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authorPeter Maydell2021-01-15 23:21:21 +0100
committerPeter Maydell2021-01-15 23:21:21 +0100
commit825a215c003cd028e26c7d19aa5049d957345f43 (patch)
treea3bedcc1d73490abbe5994b065289147f8d0b10e /audio/sdlaudio.c
parentMerge remote-tracking branch 'remotes/kraxel/tags/ui-20210115-pull-request' i... (diff)
parentaudio: space prohibited between function name and parenthesis'(' (diff)
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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.c305
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)