qemu-e2k/audio/sdlaudio.c
Thomas Huth 9399ef1683 audio/sdlaudio: Simplify the sdl_callback function
At the end of the while-loop, either "samples" or "sdl->live" is zero, so
now that we've removed the semaphore code, the content of the while-loop
is always only executed once. Thus we can remove the while-loop now to
get rid of one indentation level here.

Signed-off-by: Thomas Huth <thuth@redhat.com>
Message-id: 1549336101-17623-3-git-send-email-thuth@redhat.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-02-28 10:30:08 +01:00

380 lines
8.6 KiB
C

/*
* QEMU SDL audio driver
*
* Copyright (c) 2004-2005 Vassili Karpov (malc)
*
* 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 <SDL.h>
#include <SDL_thread.h>
#include "qemu-common.h"
#include "audio.h"
#ifndef _WIN32
#ifdef __sun__
#define _POSIX_PTHREAD_SEMANTICS 1
#elif defined(__OpenBSD__) || defined(__FreeBSD__) || defined(__DragonFly__)
#include <pthread.h>
#endif
#endif
#define AUDIO_CAP "sdl"
#include "audio_int.h"
typedef struct SDLVoiceOut {
HWVoiceOut hw;
int live;
int decr;
} SDLVoiceOut;
static struct {
int nb_samples;
} conf = {
.nb_samples = 1024
};
static struct SDLAudioState {
int exit;
int initialized;
bool driver_created;
} glob_sdl;
typedef struct SDLAudioState SDLAudioState;
static void GCC_FMT_ATTR (1, 2) sdl_logerr (const char *fmt, ...)
{
va_list ap;
va_start (ap, fmt);
AUD_vlog (AUDIO_CAP, fmt, ap);
va_end (ap);
AUD_log (AUDIO_CAP, "Reason: %s\n", SDL_GetError ());
}
static int aud_to_sdlfmt (audfmt_e fmt)
{
switch (fmt) {
case AUD_FMT_S8:
return AUDIO_S8;
case AUD_FMT_U8:
return AUDIO_U8;
case AUD_FMT_S16:
return AUDIO_S16LSB;
case AUD_FMT_U16:
return AUDIO_U16LSB;
default:
dolog ("Internal logic error: Bad audio format %d\n", fmt);
#ifdef DEBUG_AUDIO
abort ();
#endif
return AUDIO_U8;
}
}
static int sdl_to_audfmt(int sdlfmt, audfmt_e *fmt, int *endianness)
{
switch (sdlfmt) {
case AUDIO_S8:
*endianness = 0;
*fmt = AUD_FMT_S8;
break;
case AUDIO_U8:
*endianness = 0;
*fmt = AUD_FMT_U8;
break;
case AUDIO_S16LSB:
*endianness = 0;
*fmt = AUD_FMT_S16;
break;
case AUDIO_U16LSB:
*endianness = 0;
*fmt = AUD_FMT_U16;
break;
case AUDIO_S16MSB:
*endianness = 1;
*fmt = AUD_FMT_S16;
break;
case AUDIO_U16MSB:
*endianness = 1;
*fmt = AUD_FMT_U16;
break;
default:
dolog ("Unrecognized SDL audio format %d\n", sdlfmt);
return -1;
}
return 0;
}
static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt)
{
int status;
#ifndef _WIN32
int err;
sigset_t new, old;
/* Make sure potential threads created by SDL don't hog signals. */
err = sigfillset (&new);
if (err) {
dolog ("sdl_open: sigfillset failed: %s\n", strerror (errno));
return -1;
}
err = pthread_sigmask (SIG_BLOCK, &new, &old);
if (err) {
dolog ("sdl_open: pthread_sigmask failed: %s\n", strerror (err));
return -1;
}
#endif
status = SDL_OpenAudio (req, obt);
if (status) {
sdl_logerr ("SDL_OpenAudio failed\n");
}
#ifndef _WIN32
err = pthread_sigmask (SIG_SETMASK, &old, NULL);
if (err) {
dolog ("sdl_open: pthread_sigmask (restore) failed: %s\n",
strerror (errno));
/* We have failed to restore original signal mask, all bets are off,
so exit the process */
exit (EXIT_FAILURE);
}
#endif
return status;
}
static void sdl_close (SDLAudioState *s)
{
if (s->initialized) {
SDL_LockAudio();
s->exit = 1;
SDL_UnlockAudio();
SDL_PauseAudio (1);
SDL_CloseAudio ();
s->initialized = 0;
}
}
static void sdl_callback (void *opaque, Uint8 *buf, int len)
{
SDLVoiceOut *sdl = opaque;
SDLAudioState *s = &glob_sdl;
HWVoiceOut *hw = &sdl->hw;
int samples = len >> hw->info.shift;
