audio/mixeng.c - qemu - Git at Google

 /*
  * QEMU Mixing engine
  *
  * Copyright (c) 2004-2005 Vassili Karpov (malc)
  * Copyright (c) 1998 Fabrice Bellard
  *
  * Permission is hereby granted, free of charge, to any person obtaining a copy
  * of this software and associated documentation files (the "Software"), to deal
  * in the Software without restriction, including without limitation the rights
  * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
  * copies of the Software, and to permit persons to whom the Software is
  * furnished to do so, subject to the following conditions:
  *
  * The above copyright notice and this permission notice shall be included in
  * all copies or substantial portions of the Software.
  *
  * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
  * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
  * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
  * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
  * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
  * THE SOFTWARE.
  */
 #include "qemu/osdep.h"
 #include "qemu/bswap.h"
 #include "qemu/error-report.h"
 #include "audio.h"

 #define AUDIO_CAP "mixeng"
 #include "audio_int.h"

 /* 8 bit */
 #define ENDIAN_CONVERSION natural
 #define ENDIAN_CONVERT(v) (v)

 /* Signed 8 bit */
 #define BSIZE 8
 #define ITYPE int
 #define IN_MIN SCHAR_MIN
 #define IN_MAX SCHAR_MAX
 #define SIGNED
 #define SHIFT 8
 #include "mixeng_template.h"
 #undef SIGNED
 #undef IN_MAX
 #undef IN_MIN
 #undef BSIZE
 #undef ITYPE
 #undef SHIFT

 /* Unsigned 8 bit */
 #define BSIZE 8
 #define ITYPE uint
 #define IN_MIN 0
 #define IN_MAX UCHAR_MAX
 #define SHIFT 8
 #include "mixeng_template.h"
 #undef IN_MAX
 #undef IN_MIN
 #undef BSIZE
 #undef ITYPE
 #undef SHIFT

 #undef ENDIAN_CONVERT
 #undef ENDIAN_CONVERSION

 /* Signed 16 bit */
 #define BSIZE 16
 #define ITYPE int
 #define IN_MIN SHRT_MIN
 #define IN_MAX SHRT_MAX
 #define SIGNED
 #define SHIFT 16
 #define ENDIAN_CONVERSION natural
 #define ENDIAN_CONVERT(v) (v)
 #include "mixeng_template.h"
 #undef ENDIAN_CONVERT
 #undef ENDIAN_CONVERSION
 #define ENDIAN_CONVERSION swap
 #define ENDIAN_CONVERT(v) bswap16 (v)
 #include "mixeng_template.h"
 #undef ENDIAN_CONVERT
 #undef ENDIAN_CONVERSION
 #undef SIGNED
 #undef IN_MAX
 #undef IN_MIN
 #undef BSIZE
 #undef ITYPE
 #undef SHIFT

 /* Unsigned 16 bit */
 #define BSIZE 16
 #define ITYPE uint
 #define IN_MIN 0
 #define IN_MAX USHRT_MAX
 #define SHIFT 16
 #define ENDIAN_CONVERSION natural
 #define ENDIAN_CONVERT(v) (v)
 #include "mixeng_template.h"
 #undef ENDIAN_CONVERT
 #undef ENDIAN_CONVERSION
 #define ENDIAN_CONVERSION swap
 #define ENDIAN_CONVERT(v) bswap16 (v)
 #include "mixeng_template.h"
 #undef ENDIAN_CONVERT
 #undef ENDIAN_CONVERSION
 #undef IN_MAX
 #undef IN_MIN
 #undef BSIZE
 #undef ITYPE
 #undef SHIFT

 /* Signed 32 bit */
 #define BSIZE 32
 #define ITYPE int
 #define IN_MIN INT32_MIN
 #define IN_MAX INT32_MAX
 #define SIGNED
 #define SHIFT 32
 #define ENDIAN_CONVERSION natural
 #define ENDIAN_CONVERT(v) (v)
 #include "mixeng_template.h"
 #undef ENDIAN_CONVERT
 #undef ENDIAN_CONVERSION
 #define ENDIAN_CONVERSION swap
 #define ENDIAN_CONVERT(v) bswap32 (v)
 #include "mixeng_template.h"
 #undef ENDIAN_CONVERT
 #undef ENDIAN_CONVERSION
 #undef SIGNED
 #undef IN_MAX
 #undef IN_MIN
 #undef BSIZE
 #undef ITYPE
 #undef SHIFT

