The combined generic buffer management code and buffer run out code in function audio_generic_put_buffer_out has a problematic behaviour. A few hundred milliseconds after playback starts the mixing buffer and the generic buffer are nearly full and the following pattern can be seen. On first call of audio_pcm_hw_run_out the buffer run code in audio_generic_put_buffer_out writes some data to the audio hardware but the generic buffer will fill faster and is full when audio_pcm_hw_run_out returns. This is because emulated audio devices can produce playback data at a higher rate than the audio backend hardware consumes this data. On next call of audio_pcm_hw_run_out the buffer run code in audio_generic_put_buffer_out writes some data to the audio hardware but no audio data is transferred to the generic buffer because the buffer is already full. Then the pattern repeats. For the emulated audio device this looks like the audio timer period has doubled. This patch splits the combined generic buffer management code and buffer run out code and calls the buffer run out code after buffer management code to break this pattern. The bug report is for the wav audio backend. But the problem is not limited to this backend. All audio backends which use the audio_generic_put_buffer_out function show this problem. Buglink: https://bugs.launchpad.net/qemu/+bug/1858488 Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> Message-Id: <20200123074943.6699-5-vr_qemu@t-online.de> Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
350 lines
9.1 KiB
C
350 lines
9.1 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/module.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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} SDLVoiceOut;
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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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Audiodev *dev;
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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 (AudioFormat fmt)
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{
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switch (fmt) {
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case AUDIO_FORMAT_S8:
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return AUDIO_S8;
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case AUDIO_FORMAT_U8:
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return AUDIO_U8;
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case AUDIO_FORMAT_S16:
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return AUDIO_S16LSB;
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case AUDIO_FORMAT_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, AudioFormat *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 = AUDIO_FORMAT_S8;
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break;
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case AUDIO_U8:
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*endianness = 0;
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*fmt = AUDIO_FORMAT_U8;
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break;
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case AUDIO_S16LSB:
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*endianness = 0;
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*fmt = AUDIO_FORMAT_S16;
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break;
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case AUDIO_U16LSB:
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*endianness = 0;
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*fmt = AUDIO_FORMAT_U16;
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break;
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case AUDIO_S16MSB:
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*endianness = 1;
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*fmt = AUDIO_FORMAT_S16;
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break;
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case AUDIO_U16MSB:
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*endianness = 1;
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*fmt = AUDIO_FORMAT_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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if (s->exit) {
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return;
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}
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/* dolog ("in callback samples=%zu live=%zu\n", samples, sdl->live); */
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while (hw->pending_emul && len) {
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size_t write_len;
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ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
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if (start < 0) {
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start += hw->size_emul;
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}
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assert(start >= 0 && start < hw->size_emul);
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write_len = MIN(MIN(hw->pending_emul, len),
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hw->size_emul - start);
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memcpy(buf, hw->buf_emul + start, write_len);
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hw->pending_emul -= write_len;
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len -= write_len;
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buf += write_len;
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}
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/* clear remaining buffer that we couldn't fill with data */
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if (len) {
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memset(buf, 0, len);
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}
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}
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#define SDL_WRAPPER_FUNC(name, ret_type, args_decl, args, fail, unlock) \
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static ret_type glue(sdl_, name)args_decl \
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{ \
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ret_type ret; \
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\
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SDL_LockAudio(); \
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\
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ret = glue(audio_generic_, name)args; \
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\
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SDL_UnlockAudio(); \
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return ret; \
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}
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SDL_WRAPPER_FUNC(get_buffer_out, void *, (HWVoiceOut *hw, size_t *size),
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(hw, size), *size = 0, sdl_unlock)
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SDL_WRAPPER_FUNC(put_buffer_out, size_t,
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(HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size),
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/*nothing*/, sdl_unlock_and_post)
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SDL_WRAPPER_FUNC(write, size_t,
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(HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size),
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/*nothing*/, sdl_unlock_and_post)
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#undef SDL_WRAPPER_FUNC
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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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AudioFormat 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 = audio_buffer_samples(s->dev->u.sdl.out, as, 11610);
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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 void sdl_enable_out(HWVoiceOut *hw, bool enable)
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{
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SDL_PauseAudio(!enable);
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}
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static void *sdl_audio_init(Audiodev *dev)
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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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s->dev = dev;
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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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s->dev = NULL;
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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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/* wrapper for audio_generic_write */
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.write = sdl_write,
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/* wrapper for audio_generic_get_buffer_out */
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.get_buffer_out = sdl_get_buffer_out,
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/* wrapper for audio_generic_put_buffer_out */
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.put_buffer_out = sdl_put_buffer_out,
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.enable_out = sdl_enable_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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.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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