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9833438ef6
qemu/audio/sdlaudio.c
Volker Rümelin 9833438ef6 audio: restore mixing-engine playback buffer size 
Commit ff095e5231 "audio: api for mixeng code free backends"
introduced another FIFO for the audio subsystem with exactly the
same size as the mixing-engine FIFO. Most audio backends use
this generic FIFO. The generic FIFO used together with the
mixing-engine FIFO doubles the audio FIFO size, because that's
just two independent FIFOs connected together in series.

For audio playback this nearly doubles the playback latency.

This patch restores the effective mixing-engine playback buffer
size to a pre v4.2.0 size by only accepting the amount of
samples for the mixing-engine queue which the downstream queue
accepts.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Message-Id: <20220301191311.26695-10-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2022-03-04 11:05:13 +01:00

511 lines
14 KiB
C
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/*
* 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, start;
start = audio_ring_posb(hw->pos_emul, hw->pending_emul,
hw->size_emul);
assert(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(buffer_get_free, size_t, (HWVoiceOut *hw), (hw), Out)
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_buffer_get_free */
.buffer_get_free = sdl_buffer_get_free,
/* 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);
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