 /* Unsigned 32 bit */
 #define BSIZE 32
 #define ITYPE uint
 #define IN_MIN 0
 #define IN_MAX UINT32_MAX
 #define SHIFT 32
 #define ENDIAN_CONVERSION natural
 #define ENDIAN_CONVERT(v) (v)
 #include "mixeng_template.h"
 #undef ENDIAN_CONVERT
 #undef ENDIAN_CONVERSION
 #define ENDIAN_CONVERSION swap
 #define ENDIAN_CONVERT(v) bswap32 (v)
 #include "mixeng_template.h"
 #undef ENDIAN_CONVERT
 #undef ENDIAN_CONVERSION
 #undef IN_MAX
 #undef IN_MIN
 #undef BSIZE
 #undef ITYPE
 #undef SHIFT

 t_sample *mixeng_conv[2][2][2][3] = {
     {
         {
             {
                 conv_natural_uint8_t_to_mono,
                 conv_natural_uint16_t_to_mono,
                 conv_natural_uint32_t_to_mono
             },
             {
                 conv_natural_uint8_t_to_mono,
                 conv_swap_uint16_t_to_mono,
                 conv_swap_uint32_t_to_mono,
             }
         },
         {
             {
                 conv_natural_int8_t_to_mono,
                 conv_natural_int16_t_to_mono,
                 conv_natural_int32_t_to_mono
             },
             {
                 conv_natural_int8_t_to_mono,
                 conv_swap_int16_t_to_mono,
                 conv_swap_int32_t_to_mono
             }
         }
     },
     {
         {
             {
                 conv_natural_uint8_t_to_stereo,
                 conv_natural_uint16_t_to_stereo,
                 conv_natural_uint32_t_to_stereo
             },
             {
                 conv_natural_uint8_t_to_stereo,
                 conv_swap_uint16_t_to_stereo,
                 conv_swap_uint32_t_to_stereo
             }
         },
         {
             {
                 conv_natural_int8_t_to_stereo,
                 conv_natural_int16_t_to_stereo,
                 conv_natural_int32_t_to_stereo
             },
             {
                 conv_natural_int8_t_to_stereo,
                 conv_swap_int16_t_to_stereo,
                 conv_swap_int32_t_to_stereo,
             }
         }
     }
 };

 f_sample *mixeng_clip[2][2][2][3] = {
     {
         {
             {
                 clip_natural_uint8_t_from_mono,
                 clip_natural_uint16_t_from_mono,
                 clip_natural_uint32_t_from_mono
             },
             {
                 clip_natural_uint8_t_from_mono,
                 clip_swap_uint16_t_from_mono,
                 clip_swap_uint32_t_from_mono
             }
         },
         {
             {
                 clip_natural_int8_t_from_mono,
                 clip_natural_int16_t_from_mono,
                 clip_natural_int32_t_from_mono
             },
             {
                 clip_natural_int8_t_from_mono,
                 clip_swap_int16_t_from_mono,
                 clip_swap_int32_t_from_mono
             }
         }
     },
     {
         {
             {
                 clip_natural_uint8_t_from_stereo,
                 clip_natural_uint16_t_from_stereo,
                 clip_natural_uint32_t_from_stereo
             },
             {
                 clip_natural_uint8_t_from_stereo,
                 clip_swap_uint16_t_from_stereo,
                 clip_swap_uint32_t_from_stereo
             }
         },
         {
             {
                 clip_natural_int8_t_from_stereo,
                 clip_natural_int16_t_from_stereo,
                 clip_natural_int32_t_from_stereo
             },
             {
                 clip_natural_int8_t_from_stereo,
                 clip_swap_int16_t_from_stereo,
                 clip_swap_int32_t_from_stereo
             }
         }
     }
 };

