
- {
- GST_DEBUG ("could not get input rate and channels");
- return FALSE;
- }
-no_out_rate:
- {
- GST_DEBUG ("could not get output rate");
- return FALSE;
- }
-}
-
-static gboolean
-gst_speex_resample_transform_size (GstBaseTransform * base,
- GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps,
- guint * othersize)
-{
- GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
- SpeexResamplerState *state;
- GstCaps *srccaps, *sinkcaps;
- gboolean use_internal = FALSE; /* whether we use the internal state */
- gboolean ret = TRUE;
- guint32 ratio_den, ratio_num;
- gboolean fp;
-
- GST_LOG ("asked to transform size %d in direction %s",
- size, direction == GST_PAD_SINK ? "SINK" : "SRC");
- if (direction == GST_PAD_SINK) {
- sinkcaps = caps;
- srccaps = othercaps;
- } else {
- sinkcaps = othercaps;
- srccaps = caps;
- }
-
- /* if the caps are the ones that _set_caps got called with; we can use
- * our own state; otherwise we'll have to create a state */
- if (resample->state && gst_caps_is_equal (sinkcaps, resample->sinkcaps) &&
- gst_caps_is_equal (srccaps, resample->srccaps)) {
- use_internal = TRUE;
- state = resample->state;
- fp = resample->fp;
- } else {
- gint inrate, outrate, channels;
-
- GST_DEBUG ("Can't use internal state, creating state");
-
- ret =
- gst_speex_resample_parse_caps (caps, othercaps, &channels, &inrate,
- &outrate, &fp);
-
- if (!ret) {
- GST_ERROR ("Wrong caps");
- return FALSE;
- }
-
- state = gst_speex_resample_init_state (channels, inrate, outrate, 0, TRUE);
- if (!state)
- return FALSE;
- }
-
- if (resample->fp || use_internal)
- resample_float_resampler_get_ratio (state, &ratio_num, &ratio_den);
- else
- resample_int_resampler_get_ratio (state, &ratio_num, &ratio_den);
-
- if (direction == GST_PAD_SINK) {
- gint fac = (fp) ? 4 : 2;
-
- /* asked to convert size of an incoming buffer */
- size /= fac;
- *othersize = (size * ratio_den + (ratio_num >> 1)) / ratio_num;
- *othersize *= fac;
- size *= fac;
- } else {
- gint fac = (fp) ? 4 : 2;
-
- /* asked to convert size of an outgoing buffer */
- size /= fac;
- *othersize = (size * ratio_num + (ratio_den >> 1)) / ratio_den;
- *othersize *= fac;
- size *= fac;
- }
-
- GST_LOG ("transformed size %d to %d", size, *othersize);
-
- if (!use_internal)
- resample_resampler_destroy (state);
-
- return ret;
-}
-
-static gboolean
-gst_speex_resample_set_caps (GstBaseTransform * base, GstCaps * incaps,
- GstCaps * outcaps)
-{
- gboolean ret;
- gint inrate = 0, outrate = 0, channels = 0;
- gboolean fp = FALSE;
- GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
-
- GST_LOG ("incaps %" GST_PTR_FORMAT ", outcaps %"
- GST_PTR_FORMAT, incaps, outcaps);
-
- ret = gst_speex_resample_parse_caps (incaps, outcaps,
- &channels, &inrate, &outrate, &fp);
-
- g_return_val_if_fail (ret, FALSE);
-
- ret =
- gst_speex_resample_update_state (resample, channels, inrate, outrate,
- resample->quality, fp);
-
- g_return_val_if_fail (ret, FALSE);
-
- /* save caps so we can short-circuit in the size_transform if the caps
- * are the same */
- gst_caps_replace (&resample->sinkcaps, incaps);
- gst_caps_replace (&resample->srccaps, outcaps);
-
- return TRUE;
-}
-
-static void
-gst_speex_resample_push_drain (GstSpeexResample * resample)
-{
- GstBuffer *buf;
- GstBaseTransform *trans = GST_BASE_TRANSFORM (resample);
- GstFlowReturn res;
- gint outsize;
- guint out_len, out_processed;
- gint err;
-
- if (!resample->state)
- return;
-
- if (resample->fp) {
- guint num, den;
-
- resample_float_resampler_get_ratio (resample->state, &num, &den);
-
- out_len = resample_float_resampler_get_input_latency (resample->state);
- out_len = out_processed = (out_len * den + (num >> 1)) / num;
- outsize = 4 * out_len * resample->channels;
- } else {
- guint num, den;
-
- resample_int_resampler_get_ratio (resample->state, &num, &den);
-
- out_len = resample_int_resampler_get_input_latency (resample->state);
- out_len = out_processed = (out_len * den + (num >> 1)) / num;
- outsize = 2 * out_len * resample->channels;
- }
-
- res = gst_pad_alloc_buffer (trans->srcpad, GST_BUFFER_OFFSET_NONE, outsize,
- GST_PAD_CAPS (trans->srcpad), &buf);
-
- if (G_UNLIKELY (res != GST_FLOW_OK)) {
- GST_WARNING ("failed allocating buffer of %d bytes", outsize);
- return;
- }
-
- if (resample->fp) {
- guint len = resample_float_resampler_get_input_latency (resample->state);
-
- err =
- resample_float_resampler_process_interleaved_float (resample->state,
- NULL, &len, (gfloat *) GST_BUFFER_DATA (buf), &out_processed);
- } else {
- guint len = resample_int_resampler_get_input_latency (resample->state);
-
- err =
- resample_int_resampler_process_interleaved_int (resample->state, NULL,
- &len, (gint16 *) GST_BUFFER_DATA (buf), &out_processed);
- }
-
- if (err != RESAMPLER_ERR_SUCCESS) {
- GST_WARNING ("Failed to process drain: %s",
- resample_resampler_strerror (err));
- gst_buffer_unref (buf);
- return;
- }
-
- if (out_processed == 0) {
- GST_WARNING ("Failed to get drain, dropping buffer");
- gst_buffer_unref (buf);
- return;
- }
-
- GST_BUFFER_OFFSET (buf) = resample->offset;
- GST_BUFFER_TIMESTAMP (buf) = resample->next_ts;
- GST_BUFFER_SIZE (buf) =
- out_processed * resample->channels * ((resample->fp) ? 4 : 2);
-
- if (resample->ts_offset != -1) {
- resample->offset += out_processed;
- resample->ts_offset += out_processed;
- resample->next_ts =
- GST_FRAMES_TO_CLOCK_TIME (resample->ts_offset, resample->outrate);
- GST_BUFFER_OFFSET_END (buf) = resample->offset;
-
- /* we calculate DURATION as the difference between "next" timestamp
- * and current timestamp so we ensure a contiguous stream, instead of
- * having rounding errors. */
- GST_BUFFER_DURATION (buf) = resample->next_ts - GST_BUFFER_TIMESTAMP (buf);
- } else {
- /* no valid offset know, we can still sortof calculate the duration though */
- GST_BUFFER_DURATION (buf) =
- GST_FRAMES_TO_CLOCK_TIME (out_processed, resample->outrate);
- }
-
- GST_LOG ("Pushing drain buffer of %u bytes with timestamp %" GST_TIME_FORMAT
- " duration %" GST_TIME_FORMAT " offset %" G_GUINT64_FORMAT
- " offset_end %" G_GUINT64_FORMAT,
- GST_BUFFER_SIZE (buf),
- GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
- GST_TIME_ARGS (GST_BUFFER_DURATION (buf)),
- GST_BUFFER_OFFSET (buf), GST_BUFFER_OFFSET_END (buf));
-
- res = gst_pad_push (trans->srcpad, buf);
-
- if (res != GST_FLOW_OK)
- GST_WARNING ("Failed to push drain");
-
- return;
-}
-
-static gboolean
-gst_speex_resample_event (GstBaseTransform * base, GstEvent * event)
-{
- GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
-
- switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_FLUSH_START:
- break;
- case GST_EVENT_FLUSH_STOP:
- gst_speex_resample_reset_state (resample);
- resample->ts_offset = -1;
- resample->next_ts = -1;
- resample->offset = -1;
- case GST_EVENT_NEWSEGMENT:
- gst_speex_resample_push_drain (resample);
- gst_speex_resample_reset_state (resample);
- resample->ts_offset = -1;
- resample->next_ts = -1;
- resample->offset = -1;
- break;
- case GST_EVENT_EOS:{
- gst_speex_resample_push_drain (resample);
- gst_speex_resample_reset_state (resample);
- break;
- }
- default:
- break;
- }
- parent_class->event (base, event);
-
- return TRUE;
-}
-
-static gboolean
-gst_speex_resample_check_discont (GstSpeexResample * resample,
- GstClockTime timestamp)
-{
- if (timestamp != GST_CLOCK_TIME_NONE &&
- resample->prev_ts != GST_CLOCK_TIME_NONE &&
- resample->prev_duration != GST_CLOCK_TIME_NONE &&
- timestamp != resample->prev_ts + resample->prev_duration) {
- /* Potentially a discontinuous buffer. However, it turns out that many
- * elements generate imperfect streams due to rounding errors, so we permit
- * a small error (up to one sample) without triggering a filter
- * flush/restart (if triggered incorrectly, this will be audible) */
- GstClockTimeDiff diff = timestamp -
- (resample->prev_ts + resample->prev_duration);
-
- if (ABS (diff) > GST_SECOND / resample->inrate) {
- GST_WARNING ("encountered timestamp discontinuity of %" G_GINT64_FORMAT,
- diff);
- return TRUE;
- }
- }
-
- return FALSE;
-}
-
-static void
-gst_speex_fix_output_buffer (GstSpeexResample * resample, GstBuffer * outbuf,
- guint diff)
-{
- GstClockTime timediff = GST_FRAMES_TO_CLOCK_TIME (diff, resample->outrate);
-
- GST_LOG ("Adjusting buffer by %d samples", diff);
-
- GST_BUFFER_DURATION (outbuf) -= timediff;
- GST_BUFFER_SIZE (outbuf) -=
- diff * ((resample->fp) ? 4 : 2) * resample->channels;
-
- if (resample->ts_offset != -1) {
- GST_BUFFER_OFFSET_END (outbuf) -= diff;
- resample->offset -= diff;
- resample->ts_offset -= diff;
- resample->next_ts =
- GST_FRAMES_TO_CLOCK_TIME (resample->ts_offset, resample->outrate);
- }
-}
-
-static GstFlowReturn
-gst_speex_resample_process (GstSpeexResample * resample, GstBuffer * inbuf,
- GstBuffer * outbuf)
-{
- guint32 in_len, in_processed;
- guint32 out_len, out_processed;
- gint err = RESAMPLER_ERR_SUCCESS;
-
- in_len = GST_BUFFER_SIZE (inbuf) / resample->channels;
- out_len = GST_BUFFER_SIZE (outbuf) / resample->channels;
-
- if (resample->fp) {
- in_len /= 4;
- out_len /= 4;
- } else {
- in_len /= 2;
- out_len /= 2;
- }
-
- in_processed = in_len;
- out_processed = out_len;
-
- if (resample->fp)
- err = resample_float_resampler_process_interleaved_float (resample->state,
- (const gfloat *) GST_BUFFER_DATA (inbuf), &in_processed,
- (gfloat *) GST_BUFFER_DATA (outbuf), &out_processed);
- else
- err = resample_int_resampler_process_interleaved_int (resample->state,
- (const gint16 *) GST_BUFFER_DATA (inbuf), &in_processed,
