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author | Peter Maydell | 2021-01-15 23:21:21 +0100 |
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committer | Peter Maydell | 2021-01-15 23:21:21 +0100 |
commit | 825a215c003cd028e26c7d19aa5049d957345f43 (patch) | |
tree | a3bedcc1d73490abbe5994b065289147f8d0b10e /audio/sdlaudio.c | |
parent | Merge remote-tracking branch 'remotes/kraxel/tags/ui-20210115-pull-request' i... (diff) | |
parent | audio: space prohibited between function name and parenthesis'(' (diff) | |
download | qemu-825a215c003cd028e26c7d19aa5049d957345f43.tar.gz qemu-825a215c003cd028e26c7d19aa5049d957345f43.tar.xz qemu-825a215c003cd028e26c7d19aa5049d957345f43.zip |
Merge remote-tracking branch 'remotes/kraxel/tags/audio-20210115-pull-request' into staging
audio: improvements for sdl, pulse, fsound.
audio: cleanups & codestyle fixes.
# gpg: Signature made Fri 15 Jan 2021 13:20:56 GMT
# gpg: using RSA key A0328CFFB93A17A79901FE7D4CB6D8EED3E87138
# gpg: Good signature from "Gerd Hoffmann (work) <kraxel@redhat.com>" [full]
# gpg: aka "Gerd Hoffmann <gerd@kraxel.org>" [full]
# gpg: aka "Gerd Hoffmann (private) <kraxel@gmail.com>" [full]
# Primary key fingerprint: A032 8CFF B93A 17A7 9901 FE7D 4CB6 D8EE D3E8 7138
* remotes/kraxel/tags/audio-20210115-pull-request: (30 commits)
audio: space prohibited between function name and parenthesis'('
audio: Suspect code indent for conditional statements
audio: Don't use '%#' in format strings
audio: Fix lines over 90 characters
audio: foo* bar" should be "foo *bar".
audio: Add spaces around operator/delete redundant spaces
audio: Add braces for statements/fix braces' position
dsoundaudio: fix log message
dsoundaudio: enable f32 audio sample format
dsoundaudio: rename dsound_open()
dsoundaudio: replace GetForegroundWindow()
paaudio: send recorded data in smaller chunks
paaudio: limit minreq to 75% of audio timer_rate
paaudio: comment bugs in functions qpa_init_*
paaudio: remove unneeded code
paaudio: wait until the playback stream is ready
paaudio: wait for PA_STREAM_READY in qpa_write()
paaudio: avoid to clip samples multiple times
audio: remove remaining unused plive code
sdlaudio: enable (in|out).mixing-engine=off
...
Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
Diffstat (limited to 'audio/sdlaudio.c')
-rw-r--r-- | audio/sdlaudio.c | 305 |
1 files changed, 218 insertions, 87 deletions
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c index 21b7a0484b..c68c62a3e4 100644 --- a/audio/sdlaudio.c +++ b/audio/sdlaudio.c @@ -41,15 +41,19 @@ typedef struct SDLVoiceOut { HWVoiceOut hw; + int exit; + int initialized; + Audiodev *dev; + SDL_AudioDeviceID devid; } SDLVoiceOut; -static struct SDLAudioState { +typedef struct SDLVoiceIn { + HWVoiceIn hw; int exit; int initialized; - bool driver_created; Audiodev *dev; -} glob_sdl; -typedef struct SDLAudioState SDLAudioState; + SDL_AudioDeviceID devid; +} SDLVoiceIn; static void GCC_FMT_ATTR (1, 2) sdl_logerr (const char *fmt, ...) { @@ -155,9 +159,10 @@ static int sdl_to_audfmt(int sdlfmt, AudioFormat *fmt, int *endianness) return 0; } -static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt) +static SDL_AudioDeviceID sdl_open(SDL_AudioSpec *req, SDL_AudioSpec *obt, + int rec) { - int status; + SDL_AudioDeviceID devid; #ifndef _WIN32 int err; sigset_t new, old; @@ -166,18 +171,19 @@ static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt) err = sigfillset (&new); if (err) { dolog ("sdl_open: sigfillset failed: %s\n", strerror (errno)); - return -1; + return 0; } err = pthread_sigmask (SIG_BLOCK, &new, &old); if (err) { dolog ("sdl_open: pthread_sigmask failed: %s\n", strerror (err)); - return -1; + return 0; } #endif - status = SDL_OpenAudio (req, obt); - if (status) { - sdl_logerr ("SDL_OpenAudio failed\n"); + devid = SDL_OpenAudioDevice(NULL, rec, req, obt, 