 #ifdef FLOAT_MIXENG
 #define CONV_NATURAL_FLOAT(x) (x)
 #define CLIP_NATURAL_FLOAT(x) (x)
 #else
 /* macros to map [-1.f, 1.f] <-> [INT32_MIN, INT32_MAX + 1] */
 static const float float_scale = (int64_t)INT32_MAX + 1;
 #define CONV_NATURAL_FLOAT(x) ((x) * float_scale)

 #ifdef RECIPROCAL
 static const float float_scale_reciprocal = 1.f / ((int64_t)INT32_MAX + 1);
 #define CLIP_NATURAL_FLOAT(x) ((x) * float_scale_reciprocal)
 #else
 #define CLIP_NATURAL_FLOAT(x) ((x) / float_scale)
 #endif
 #endif

 static void conv_natural_float_to_mono(struct st_sample *dst, const void *src,
                                        int samples)
 {
     float *in = (float *)src;

     while (samples--) {
         dst->r = dst->l = CONV_NATURAL_FLOAT(*in++);
         dst++;
     }
 }

 static void conv_natural_float_to_stereo(struct st_sample *dst, const void *src,
                                          int samples)
 {
     float *in = (float *)src;

     while (samples--) {
         dst->l = CONV_NATURAL_FLOAT(*in++);
         dst->r = CONV_NATURAL_FLOAT(*in++);
         dst++;
     }
 }

 t_sample *mixeng_conv_float[2] = {
     conv_natural_float_to_mono,
     conv_natural_float_to_stereo,
 };

 static void clip_natural_float_from_mono(void *dst, const struct st_sample *src,
                                          int samples)
 {
     float *out = (float *)dst;

     while (samples--) {
         *out++ = CLIP_NATURAL_FLOAT(src->l + src->r);
         src++;
     }
 }

 static void clip_natural_float_from_stereo(
     void *dst, const struct st_sample *src, int samples)
 {
     float *out = (float *)dst;

     while (samples--) {
         *out++ = CLIP_NATURAL_FLOAT(src->l);
         *out++ = CLIP_NATURAL_FLOAT(src->r);
         src++;
     }
 }

 f_sample *mixeng_clip_float[2] = {
     clip_natural_float_from_mono,
     clip_natural_float_from_stereo,
 };

 void audio_sample_to_uint64(const void *samples, int pos,
                             uint64_t *left, uint64_t *right)
 {
 #ifdef FLOAT_MIXENG
     error_report(
         "Coreaudio and floating point samples are not supported by replay yet");
     abort();
 #else
     const struct st_sample *sample = samples;
     sample += pos;
     *left = sample->l;
     *right = sample->r;
 #endif
 }

 void audio_sample_from_uint64(void *samples, int pos,
                             uint64_t left, uint64_t right)
 {
 #ifdef FLOAT_MIXENG
     error_report(
         "Coreaudio and floating point samples are not supported by replay yet");
     abort();
 #else
     struct st_sample *sample = samples;
     sample += pos;
     sample->l = left;
     sample->r = right;
 #endif
 }

 /*
  * August 21, 1998
  * Copyright 1998 Fabrice Bellard.
  *
  * [Rewrote completely the code of Lance Norskog And Sundry
  * Contributors with a more efficient algorithm.]
  *
  * This source code is freely redistributable and may be used for
  * any purpose.  This copyright notice must be maintained.
  * Lance Norskog And Sundry Contributors are not responsible for
  * the consequences of using this software.
  */

 /*
  * Sound Tools rate change effect file.
  */
 /*
  * Linear Interpolation.
  *
  * The use of fractional increment allows us to use no buffer. It
  * avoid the problems at the end of the buffer we had with the old
  * method which stored a possibly big buffer of size
  * lcm(in_rate,out_rate).
  *
  * Limited to 16 bit samples and sampling frequency <= 65535 Hz. If
  * the input & output frequencies are equal, a delay of one sample is
  * introduced.  Limited to processing 32-bit count worth of samples.
  *
  * 1 << FRAC_BITS evaluating to zero in several places.  Changed with
  * an (unsigned long) cast to make it safe.  MarkMLl 2/1/99
  */