- (gint16 *) GST_BUFFER_DATA (outbuf), &out_processed);
-
- if (in_len != in_processed)
- GST_WARNING ("Converted %d of %d input samples", in_processed, in_len);
-
- if (out_len != out_processed) {
- /* One sample difference is allowed as this will happen
- * because of rounding errors */
- if (out_processed == 0) {
- GST_DEBUG ("Converted to 0 samples, buffer dropped");
-
- if (resample->ts_offset != -1) {
- GST_BUFFER_OFFSET_END (outbuf) -= out_len;
- resample->offset -= out_len;
- resample->ts_offset -= out_len;
- resample->next_ts =
- GST_FRAMES_TO_CLOCK_TIME (resample->ts_offset, resample->outrate);
- }
-
- return GST_BASE_TRANSFORM_FLOW_DROPPED;
- } else if (out_len - out_processed != 1)
- GST_WARNING ("Converted to %d instead of %d output samples",
- out_processed, out_len);
- if (out_len > out_processed) {
- gst_speex_fix_output_buffer (resample, outbuf, out_len - out_processed);
- } else {
- GST_ERROR ("Wrote more output than allocated!");
- return GST_FLOW_ERROR;
- }
- }
-
- if (err != RESAMPLER_ERR_SUCCESS) {
- GST_ERROR ("Failed to convert data: %s", resample_resampler_strerror (err));
- return GST_FLOW_ERROR;
- } else {
- GST_LOG ("Converted to buffer of %u bytes with timestamp %" GST_TIME_FORMAT
- ", duration %" GST_TIME_FORMAT ", offset %" G_GUINT64_FORMAT
- ", offset_end %" G_GUINT64_FORMAT,
- GST_BUFFER_SIZE (outbuf),
- GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
- GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
- GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf));
- return GST_FLOW_OK;
- }
-}
-
-static GstFlowReturn
-gst_speex_resample_transform (GstBaseTransform * base, GstBuffer * inbuf,
- GstBuffer * outbuf)
-{
- GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
- guint8 *data;
- gulong size;
- GstClockTime timestamp;
- gint outsamples;
-
- if (resample->state == NULL)
- if (!(resample->state = gst_speex_resample_init_state (resample->channels,
- resample->inrate, resample->outrate, resample->quality,
- resample->fp)))
- return GST_FLOW_ERROR;
-
- data = GST_BUFFER_DATA (inbuf);
- size = GST_BUFFER_SIZE (inbuf);
- timestamp = GST_BUFFER_TIMESTAMP (inbuf);
-
- GST_LOG ("transforming buffer of %ld bytes, ts %"
- GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
- G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
- size, GST_TIME_ARGS (timestamp),
- GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)),
- GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf));
-
- /* check for timestamp discontinuities and flush/reset if needed */
- if (G_UNLIKELY (gst_speex_resample_check_discont (resample, timestamp)
- || GST_BUFFER_IS_DISCONT (inbuf))) {
- /* Flush internal samples */
- gst_speex_resample_reset_state (resample);
- /* Inform downstream element about discontinuity */
- resample->need_discont = TRUE;
- /* We want to recalculate the offset */
- resample->ts_offset = -1;
- }
-
- outsamples = GST_BUFFER_SIZE (outbuf) / resample->channels;
- outsamples /= (resample->fp) ? 4 : 2;
-
- if (resample->ts_offset == -1) {
- /* if we don't know the initial offset yet, calculate it based on the
- * input timestamp. */
- if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
- GstClockTime stime;
-
- /* offset used to calculate the timestamps. We use the sample offset for
- * this to make it more accurate. We want the first buffer to have the
- * same timestamp as the incoming timestamp. */
- resample->next_ts = timestamp;
- resample->ts_offset =
- GST_CLOCK_TIME_TO_FRAMES (timestamp, resample->outrate);
- /* offset used to set as the buffer offset, this offset is always
- * relative to the stream time, note that timestamp is not... */
- stime = (timestamp - base->segment.start) + base->segment.time;
- resample->offset = GST_CLOCK_TIME_TO_FRAMES (stime, resample->outrate);
- }
- }
- resample->prev_ts = timestamp;
- resample->prev_duration = GST_BUFFER_DURATION (inbuf);
-
- GST_BUFFER_OFFSET (outbuf) = resample->offset;
- GST_BUFFER_TIMESTAMP (outbuf) = resample->next_ts;
-
- if (resample->ts_offset != -1) {
- resample->offset += outsamples;
- resample->ts_offset += outsamples;
- resample->next_ts =
- GST_FRAMES_TO_CLOCK_TIME (resample->ts_offset, resample->outrate);
- GST_BUFFER_OFFSET_END (outbuf) = resample->offset;
-
- /* we calculate DURATION as the difference between "next" timestamp
- * and current timestamp so we ensure a contiguous stream, instead of
- * having rounding errors. */
- GST_BUFFER_DURATION (outbuf) = resample->next_ts -
- GST_BUFFER_TIMESTAMP (outbuf);
- } else {
- /* no valid offset know, we can still sortof calculate the duration though */
- GST_BUFFER_DURATION (outbuf) =
- GST_FRAMES_TO_CLOCK_TIME (outsamples, resample->outrate);
- }
-
- if (G_UNLIKELY (resample->need_discont)) {
- GST_DEBUG ("marking this buffer with the DISCONT flag");
- GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
- resample->need_discont = FALSE;
- }
-
- return gst_speex_resample_process (resample, inbuf, outbuf);
-}
-
-static gboolean
-gst_speex_resample_query (GstPad * pad, GstQuery * query)
-{
- GstSpeexResample *resample = GST_SPEEX_RESAMPLE (gst_pad_get_parent (pad));
- GstBaseTransform *trans = GST_BASE_TRANSFORM (resample);
- gboolean res = TRUE;
-
- switch (GST_QUERY_TYPE (query)) {
- case GST_QUERY_LATENCY:
- {
- GstClockTime min, max;
- gboolean live;
- guint64 latency;
- GstPad *peer;
- gint rate = resample->inrate;
- gint resampler_latency;
-
- if (resample->state && resample->fp)
- resampler_latency =
- resample_float_resampler_get_input_latency (resample->state);
- else if (resample->state && !resample->fp)
- resampler_latency =
- resample_int_resampler_get_input_latency (resample->state);
- else
- resampler_latency = 0;
-
- if (gst_base_transform_is_passthrough (trans))
- resampler_latency = 0;
-
- if ((peer = gst_pad_get_peer (trans->sinkpad))) {
- if ((res = gst_pad_query (peer, query))) {
- gst_query_parse_latency (query, &live, &min, &max);
-
- GST_DEBUG ("Peer latency: min %"
- GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
- GST_TIME_ARGS (min), GST_TIME_ARGS (max));
-
- /* add our own latency */
- if (rate != 0 && resampler_latency != 0)
- latency =
- gst_util_uint64_scale (resampler_latency, GST_SECOND, rate);
- else
- latency = 0;
-
- GST_DEBUG ("Our latency: %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
-
- min += latency;
- if (max != GST_CLOCK_TIME_NONE)
- max += latency;
-
- GST_DEBUG ("Calculated total latency : min %"
- GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
- GST_TIME_ARGS (min), GST_TIME_ARGS (max));
-
- gst_query_set_latency (query, live, min, max);
- }
- gst_object_unref (peer);
- }
- break;
- }
- default:
- res = gst_pad_query_default (pad, query);
- break;
- }
- gst_object_unref (resample);
- return res;
-}
-
-static const GstQueryType *
-gst_speex_resample_query_type (GstPad * pad)
-{
- static const GstQueryType types[] = {
- GST_QUERY_LATENCY,
- 0
- };
-
- return types;
-}
-
-static void
-gst_speex_resample_set_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec)
-{
- GstSpeexResample *resample;
-
- resample = GST_SPEEX_RESAMPLE (object);
-
- switch (prop_id) {
- case PROP_QUALITY:
- resample->quality = g_value_get_int (value);
- GST_DEBUG ("new quality %d", resample->quality);
-
- gst_speex_resample_update_state (resample, resample->channels,
- resample->inrate, resample->outrate, resample->quality, resample->fp);
- break;
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
- }
-}
-
-static void
-gst_speex_resample_get_property (GObject * object, guint prop_id,
- GValue * value, GParamSpec * pspec)
-{
- GstSpeexResample *resample;
-
- resample = GST_SPEEX_RESAMPLE (object);
-
- switch (prop_id) {
- case PROP_QUALITY:
- g_value_set_int (value, resample->quality);
- break;
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
- }
-}
-
-
-static gboolean
-plugin_init (GstPlugin * plugin)
-{
- if (!gst_element_register (plugin, "speexresample", GST_RANK_NONE,
- GST_TYPE_SPEEX_RESAMPLE)) {
- return FALSE;
- }
-
- return TRUE;
-}
-
-GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
- GST_VERSION_MINOR,
- "speexresample",
- "Resamples audio", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME,
- GST_PACKAGE_ORIGIN);
diff --git a/gst/speexresample/gstspeexresample.h b/gst/speexresample/gstspeexresample.h
deleted file mode 100644
index b5abf0e6..00000000
--- a/gst/speexresample/gstspeexresample.h
+++ /dev/null
@@ -1,82 +0,0 @@
-/* GStreamer
- * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
- * Copyright (C) <2007> Sebastian Dröge <slomo@circular-chaos.org>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-
-#ifndef __SPEEX_RESAMPLE_H__
-#define __SPEEX_RESAMPLE_H__
-
-#include <gst/gst.h>
-#include <gst/base/gstbasetransform.h>
-
-#include "speex_resampler_wrapper.h"
-
-G_BEGIN_DECLS
-
-#define GST_TYPE_SPEEX_RESAMPLE \
- (gst_speex_resample_get_type())
-#define GST_SPEEX_RESAMPLE(obj) \
- (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_SPEEX_RESAMPLE,GstSpeexResample))
-#define GST_SPEEX_RESAMPLE_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_SPEEX_RESAMPLE,GstSpeexResampleClass))
-#define GST_IS_SPEEX_RESAMPLE(obj) \
- (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_SPEEX_RESAMPLE))
-#define GST_IS_SPEEX_RESAMPLE_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_SPEEX_RESAMPLE))
-
-typedef struct _GstSpeexResample GstSpeexResample;
-typedef struct _GstSpeexResampleClass GstSpeexResampleClass;
-
-/**
- * GstSpeexResample:
- *
- * Opaque data structure.
