mirror of
https://gitlab.com/qemu-project/qemu
synced 2024-11-05 20:35:44 +00:00
bd37ede4eb
Enable the SDL2 backend options -audiodev sdl,out.mixing- engine=off,in.mixing-engine=off. Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-id: 9315afe5-5958-c0b4-ea1e-14769511a9d5@t-online.de Message-Id: <20210110100239.27588-11-vr_qemu@t-online.de> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
508 lines
14 KiB
C
508 lines
14 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/module.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 exit;
|
|
int initialized;
|
|
Audiodev *dev;
|
|
SDL_AudioDeviceID devid;
|
|
} SDLVoiceOut;
|
|
|
|
typedef struct SDLVoiceIn {
|
|
HWVoiceIn hw;
|
|
int exit;
|
|
int initialized;
|
|
Audiodev *dev;
|
|
SDL_AudioDeviceID devid;
|
|
} SDLVoiceIn;
|
|
|
|
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 (AudioFormat fmt)
|
|
{
|
|
switch (fmt) {
|
|
case AUDIO_FORMAT_S8:
|
|
return AUDIO_S8;
|
|
|
|
case AUDIO_FORMAT_U8:
|
|
return AUDIO_U8;
|
|
|
|
case AUDIO_FORMAT_S16:
|
|
return AUDIO_S16LSB;
|
|
|
|
case AUDIO_FORMAT_U16:
|
|
return AUDIO_U16LSB;
|
|
|
|
case AUDIO_FORMAT_S32:
|
|
return AUDIO_S32LSB;
|
|
|
|
/* no unsigned 32-bit support in SDL */
|
|
|
|
case AUDIO_FORMAT_F32:
|
|
return AUDIO_F32LSB;
|
|
|
|
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, AudioFormat *fmt, int *endianness)
|
|
{
|
|
switch (sdlfmt) {
|
|
case AUDIO_S8:
|
|
*endianness = 0;
|
|
*fmt = AUDIO_FORMAT_S8;
|
|
break;
|
|
|
|
case AUDIO_U8:
|
|
*endianness = 0;
|
|
*fmt = AUDIO_FORMAT_U8;
|
|
break;
|
|
|
|
case AUDIO_S16LSB:
|
|
*endianness = 0;
|
|
*fmt = AUDIO_FORMAT_S16;
|
|
break;
|
|
|
|
case AUDIO_U16LSB:
|
|
*endianness = 0;
|
|
*fmt = AUDIO_FORMAT_U16;
|
|
break;
|
|
|
|
case AUDIO_S16MSB:
|
|
*endianness = 1;
|
|
*fmt = AUDIO_FORMAT_S16;
|
|
break;
|
|
|
|
case AUDIO_U16MSB:
|
|
*endianness = 1;
|
|
*fmt = AUDIO_FORMAT_U16;
|
|
break;
|
|
|
|
case AUDIO_S32LSB:
|
|
*endianness = 0;
|
|
*fmt = AUDIO_FORMAT_S32;
|
|
break;
|
|
|
|
case AUDIO_S32MSB:
|
|
*endianness = 1;
|
|
*fmt = AUDIO_FORMAT_S32;
|
|
break;
|
|
|
|
case AUDIO_F32LSB:
|
|
*endianness = 0;
|
|
*fmt = AUDIO_FORMAT_F32;
|
|
break;
|
|
|
|
case AUDIO_F32MSB:
|
|
*endianness = 1;
|
|
*fmt = AUDIO_FORMAT_F32;
|
|
break;
|
|
|
|
default:
|
|
dolog ("Unrecognized SDL audio format %d\n", sdlfmt);
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static SDL_AudioDeviceID sdl_open(SDL_AudioSpec *req, SDL_AudioSpec *obt,
|
|
int rec)
|
|
{
|
|
SDL_AudioDeviceID devid;
|
|
#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 0;
|
|
}
|
|
err = pthread_sigmask (SIG_BLOCK, &new, &old);
|
|
if (err) {
|
|
dolog ("sdl_open: pthread_sigmask failed: %s\n", strerror (err));
|
|
return 0;
|
|
}
|
|
#endif
|
|
|
|
devid = SDL_OpenAudioDevice(NULL, rec, req, obt, 0);
