audio: implemented higher level infrastructure for running capture devices.

This commit is contained in:
Ryan C. Gordon 2016-08-02 13:50:21 -04:00
parent 6d5c9c1e67
commit 0d0f7080a3
3 changed files with 104 additions and 8 deletions

View file

@ -206,6 +206,17 @@ SDL_AudioWaitDone_Default(_THIS)
{ /* no-op. */
}
static int
SDL_AudioCaptureFromDevice_Default(_THIS, void *buffer, int buflen)
{
return -1; /* just fail immediately. */
}
static void
SDL_AudioFlushCapture_Default(_THIS)
{ /* no-op. */
}
static void
SDL_AudioCloseDevice_Default(_THIS)
{ /* no-op. */
@ -279,6 +290,8 @@ finalize_audio_entry_points(void)
FILL_STUB(GetPendingBytes);
FILL_STUB(GetDeviceBuf);
FILL_STUB(WaitDone);
FILL_STUB(CaptureFromDevice);
FILL_STUB(FlushCapture);
FILL_STUB(CloseDevice);
FILL_STUB(LockDevice);
FILL_STUB(UnlockDevice);
@ -592,7 +605,7 @@ SDL_ClearQueuedAudio(SDL_AudioDeviceID devid)
/* The general mixing thread function */
int SDLCALL
static int SDLCALL
SDL_RunAudio(void *devicep)
{
SDL_AudioDevice *device = (SDL_AudioDevice *) devicep;
@ -601,7 +614,9 @@ SDL_RunAudio(void *devicep)
const int stream_len = (device->convert.needed) ? device->convert.len : device->spec.size;
Uint8 *stream;
void *udata = device->spec.userdata;
void (SDLCALL *fill) (void *, Uint8 *, int) = device->spec.callback;
void (SDLCALL *callback) (void *, Uint8 *, int) = device->spec.callback;
SDL_assert(!device->iscapture);
/* The audio mixing is always a high priority thread */
SDL_SetThreadPriority(SDL_THREAD_PRIORITY_HIGH);
@ -635,7 +650,7 @@ SDL_RunAudio(void *devicep)
if (SDL_AtomicGet(&device->paused)) {
SDL_memset(stream, silence, stream_len);
} else {
(*fill) (udata, stream, stream_len);
(*callback) (udata, stream, stream_len);
}
SDL_UnlockMutex(device->mixer_lock);
@ -661,11 +676,92 @@ SDL_RunAudio(void *devicep)
}
/* Wait for the audio to drain. */
/* !!! FIXME: can we rename this WaitDrain? */
current_audio.impl.WaitDone(device);
return 0;
}
/* The general capture thread function */
static int SDLCALL
SDL_CaptureAudio(void *devicep)
{
SDL_AudioDevice *device = (SDL_AudioDevice *) devicep;
const int silence = (int) device->spec.silence;
const Uint32 delay = ((device->spec.samples * 1000) / device->spec.freq);
const int stream_len = (device->convert.needed) ? device->convert.len : device->spec.size;
Uint8 *stream;
void *udata = device->spec.userdata;
void (SDLCALL *callback) (void *, Uint8 *, int) = device->spec.callback;
SDL_assert(device->iscapture);
/* The audio mixing is always a high priority thread */
SDL_SetThreadPriority(SDL_THREAD_PRIORITY_HIGH);
/* Perform any thread setup */
device->threadid = SDL_ThreadID();
current_audio.impl.ThreadInit(device);
/* Loop, filling the audio buffers */
while (!SDL_AtomicGet(&device->shutdown)) {
int still_need;
Uint8 *ptr;
if (!SDL_AtomicGet(&device->enabled) || SDL_AtomicGet(&device->paused)) {
SDL_Delay(delay); /* just so we don't cook the CPU. */
current_audio.impl.FlushCapture(device); /* dump anything pending. */
continue;
}
/* Fill the current buffer with sound */
still_need = stream_len;
if (device->convert.needed) {
ptr = stream = device->convert.buf;
} else {
/* just use the "fake" stream to hold data read from the device. */
ptr = stream = device->fake_stream;
}
/* We still read from the device when "paused" to keep the state sane,
and block when there isn't data so this thread isn't eating CPU.
But we don't process it further or call the app's callback. */
while (still_need > 0) {
const int rc = current_audio.impl.CaptureFromDevice(device, ptr, still_need);
SDL_assert(rc != 0); /* device should have blocked, failed, or returned data. */
SDL_assert(rc <= still_need); /* device should not overflow buffer. :) */
if (rc > 0) {
still_need -= rc;
ptr += rc;
} else { /* uhoh, device failed for some reason! */
SDL_OpenedAudioDeviceDisconnected(device);
break;
}
}
if (still_need > 0) {
/* Keep any data we already read, silence the rest. */
SDL_memset(ptr, silence, still_need);
}
if (device->convert.needed) {
SDL_ConvertAudio(&device->convert);
}
/* !!! FIXME: this should be LockDevice. */
SDL_LockMutex(device->mixer_lock);
if (!SDL_AtomicGet(&device->paused)) {
(*callback)(udata, stream, stream_len);
}
SDL_UnlockMutex(device->mixer_lock);
}
current_audio.impl.FlushCapture(device);
return 0;
}
static SDL_AudioFormat
SDL_ParseAudioFormat(const char *string)
@ -1198,10 +1294,11 @@ open_audio_device(const char *devname, int iscapture,
/* !!! FIXME: we don't force the audio thread stack size here because it calls into user code, but maybe we should? */
/* buffer queueing callback only needs a few bytes, so make the stack tiny. */
char name[64];
const size_t stacksize = (device->spec.callback == SDL_BufferQueueDrainCallback) ? 64 * 1024 : 0;
const SDL_bool is_internal_thread = (device->spec.callback == SDL_BufferQueueDrainCallback);
const size_t stacksize = is_internal_thread ? 64 * 1024 : 0;
SDL_snprintf(name, sizeof (name), "SDLAudioDev%d", (int) device->id);
device->thread = SDL_CreateThreadInternal(SDL_RunAudio, name, stacksize, device);
device->thread = SDL_CreateThreadInternal(iscapture ? SDL_CaptureAudio : SDL_RunAudio, name, stacksize, device);
if (device->thread == NULL) {
SDL_CloseAudioDevice(device->id);

View file

@ -29,9 +29,6 @@ extern SDL_AudioFormat SDL_NextAudioFormat(void);
/* Function to calculate the size and silence for a SDL_AudioSpec */
extern void SDL_CalculateAudioSpec(SDL_AudioSpec * spec);
/* The actual mixing thread function */
extern int SDLCALL SDL_RunAudio(void *audiop);
/* this is used internally to access some autogenerated code. */
typedef struct
{

View file

@ -76,6 +76,8 @@ typedef struct SDL_AudioDriverImpl
int (*GetPendingBytes) (_THIS);
Uint8 *(*GetDeviceBuf) (_THIS);
void (*WaitDone) (_THIS);
int (*CaptureFromDevice) (_THIS, void *buffer, int buflen);
void (*FlushCapture) (_THIS);
void (*CloseDevice) (_THIS);
void (*LockDevice) (_THIS);
void (*UnlockDevice) (_THIS);