- */
-struct _GstSpeexResample {
- GstBaseTransform element;
-
- /* <private> */
-
- GstCaps *srccaps, *sinkcaps;
-
- gboolean need_discont;
-
- guint64 offset;
- guint64 ts_offset;
- GstClockTime next_ts;
- GstClockTime prev_ts, prev_duration;
-
- gboolean fp;
- int channels;
- int inrate;
- int outrate;
- int quality;
-
- SpeexResamplerState *state;
-};
-
-struct _GstSpeexResampleClass {
- GstBaseTransformClass parent_class;
-};
-
-GType gst_speex_resample_get_type(void);
-
-G_END_DECLS
-
-#endif /* __SPEEX_RESAMPLE_H__ */
diff --git a/gst/speexresample/resample.c b/gst/speexresample/resample.c
deleted file mode 100644
index 29b95f23..00000000
--- a/gst/speexresample/resample.c
+++ /dev/null
@@ -1,1372 +0,0 @@
-/* Copyright (C) 2007 Jean-Marc Valin
-
- 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.
-*/
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#ifdef OUTSIDE_SPEEX
-#include <stdlib.h>
-#include <glib.h>
-
-static inline void *
-speex_alloc (int size)
-{
- return g_malloc0 (size);
-}
-static inline void *
-speex_realloc (void *ptr, int size)
-{
- return g_realloc (ptr, size);
-}
-
-static inline 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 <math.h>
-
-#ifndef M_PI
-#define M_PI 3.14159263
-#endif
-
-#ifdef FIXED_POINT
-#define WORD2INT(x) ((x) < -32767 ? -32768 : ((x) > 32766 ? 32767 : (x)))
-#else
-#define WORD2INT(x) ((x) < -32767.5f ? -32768 : ((x) > 32766.5f ? 32767 : floor(.5+(x))))
-#endif
-
-/*#define float double*/
-#define FILTER_SIZE 64
-#define OVERSAMPLE 8
-
-#define IMAX(a,b) ((a) > (b) ? (a) : (b))
-#define IMIN(a,b) ((a) < (b) ? (a) : (b))
-
-#ifndef NULL
-#define NULL 0
-#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;
- int int_advance;
- int frac_advance;
- float cutoff;
- spx_uint32_t oversample;
- int initialised;
- int started;
-
- /* These are per-channel */
- spx_int32_t *last_sample;
- spx_uint32_t *samp_frac_num;
- spx_uint32_t *magic_samples;
-
- spx_word16_t *mem;
- spx_word16_t *sinc_table;
- spx_uint32_t sinc_table_length;
- resampler_basic_func resampler_ptr;
-
- int in_stride;
- int out_stride;
-};
-
-static double kaiser12_table[68] = {
- 0.99859849, 1.00000000, 0.99859849, 0.99440475, 0.98745105, 0.97779076,
- 0.96549770, 0.95066529, 0.93340547, 0.91384741, 0.89213598, 0.86843014,
- 0.84290116, 0.81573067, 0.78710866, 0.75723148, 0.72629970, 0.69451601,
- 0.66208321, 0.62920216, 0.59606986, 0.56287762, 0.52980938, 0.49704014,
- 0.46473455, 0.43304576, 0.40211431, 0.37206735, 0.34301800, 0.31506490,
- 0.28829195, 0.26276832, 0.23854851, 0.21567274, 0.19416736, 0.17404546,
- 0.15530766, 0.13794294, 0.12192957, 0.10723616, 0.09382272, 0.08164178,
- 0.07063950, 0.06075685, 0.05193064, 0.04409466, 0.03718069, 0.03111947,
- 0.02584161, 0.02127838, 0.01736250, 0.01402878, 0.01121463, 0.00886058,
- 0.00691064, 0.00531256, 0.00401805, 0.00298291, 0.00216702, 0.00153438,
- 0.00105297, 0.00069463, 0.00043489, 0.00025272, 0.00013031, 0.0000527734,
- 0.00001000, 0.00000000
-};
-
-/*
-static double kaiser12_table[36] = {
- 0.99440475, 1.00000000, 0.99440475, 0.97779076, 0.95066529, 0.91384741,
- 0.86843014, 0.81573067, 0.75723148, 0.69451601, 0.62920216, 0.56287762,
- 0.49704014, 0.43304576, 0.37206735, 0.31506490, 0.26276832, 0.21567274,
- 0.17404546, 0.13794294, 0.10723616, 0.08164178, 0.06075685, 0.04409466,
- 0.03111947, 0.02127838, 0.01402878, 0.00886058, 0.00531256, 0.00298291,
- 0.00153438, 0.00069463, 0.00025272, 0.0000527734, 0.00000500, 0.00000000};
-*/
-static double kaiser10_table[36] = {
- 0.99537781, 1.00000000, 0.99537781, 0.98162644, 0.95908712, 0.92831446,
- 0.89005583, 0.84522401, 0.79486424, 0.74011713, 0.68217934, 0.62226347,
- 0.56155915, 0.50119680, 0.44221549, 0.38553619, 0.33194107, 0.28205962,
- 0.23636152, 0.19515633, 0.15859932, 0.12670280, 0.09935205, 0.07632451,
- 0.05731132, 0.04193980, 0.02979584, 0.02044510, 0.01345224, 0.00839739,
- 0.00488951, 0.00257636, 0.00115101, 0.00035515, 0.00000000, 0.00000000
-};
-
-static double kaiser8_table[36] = {
- 0.99635258, 1.00000000, 0.99635258, 0.98548012, 0.96759014, 0.94302200,
- 0.91223751, 0.87580811, 0.83439927, 0.78875245, 0.73966538, 0.68797126,
- 0.63451750, 0.58014482, 0.52566725, 0.47185369, 0.41941150, 0.36897272,
- 0.32108304, 0.27619388, 0.23465776, 0.19672670, 0.16255380, 0.13219758,
- 0.10562887, 0.08273982, 0.06335451, 0.04724088, 0.03412321, 0.02369490,
- 0.01563093, 0.00959968, 0.00527363, 0.00233883, 0.00050000, 0.00000000
-};
-
-static double kaiser6_table[36] = {
- 0.99733006, 1.00000000, 0.99733006, 0.98935595, 0.97618418, 0.95799003,
- 0.93501423, 0.90755855, 0.87598009, 0.84068475, 0.80211977, 0.76076565,
- 0.71712752, 0.67172623, 0.62508937, 0.57774224, 0.53019925, 0.48295561,
- 0.43647969, 0.39120616, 0.34752997, 0.30580127, 0.26632152, 0.22934058,
- 0.19505503, 0.16360756, 0.13508755, 0.10953262, 0.08693120, 0.06722600,
- 0.05031820, 0.03607231, 0.02432151, 0.01487334, 0.00752000, 0.00000000
-};
-
-struct FuncDef
-{
- double *table;
- int oversample;
-};
-
-static struct FuncDef _KAISER12 = { kaiser12_table, 64 };
-
-#define KAISER12 (&_KAISER12)
-/*static struct FuncDef _KAISER12 = {kaiser12_table, 32};
-#define KAISER12 (&_KAISER12)*/
-static struct FuncDef _KAISER10 = { kaiser10_table, 32 };
-
-#define KAISER10 (&_KAISER10)
-static struct FuncDef _KAISER8 = { kaiser8_table, 32 };
-
-#define KAISER8 (&_KAISER8)
-static struct FuncDef _KAISER6 = { kaiser6_table, 32 };
-
-#define KAISER6 (&_KAISER6)
-
-struct QualityMapping
-{
- int base_length;
- int oversample;
- float downsample_bandwidth;
- float upsample_bandwidth;
- struct FuncDef *window_func;
-};
-
-
-/* This table maps conversion quality to internal parameters. There are two
- reasons that explain why the up-sampling bandwidth is larger than the
- down-sampling bandwidth:
- 1) When up-sampling, we can assume that the spectrum is already attenuated
- close to the Nyquist rate (from an A/D or a previous resampling filter)
- 2) Any aliasing that occurs very close to the Nyquist rate will be masked
- by the sinusoids/noise just below the Nyquist rate (guaranteed only for
- up-sampling).