|
|
if (!devid) {
|
|
sdl_logerr("SDL_OpenAudioDevice for %s failed\n",
|
|
rec ? "recording" : "playback");
|
|
}
|
|
|
|
#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 devid;
|
|
}
|
|
|
|
static void sdl_close_out(SDLVoiceOut *sdl)
|
|
{
|
|
if (sdl->initialized) {
|
|
SDL_LockAudioDevice(sdl->devid);
|
|
sdl->exit = 1;
|
|
SDL_UnlockAudioDevice(sdl->devid);
|
|
SDL_PauseAudioDevice(sdl->devid, 1);
|
|
sdl->initialized = 0;
|
|
}
|
|
if (sdl->devid) {
|
|
SDL_CloseAudioDevice(sdl->devid);
|
|
sdl->devid = 0;
|
|
}
|
|
}
|
|
|
|
static void sdl_callback_out(void *opaque, Uint8 *buf, int len)
|
|
{
|
|
SDLVoiceOut *sdl = opaque;
|
|
HWVoiceOut *hw = &sdl->hw;
|
|
|
|
if (!sdl->exit) {
|
|
|
|
/* dolog("callback_out: len=%d avail=%zu\n", len, hw->pending_emul); */
|
|
|
|
while (hw->pending_emul && len) {
|
|
size_t write_len;
|
|
ssize_t start = (ssize_t)hw->pos_emul - hw->pending_emul;
|
|
if (start < 0) {
|
|
start += hw->size_emul;
|
|
}
|
|
assert(start >= 0 && start < hw->size_emul);
|
|
|
|
write_len = MIN(MIN(hw->pending_emul, len),
|
|
hw->size_emul - start);
|
|
|
|
memcpy(buf, hw->buf_emul + start, write_len);
|
|
hw->pending_emul -= write_len;
|
|
len -= write_len;
|
|
buf += write_len;
|
|
}
|
|
}
|
|
|
|
/* clear remaining buffer that we couldn't fill with data */
|
|
if (len) {
|
|
audio_pcm_info_clear_buf(&hw->info, buf,
|
|
len / hw->info.bytes_per_frame);
|
|
}
|
|
}
|
|
|
|
static void sdl_close_in(SDLVoiceIn *sdl)
|
|
{
|
|
if (sdl->initialized) {
|
|
SDL_LockAudioDevice(sdl->devid);
|
|
sdl->exit = 1;
|
|
SDL_UnlockAudioDevice(sdl->devid);
|
|
SDL_PauseAudioDevice(sdl->devid, 1);
|
|
sdl->initialized = 0;
|
|
}
|
|
if (sdl->devid) {
|
|
SDL_CloseAudioDevice(sdl->devid);
|
|
sdl->devid = 0;
|
|
}
|
|
}
|
|
|
|
static void sdl_callback_in(void *opaque, Uint8 *buf, int len)
|
|
{
|
|
SDLVoiceIn *sdl = opaque;
|
|
HWVoiceIn *hw = &sdl->hw;
|
|
|
|
if (sdl->exit) {
|
|
return;
|
|
}
|
|
|
|
/* dolog("callback_in: len=%d pending=%zu\n", len, hw->pending_emul); */
|
|
|
|
while (hw->pending_emul < hw->size_emul && len) {
|
|
size_t read_len = MIN(len, MIN(hw->size_emul - hw->pos_emul,
|
|
hw->size_emul - hw->pending_emul));
|
|
|
|
memcpy(hw->buf_emul + hw->pos_emul, buf, read_len);
|
|
|
|
hw->pending_emul += read_len;
|
|
hw->pos_emul = (hw->pos_emul + read_len) % hw->size_emul;
|
|
len -= read_len;
|
|
buf += read_len;
|
|
}
|
|
}
|
|
|
|
#define SDL_WRAPPER_FUNC(name, ret_type, args_decl, args, dir) \
|
|
static ret_type glue(sdl_, name)args_decl \
|
|
{ \
|
|
ret_type ret; \
|
|
glue(SDLVoice, dir) *sdl = (glue(SDLVoice, dir) *)hw; \
|
|
\
|
|
SDL_LockAudioDevice(sdl->devid); \
|
|
ret = glue(audio_generic_, name)args; \
|
|
SDL_UnlockAudioDevice(sdl->devid); \
|
|
\
|
|
return ret; \