 /* Private data */
 struct rate {
     uint64_t opos;
     uint64_t opos_inc;
     uint32_t ipos;              /* position in the input stream (integer) */
     struct st_sample ilast;          /* last sample in the input stream */
 };

 /*
  * Prepare processing.
  */
 void *st_rate_start (int inrate, int outrate)
 {
     struct rate *rate = g_new0(struct rate, 1);

     rate->opos = 0;

     /* increment */
     rate->opos_inc = ((uint64_t) inrate << 32) / outrate;

     rate->ipos = 0;
     rate->ilast.l = 0;
     rate->ilast.r = 0;
     return rate;
 }

 #define NAME st_rate_flow_mix
 #define OP(a, b) a += b
 #include "rate_template.h"

 #define NAME st_rate_flow
 #define OP(a, b) a = b
 #include "rate_template.h"

 void st_rate_stop (void *opaque)
 {
     g_free (opaque);
 }

 /**
  * st_rate_frames_out() - returns the number of frames the resampling code
  * generates from frames_in frames
  *
  * @opaque: pointer to struct rate
  * @frames_in: number of frames
  *
  * When upsampling, there may be more than one correct result. In this case,
  * the function returns the maximum number of output frames the resampling
  * code can generate.
  */
 uint32_t st_rate_frames_out(void *opaque, uint32_t frames_in)
 {
     struct rate *rate = opaque;
     uint64_t opos_end, opos_delta;
     uint32_t ipos_end;
     uint32_t frames_out;

     if (rate->opos_inc == 1ULL << 32) {
         return frames_in;
     }

     /* no output frame without at least one input frame */
     if (!frames_in) {
         return 0;
     }

     /* last frame read was at rate->ipos - 1 */
     ipos_end = rate->ipos - 1 + frames_in;
     opos_end = (uint64_t)ipos_end << 32;

     /* last frame written was at rate->opos - rate->opos_inc */
     if (opos_end + rate->opos_inc <= rate->opos) {
         return 0;
     }
     opos_delta = opos_end - rate->opos + rate->opos_inc;
     frames_out = opos_delta / rate->opos_inc;

     return opos_delta % rate->opos_inc ? frames_out : frames_out - 1;
 }

 /**
  * st_rate_frames_in() - returns the number of frames needed to
  * get frames_out frames after resampling
  *
  * @opaque: pointer to struct rate
  * @frames_out: number of frames
  *
  * When downsampling, there may be more than one correct result. In this
  * case, the function returns the maximum number of input frames needed.
  */
 uint32_t st_rate_frames_in(void *opaque, uint32_t frames_out)
 {
     struct rate *rate = opaque;
     uint64_t opos_start, opos_end;
     uint32_t ipos_start, ipos_end;

     if (rate->opos_inc == 1ULL << 32) {
         return frames_out;
     }

     if (frames_out) {
         opos_start = rate->opos;
         ipos_start = rate->ipos;
     } else {
         uint64_t offset;

         /* add offset = ceil(opos_inc) to opos and ipos to avoid an underflow */
         offset = (rate->opos_inc + (1ULL << 32) - 1) & ~((1ULL << 32) - 1);
         opos_start = rate->opos + offset;
         ipos_start = rate->ipos + (offset >> 32);
     }
     /* last frame written was at opos_start - rate->opos_inc */
     opos_end = opos_start - rate->opos_inc + rate->opos_inc * frames_out;
     ipos_end = (opos_end >> 32) + 1;

     /* last frame read was at ipos_start - 1 */
     return ipos_end + 1 > ipos_start ? ipos_end + 1 - ipos_start : 0;
 }

 void mixeng_clear (struct st_sample *buf, int len)
 {
     memset (buf, 0, len * sizeof (struct st_sample));
 }

 void mixeng_volume (struct st_sample *buf, int len, struct mixeng_volume *vol)
 {
     if (vol->mute) {
         mixeng_clear (buf, len);
         return;
     }