-*/
-static const struct QualityMapping quality_map[11] = {
- {8, 4, 0.830f, 0.860f, KAISER6}, /* Q0 */
- {16, 4, 0.850f, 0.880f, KAISER6}, /* Q1 */
- {32, 4, 0.882f, 0.910f, KAISER6}, /* Q2 *//* 82.3% cutoff ( ~60 dB stop) 6 */
- {48, 8, 0.895f, 0.917f, KAISER8}, /* Q3 *//* 84.9% cutoff ( ~80 dB stop) 8 */
- {64, 8, 0.921f, 0.940f, KAISER8}, /* Q4 *//* 88.7% cutoff ( ~80 dB stop) 8 */
- {80, 16, 0.922f, 0.940f, KAISER10}, /* Q5 *//* 89.1% cutoff (~100 dB stop) 10 */
- {96, 16, 0.940f, 0.945f, KAISER10}, /* Q6 *//* 91.5% cutoff (~100 dB stop) 10 */
- {128, 16, 0.950f, 0.950f, KAISER10}, /* Q7 *//* 93.1% cutoff (~100 dB stop) 10 */
- {160, 16, 0.960f, 0.960f, KAISER10}, /* Q8 *//* 94.5% cutoff (~100 dB stop) 10 */
- {192, 32, 0.968f, 0.968f, KAISER12}, /* Q9 *//* 95.5% cutoff (~100 dB stop) 10 */
- {256, 32, 0.975f, 0.975f, KAISER12}, /* Q10 *//* 96.6% cutoff (~100 dB stop) 10 */
-};
-
-/*8,24,40,56,80,104,128,160,200,256,320*/
-static double
-compute_func (float x, struct FuncDef *func)
-{
- float y, frac;
- double interp[4];
- int ind;
-
- y = x * func->oversample;
- ind = (int) floor (y);
- frac = (y - ind);
- /* CSE with handle the repeated powers */
- interp[3] = -0.1666666667 * frac + 0.1666666667 * (frac * frac * frac);
- interp[2] = frac + 0.5 * (frac * frac) - 0.5 * (frac * frac * frac);
- /*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac; */
- interp[0] =
- -0.3333333333 * frac + 0.5 * (frac * frac) -
- 0.1666666667 * (frac * frac * frac);
- /* Just to make sure we don't have rounding problems */
- interp[1] = 1.f - interp[3] - interp[2] - interp[0];
-
- /*sum = frac*accum[1] + (1-frac)*accum[2]; */
- return interp[0] * func->table[ind] + interp[1] * func->table[ind + 1] +
- interp[2] * func->table[ind + 2] + interp[3] * func->table[ind + 3];
-}
-
-#if 0
-#include <stdio.h>
-int
-main (int argc, char **argv)
-{
- int i;
-
- for (i = 0; i < 256; i++) {
- printf ("%f\n", compute_func (i / 256., KAISER12));
- }
- return 0;
-}
-#endif
-
-#ifdef FIXED_POINT
-/* The slow way of computing a sinc for the table. Should improve that some day */
-static spx_word16_t
-sinc (float cutoff, float x, int N, struct FuncDef *window_func)
-{
- /*fprintf (stderr, "%f ", x); */
- float xx = x * cutoff;
-
- if (fabs (x) < 1e-6f)
- return WORD2INT (32768. * cutoff);
- else if (fabs (x) > .5f * N)
- return 0;
- /*FIXME: Can it really be any slower than this? */
- return WORD2INT (32768. * cutoff * sin (M_PI * xx) / (M_PI * xx) *
- compute_func (fabs (2. * x / N), window_func));
-}
-#else
-/* The slow way of computing a sinc for the table. Should improve that some day */
-static spx_word16_t
-sinc (float cutoff, float x, int N, struct FuncDef *window_func)
-{
- /*fprintf (stderr, "%f ", x); */
- float xx = x * cutoff;
-
- if (fabs (x) < 1e-6)
- return cutoff;
- else if (fabs (x) > .5 * N)
- return 0;
- /*FIXME: Can it really be any slower than this? */
- return cutoff * sin (M_PI * xx) / (M_PI * xx) * compute_func (fabs (2. * x /
- N), window_func);
-}
-#endif
-
-#ifdef FIXED_POINT
-static void
-cubic_coef (spx_word16_t x, spx_word16_t interp[4])
-{
- /* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation
- but I know it's MMSE-optimal on a sinc */
- spx_word16_t x2, x3;
-
- x2 = MULT16_16_P15 (x, x);
- x3 = MULT16_16_P15 (x, x2);
- interp[0] =
- PSHR32 (MULT16_16 (QCONST16 (-0.16667f, 15),
- x) + MULT16_16 (QCONST16 (0.16667f, 15), x3), 15);
- interp[1] =
- EXTRACT16 (EXTEND32 (x) + SHR32 (SUB32 (EXTEND32 (x2), EXTEND32 (x3)),
- 1));
- interp[3] =
- PSHR32 (MULT16_16 (QCONST16 (-0.33333f, 15),
- x) + MULT16_16 (QCONST16 (.5f, 15),
- x2) - MULT16_16 (QCONST16 (0.16667f, 15), x3), 15);
- /* Just to make sure we don't have rounding problems */
- interp[2] = Q15_ONE - interp[0] - interp[1] - interp[3];
- if (interp[2] < 32767)
- interp[2] += 1;
-}
-#else
-static void
-cubic_coef (spx_word16_t frac, spx_word16_t interp[4])
-{
- /* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation
- but I know it's MMSE-optimal on a sinc */
- interp[0] = -0.16667f * frac + 0.16667f * frac * frac * frac;
- interp[1] = frac + 0.5f * frac * frac - 0.5f * frac * frac * frac;
- /*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac; */
- interp[3] =
- -0.33333f * frac + 0.5f * frac * frac - 0.16667f * frac * frac * frac;
- /* Just to make sure we don't have rounding problems */
- interp[2] = 1. - interp[0] - interp[1] - interp[3];
-}
-#endif
-
-static int
-resampler_basic_direct_single (SpeexResamplerState * st,
- spx_uint32_t channel_index, const spx_word16_t * in, spx_uint32_t * in_len,
- spx_word16_t * out, spx_uint32_t * out_len)
-{
- int N = st->filt_len;
- int out_sample = 0;
- spx_word16_t *mem;
- int last_sample = st->last_sample[channel_index];
- spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
-
- mem = st->mem + channel_index * st->mem_alloc_size;
- while (!(last_sample >= (spx_int32_t) * in_len
- || out_sample >= (spx_int32_t) * out_len)) {
- int j;
- spx_word32_t sum = 0;
-
- /* We already have all the filter coefficients pre-computed in the table */
- const spx_word16_t *ptr;
-
- /* Do the memory part */
- for (j = 0; last_sample - N + 1 + j < 0; j++) {
- sum +=
- MULT16_16 (mem[last_sample + j],
- st->sinc_table[samp_frac_num * st->filt_len + j]);
- }
-
- /* Do the new part */
- if (in != NULL) {
- ptr = in + st->in_stride * (last_sample - N + 1 + j);
- for (; j < N; j++) {
- sum +=
- MULT16_16 (*ptr, st->sinc_table[samp_frac_num * st->filt_len + j]);
- ptr += st->in_stride;
- }
- }
-
- *out = PSHR32 (sum, 15);
- out += st->out_stride;
- out_sample++;
- 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++;
- }
- }
- 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)
-{
- int N = st->filt_len;
- int out_sample = 0;
- spx_word16_t *mem;
- int last_sample = st->last_sample[channel_index];
- spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
-
- mem = st->mem + channel_index * st->mem_alloc_size;
- while (!(last_sample >= (spx_int32_t) * in_len
- || out_sample >= (spx_int32_t) * out_len)) {
- int j;
- double sum = 0;
-
- /* We already have all the filter coefficients pre-computed in the table */
- const spx_word16_t *ptr;
-
- /* Do the memory part */
- for (j = 0; last_sample - N + 1 + j < 0; j++) {
- sum +=
- MULT16_16 (mem[last_sample + j],
- (double) st->sinc_table[samp_frac_num * st->filt_len + j]);
- }
-
- /* Do the new part */
- if (in != NULL) {
- ptr = in + st->in_stride * (last_sample - N + 1 + j);
- for (; j < N; j++) {
- sum +=
- MULT16_16 (*ptr,
- (double) st->sinc_table[samp_frac_num * st->filt_len + j]);
- ptr += st->in_stride;
- }
- }
-
- *out = sum;
- out += st->out_stride;
- out_sample++;
- 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++;
- }
- }
- 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)
-{
- int N = st->filt_len;
- int out_sample = 0;
- spx_word16_t *mem;
- int last_sample = st->last_sample[channel_index];
- spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
-
- mem = st->mem + channel_index * st->mem_alloc_size;
- while (!(last_sample >= (spx_int32_t) * in_len
- || out_sample >= (spx_int32_t) * out_len)) {
- int j;
- spx_word32_t sum = 0;
-
- /* We need to interpolate the sinc filter */
- spx_word32_t accum[4] = { 0.f, 0.f, 0.f, 0.f };
- spx_word16_t interp[4];
- const spx_word16_t *ptr;
- int offset;
- spx_word16_t frac;
-
- offset = samp_frac_num * st->oversample / st->den_rate;
-#ifdef FIXED_POINT
- frac =
- PDIV32 (SHL32 ((samp_frac_num * st->oversample) % st->den_rate, 15),
- st->den_rate);
-#else
- frac =
- ((float) ((samp_frac_num * st->oversample) % st->den_rate)) /
- st->den_rate;
-#endif
- /* This code is written like this to make it easy to optimise with SIMD.
