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https://github.com/Ryujinx/SDL.git
synced 2024-12-24 02:15:36 +00:00
audio: implemented higher level infrastructure for running capture devices.
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6d5c9c1e67
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0d0f7080a3
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@ -206,6 +206,17 @@ SDL_AudioWaitDone_Default(_THIS)
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{ /* no-op. */
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}
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static int
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SDL_AudioCaptureFromDevice_Default(_THIS, void *buffer, int buflen)
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{
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return -1; /* just fail immediately. */
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}
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static void
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SDL_AudioFlushCapture_Default(_THIS)
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{ /* no-op. */
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}
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static void
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SDL_AudioCloseDevice_Default(_THIS)
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{ /* no-op. */
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@ -279,6 +290,8 @@ finalize_audio_entry_points(void)
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FILL_STUB(GetPendingBytes);
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FILL_STUB(GetDeviceBuf);
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FILL_STUB(WaitDone);
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FILL_STUB(CaptureFromDevice);
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FILL_STUB(FlushCapture);
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FILL_STUB(CloseDevice);
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FILL_STUB(LockDevice);
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FILL_STUB(UnlockDevice);
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@ -592,7 +605,7 @@ SDL_ClearQueuedAudio(SDL_AudioDeviceID devid)
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/* The general mixing thread function */
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int SDLCALL
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static int SDLCALL
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SDL_RunAudio(void *devicep)
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{
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SDL_AudioDevice *device = (SDL_AudioDevice *) devicep;
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@ -601,7 +614,9 @@ SDL_RunAudio(void *devicep)
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const int stream_len = (device->convert.needed) ? device->convert.len : device->spec.size;
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Uint8 *stream;
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void *udata = device->spec.userdata;
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void (SDLCALL *fill) (void *, Uint8 *, int) = device->spec.callback;
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void (SDLCALL *callback) (void *, Uint8 *, int) = device->spec.callback;
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SDL_assert(!device->iscapture);
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/* The audio mixing is always a high priority thread */
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SDL_SetThreadPriority(SDL_THREAD_PRIORITY_HIGH);
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@ -635,7 +650,7 @@ SDL_RunAudio(void *devicep)
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if (SDL_AtomicGet(&device->paused)) {
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SDL_memset(stream, silence, stream_len);
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} else {
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(*fill) (udata, stream, stream_len);
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(*callback) (udata, stream, stream_len);
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}
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SDL_UnlockMutex(device->mixer_lock);
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@ -661,11 +676,92 @@ SDL_RunAudio(void *devicep)
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}
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/* Wait for the audio to drain. */
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/* !!! FIXME: can we rename this WaitDrain? */
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current_audio.impl.WaitDone(device);
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return 0;
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}
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/* The general capture thread function */
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static int SDLCALL
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SDL_CaptureAudio(void *devicep)
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{
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SDL_AudioDevice *device = (SDL_AudioDevice *) devicep;
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const int silence = (int) device->spec.silence;
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const Uint32 delay = ((device->spec.samples * 1000) / device->spec.freq);
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const int stream_len = (device->convert.needed) ? device->convert.len : device->spec.size;
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Uint8 *stream;
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void *udata = device->spec.userdata;
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void (SDLCALL *callback) (void *, Uint8 *, int) = device->spec.callback;
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SDL_assert(device->iscapture);
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/* The audio mixing is always a high priority thread */
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SDL_SetThreadPriority(SDL_THREAD_PRIORITY_HIGH);
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/* Perform any thread setup */
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device->threadid = SDL_ThreadID();
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current_audio.impl.ThreadInit(device);
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/* Loop, filling the audio buffers */
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while (!SDL_AtomicGet(&device->shutdown)) {
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int still_need;
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Uint8 *ptr;
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if (!SDL_AtomicGet(&device->enabled) || SDL_AtomicGet(&device->paused)) {
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SDL_Delay(delay); /* just so we don't cook the CPU. */
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current_audio.impl.FlushCapture(device); /* dump anything pending. */
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continue;
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}
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/* Fill the current buffer with sound */
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still_need = stream_len;
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if (device->convert.needed) {
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ptr = stream = device->convert.buf;
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} else {
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/* just use the "fake" stream to hold data read from the device. */
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ptr = stream = device->fake_stream;
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}
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/* We still read from the device when "paused" to keep the state sane,
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and block when there isn't data so this thread isn't eating CPU.
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But we don't process it further or call the app's callback. */
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while (still_need > 0) {
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const int rc = current_audio.impl.CaptureFromDevice(device, ptr, still_need);
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SDL_assert(rc != 0); /* device should have blocked, failed, or returned data. */
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SDL_assert(rc <= still_need); /* device should not overflow buffer. :) */
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if (rc > 0) {
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still_need -= rc;
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ptr += rc;
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} else { /* uhoh, device failed for some reason! */
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SDL_OpenedAudioDeviceDisconnected(device);
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break;
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}
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}
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if (still_need > 0) {
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/* Keep any data we already read, silence the rest. */
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SDL_memset(ptr, silence, still_need);
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}
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if (device->convert.needed) {
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SDL_ConvertAudio(&device->convert);
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}
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/* !!! FIXME: this should be LockDevice. */
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SDL_LockMutex(device->mixer_lock);
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if (!SDL_AtomicGet(&device->paused)) {
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(*callback)(udata, stream, stream_len);
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}
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SDL_UnlockMutex(device->mixer_lock);
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}
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current_audio.impl.FlushCapture(device);
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return 0;
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}
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static SDL_AudioFormat
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SDL_ParseAudioFormat(const char *string)
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@ -1198,10 +1294,11 @@ open_audio_device(const char *devname, int iscapture,
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/* !!! FIXME: we don't force the audio thread stack size here because it calls into user code, but maybe we should? */
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/* buffer queueing callback only needs a few bytes, so make the stack tiny. */
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char name[64];
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const size_t stacksize = (device->spec.callback == SDL_BufferQueueDrainCallback) ? 64 * 1024 : 0;
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const SDL_bool is_internal_thread = (device->spec.callback == SDL_BufferQueueDrainCallback);
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const size_t stacksize = is_internal_thread ? 64 * 1024 : 0;
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SDL_snprintf(name, sizeof (name), "SDLAudioDev%d", (int) device->id);
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device->thread = SDL_CreateThreadInternal(SDL_RunAudio, name, stacksize, device);
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device->thread = SDL_CreateThreadInternal(iscapture ? SDL_CaptureAudio : SDL_RunAudio, name, stacksize, device);
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if (device->thread == NULL) {
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SDL_CloseAudioDevice(device->id);
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@ -29,9 +29,6 @@ extern SDL_AudioFormat SDL_NextAudioFormat(void);
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/* Function to calculate the size and silence for a SDL_AudioSpec */
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extern void SDL_CalculateAudioSpec(SDL_AudioSpec * spec);
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/* The actual mixing thread function */
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extern int SDLCALL SDL_RunAudio(void *audiop);
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/* this is used internally to access some autogenerated code. */
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typedef struct
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{
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@ -76,6 +76,8 @@ typedef struct SDL_AudioDriverImpl
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int (*GetPendingBytes) (_THIS);
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Uint8 *(*GetDeviceBuf) (_THIS);
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void (*WaitDone) (_THIS);
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int (*CaptureFromDevice) (_THIS, void *buffer, int buflen);
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void (*FlushCapture) (_THIS);
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void (*CloseDevice) (_THIS);
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void (*LockDevice) (_THIS);
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void (*UnlockDevice) (_THIS);
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