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Update SDL to 2.0.7

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This commit is contained in:
Alex Szpakowski
2017-10-28 22:36:09 -03:00
parent 7d27167be2
commit 4a2fd6d703
1400 changed files with 52242 additions and 127752 deletions
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libs/SDL2/src/audio/SDL_audio.c
+300 -340
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@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -33,36 +33,6 @@
static SDL_AudioDriver current_audio;
static SDL_AudioDevice *open_devices[16];
/*
* Not all of these will be compiled and linked in, but it's convenient
* to have a complete list here and saves yet-another block of #ifdefs...
* Please see bootstrap[], below, for the actual #ifdef mess.
*/
extern AudioBootStrap PULSEAUDIO_bootstrap;
extern AudioBootStrap ALSA_bootstrap;
extern AudioBootStrap SNDIO_bootstrap;
extern AudioBootStrap BSD_AUDIO_bootstrap;
extern AudioBootStrap DSP_bootstrap;
extern AudioBootStrap QSAAUDIO_bootstrap;
extern AudioBootStrap SUNAUDIO_bootstrap;
extern AudioBootStrap ARTS_bootstrap;
extern AudioBootStrap ESD_bootstrap;
extern AudioBootStrap NACLAUDIO_bootstrap;
extern AudioBootStrap NAS_bootstrap;
extern AudioBootStrap XAUDIO2_bootstrap;
extern AudioBootStrap DSOUND_bootstrap;
extern AudioBootStrap WINMM_bootstrap;
extern AudioBootStrap PAUDIO_bootstrap;
extern AudioBootStrap HAIKUAUDIO_bootstrap;
extern AudioBootStrap COREAUDIO_bootstrap;
extern AudioBootStrap DISKAUDIO_bootstrap;
extern AudioBootStrap DUMMYAUDIO_bootstrap;
extern AudioBootStrap FUSIONSOUND_bootstrap;
extern AudioBootStrap ANDROIDAUDIO_bootstrap;
extern AudioBootStrap PSPAUDIO_bootstrap;
extern AudioBootStrap SNDIO_bootstrap;
extern AudioBootStrap EMSCRIPTENAUDIO_bootstrap;
/* Available audio drivers */
static const AudioBootStrap *const bootstrap[] = {
#if SDL_AUDIO_DRIVER_PULSEAUDIO
@@ -74,8 +44,8 @@ static const AudioBootStrap *const bootstrap[] = {
#if SDL_AUDIO_DRIVER_SNDIO
&SNDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_BSD
&BSD_AUDIO_bootstrap,
#if SDL_AUDIO_DRIVER_NETBSD
&NETBSDAUDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_OSS
&DSP_bootstrap,
@@ -98,6 +68,9 @@ static const AudioBootStrap *const bootstrap[] = {
#if SDL_AUDIO_DRIVER_NAS
&NAS_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_WASAPI
&WASAPI_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_XAUDIO2
&XAUDIO2_bootstrap,
#endif
@@ -116,12 +89,6 @@ static const AudioBootStrap *const bootstrap[] = {
#if SDL_AUDIO_DRIVER_COREAUDIO
&COREAUDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_DISK
&DISKAUDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_DUMMY
&DUMMYAUDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_FUSIONSOUND
&FUSIONSOUND_bootstrap,
#endif
@@ -133,10 +100,102 @@ static const AudioBootStrap *const bootstrap[] = {
#endif
#if SDL_AUDIO_DRIVER_EMSCRIPTEN
&EMSCRIPTENAUDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_JACK
&JACK_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_DISK
&DISKAUDIO_bootstrap,
#endif
#if SDL_AUDIO_DRIVER_DUMMY
&DUMMYAUDIO_bootstrap,
#endif
NULL
};