- For most DSPs, it would be best to split the loops in two because most DSPs
- have only two accumulators */
- for (j = 0; last_sample - N + 1 + j < 0; j++) {
- spx_word16_t curr_mem = mem[last_sample + j];
-
- accum[0] +=
- MULT16_16 (curr_mem,
- st->sinc_table[4 + (j + 1) * st->oversample - offset - 2]);
- accum[1] +=
- MULT16_16 (curr_mem,
- st->sinc_table[4 + (j + 1) * st->oversample - offset - 1]);
- accum[2] +=
- MULT16_16 (curr_mem,
- st->sinc_table[4 + (j + 1) * st->oversample - offset]);
- accum[3] +=
- MULT16_16 (curr_mem,
- st->sinc_table[4 + (j + 1) * st->oversample - offset + 1]);
- }
-
- if (in != NULL) {
- ptr = in + st->in_stride * (last_sample - N + 1 + j);
- /* Do the new part */
- for (; j < N; j++) {
- spx_word16_t curr_in = *ptr;
-
- ptr += st->in_stride;
- accum[0] +=
- MULT16_16 (curr_in,
- st->sinc_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]);
-
- *out = PSHR32 (sum, 15);
- out += st->out_stride;
- out_sample++;
- 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++;
- }
- }
- 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)
-{
- int N = st->filt_len;
- int out_sample = 0;
- spx_word16_t *mem;
- int last_sample = st->last_sample[channel_index];
- spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
-
- mem = st->mem + channel_index * st->mem_alloc_size;
- while (!(last_sample >= (spx_int32_t) * in_len
- || out_sample >= (spx_int32_t) * out_len)) {
- int j;
- spx_word32_t sum = 0;
-
- /* We need to interpolate the sinc filter */
- double accum[4] = { 0.f, 0.f, 0.f, 0.f };
- float interp[4];
- const spx_word16_t *ptr;
- float alpha = ((float) samp_frac_num) / st->den_rate;
- int offset = samp_frac_num * st->oversample / st->den_rate;
- float frac = alpha * st->oversample - offset;
-
- /* This code is written like this to make it easy to optimise with SIMD.
- For most DSPs, it would be best to split the loops in two because most DSPs
- have only two accumulators */
- for (j = 0; last_sample - N + 1 + j < 0; j++) {
- double curr_mem = mem[last_sample + j];
-
- accum[0] +=
- MULT16_16 (curr_mem,
- st->sinc_table[4 + (j + 1) * st->oversample - offset - 2]);
- accum[1] +=
- MULT16_16 (curr_mem,
- st->sinc_table[4 + (j + 1) * st->oversample - offset - 1]);
- accum[2] +=
- MULT16_16 (curr_mem,
- st->sinc_table[4 + (j + 1) * st->oversample - offset]);
- accum[3] +=
- MULT16_16 (curr_mem,
- st->sinc_table[4 + (j + 1) * st->oversample - offset + 1]);
- }
- if (in != NULL) {
- ptr = in + st->in_stride * (last_sample - N + 1 + j);
- /* Do the new part */
- for (; j < N; j++) {
- double curr_in = *ptr;
-
- ptr += st->in_stride;
- accum[0] +=
- MULT16_16 (curr_in,
- st->sinc_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 =
- interp[0] * accum[0] + interp[1] * accum[1] + interp[2] * accum[2] +
- interp[3] * accum[3];
-
- *out = PSHR32 (sum, 15);
- out += st->out_stride;
- out_sample++;
- 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++;
- }
- }
- st->last_sample[channel_index] = last_sample;
- st->samp_frac_num[channel_index] = samp_frac_num;
- return out_sample;
-}
-#endif
-
-static void
-update_filter (SpeexResamplerState * st)
-{
- spx_uint32_t old_length;
-
- old_length = st->filt_len;
- st->oversample = quality_map[st->quality].oversample;
- st->filt_len = quality_map[st->quality].base_length;
-
- if (st->num_rate > st->den_rate) {
- /* down-sampling */
- st->cutoff =
- quality_map[st->quality].downsample_bandwidth * st->den_rate /
- st->num_rate;
- /* FIXME: divide the numerator and denominator by a certain amount if they're too large */
- st->filt_len = st->filt_len * st->num_rate / st->den_rate;
- /* Round down to make sure we have a multiple of 4 */
- st->filt_len &= (~0x3);
- if (2 * st->den_rate < st->num_rate)
- st->oversample >>= 1;
- if (4 * st->den_rate < st->num_rate)
- st->oversample >>= 1;
- if (8 * st->den_rate < st->num_rate)
- st->oversample >>= 1;
- if (16 * st->den_rate < st->num_rate)
- st->oversample >>= 1;
- if (st->oversample < 1)
- st->oversample = 1;
- } else {
- /* up-sampling */
- st->cutoff = quality_map[st->quality].upsample_bandwidth;
- }
-
- /* Choose the resampling type that requires the least amount of memory */
- if (st->den_rate <= st->oversample) {
- spx_uint32_t i;
-
- if (!st->sinc_table)
- st->sinc_table =
- (spx_word16_t *) speex_alloc (st->filt_len * st->den_rate *
- sizeof (spx_word16_t));
- else if (st->sinc_table_length < st->filt_len * st->den_rate) {
- st->sinc_table =
- (spx_word16_t *) speex_realloc (st->sinc_table,
- st->filt_len * st->den_rate * sizeof (spx_word16_t));
- st->sinc_table_length = st->filt_len * st->den_rate;
- }
- for (i = 0; i < st->den_rate; i++) {
- spx_int32_t j;
-
- for (j = 0; j < st->filt_len; j++) {
- st->sinc_table[i * st->filt_len + j] =
- sinc (st->cutoff,
- ((j - (spx_int32_t) st->filt_len / 2 + 1) -
- ((float) i) / st->den_rate), st->filt_len,
- quality_map[st->quality].window_func);
- }
- }
-#ifdef FIXED_POINT
- st->resampler_ptr = resampler_basic_direct_single;
-#else
- if (st->quality > 8)
- st->resampler_ptr = resampler_basic_direct_double;
- else
- st->resampler_ptr = resampler_basic_direct_single;
-#endif
- /*fprintf (stderr, "resampler uses direct sinc table and normalised cutoff %f\n", cutoff); */
- } else {
- spx_int32_t i;
-
- if (!st->sinc_table)
- st->sinc_table =
- (spx_word16_t *) speex_alloc ((st->filt_len * st->oversample +
- 8) * sizeof (spx_word16_t));
- else if (st->sinc_table_length < st->filt_len * st->oversample + 8) {
- st->sinc_table =
- (spx_word16_t *) speex_realloc (st->sinc_table,
- (st->filt_len * st->oversample + 8) * sizeof (spx_word16_t));
- st->sinc_table_length = st->filt_len * st->oversample + 8;
- }
- for (i = -4; i < (spx_int32_t) (st->oversample * st->filt_len + 4); i++)
- st->sinc_table[i + 4] =
- sinc (st->cutoff, (i / (float) st->oversample - st->filt_len / 2),
- st->filt_len, quality_map[st->quality].window_func);
-#ifdef FIXED_POINT
- st->resampler_ptr = resampler_basic_interpolate_single;
-#else
- if (st->quality > 8)
- st->resampler_ptr = resampler_basic_interpolate_double;
- else
- st->resampler_ptr = resampler_basic_interpolate_single;
-#endif
- /*fprintf (stderr, "resampler uses interpolated sinc table and normalised cutoff %f\n", cutoff); */
- }
- st->int_advance = st->num_rate / st->den_rate;
- st->frac_advance = st->num_rate % st->den_rate;
-
-
- /* Here's the place where we update the filter memory to take into account
- the change in filter length. It's probably the messiest part of the code
- due to handling of lots of corner cases. */
- if (!st->mem) {
- spx_uint32_t i;
-
- st->mem =
- (spx_word16_t *) speex_alloc (st->nb_channels * (st->filt_len -
- 1) * sizeof (spx_word16_t));
- for (i = 0; i < st->nb_channels * (st->filt_len - 1); i++)
- st->mem[i] = 0;
- st->mem_alloc_size = st->filt_len - 1;
- /*speex_warning("init filter"); */
- } else if (!st->started) {
- spx_uint32_t i;
-
- st->mem =
- (spx_word16_t *) speex_realloc (st->mem,
- st->nb_channels * (st->filt_len - 1) * sizeof (spx_word16_t));
- for (i = 0; i < st->nb_channels * (st->filt_len - 1); i++)
- st->mem[i] = 0;
- st->mem_alloc_size = st->filt_len - 1;
- /*speex_warning("reinit filter"); */
- } else if (st->filt_len > old_length) {
- spx_int32_t i;
-
- /* Increase the filter length */
- /*speex_warning("increase filter size"); */
- int old_alloc_size = st->mem_alloc_size;
-
- if (st->filt_len - 1 > st->mem_alloc_size) {
- st->mem =
- (spx_word16_t *) speex_realloc (st->mem,
- st->nb_channels * (st->filt_len - 1) * sizeof (spx_word16_t));
- st->mem_alloc_size = st->filt_len - 1;
- }
- for (i = st->nb_channels - 1; i >= 0; i--) {
- spx_int32_t j;
- spx_uint32_t olen = old_length;
-
- /*if (st->magic_samples[i]) */
- {
- /* Try and remove the magic samples as if nothing had happened */
-
- /* FIXME: This is wrong but for now we need it to avoid going over the array bounds */
- olen = old_length + 2 * st->magic_samples[i];
- for (j = old_length - 2 + st->magic_samples[i]; j >= 0; j--)
- st->mem[i * st->mem_alloc_size + j + st->magic_samples[i]] =
- st->mem[i * old_alloc_size + j];
- for (j = 0; j < st->magic_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; j < olen - 1; j++)
- st->mem[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 (; j < st->filt_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; j < st->filt_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; i < st->nb_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; j < st->filt_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;