0); + if (!devid) { + sdl_logerr("SDL_OpenAudioDevice for %s failed\n", + rec ? "recording" : "playback"); } #ifndef _WIN32 @@ -190,112 +196,175 @@ static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt) exit (EXIT_FAILURE); } #endif - return status; + return devid; } -static void sdl_close (SDLAudioState *s) +static void sdl_close_out(SDLVoiceOut *sdl) { - if (s->initialized) { - SDL_LockAudio(); - s->exit = 1; - SDL_UnlockAudio(); - SDL_PauseAudio (1); - SDL_CloseAudio (); - s->initialized = 0; + 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 (void *opaque, Uint8 *buf, int len) +static void sdl_callback_out(void *opaque, Uint8 *buf, int len) { SDLVoiceOut *sdl = opaque; - SDLAudioState *s = &glob_sdl; HWVoiceOut *hw = &sdl->hw; - if (s->exit) { - return; - } + if (!sdl->exit) { - /* dolog ("in callback samples=%zu live=%zu\n", samples, sdl->live); */ + /* 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); + 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); + 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; + 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) { - memset(buf, 0, 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, fail, unlock) \ - static ret_type glue(sdl_, name)args_decl \ - { \ - ret_type ret; \ - \ - SDL_LockAudio(); \ - \ - ret = glue(audio_generic_, name)args; \ - \ - SDL_UnlockAudio(); \ - return ret; \ +#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), *size = 0, sdl_unlock) + (hw, size), Out) SDL_WRAPPER_FUNC(put_buffer_out, size_t, - (HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size), - /*nothing*/, sdl_unlock_and_post) + (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), - /*nothing*/, sdl_unlock_and_post) - + (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) +static void sdl_fini_out(HWVoiceOut *hw) { - (void) hw; + SDLVoiceOut *sdl = (SDLVoiceOut *)hw; - sdl_close (&glob_sdl); + sdl_close_out(sdl); } static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque) { - SDLVoiceOut *sdl = (SDLVoiceOut *) hw; - SDLAudioState *s = &glob_sdl; + 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; - req.samples = audio_buffer_samples(s->dev->u.sdl.out, as, 11610); - req.callback = sdl_callback; + /* + * 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; - if (sdl_open (&req, &obt)) { + 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 (s); + sdl_close_out(sdl); return -1; } @@ -305,44 +374,97 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as, obt_as.endianness = endianness; audio_pcm_init_info (&hw->info, &obt_as); - hw->samples = obt.samples; + hw->samples = (spdo->has_buffer_count ? spdo->buffer_count : 4) * + obt.samples; - s->initialized = 1; - s->exit = 0; - SDL_PauseAudio (0); + sdl->initialized = 1; + sdl->exit = 0; return 0; } static void sdl_enable_out(HWVoiceOut *hw, bool enable) { - SDL_PauseAudio(!enable); + SDLVoiceOut *sdl = (SDLVoiceOut *)hw; + + SDL_PauseAudioDevice(sdl->devid, !enable); } -static void *sdl_audio_init(Audiodev *dev) +static void sdl_fini_in(HWVoiceIn *hw) { - SDLAudioState *s = &glob_sdl; - if (s->driver_created) { - sdl_logerr("Can't create multiple sdl backends\n"); - return NULL; + 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; } - s->driver_created = true; - s->dev = dev; - return s; + return dev; } static void sdl_audio_fini (void *opaque) { - SDLAudioState *s = opaque; - sdl_close (s); SDL_QuitSubSystem (SDL_INIT_AUDIO); - s->driver_created = false; - s->dev = NULL; } static struct audio_pcm_ops sdl_pcm_ops = { @@ -355,6 +477,15 @@ static struct audio_pcm_ops sdl_pcm_ops = { /* 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 = { @@ -364,10 +495,10 @@ static struct audio_driver sdl_audio_driver = { .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 + .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) |