#ifdef HAVE_LIBSAMPLERATE_H
#ifdef SDL_LIBSAMPLERATE_DYNAMIC
static void *SRC_lib = NULL;
#endif
SDL_bool SRC_available = SDL_FALSE;
int SRC_converter = 0;
SRC_STATE* (*SRC_src_new)(int converter_type, int channels, int *error) = NULL;
int (*SRC_src_process)(SRC_STATE *state, SRC_DATA *data) = NULL;
int (*SRC_src_reset)(SRC_STATE *state) = NULL;
SRC_STATE* (*SRC_src_delete)(SRC_STATE *state) = NULL;
const char* (*SRC_src_strerror)(int error) = NULL;
static SDL_bool
LoadLibSampleRate(void)
{
const char *hint = SDL_GetHint(SDL_HINT_AUDIO_RESAMPLING_MODE);
SRC_available = SDL_FALSE;
SRC_converter = 0;
if (!hint || *hint == '0' || SDL_strcasecmp(hint, "default") == 0) {
return SDL_FALSE; /* don't load anything. */
} else if (*hint == '1' || SDL_strcasecmp(hint, "fast") == 0) {
SRC_converter = SRC_SINC_FASTEST;
} else if (*hint == '2' || SDL_strcasecmp(hint, "medium") == 0) {
SRC_converter = SRC_SINC_MEDIUM_QUALITY;
} else if (*hint == '3' || SDL_strcasecmp(hint, "best") == 0) {
SRC_converter = SRC_SINC_BEST_QUALITY;
} else {
return SDL_FALSE; /* treat it like "default", don't load anything. */
}
#ifdef SDL_LIBSAMPLERATE_DYNAMIC
SDL_assert(SRC_lib == NULL);
SRC_lib = SDL_LoadObject(SDL_LIBSAMPLERATE_DYNAMIC);
if (!SRC_lib) {
SDL_ClearError();
return SDL_FALSE;
}
SRC_src_new = (SRC_STATE* (*)(int converter_type, int channels, int *error))SDL_LoadFunction(SRC_lib, "src_new");
SRC_src_process = (int (*)(SRC_STATE *state, SRC_DATA *data))SDL_LoadFunction(SRC_lib, "src_process");
SRC_src_reset = (int(*)(SRC_STATE *state))SDL_LoadFunction(SRC_lib, "src_reset");
SRC_src_delete = (SRC_STATE* (*)(SRC_STATE *state))SDL_LoadFunction(SRC_lib, "src_delete");
SRC_src_strerror = (const char* (*)(int error))SDL_LoadFunction(SRC_lib, "src_strerror");
if (!SRC_src_new || !SRC_src_process || !SRC_src_reset || !SRC_src_delete || !SRC_src_strerror) {
SDL_UnloadObject(SRC_lib);
SRC_lib = NULL;
return SDL_FALSE;
}
#else
SRC_src_new = src_new;
SRC_src_process = src_process;
SRC_src_reset = src_reset;
SRC_src_delete = src_delete;
SRC_src_strerror = src_strerror;
#endif
SRC_available = SDL_TRUE;
return SDL_TRUE;
}
static void
UnloadLibSampleRate(void)
{
#ifdef SDL_LIBSAMPLERATE_DYNAMIC
if (SRC_lib != NULL) {
SDL_UnloadObject(SRC_lib);
}
SRC_lib = NULL;
#endif
SRC_available = SDL_FALSE;
SRC_src_new = NULL;
SRC_src_process = NULL;
SRC_src_reset = NULL;
SRC_src_delete = NULL;
SRC_src_strerror = NULL;
}
#endif
static SDL_AudioDevice *
get_audio_device(SDL_AudioDeviceID id)
{
@@ -169,6 +228,11 @@ SDL_AudioThreadInit_Default(_THIS)
{ /* no-op. */
}
static void
SDL_AudioThreadDeinit_Default(_THIS)
{ /* no-op. */
}
static void
SDL_AudioWaitDevice_Default(_THIS)
{ /* no-op. */
@@ -288,6 +352,7 @@ finish_audio_entry_points_init(void)
FILL_STUB(DetectDevices);
FILL_STUB(OpenDevice);
FILL_STUB(ThreadInit);
FILL_STUB(ThreadDeinit);
FILL_STUB(WaitDevice);
FILL_STUB(PlayDevice);
FILL_STUB(GetPendingBytes);
@@ -448,136 +513,23 @@ SDL_RemoveAudioDevice(const int iscapture, void *handle)
/* buffer queueing support... */
/* this expects that you managed thread safety elsewhere. */
static void
free_audio_queue(SDL_AudioBufferQueue *packet)
{