- }
- }
-
-}
-
-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);
-}
-
-SpeexResamplerState *
-speex_resampler_init_frac (spx_uint32_t nb_channels, spx_uint32_t ratio_num,
- spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate,
- int quality, int *err)
-{
- spx_uint32_t i;
- SpeexResamplerState *st;
-
- if (quality > 10 || quality < 0) {
- if (err)
- *err = RESAMPLER_ERR_INVALID_ARG;
- return NULL;
- }
- st = (SpeexResamplerState *) speex_alloc (sizeof (SpeexResamplerState));
- st->initialised = 0;
- st->started = 0;
- st->in_rate = 0;
- st->out_rate = 0;
- st->num_rate = 0;
- st->den_rate = 0;
- st->quality = -1;
- st->sinc_table_length = 0;
- st->mem_alloc_size = 0;
- st->filt_len = 0;
- st->mem = 0;
- st->resampler_ptr = 0;
-
- st->cutoff = 1.f;
- st->nb_channels = nb_channels;
- st->in_stride = 1;
- st->out_stride = 1;
-
- /* Per channel data */
- st->last_sample = (spx_int32_t *) speex_alloc (nb_channels * sizeof (int));
- st->magic_samples = (spx_uint32_t *) speex_alloc (nb_channels * sizeof (int));
- st->samp_frac_num = (spx_uint32_t *) speex_alloc (nb_channels * sizeof (int));
- for (i = 0; i < nb_channels; i++) {
- st->last_sample[i] = 0;
- st->magic_samples[i] = 0;
- st->samp_frac_num[i] = 0;
- }
-
- speex_resampler_set_quality (st, quality);
- speex_resampler_set_rate_frac (st, ratio_num, ratio_den, in_rate, out_rate);
-
-
- update_filter (st);
-
- st->initialised = 1;
- if (err)
- *err = RESAMPLER_ERR_SUCCESS;
-
- return st;
-}
-
-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, const spx_word16_t * in, spx_uint32_t * in_len,
- spx_word16_t * out, spx_uint32_t * out_len)
-{
- int j = 0;
- int N = st->filt_len;
- int out_sample = 0;
- spx_word16_t *mem;
- spx_uint32_t tmp_out_len = 0;
-
- mem = st->mem + channel_index * st->mem_alloc_size;
- st->started = 1;
-
- /* Handle the case where we have samples left from a reduction in filter length */
- if (st->magic_samples[channel_index]) {
- int istride_save;
- spx_uint32_t tmp_in_len;
- spx_uint32_t tmp_magic;
-
- istride_save = st->in_stride;
- tmp_in_len = st->magic_samples[channel_index];
- tmp_out_len = *out_len;
- /* magic_samples needs to be set to zero to avoid infinite recursion */
- tmp_magic = st->magic_samples[channel_index];
- st->magic_samples[channel_index] = 0;
- st->in_stride = 1;
- speex_resampler_process_native (st, channel_index, mem + N - 1, &tmp_in_len,
- out, &tmp_out_len);
- st->in_stride = istride_save;
- /*speex_warning_int("extra samples:", tmp_out_len); */
- /* If we couldn't process all "magic" input samples, save the rest for next time */
- if (tmp_in_len < tmp_magic) {
- spx_uint32_t i;
-
- st->magic_samples[channel_index] = tmp_magic - tmp_in_len;
- for (i = 0; i < st->magic_samples[channel_index]; i++)
- mem[N - 1 + i] = mem[N - 1 + i + tmp_in_len];
- }
- out += tmp_out_len * st->out_stride;
- *out_len -= tmp_out_len;
- }
-
- /* Call the right resampler through the function ptr */
- out_sample = st->resampler_ptr (st, channel_index, in, in_len, out, out_len);
-
- if (st->last_sample[channel_index] < (spx_int32_t) * in_len)
- *in_len = st->last_sample[channel_index];
- *out_len = out_sample + tmp_out_len;
- st->last_sample[channel_index] -= *in_len;
-
- for (j = 0; j < N - 1 - (spx_int32_t) * in_len; j++)
- mem[j] = mem[j + *in_len];
- if (in != NULL) {
- for (; j < N - 1; j++)
- mem[j] = in[st->in_stride * (j + *in_len - N + 1)];
- } else {
- for (; j < N - 1; j++)
- mem[j] = 0;
- }
- return RESAMPLER_ERR_SUCCESS;
-}
-
-#define FIXED_STACK_ALLOC 1024
-
-#ifdef FIXED_POINT
-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)
-{
- spx_uint32_t i;
- int istride_save, ostride_save;
-
-#ifdef VAR_ARRAYS
- spx_word16_t x[*in_len];
- spx_word16_t y[*out_len];
-
- /*VARDECL(spx_word16_t *x);
- VARDECL(spx_word16_t *y);
- ALLOC(x, *in_len, spx_word16_t);
- ALLOC(y, *out_len, spx_word16_t); */
- istride_save = st->in_stride;
- ostride_save = st->out_stride;
- if (in != NULL) {
- for (i = 0; i < *in_len; i++)
- x[i] = WORD2INT (in[i * st->in_stride]);
- st->in_stride = st->out_stride = 1;
- speex_resampler_process_native (st, channel_index, x, in_len, y, out_len);
- } else {
- st->in_stride = st->out_stride = 1;
- speex_resampler_process_native (st, channel_index, NULL, in_len, y,
- out_len);
- }
- st->in_stride = istride_save;
- st->out_stride = ostride_save;
- for (i = 0; i < *out_len; i++)
- out[i * st->out_stride] = y[i];
-#else
- spx_word16_t x[FIXED_STACK_ALLOC];
- spx_word16_t y[FIXED_STACK_ALLOC];
- spx_uint32_t ilen = *in_len, olen = *out_len;
-
- istride_save = st->in_stride;
- ostride_save = st->out_stride;
- while (ilen && olen) {
- spx_uint32_t ichunk, ochunk;
-
- ichunk = ilen;
- ochunk = olen;
- if (ichunk > FIXED_STACK_ALLOC)
- ichunk = FIXED_STACK_ALLOC;
- if (ochunk > FIXED_STACK_ALLOC)
- ochunk = FIXED_STACK_ALLOC;
- if (in != NULL) {
- for (i = 0; i < ichunk; i++)
- x[i] = WORD2INT (in[i * st->in_stride]);
- st->in_stride = st->out_stride = 1;
- speex_resampler_process_native (st, channel_index, x, &ichunk, y,
- &ochunk);
- } else {
- st->in_stride = st->out_stride = 1;
- speex_resampler_process_native (st, channel_index, NULL, &ichunk, y,
- &ochunk);
- }
- st->in_stride = istride_save;
- st->out_stride = ostride_save;
- for (i = 0; i < ochunk; i++)
- out[i * st->out_stride] = y[i];
- out += ochunk;
- in += ichunk;
- ilen -= ichunk;
- olen -= ochunk;
- }
- *in_len -= ilen;
- *out_len -= olen;
-#endif
- return RESAMPLER_ERR_SUCCESS;
-}
-
-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)
-{
- return speex_resampler_process_native (st, channel_index, in, in_len, out,
- out_len);
-}
-#else
-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)
-{
- return speex_resampler_process_native (st, channel_index, in, in_len, out,
- out_len);
-}
-
-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)
-{
- spx_uint32_t i;
- int istride_save, ostride_save;
-
-#ifdef VAR_ARRAYS
- spx_word16_t x[*in_len];
- spx_word16_t y[*out_len];
-
- /*VARDECL(spx_word16_t *x);
- VARDECL(spx_word16_t *y);
- ALLOC(x, *in_len, spx_word16_t);
- ALLOC(y, *out_len, spx_word16_t); */
- istride_save = st->in_stride;
- ostride_save = st->out_stride;
- if (in != NULL) {
- for (i = 0; i < *in_len; i++)
- x[i] = in[i * st->in_stride];
- st->in_stride = st->out_stride = 1;
- speex_resampler_process_native (st, channel_index, x, in_len, y, out_len);
- } else {
- st->in_stride = st->out_stride = 1;
- speex_resampler_process_native (st, channel_index, NULL, in_len, y,
- out_len);
- }
- st->in_stride = istride_save;
- st->out_stride = ostride_save;
- for (i = 0; i < *out_len; i++)
- out[i * st->out_stride] = WORD2INT (y[i]);
-#else
- spx_word16_t x[FIXED_STACK_ALLOC];
- spx_word16_t y[FIXED_STACK_ALLOC];
- spx_uint32_t ilen = *in_len, olen = *out_len;
-
- istride_save = st->in_stride;
- ostride_save = st->out_stride;
- while (ilen && olen) {
- spx_uint32_t ichunk, ochunk;
-
- ichunk = ilen;
- ochunk = olen;
- if (ichunk > FIXED_STACK_ALLOC)
- ichunk = FIXED_STACK_ALLOC;
- if (ochunk > FIXED_STACK_ALLOC)
- ochunk = FIXED_STACK_ALLOC;
- if (in != NULL) {
- for (i = 0; i < ichunk; i++)
- x[i] = in[i * st->in_stride];
- st->in_stride = st->out_stride = 1;
- speex_resampler_process_native (st, channel_index, x, &ichunk, y,
- &ochunk);
- } else {
- st->in_stride = st->out_stride = 1;
- speex_resampler_process_native (st, channel_index, NULL, &ichunk, y,
- &ochunk);
- }
- st->in_stride = istride_save;
- st->out_stride = ostride_save;
- for (i = 0; i < ochunk; i++)
- out[i * st->out_stride] = WORD2INT (y[i]);
- out += ochunk;
- in += ichunk;
- ilen -= ichunk;
- olen -= ochunk;
- }
- *in_len -= ilen;
- *out_len -= olen;
-#endif
- return RESAMPLER_ERR_SUCCESS;
-}
-#endif
-
-int
-speex_resampler_process_interleaved_float (SpeexResamplerState * st,
- const float *in, spx_uint32_t * in_len, float *out, spx_uint32_t * out_len)
-{
- spx_uint32_t i;
- int istride_save, ostride_save;
- spx_uint32_t bak_len = *out_len;
-
- istride_save = st->in_stride;
- ostride_save = st->out_stride;
- st->in_stride = st->out_stride = st->nb_channels;
- for (i = 0; i < st->nb_channels; i++) {
- *out_len = bak_len;
- if (in != NULL)
- speex_resampler_process_float (st, i, in + i, in_len, out + i, out_len);
- else
- speex_resampler_process_float (st, i, NULL, in_len, out + i, out_len);
- }
- st->in_stride = istride_save;
- st->out_stride = ostride_save;
- return RESAMPLER_ERR_SUCCESS;
-}
-
-
-int
-speex_resampler_process_interleaved_int (SpeexResamplerState * st,