|
|
}
|
|
|
|
#define SDL_WRAPPER_VOID_FUNC(name, args_decl, args, dir) \
|
|
static void glue(sdl_, name)args_decl \
|
|
{ \
|
|
glue(SDLVoice, dir) *sdl = (glue(SDLVoice, dir) *)hw; \
|
|
\
|
|
SDL_LockAudioDevice(sdl->devid); \
|
|
glue(audio_generic_, name)args; \
|
|
SDL_UnlockAudioDevice(sdl->devid); \
|
|
}
|
|
|
|
SDL_WRAPPER_FUNC(get_buffer_out, void *, (HWVoiceOut *hw, size_t *size),
|
|
(hw, size), Out)
|
|
SDL_WRAPPER_FUNC(put_buffer_out, size_t,
|
|
(HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size), Out)
|
|
SDL_WRAPPER_FUNC(write, size_t,
|
|
(HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size), Out)
|
|
SDL_WRAPPER_FUNC(read, size_t, (HWVoiceIn *hw, void *buf, size_t size),
|
|
(hw, buf, size), In)
|
|
SDL_WRAPPER_FUNC(get_buffer_in, void *, (HWVoiceIn *hw, size_t *size),
|
|
(hw, size), In)
|
|
SDL_WRAPPER_VOID_FUNC(put_buffer_in, (HWVoiceIn *hw, void *buf, size_t size),
|
|
(hw, buf, size), In)
|
|
#undef SDL_WRAPPER_FUNC
|
|
#undef SDL_WRAPPER_VOID_FUNC
|
|
|
|
static void sdl_fini_out(HWVoiceOut *hw)
|
|
{
|
|
SDLVoiceOut *sdl = (SDLVoiceOut *)hw;
|
|
|
|
sdl_close_out(sdl);
|
|
}
|
|
|
|
static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
|
|
void *drv_opaque)
|
|
{
|
|
SDLVoiceOut *sdl = (SDLVoiceOut *)hw;
|
|
SDL_AudioSpec req, obt;
|
|
int endianness;
|
|
int err;
|
|
AudioFormat effective_fmt;
|
|
Audiodev *dev = drv_opaque;
|
|
AudiodevSdlPerDirectionOptions *spdo = dev->u.sdl.out;
|
|
struct audsettings obt_as;
|
|
|
|
req.freq = as->freq;
|
|
req.format = aud_to_sdlfmt (as->fmt);
|
|
req.channels = as->nchannels;
|
|
/*
|
|
* This is wrong. SDL samples are QEMU frames. The buffer size will be
|
|
* the requested buffer size multiplied by the number of channels.
|
|
*/
|
|
req.samples = audio_buffer_samples(
|
|
qapi_AudiodevSdlPerDirectionOptions_base(spdo), as, 11610);
|
|
req.callback = sdl_callback_out;
|
|
req.userdata = sdl;
|
|
|
|
sdl->dev = dev;
|
|
sdl->devid = sdl_open(&req, &obt, 0);
|
|
if (!sdl->devid) {
|
|
return -1;
|
|
}
|
|
|
|
err = sdl_to_audfmt(obt.format, &effective_fmt, &endianness);
|
|
if (err) {
|
|
sdl_close_out(sdl);
|
|
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 = (spdo->has_buffer_count ? spdo->buffer_count : 4) *
|
|
obt.samples;
|
|
|
|
sdl->initialized = 1;
|
|
sdl->exit = 0;
|
|
return 0;
|
|
}
|
|
|
|
static void sdl_enable_out(HWVoiceOut *hw, bool enable)
|
|
{
|
|
SDLVoiceOut *sdl = (SDLVoiceOut *)hw;
|
|
|
|
SDL_PauseAudioDevice(sdl->devid, !enable);
|
|
}
|
|
|
|
static void sdl_fini_in(HWVoiceIn *hw)
|
|
{
|
|
SDLVoiceIn *sdl = (SDLVoiceIn *)hw;
|
|
|
|
sdl_close_in(sdl);
|
|
}
|
|
|
|
static int sdl_init_in(HWVoiceIn *hw, audsettings *as, void *drv_opaque)
|
|
{
|
|
SDLVoiceIn *sdl = (SDLVoiceIn *)hw;
|
|
SDL_AudioSpec req, obt;
|