     while (len--) {
 #ifdef FLOAT_MIXENG
         buf->l = buf->l * vol->l;
         buf->r = buf->r * vol->r;
 #else
         buf->l = (buf->l * vol->l) >> 32;
         buf->r = (buf->r * vol->r) >> 32;
 #endif
         buf += 1;
     }
 }
	/*
	* QEMU Mixing engine
	*
	* Copyright (c) 2004-2005 Vassili Karpov (malc)
	* Copyright (c) 1998 Fabrice Bellard
	*
	* Permission is hereby granted, free of charge, to any person obtaining a copy
	* of this software and associated documentation files (the "Software"), to deal
	* in the Software without restriction, including without limitation the rights
	* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
	* copies of the Software, and to permit persons to whom the Software is
	* furnished to do so, subject to the following conditions:
	*
	* The above copyright notice and this permission notice shall be included in
	* all copies or substantial portions of the Software.
	*
	* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
	* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
	* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
	* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
	* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
	* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
	* THE SOFTWARE.
	*/
	#include "qemu/osdep.h"
	#include "qemu/bswap.h"
	#include "qemu/error-report.h"
	#include "audio.h"

	#define AUDIO_CAP "mixeng"
	#include "audio_int.h"

	/* 8 bit */
	#define ENDIAN_CONVERSION natural
	#define ENDIAN_CONVERT(v) (v)

	/* Signed 8 bit */
	#define BSIZE 8
	#define ITYPE int
	#define IN_MIN SCHAR_MIN
	#define IN_MAX SCHAR_MAX
	#define SIGNED
	#define SHIFT 8
	#include "mixeng_template.h"
	#undef SIGNED
	#undef IN_MAX
	#undef IN_MIN
	#undef BSIZE
	#undef ITYPE
	#undef SHIFT

	/* Unsigned 8 bit */
	#define BSIZE 8
	#define ITYPE uint
	#define IN_MIN 0
	#define IN_MAX UCHAR_MAX
	#define SHIFT 8
	#include "mixeng_template.h"
	#undef IN_MAX
	#undef IN_MIN
	#undef BSIZE
	#undef ITYPE
	#undef SHIFT

	#undef ENDIAN_CONVERT
	#undef ENDIAN_CONVERSION

	/* Signed 16 bit */
	#define BSIZE 16
	#define ITYPE int
	#define IN_MIN SHRT_MIN
	#define IN_MAX SHRT_MAX
	#define SIGNED
	#define SHIFT 16
	#define ENDIAN_CONVERSION natural
	#define ENDIAN_CONVERT(v) (v)
	#include "mixeng_template.h"
	#undef ENDIAN_CONVERT
	#undef ENDIAN_CONVERSION
	#define ENDIAN_CONVERSION swap
	#define ENDIAN_CONVERT(v) bswap16 (v)
	#include "mixeng_template.h"
	#undef ENDIAN_CONVERT
	#undef ENDIAN_CONVERSION
	#undef SIGNED
	#undef IN_MAX
	#undef IN_MIN
	#undef BSIZE
	#undef ITYPE
	#undef SHIFT

	/* Unsigned 16 bit */
	#define BSIZE 16
	#define ITYPE uint
	#define IN_MIN 0
	#define IN_MAX USHRT_MAX
	#define SHIFT 16
	#define ENDIAN_CONVERSION natural
	#define ENDIAN_CONVERT(v) (v)
	#include "mixeng_template.h"
	#undef ENDIAN_CONVERT
	#undef ENDIAN_CONVERSION
	#define ENDIAN_CONVERSION swap
	#define ENDIAN_CONVERT(v) bswap16 (v)
	#include "mixeng_template.h"
	#undef ENDIAN_CONVERT
	#undef ENDIAN_CONVERSION
	#undef IN_MAX
	#undef IN_MIN
	#undef BSIZE
	#undef ITYPE
	#undef SHIFT

	/* Signed 32 bit */
	#define BSIZE 32
	#define ITYPE int
	#define IN_MIN INT32_MIN
	#define IN_MAX INT32_MAX
	#define SIGNED
	#define SHIFT 32
	#define ENDIAN_CONVERSION natural
	#define ENDIAN_CONVERT(v) (v)
	#include "mixeng_template.h"
	#undef ENDIAN_CONVERT
	#undef ENDIAN_CONVERSION
	#define ENDIAN_CONVERSION swap
	#define ENDIAN_CONVERT(v) bswap32 (v)
	#include "mixeng_template.h"
	#undef ENDIAN_CONVERT
	#undef ENDIAN_CONVERSION
	#undef SIGNED
	#undef IN_MAX
	#undef IN_MIN
	#undef BSIZE
	#undef ITYPE
	#undef SHIFT

