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qemu/audio/mixeng.c
Volker Rümelin 1a01df3db8 audio: make playback packet length calculation exact 
Introduce the new function st_rate_frames_in() to calculate the
exact number of audio input frames needed to get a given number
of audio output frames. The exact number of frames depends only
on the difference of opos - ipos and the number of output frames.
When downsampling, this function returns the maximum number of
input frames needed.

This new function replaces the audio_frontend_frames_out() function,
which calculated the average number of input frames rounded down
to the nearest integer. Because audio_frontend_frames_out() also
limited the number of input frames to the size of the resample
buffer, st_rate_frames_in() is not a direct replacement and two
additional MIN() functions are needed. One to prevent resample
buffer overflows and one to limit the available bytes for the audio
frontends.

After this patch the audio packet length calculation for playback is
exact. When upsampling, it's still possible that the audio frontends
can't write the last audio frame. This will be fixed later.

Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Message-Id: <20230224190555.7409-9-vr_qemu@t-online.de>
2023-03-06 10:30:23 +04:00

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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/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_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;
}
}
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