- const spx_int16_t * in, spx_uint32_t * in_len, spx_int16_t * out,
- spx_uint32_t * out_len)
-{
- spx_uint32_t i;
- int istride_save, ostride_save;
- spx_uint32_t bak_len = *out_len;
-
- istride_save = st->in_stride;
- ostride_save = st->out_stride;
- st->in_stride = st->out_stride = st->nb_channels;
- for (i = 0; i < st->nb_channels; i++) {
- *out_len = bak_len;
- if (in != NULL)
- speex_resampler_process_int (st, i, in + i, in_len, out + i, out_len);
- else
- speex_resampler_process_int (st, i, NULL, in_len, out + i, out_len);
- }
- st->in_stride = istride_save;
- st->out_stride = ostride_save;
- return RESAMPLER_ERR_SUCCESS;
-}
-
-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);
-}
-
-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;
-}
-
-int
-speex_resampler_set_rate_frac (SpeexResamplerState * st, spx_uint32_t ratio_num,
- spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate)
-{
- spx_uint32_t fact;
- spx_uint32_t old_den;
- spx_uint32_t i;
-
- if (st->in_rate == in_rate && st->out_rate == out_rate
- && st->num_rate == ratio_num && st->den_rate == ratio_den)
- return RESAMPLER_ERR_SUCCESS;
-
- old_den = st->den_rate;
- st->in_rate = in_rate;
- st->out_rate = out_rate;
- st->num_rate = ratio_num;
- st->den_rate = ratio_den;
- /* FIXME: This is terribly inefficient, but who cares (at least for now)? */
- for (fact = 2; fact <= IMIN (st->num_rate, st->den_rate); fact++) {
- while ((st->num_rate % fact == 0) && (st->den_rate % fact == 0)) {
- st->num_rate /= fact;
- st->den_rate /= fact;
- }
- }
-
- if (old_den > 0) {
- for (i = 0; i < st->nb_channels; i++) {
- st->samp_frac_num[i] = st->samp_frac_num[i] * st->den_rate / old_den;
- /* Safety net */
- if (st->samp_frac_num[i] >= st->den_rate)
- st->samp_frac_num[i] = st->den_rate - 1;
- }
- }
-
- if (st->initialised)
- update_filter (st);
- return RESAMPLER_ERR_SUCCESS;
-}
-
-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;
-}
-
-int
-speex_resampler_set_quality (SpeexResamplerState * st, int quality)
-{
- if (quality > 10 || quality < 0)
- return RESAMPLER_ERR_INVALID_ARG;
- if (st->quality == quality)
- return RESAMPLER_ERR_SUCCESS;
- st->quality = quality;
- if (st->initialised)
- update_filter (st);
- return RESAMPLER_ERR_SUCCESS;
-}
-
-void
-speex_resampler_get_quality (SpeexResamplerState * st, int *quality)
-{
- *quality = st->quality;
-}
-
-void
-speex_resampler_set_input_stride (SpeexResamplerState * st, spx_uint32_t stride)
-{
- st->in_stride = stride;
-}
-
-void
-speex_resampler_get_input_stride (SpeexResamplerState * st,
- spx_uint32_t * stride)
-{
- *stride = st->in_stride;
-}
-
-void
-speex_resampler_set_output_stride (SpeexResamplerState * st,
- spx_uint32_t stride)
-{
- st->out_stride = stride;
-}
-
-void
-speex_resampler_get_output_stride (SpeexResamplerState * st,
- spx_uint32_t * stride)
-{
- *stride = st->out_stride;
-}
-
-int
-speex_resampler_get_input_latency (SpeexResamplerState * st)
-{
- return st->filt_len / 2;
-}
-
-int
-speex_resampler_get_output_latency (SpeexResamplerState * st)
-{
- return ((st->filt_len / 2) * st->den_rate +
- (st->num_rate >> 1)) / st->num_rate;
-}
-
-int
-speex_resampler_skip_zeros (SpeexResamplerState * st)
-{
- spx_uint32_t i;
-
- for (i = 0; i < st->nb_channels; i++)
- st->last_sample[i] = st->filt_len / 2;
- return RESAMPLER_ERR_SUCCESS;
-}
-
-int
-speex_resampler_reset_mem (SpeexResamplerState * st)
-{
- spx_uint32_t i;
-
- for (i = 0; i < st->nb_channels * (st->filt_len - 1); i++)
- st->mem[i] = 0;
- return RESAMPLER_ERR_SUCCESS;
-}
-
-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/gst/speexresample/speex_resampler.h b/gst/speexresample/speex_resampler.h
deleted file mode 100644
index ad832c54..00000000
--- a/gst/speexresample/speex_resampler.h
+++ /dev/null
@@ -1,342 +0,0 @@
-/* Copyright (C) 2007 Jean-Marc Valin
-
- File: speex_resampler.h
- Resampling code
-
- The design goals of this code are:
- - Very fast algorithm
- - Low memory requirement
- - Good *perceptual* quality (and not best SNR)
-
- Redistribution and use in source and binary forms, with or without
- modification, are permitted provided that the following conditions are
- met:
-
- 1. Redistributions of source code must retain the above copyright notice,
- this list of conditions and the following disclaimer.
-
- 2. Redistributions in binary form must reproduce the above copyright
- notice, this list of conditions and the following disclaimer in the
- documentation and/or other materials provided with the distribution.
-
- 3. The name of the author may not be used to endorse or promote products
- derived from this software without specific prior written permission.
-
- THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
- IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
- OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
- DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
- INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
- (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
- SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
- HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
- STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
- ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
- POSSIBILITY OF SUCH DAMAGE.
-*/
-
-
-#ifndef SPEEX_RESAMPLER_H
-#define SPEEX_RESAMPLER_H
-
-#ifdef OUTSIDE_SPEEX
-
-#include <glib.h>
-
-/********* 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 gint16
-#define spx_int32_t gint32
-#define spx_uint16_t guint16
-#define spx_uint32_t guint32
-
-#else /* OUTSIDE_SPEEX */
-
-#include "speex/speex_types.h"
-
-#endif /* OUTSIDE_SPEEX */
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-#define SPEEX_RESAMPLER_QUALITY_MAX 10
-#define SPEEX_RESAMPLER_QUALITY_MIN 0
-#define SPEEX_RESAMPLER_QUALITY_DEFAULT 4
-#define SPEEX_RESAMPLER_QUALITY_VOIP 3
-#define SPEEX_RESAMPLER_QUALITY_DESKTOP 5
-
-enum {
- RESAMPLER_ERR_SUCCESS = 0,
- RESAMPLER_ERR_ALLOC_FAILED = 1,
- RESAMPLER_ERR_BAD_STATE = 2,
- RESAMPLER_ERR_INVALID_ARG = 3,
- RESAMPLER_ERR_PTR_OVERLAP = 4,
-
- RESAMPLER_ERR_MAX_ERROR
-};
-
-struct SpeexResamplerState_;
-typedef struct SpeexResamplerState_ SpeexResamplerState;
-
-/** Create a new resampler with integer input and output rates.
- * @param nb_channels Number of channels to be processed
- * @param in_rate Input sampling rate (integer number of Hz).
- * @param out_rate Output sampling rate (integer number of Hz).
- * @param quality Resampling quality between 0 and 10, where 0 has poor quality
- * and 10 has very high quality.
- * @return Newly created resampler state
- * @retval NULL Error: not enough memory
- */
-SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels,
- spx_uint32_t in_rate,
- spx_uint32_t out_rate,
- int quality,
- int *err);
-
-/** Create a new resampler with fractional input/output rates. The sampling
- * rate ratio is an arbitrary rational number with both the numerator and
- * denominator being 32-bit integers.
- * @param nb_channels Number of channels to be processed
- * @param ratio_num Numerator of the sampling rate ratio
- * @param ratio_den Denominator of the sampling rate ratio
- * @param in_rate Input sampling rate rounded to the nearest integer (in Hz).
- * @param out_rate Output sampling rate rounded to the nearest integer (in Hz).
- * @param quality Resampling quality between 0 and 10, where 0 has poor quality
- * and 10 has very high quality.
- * @return Newly created resampler state
- * @retval NULL Error: not enough memory
- */
-SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels,
- spx_uint32_t ratio_num,
- spx_uint32_t ratio_den,
- spx_uint32_t in_rate,
- spx_uint32_t out_rate,
- int quality,
- int *err);
-
-/** Destroy a resampler state.
- * @param st Resampler state
- */
-void speex_resampler_destroy(SpeexResamplerState *st);
-
-/** Resample a float array. The input and output buffers must *not* overlap.
- * @param st Resampler state
- * @param channel_index Index of the channel to process for the multi-channel
- * base (0 otherwise)
- * @param in Input buffer
- * @param in_len Number of input samples in the input buffer. Returns the
- * number of samples processed
- * @param out Output buffer
- * @param out_len Size of the output buffer. Returns the number of samples written
- */
-int speex_resampler_process_float(SpeexResamplerState *st,
- spx_uint32_t channel_index,
- const float *in,
- spx_uint32_t *in_len,
- float *out,
- spx_uint32_t *out_len);
-
-/** Resample an int array. The input and output buffers must *not* overlap.