	/* Unsigned 32 bit */
	#define BSIZE 32
	#define ITYPE uint
	#define IN_MIN 0
	#define IN_MAX UINT32_MAX
	#define SHIFT 32
	#define ENDIAN_CONVERSION natural
	#define ENDIAN_CONVERT(v) (v)
	#include "mixeng_template.h"
	#undef ENDIAN_CONVERT
	#undef ENDIAN_CONVERSION
	#define ENDIAN_CONVERSION swap
	#define ENDIAN_CONVERT(v) bswap32 (v)
	#include "mixeng_template.h"
	#undef ENDIAN_CONVERT
	#undef ENDIAN_CONVERSION
	#undef IN_MAX
	#undef IN_MIN
	#undef BSIZE
	#undef ITYPE
	#undef SHIFT

	t_sample *mixeng_conv[2][2][2][3] = {
	{
	{
	{
	conv_natural_uint8_t_to_mono,
	conv_natural_uint16_t_to_mono,
	conv_natural_uint32_t_to_mono
	},
	{
	conv_natural_uint8_t_to_mono,
	conv_swap_uint16_t_to_mono,
	conv_swap_uint32_t_to_mono,
	}
	},
	{
	{
	conv_natural_int8_t_to_mono,
	conv_natural_int16_t_to_mono,
	conv_natural_int32_t_to_mono
	},
	{
	conv_natural_int8_t_to_mono,
	conv_swap_int16_t_to_mono,
	conv_swap_int32_t_to_mono
	}
	}
	},
	{
	{
	{
	conv_natural_uint8_t_to_stereo,
	conv_natural_uint16_t_to_stereo,
	conv_natural_uint32_t_to_stereo
	},
	{
	conv_natural_uint8_t_to_stereo,
	conv_swap_uint16_t_to_stereo,
	conv_swap_uint32_t_to_stereo
	}
	},
	{
	{
	conv_natural_int8_t_to_stereo,
	conv_natural_int16_t_to_stereo,
	conv_natural_int32_t_to_stereo
	},
	{
	conv_natural_int8_t_to_stereo,
	conv_swap_int16_t_to_stereo,
	conv_swap_int32_t_to_stereo,
	}
	}
	}
	};

	f_sample *mixeng_clip[2][2][2][3] = {
	{
	{
	{
	clip_natural_uint8_t_from_mono,
	clip_natural_uint16_t_from_mono,
	clip_natural_uint32_t_from_mono
	},
	{
	clip_natural_uint8_t_from_mono,
	clip_swap_uint16_t_from_mono,
	clip_swap_uint32_t_from_mono
	}
	},
	{
	{
	clip_natural_int8_t_from_mono,
	clip_natural_int16_t_from_mono,
	clip_natural_int32_t_from_mono
	},
	{
	clip_natural_int8_t_from_mono,
	clip_swap_int16_t_from_mono,
	clip_swap_int32_t_from_mono
	}
	}
	},
	{
	{
	{
	clip_natural_uint8_t_from_stereo,
	clip_natural_uint16_t_from_stereo,
	clip_natural_uint32_t_from_stereo
	},
	{
	clip_natural_uint8_t_from_stereo,
	clip_swap_uint16_t_from_stereo,
	clip_swap_uint32_t_from_stereo
	}
	},
	{
	{
	clip_natural_int8_t_from_stereo,
	clip_natural_int16_t_from_stereo,
	clip_natural_int32_t_from_stereo
	},
	{
	clip_natural_int8_t_from_stereo,
	clip_swap_int16_t_from_stereo,
	clip_swap_int32_t_from_stereo
	}
	}
	}
	};

