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xemu/audio/mixeng.c
Alistair Francis 470bcabd8f audio: Replace AUDIO_FUNC with __func__ 
Apparently we don't use __MSC_VER as a compiler anymore and we always
require a C99 compiler (which means we always have __func__) so we don't
need a special AUDIO_FUNC macro. We can just replace AUDIO_FUNC with
__func__ instead.

Checkpatch failures were manually fixed.

Signed-off-by: Alistair Francis <alistair.francis@xilinx.com>
Cc: Gerd Hoffmann <kraxel@redhat.com>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Reviewed-by: Eric Blake <eblake@redhat.com>
Reviewed-by: Gerd Hoffmann <kraxel@redhat.com>
Signed-off-by: Markus Armbruster <armbru@redhat.com>
Reviewed-by: Philippe Mathieu-Daudé <f4bug@amsat.org>
Message-Id: <20180203084315.20497-2-armbru@redhat.com>
2018-02-06 18:26:42 +01:00

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/*
* 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-common.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
}
}
}
};
void audio_sample_to_uint64(void *samples, int pos,
uint64_t *left, uint64_t *right)
{
struct st_sample *sample = samples;
sample += pos;
#ifdef FLOAT_MIXENG
error_report(
"Coreaudio and floating point samples are not supported by replay yet");
abort();
#else
*left = sample->l;
*right = sample->r;
#endif
}
void audio_sample_from_uint64(void *samples, int pos,
uint64_t left, uint64_t right)
{
struct st_sample *sample = samples;
sample += pos;
#ifdef FLOAT_MIXENG
error_report(
"Coreaudio and floating point samples are not supported by replay yet");
abort();
#else
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 = audio_calloc(__func__, 1, sizeof(*rate));
if (!rate) {
dolog ("Could not allocate resampler (%zu bytes)\n", sizeof (*rate));
return NULL;
}
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);
}
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;
}
}
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