- * @param st Resampler state
- * @param channel_index Index of the channel to process for the multi-channel
- * base (0 otherwise)
- * @param in Input buffer
- * @param in_len Number of input samples in the input buffer. Returns the number
- * of samples processed
- * @param out Output buffer
- * @param out_len Size of the output buffer. Returns the number of samples written
- */
-int speex_resampler_process_int(SpeexResamplerState *st,
- spx_uint32_t channel_index,
- const spx_int16_t *in,
- spx_uint32_t *in_len,
- spx_int16_t *out,
- spx_uint32_t *out_len);
-
-/** Resample an interleaved float array. The input and output buffers must *not* overlap.
- * @param st Resampler state
- * @param in Input buffer
- * @param in_len Number of input samples in the input buffer. Returns the number
- * of samples processed. This is all per-channel.
- * @param out Output buffer
- * @param out_len Size of the output buffer. Returns the number of samples written.
- * This is all per-channel.
- */
-int speex_resampler_process_interleaved_float(SpeexResamplerState *st,
- const float *in,
- spx_uint32_t *in_len,
- float *out,
- spx_uint32_t *out_len);
-
-/** Resample an interleaved int array. The input and output buffers must *not* overlap.
- * @param st Resampler state
- * @param in Input buffer
- * @param in_len Number of input samples in the input buffer. Returns the number
- * of samples processed. This is all per-channel.
- * @param out Output buffer
- * @param out_len Size of the output buffer. Returns the number of samples written.
- * This is all per-channel.
- */
-int speex_resampler_process_interleaved_int(SpeexResamplerState *st,
- const spx_int16_t *in,
- spx_uint32_t *in_len,
- spx_int16_t *out,
- spx_uint32_t *out_len);
-
-/** Set (change) the input/output sampling rates (integer value).
- * @param st Resampler state
- * @param in_rate Input sampling rate (integer number of Hz).
- * @param out_rate Output sampling rate (integer number of Hz).
- */
-int speex_resampler_set_rate(SpeexResamplerState *st,
- spx_uint32_t in_rate,
- spx_uint32_t out_rate);
-
-/** Get the current input/output sampling rates (integer value).
- * @param st Resampler state
- * @param in_rate Input sampling rate (integer number of Hz) copied.
- * @param out_rate Output sampling rate (integer number of Hz) copied.
- */
-void speex_resampler_get_rate(SpeexResamplerState *st,
- spx_uint32_t *in_rate,
- spx_uint32_t *out_rate);
-
-/** Set (change) the input/output sampling rates and resampling ratio
- * (fractional values in Hz supported).
- * @param st Resampler state
- * @param ratio_num Numerator of the sampling rate ratio
- * @param ratio_den Denominator of the sampling rate ratio
- * @param in_rate Input sampling rate rounded to the nearest integer (in Hz).
- * @param out_rate Output sampling rate rounded to the nearest integer (in Hz).
- */
-int speex_resampler_set_rate_frac(SpeexResamplerState *st,
- spx_uint32_t ratio_num,
- spx_uint32_t ratio_den,
- spx_uint32_t in_rate,
- spx_uint32_t out_rate);
-
-/** Get the current resampling ratio. This will be reduced to the least
- * common denominator.
- * @param st Resampler state
- * @param ratio_num Numerator of the sampling rate ratio copied
- * @param ratio_den Denominator of the sampling rate ratio copied
- */
-void speex_resampler_get_ratio(SpeexResamplerState *st,
- spx_uint32_t *ratio_num,
- spx_uint32_t *ratio_den);
-
-/** Set (change) the conversion quality.
- * @param st Resampler state
- * @param quality Resampling quality between 0 and 10, where 0 has poor
- * quality and 10 has very high quality.
- */
-int speex_resampler_set_quality(SpeexResamplerState *st,
- int quality);
-
-/** Get the conversion quality.
- * @param st Resampler state
- * @param quality Resampling quality between 0 and 10, where 0 has poor
- * quality and 10 has very high quality.
- */
-void speex_resampler_get_quality(SpeexResamplerState *st,
- int *quality);
-
-/** Set (change) the input stride.
- * @param st Resampler state
- * @param stride Input stride
- */
-void speex_resampler_set_input_stride(SpeexResamplerState *st,
- spx_uint32_t stride);
-
-/** Get the input stride.
- * @param st Resampler state
- * @param stride Input stride copied
- */
-void speex_resampler_get_input_stride(SpeexResamplerState *st,
- spx_uint32_t *stride);
-
-/** Set (change) the output stride.
- * @param st Resampler state
- * @param stride Output stride
- */
-void speex_resampler_set_output_stride(SpeexResamplerState *st,
- spx_uint32_t stride);
-
-/** Get the output stride.
- * @param st Resampler state copied
- * @param stride Output stride
- */
-void speex_resampler_get_output_stride(SpeexResamplerState *st,
- spx_uint32_t *stride);
-
-/** Get the latency in input samples introduced by the resampler.
- * @param st Resampler state
- */
-int speex_resampler_get_input_latency(SpeexResamplerState *st);
-
-/** Get the latency in output samples introduced by the resampler.
- * @param st Resampler state
- */
-int speex_resampler_get_output_latency(SpeexResamplerState *st);
-
-/** Make sure that the first samples to go out of the resamplers don't have
- * leading zeros. This is only useful before starting to use a newly created
- * resampler. It is recommended to use that when resampling an audio file, as
- * it will generate a file with the same length. For real-time processing,
- * it is probably easier not to use this call (so that the output duration
- * is the same for the first frame).
- * @param st Resampler state
- */
-int speex_resampler_skip_zeros(SpeexResamplerState *st);
-
-/** Reset a resampler so a new (unrelated) stream can be processed.
- * @param st Resampler state
- */
-int speex_resampler_reset_mem(SpeexResamplerState *st);
-
-/** Returns the English meaning for an error code
- * @param err Error code
- * @return English string
- */
-const char *speex_resampler_strerror(int err);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif
diff --git a/gst/speexresample/speex_resampler_float.c b/gst/speexresample/speex_resampler_float.c
deleted file mode 100644
index 281e52d3..00000000
--- a/gst/speexresample/speex_resampler_float.c
+++ /dev/null
@@ -1,24 +0,0 @@
-/* GStreamer
- * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#define FLOATING_POINT
-#define OUTSIDE_SPEEX
-#define RANDOM_PREFIX resample_float
-
-#include "resample.c"
diff --git a/gst/speexresample/speex_resampler_int.c b/gst/speexresample/speex_resampler_int.c
deleted file mode 100644
index c992f0a6..00000000
--- a/gst/speexresample/speex_resampler_int.c
+++ /dev/null
@@ -1,24 +0,0 @@
-/* GStreamer
- * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#define FIXED_POINT 1
-#define OUTSIDE_SPEEX 1
-#define RANDOM_PREFIX resample_int
-
-#include "resample.c"
diff --git a/gst/speexresample/speex_resampler_wrapper.h b/gst/speexresample/speex_resampler_wrapper.h
deleted file mode 100644
index 6d7c17d0..00000000
--- a/gst/speexresample/speex_resampler_wrapper.h
+++ /dev/null
@@ -1,86 +0,0 @@
-/* GStreamer
- * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#ifndef __SPEEX_RESAMPLER_WRAPPER_H__
-#define __SPEEX_RESAMPLER_WRAPPER_H__
-
-#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
-};
-
-typedef struct SpeexResamplerState_ SpeexResamplerState;
-
-SpeexResamplerState *resample_float_resampler_init (guint32 nb_channels,
- guint32 in_rate, guint32 out_rate, gint quality, gint * err);
-SpeexResamplerState *resample_int_resampler_init (guint32 nb_channels,
- guint32 in_rate, guint32 out_rate, gint quality, gint * err);
-
-#define resample_resampler_destroy resample_int_resampler_destroy
-void resample_resampler_destroy (SpeexResamplerState * st);
-
-int resample_float_resampler_process_interleaved_float (SpeexResamplerState *
- st, const gfloat * in, guint32 * in_len, gfloat * out, guint32 * out_len);
-int resample_int_resampler_process_interleaved_int (SpeexResamplerState * st,
- const gint16 * in, guint32 * in_len, gint16 * out, guint32 * out_len);
-
-int resample_float_resampler_set_rate (SpeexResamplerState * st,
- guint32 in_rate, guint32 out_rate);
-int resample_int_resampler_set_rate (SpeexResamplerState * st,
- guint32 in_rate, guint32 out_rate);
-
-void resample_float_resampler_get_rate (SpeexResamplerState * st,
- guint32 * in_rate, guint32 * out_rate);
-void resample_int_resampler_get_rate (SpeexResamplerState * st,
- guint32 * in_rate, guint32 * out_rate);
-
-void resample_float_resampler_get_ratio (SpeexResamplerState * st,
- guint32 * ratio_num, guint32 * ratio_den);
-void resample_int_resampler_get_ratio (SpeexResamplerState * st,
- guint32 * ratio_num, guint32 * ratio_den);
-
-int resample_float_resampler_get_input_latency (SpeexResamplerState * st);
-int resample_int_resampler_get_input_latency (SpeexResamplerState * st);
-
-int resample_float_resampler_set_quality (SpeexResamplerState * st,
- gint quality);
-int resample_int_resampler_set_quality (SpeexResamplerState * st, gint quality);
-
-int resample_float_resampler_reset_mem (SpeexResamplerState * st);
-int resample_int_resampler_reset_mem (SpeexResamplerState * st);
-
-int resample_float_resampler_skip_zeros (SpeexResamplerState * st);
-int resample_int_resampler_skip_zeros (SpeexResamplerState * st);
-
-#define resample_resampler_strerror resample_int_resampler_strerror
-const char *resample_resampler_strerror (gint err);
-
-#endif /* __SPEEX_RESAMPLER_WRAPPER_H__ */
generated by cgit v1.2.1 (git 2.18.0) at 2024-06-29 03:46:59 +0000