	#ifdef FLOAT_MIXENG
	#define CONV_NATURAL_FLOAT(x) (x)
	#define CLIP_NATURAL_FLOAT(x) (x)
	#else
	/* macros to map [-1.f, 1.f] <-> [INT32_MIN, INT32_MAX + 1] */
	static const float float_scale = (int64_t)INT32_MAX + 1;
	#define CONV_NATURAL_FLOAT(x) ((x) * float_scale)

	#ifdef RECIPROCAL
	static const float float_scale_reciprocal = 1.f / ((int64_t)INT32_MAX + 1);
	#define CLIP_NATURAL_FLOAT(x) ((x) * float_scale_reciprocal)
	#else
	#define CLIP_NATURAL_FLOAT(x) ((x) / float_scale)
	#endif
	#endif

	static void conv_natural_float_to_mono(struct st_sample dst, const void src,
	int samples)
	{
	float in = (float )src;

	while (samples--) {
	dst->r = dst->l = CONV_NATURAL_FLOAT(*in++);
	dst++;
	}
	}

	static void conv_natural_float_to_stereo(struct st_sample dst, const void src,
	int samples)
	{
	float in = (float )src;

	while (samples--) {
	dst->l = CONV_NATURAL_FLOAT(*in++);
	dst->r = CONV_NATURAL_FLOAT(*in++);
	dst++;
	}
	}

	t_sample *mixeng_conv_float[2] = {
	conv_natural_float_to_mono,
	conv_natural_float_to_stereo,
	};

	static void clip_natural_float_from_mono(void dst, const struct st_sample src,
	int samples)
	{
	float out = (float )dst;

	while (samples--) {
	*out++ = CLIP_NATURAL_FLOAT(src->l + src->r);
	src++;
	}
	}

	static void clip_natural_float_from_stereo(
	void dst, const struct st_sample src, int samples)
	{
	float out = (float )dst;

	while (samples--) {
	*out++ = CLIP_NATURAL_FLOAT(src->l);
	*out++ = CLIP_NATURAL_FLOAT(src->r);
	src++;
	}
	}

	f_sample *mixeng_clip_float[2] = {
	clip_natural_float_from_mono,
	clip_natural_float_from_stereo,
	};

	void audio_sample_to_uint64(const void *samples, int pos,
	uint64_t left, uint64_t right)
	{
	#ifdef FLOAT_MIXENG
	error_report(
	"Coreaudio and floating point samples are not supported by replay yet");
	abort();
	#else
	const struct st_sample *sample = samples;
	sample += pos;
	*left = sample->l;
	*right = sample->r;
	#endif
	}

	void audio_sample_from_uint64(void *samples, int pos,
	uint64_t left, uint64_t right)
	{
	#ifdef FLOAT_MIXENG
	error_report(
	"Coreaudio and floating point samples are not supported by replay yet");
	abort();
	#else
	struct st_sample *sample = samples;
	sample += pos;
	sample->l = left;
	sample->r = right;
	#endif
	}

	/*
	* August 21, 1998
	* Copyright 1998 Fabrice Bellard.
	*
	* [Rewrote completely the code of Lance Norskog And Sundry
	* Contributors with a more efficient algorithm.]
	*
	* This source code is freely redistributable and may be used for
	* any purpose. This copyright notice must be maintained.
	* Lance Norskog And Sundry Contributors are not responsible for
	* the consequences of using this software.
	*/

	/*
	* Sound Tools rate change effect file.
	*/
	/*
	* Linear Interpolation.
	*
	* The use of fractional increment allows us to use no buffer. It
	* avoid the problems at the end of the buffer we had with the old
	* method which stored a possibly big buffer of size
	* lcm(in_rate,out_rate).
	*
	* Limited to 16 bit samples and sampling frequency <= 65535 Hz. If
	* the input & output frequencies are equal, a delay of one sample is
	* introduced. Limited to processing 32-bit count worth of samples.
	*
	* 1 << FRAC_BITS evaluating to zero in several places. Changed with
	* an (unsigned long) cast to make it safe. MarkMLl 2/1/99
	*/

	/* Private data */
	struct rate {
	uint64_t opos;
	uint64_t opos_inc;
	uint32_t ipos; /* position in the input stream (integer) */
	struct st_sample ilast; /* last sample in the input stream */
	};

