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>
380 lines
8.6 KiB
C
380 lines
8.6 KiB
C
/*
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* QEMU SDL audio driver
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*
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* Copyright (c) 2004-2005 Vassili Karpov (malc)
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
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* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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* THE SOFTWARE.
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*/
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#include "qemu/osdep.h"
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#include <SDL.h>
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#include <SDL_thread.h>
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#include "qemu-common.h"
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#include "audio.h"
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#ifndef _WIN32
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#ifdef __sun__
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#define _POSIX_PTHREAD_SEMANTICS 1
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#elif defined(__OpenBSD__) || defined(__FreeBSD__) || defined(__DragonFly__)
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#include <pthread.h>
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#endif
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#endif
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#define AUDIO_CAP "sdl"
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#include "audio_int.h"
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typedef struct SDLVoiceOut {
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HWVoiceOut hw;
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int live;
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int decr;
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} SDLVoiceOut;
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static struct {
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int nb_samples;
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} conf = {
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.nb_samples = 1024
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};
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static struct SDLAudioState {
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int exit;
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int initialized;
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bool driver_created;
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} glob_sdl;
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typedef struct SDLAudioState SDLAudioState;
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static void GCC_FMT_ATTR (1, 2) sdl_logerr (const char *fmt, ...)
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{
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va_list ap;
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va_start (ap, fmt);
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AUD_vlog (AUDIO_CAP, fmt, ap);
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va_end (ap);
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AUD_log (AUDIO_CAP, "Reason: %s\n", SDL_GetError ());
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}
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static int aud_to_sdlfmt (audfmt_e fmt)
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{
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switch (fmt) {
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case AUD_FMT_S8:
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return AUDIO_S8;
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case AUD_FMT_U8:
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return AUDIO_U8;
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case AUD_FMT_S16:
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return AUDIO_S16LSB;
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case AUD_FMT_U16:
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return AUDIO_U16LSB;
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default:
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dolog ("Internal logic error: Bad audio format %d\n", fmt);
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#ifdef DEBUG_AUDIO
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abort ();
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#endif
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return AUDIO_U8;
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}
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}
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static int sdl_to_audfmt(int sdlfmt, audfmt_e *fmt, int *endianness)
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{
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switch (sdlfmt) {
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case AUDIO_S8:
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*endianness = 0;
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*fmt = AUD_FMT_S8;
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break;
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case AUDIO_U8:
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*endianness = 0;
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*fmt = AUD_FMT_U8;
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break;
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case AUDIO_S16LSB:
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*endianness = 0;
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*fmt = AUD_FMT_S16;
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break;
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case AUDIO_U16LSB:
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*endianness = 0;
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*fmt = AUD_FMT_U16;
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break;
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case AUDIO_S16MSB:
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*endianness = 1;
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*fmt = AUD_FMT_S16;
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break;
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case AUDIO_U16MSB:
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*endianness = 1;
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*fmt = AUD_FMT_U16;
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break;
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default:
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dolog ("Unrecognized SDL audio format %d\n", sdlfmt);
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return -1;
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}
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return 0;
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}
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static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt)
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{
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int status;
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#ifndef _WIN32
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int err;
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sigset_t new, old;
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/* Make sure potential threads created by SDL don't hog signals. */
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err = sigfillset (&new);
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if (err) {
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dolog ("sdl_open: sigfillset failed: %s\n", strerror (errno));
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return -1;
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}
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err = pthread_sigmask (SIG_BLOCK, &new, &old);
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if (err) {
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dolog ("sdl_open: pthread_sigmask failed: %s\n", strerror (err));
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return -1;
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}
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#endif
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status = SDL_OpenAudio (req, obt);
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if (status) {
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sdl_logerr ("SDL_OpenAudio failed\n");
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}
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#ifndef _WIN32
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err = pthread_sigmask (SIG_SETMASK, &old, NULL);
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if (err) {
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dolog ("sdl_open: pthread_sigmask (restore) failed: %s\n",
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strerror (errno));
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/* We have failed to restore original signal mask, all bets are off,
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so exit the process */
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exit (EXIT_FAILURE);
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}
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#endif
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return status;
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}
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static void sdl_close (SDLAudioState *s)
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{
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if (s->initialized) {
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SDL_LockAudio();
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s->exit = 1;
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SDL_UnlockAudio();
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SDL_PauseAudio (1);
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SDL_CloseAudio ();
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s->initialized = 0;
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}
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}
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static void sdl_callback (void *opaque, Uint8 *buf, int len)
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{
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SDLVoiceOut *sdl = opaque;
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SDLAudioState *s = &glob_sdl;
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HWVoiceOut *hw = &sdl->hw;
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int samples = len >> hw->info.shift;
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int to_mix, decr;
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if (s->exit || !sdl->live) {
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return;
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}
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/* dolog ("in callback samples=%d live=%d\n", samples, sdl->live); */