int to_mix, decr;
if (s->exit || !sdl->live) {
return;
}
/* dolog ("in callback samples=%d live=%d\n", samples, sdl->live); */
to_mix = audio_MIN(samples, sdl->live);
decr = to_mix;
while (to_mix) {
int chunk = audio_MIN(to_mix, hw->samples - hw->rpos);
struct st_sample *src = hw->mix_buf + hw->rpos;
/* dolog ("in callback to_mix %d, chunk %d\n", to_mix, chunk); */
hw->clip(buf, src, chunk);
hw->rpos = (hw->rpos + chunk) % hw->samples;
to_mix -= chunk;
buf += chunk << hw->info.shift;
}
samples -= decr;
sdl->live -= decr;
sdl->decr += decr;
/* dolog ("done len=%d\n", len); */
/* SDL2 does not clear the remaining buffer for us, so do it on our own */
if (samples) {
memset(buf, 0, samples << hw->info.shift);
}
}
static int sdl_write_out (SWVoiceOut *sw, void *buf, int len)
{
return audio_pcm_sw_write (sw, buf, len);
}
static int sdl_run_out (HWVoiceOut *hw, int live)
{
int decr;
SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
SDL_LockAudio();
if (sdl->decr > live) {
ldebug ("sdl->decr %d live %d sdl->live %d\n",
sdl->decr,
live,
sdl->live);
}
decr = audio_MIN (sdl->decr, live);
sdl->decr -= decr;
sdl->live = live;
SDL_UnlockAudio();
return decr;
}
static void sdl_fini_out (HWVoiceOut *hw)
{
(void) hw;
sdl_close (&glob_sdl);
}
static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
void *drv_opaque)
{
SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
SDLAudioState *s = &glob_sdl;
SDL_AudioSpec req, obt;
int endianness;
int err;
audfmt_e effective_fmt;
struct audsettings obt_as;
req.freq = as->freq;
req.format = aud_to_sdlfmt (as->fmt);
req.channels = as->nchannels;
req.samples = conf.nb_samples;
req.callback = sdl_callback;
req.userdata = sdl;
if (sdl_open (&req, &obt)) {
return -1;
}
err = sdl_to_audfmt(obt.format, &effective_fmt, &endianness);
if (err) {
sdl_close (s);
return -1;
}
obt_as.freq = obt.freq;
obt_as.nchannels = obt.channels;
obt_as.fmt = effective_fmt;
obt_as.endianness = endianness;
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
s->initialized = 1;
s->exit = 0;
SDL_PauseAudio (0);
return 0;
}
static int sdl_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
(void) hw;
switch (cmd) {
case VOICE_ENABLE:
SDL_PauseAudio (0);
break;
case VOICE_DISABLE:
SDL_PauseAudio (1);
break;
}
return 0;
}
static void *sdl_audio_init (void)
{
SDLAudioState *s = &glob_sdl;
if (s->driver_created) {
sdl_logerr("Can't create multiple sdl backends\n");
return NULL;
}
if (SDL_InitSubSystem (SDL_INIT_AUDIO)) {
sdl_logerr ("SDL failed to initialize audio subsystem\n");
return NULL;
}
s->driver_created = true;
return s;
}
static void sdl_audio_fini (void *opaque)
{
SDLAudioState *s = opaque;
sdl_close (s);
SDL_QuitSubSystem (SDL_INIT_AUDIO);
s->driver_created = false;
}
static struct audio_option sdl_options[] = {
{
.name = "SAMPLES",
.tag = AUD_OPT_INT,
.valp = &conf.nb_samples,
.descr = "Size of SDL buffer in samples"
},
{ /* End of list */ }
};
static struct audio_pcm_ops sdl_pcm_ops = {
.init_out = sdl_init_out,
.fini_out = sdl_fini_out,
.run_out = sdl_run_out,
.write = sdl_write_out,
.ctl_out = sdl_ctl_out,
};
static struct audio_driver sdl_audio_driver = {
.name = "sdl",
.descr = "SDL http://www.libsdl.org",
.options = sdl_options,
.init = sdl_audio_init,
.fini = sdl_audio_fini,
.pcm_ops = &sdl_pcm_ops,
.can_be_default = 1,
.max_voices_out = 1,
.max_voices_in = 0,
.voice_size_out = sizeof (SDLVoiceOut),
.voice_size_in = 0
};
static void register_audio_sdl(void)
{
audio_driver_register(&sdl_audio_driver);
}
type_init(register_audio_sdl);