	/*
	* Prepare processing.
	*/
	void *st_rate_start (int inrate, int outrate)
	{
	struct rate *rate = g_new0(struct rate, 1);

	rate->opos = 0;

	/* increment */
	rate->opos_inc = ((uint64_t) inrate << 32) / outrate;

	rate->ipos = 0;
	rate->ilast.l = 0;
	rate->ilast.r = 0;
	return rate;
	}

	#define NAME st_rate_flow_mix
	#define OP(a, b) a += b
	#include "rate_template.h"

	#define NAME st_rate_flow
	#define OP(a, b) a = b
	#include "rate_template.h"

	void st_rate_stop (void *opaque)
	{
	g_free (opaque);
	}

	/**
	* st_rate_frames_out() - returns the number of frames the resampling code
	* generates from frames_in frames
	*
	* @opaque: pointer to struct rate
	* @frames_in: number of frames
	*
	* When upsampling, there may be more than one correct result. In this case,
	* the function returns the maximum number of output frames the resampling
	* code can generate.
	*/
	uint32_t st_rate_frames_out(void *opaque, uint32_t frames_in)
	{
	struct rate *rate = opaque;
	uint64_t opos_end, opos_delta;
	uint32_t ipos_end;
	uint32_t frames_out;

	if (rate->opos_inc == 1ULL << 32) {
	return frames_in;
	}

	/* no output frame without at least one input frame */
	if (!frames_in) {
	return 0;
	}

	/* last frame read was at rate->ipos - 1 */
	ipos_end = rate->ipos - 1 + frames_in;
	opos_end = (uint64_t)ipos_end << 32;

	/* last frame written was at rate->opos - rate->opos_inc */
	if (opos_end + rate->opos_inc <= rate->opos) {
	return 0;
	}
	opos_delta = opos_end - rate->opos + rate->opos_inc;
	frames_out = opos_delta / rate->opos_inc;

	return opos_delta % rate->opos_inc ? frames_out : frames_out - 1;
	}

	/**
	* st_rate_frames_in() - returns the number of frames needed to
	* get frames_out frames after resampling
	*
	* @opaque: pointer to struct rate
	* @frames_out: number of frames
	*
	* When downsampling, there may be more than one correct result. In this
	* case, the function returns the maximum number of input frames needed.
	*/
	uint32_t st_rate_frames_in(void *opaque, uint32_t frames_out)
	{
	struct rate *rate = opaque;
	uint64_t opos_start, opos_end;
	uint32_t ipos_start, ipos_end;

	if (rate->opos_inc == 1ULL << 32) {
	return frames_out;
	}

	if (frames_out) {
	opos_start = rate->opos;
	ipos_start = rate->ipos;
	} else {
	uint64_t offset;

	/* add offset = ceil(opos_inc) to opos and ipos to avoid an underflow */
	offset = (rate->opos_inc + (1ULL << 32) - 1) & ~((1ULL << 32) - 1);
	opos_start = rate->opos + offset;
	ipos_start = rate->ipos + (offset >> 32);
	}
	/* last frame written was at opos_start - rate->opos_inc */
	opos_end = opos_start - rate->opos_inc + rate->opos_inc * frames_out;
	ipos_end = (opos_end >> 32) + 1;

	/* last frame read was at ipos_start - 1 */
	return ipos_end + 1 > ipos_start ? ipos_end + 1 - ipos_start : 0;
	}

	void mixeng_clear (struct st_sample *buf, int len)
	{
	memset (buf, 0, len * sizeof (struct st_sample));
	}

	void mixeng_volume (struct st_sample buf, int len, struct mixeng_volume vol)
	{
	if (vol->mute) {
	mixeng_clear (buf, len);
	return;
	}

	while (len--) {
	#ifdef FLOAT_MIXENG
	buf->l = buf->l * vol->l;
	buf->r = buf->r * vol->r;
	#else
	buf->l = (buf->l * vol->l) >> 32;
	buf->r = (buf->r * vol->r) >> 32;
	#endif
	buf += 1;
	}
	}