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to_mix = audio_MIN(samples, sdl->live);
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decr = to_mix;
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while (to_mix) {
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int chunk = audio_MIN(to_mix, hw->samples - hw->rpos);
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struct st_sample *src = hw->mix_buf + hw->rpos;
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/* dolog ("in callback to_mix %d, chunk %d\n", to_mix, chunk); */
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hw->clip(buf, src, chunk);
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hw->rpos = (hw->rpos + chunk) % hw->samples;
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to_mix -= chunk;
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buf += chunk << hw->info.shift;
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}
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samples -= decr;
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sdl->live -= decr;
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sdl->decr += decr;
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/* dolog ("done len=%d\n", len); */
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/* SDL2 does not clear the remaining buffer for us, so do it on our own */
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if (samples) {
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memset(buf, 0, samples << hw->info.shift);
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}
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}
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static int sdl_write_out (SWVoiceOut *sw, void *buf, int len)
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{
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return audio_pcm_sw_write (sw, buf, len);
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}
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static int sdl_run_out (HWVoiceOut *hw, int live)
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{
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int decr;
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SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
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SDL_LockAudio();
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if (sdl->decr > live) {
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ldebug ("sdl->decr %d live %d sdl->live %d\n",
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sdl->decr,
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live,
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sdl->live);
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}
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decr = audio_MIN (sdl->decr, live);
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sdl->decr -= decr;
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sdl->live = live;
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SDL_UnlockAudio();
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return decr;
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}
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static void sdl_fini_out (HWVoiceOut *hw)
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{
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(void) hw;
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sdl_close (&glob_sdl);
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}
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static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
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void *drv_opaque)
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{
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SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
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SDLAudioState *s = &glob_sdl;
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SDL_AudioSpec req, obt;
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int endianness;
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int err;
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audfmt_e effective_fmt;
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struct audsettings obt_as;
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req.freq = as->freq;
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req.format = aud_to_sdlfmt (as->fmt);
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req.channels = as->nchannels;
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req.samples = conf.nb_samples;
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req.callback = sdl_callback;
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req.userdata = sdl;
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if (sdl_open (&req, &obt)) {
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return -1;
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}
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err = sdl_to_audfmt(obt.format, &effective_fmt, &endianness);
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if (err) {
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sdl_close (s);
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return -1;
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}
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obt_as.freq = obt.freq;
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obt_as.nchannels = obt.channels;
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obt_as.fmt = effective_fmt;
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obt_as.endianness = endianness;
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audio_pcm_init_info (&hw->info, &obt_as);
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hw->samples = obt.samples;
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s->initialized = 1;
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s->exit = 0;
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SDL_PauseAudio (0);
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return 0;
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}
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static int sdl_ctl_out (HWVoiceOut *hw, int cmd, ...)
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{
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(void) hw;
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switch (cmd) {
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case VOICE_ENABLE:
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SDL_PauseAudio (0);
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break;
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case VOICE_DISABLE:
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SDL_PauseAudio (1);
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break;
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}
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return 0;
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}
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static void *sdl_audio_init (void)
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{
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SDLAudioState *s = &glob_sdl;
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if (s->driver_created) {
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sdl_logerr("Can't create multiple sdl backends\n");
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return NULL;
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}
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if (SDL_InitSubSystem (SDL_INIT_AUDIO)) {
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sdl_logerr ("SDL failed to initialize audio subsystem\n");
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return NULL;
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}
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s->driver_created = true;
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return s;
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}
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static void sdl_audio_fini (void *opaque)
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{
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SDLAudioState *s = opaque;
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sdl_close (s);
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SDL_QuitSubSystem (SDL_INIT_AUDIO);
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s->driver_created = false;
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}
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static struct audio_option sdl_options[] = {
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{
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.name = "SAMPLES",
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.tag = AUD_OPT_INT,
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.valp = &conf.nb_samples,
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.descr = "Size of SDL buffer in samples"
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},
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{ /* End of list */ }
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};
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static struct audio_pcm_ops sdl_pcm_ops = {
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.init_out = sdl_init_out,
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.fini_out = sdl_fini_out,
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.run_out = sdl_run_out,
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.write = sdl_write_out,
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.ctl_out = sdl_ctl_out,
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};
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static struct audio_driver sdl_audio_driver = {
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.name = "sdl",
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.descr = "SDL http://www.libsdl.org",
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.options = sdl_options,
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.init = sdl_audio_init,
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.fini = sdl_audio_fini,
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.pcm_ops = &sdl_pcm_ops,
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.can_be_default = 1,
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.max_voices_out = 1,
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.max_voices_in = 0,
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.voice_size_out = sizeof (SDLVoiceOut),
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.voice_size_in = 0
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};
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static void register_audio_sdl(void)
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{
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audio_driver_register(&sdl_audio_driver);
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}
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type_init(register_audio_sdl);
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