|
int endianness;
|
|
int err;
|
|
AudioFormat effective_fmt;
|
|
Audiodev *dev = drv_opaque;
|
|
AudiodevSdlPerDirectionOptions *spdo = dev->u.sdl.in;
|
|
struct audsettings obt_as;
|
|
|
|
req.freq = as->freq;
|
|
req.format = aud_to_sdlfmt(as->fmt);
|
|
req.channels = as->nchannels;
|
|
/* SDL samples are QEMU frames */
|
|
req.samples = audio_buffer_frames(
|
|
qapi_AudiodevSdlPerDirectionOptions_base(spdo), as, 11610);
|
|
req.callback = sdl_callback_in;
|
|
req.userdata = sdl;
|
|
|
|
sdl->dev = dev;
|
|
sdl->devid = sdl_open(&req, &obt, 1);
|
|
if (!sdl->devid) {
|
|
return -1;
|
|
}
|
|
|
|
err = sdl_to_audfmt(obt.format, &effective_fmt, &endianness);
|
|
if (err) {
|
|
sdl_close_in(sdl);
|
|
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 = (spdo->has_buffer_count ? spdo->buffer_count : 4) *
|
|
obt.samples;
|
|
hw->size_emul = hw->samples * hw->info.bytes_per_frame;
|
|
hw->buf_emul = g_malloc(hw->size_emul);
|
|
hw->pos_emul = hw->pending_emul = 0;
|
|
|
|
sdl->initialized = 1;
|
|
sdl->exit = 0;
|
|
return 0;
|
|
}
|
|
|
|
static void sdl_enable_in(HWVoiceIn *hw, bool enable)
|
|
{
|
|
SDLVoiceIn *sdl = (SDLVoiceIn *)hw;
|
|
|
|
SDL_PauseAudioDevice(sdl->devid, !enable);
|
|
}
|
|
|
|
static void *sdl_audio_init(Audiodev *dev)
|
|
{
|
|
if (SDL_InitSubSystem (SDL_INIT_AUDIO)) {
|
|
sdl_logerr ("SDL failed to initialize audio subsystem\n");
|
|
return NULL;
|
|
}
|
|
|
|
return dev;
|
|
}
|
|
|
|
static void sdl_audio_fini (void *opaque)
|
|
{
|
|
SDL_QuitSubSystem (SDL_INIT_AUDIO);
|
|
}
|
|
|
|
static struct audio_pcm_ops sdl_pcm_ops = {
|
|
.init_out = sdl_init_out,
|
|
.fini_out = sdl_fini_out,
|
|
/* wrapper for audio_generic_write */
|
|
.write = sdl_write,
|
|
/* wrapper for audio_generic_get_buffer_out */
|
|
.get_buffer_out = sdl_get_buffer_out,
|
|
/* wrapper for audio_generic_put_buffer_out */
|
|
.put_buffer_out = sdl_put_buffer_out,
|
|
.enable_out = sdl_enable_out,
|
|
.init_in = sdl_init_in,
|
|
.fini_in = sdl_fini_in,
|
|
/* wrapper for audio_generic_read */
|
|
.read = sdl_read,
|
|
/* wrapper for audio_generic_get_buffer_in */
|
|
.get_buffer_in = sdl_get_buffer_in,
|
|
/* wrapper for audio_generic_put_buffer_in */
|
|
.put_buffer_in = sdl_put_buffer_in,
|
|
.enable_in = sdl_enable_in,
|
|
};
|
|
|
|
static struct audio_driver sdl_audio_driver = {
|
|
.name = "sdl",
|
|
.descr = "SDL http://www.libsdl.org",
|
|
.init = sdl_audio_init,
|
|
.fini = sdl_audio_fini,
|
|
.pcm_ops = &sdl_pcm_ops,
|
|
.can_be_default = 1,
|
|
.max_voices_out = INT_MAX,
|
|
.max_voices_in = INT_MAX,
|
|
.voice_size_out = sizeof(SDLVoiceOut),
|
|
.voice_size_in = sizeof(SDLVoiceIn),
|
|
};
|
|
|
|
static void register_audio_sdl(void)
|
|
{
|
|
audio_driver_register(&sdl_audio_driver);
|
|
}
|
|
type_init(register_audio_sdl);
|