while (packet) {
SDL_AudioBufferQueue *next = packet->next;
SDL_free(packet);
packet = next;
}
}
/* NOTE: This assumes you'll hold the mixer lock before calling! */
static int
queue_audio_to_device(SDL_AudioDevice *device, const Uint8 *data, Uint32 len)
{
SDL_AudioBufferQueue *orighead;
SDL_AudioBufferQueue *origtail;
Uint32 origlen;
Uint32 datalen;
orighead = device->buffer_queue_head;
origtail = device->buffer_queue_tail;
origlen = origtail ? origtail->datalen : 0;
while (len > 0) {
SDL_AudioBufferQueue *packet = device->buffer_queue_tail;
SDL_assert(!packet || (packet->datalen <= SDL_AUDIOBUFFERQUEUE_PACKETLEN));
if (!packet || (packet->datalen >= SDL_AUDIOBUFFERQUEUE_PACKETLEN)) {
/* tail packet missing or completely full; we need a new packet. */
packet = device->buffer_queue_pool;
if (packet != NULL) {
/* we have one available in the pool. */
device->buffer_queue_pool = packet->next;
} else {
/* Have to allocate a new one! */
packet = (SDL_AudioBufferQueue *) SDL_malloc(sizeof (SDL_AudioBufferQueue));
if (packet == NULL) {
/* uhoh, reset so we've queued nothing new, free what we can. */
if (!origtail) {
packet = device->buffer_queue_head; /* whole queue. */
} else {
packet = origtail->next; /* what we added to existing queue. */
origtail->next = NULL;
origtail->datalen = origlen;
}
device->buffer_queue_head = orighead;
device->buffer_queue_tail = origtail;
device->buffer_queue_pool = NULL;
free_audio_queue(packet); /* give back what we can. */
return SDL_OutOfMemory();
}
}
packet->datalen = 0;
packet->startpos = 0;
packet->next = NULL;
SDL_assert((device->buffer_queue_head != NULL) == (device->queued_bytes != 0));
if (device->buffer_queue_tail == NULL) {
device->buffer_queue_head = packet;
} else {
device->buffer_queue_tail->next = packet;
}
device->buffer_queue_tail = packet;
}
datalen = SDL_min(len, SDL_AUDIOBUFFERQUEUE_PACKETLEN - packet->datalen);
SDL_memcpy(packet->data + packet->datalen, data, datalen);
data += datalen;
len -= datalen;
packet->datalen += datalen;
device->queued_bytes += datalen;
}
return 0;
}
/* NOTE: This assumes you'll hold the mixer lock before calling! */
static Uint32
dequeue_audio_from_device(SDL_AudioDevice *device, Uint8 *stream, Uint32 len)
{
SDL_AudioBufferQueue *packet;
Uint8 *ptr = stream;
while ((len > 0) && ((packet = device->buffer_queue_head) != NULL)) {
const Uint32 avail = packet->datalen - packet->startpos;
const Uint32 cpy = SDL_min(len, avail);
SDL_assert(device->queued_bytes >= avail);
SDL_memcpy(ptr, packet->data + packet->startpos, cpy);
packet->startpos += cpy;
ptr += cpy;
device->queued_bytes -= cpy;
len -= cpy;
if (packet->startpos == packet->datalen) { /* packet is done, put it in the pool. */
device->buffer_queue_head = packet->next;
SDL_assert((packet->next != NULL) || (packet == device->buffer_queue_tail));
packet->next = device->buffer_queue_pool;
device->buffer_queue_pool = packet;
}
}
SDL_assert((device->buffer_queue_head != NULL) == (device->queued_bytes != 0));
if (device->buffer_queue_head == NULL) {
device->buffer_queue_tail = NULL; /* in case we drained the queue entirely. */
}
return (Uint32) (ptr - stream);
}
static void SDLCALL
SDL_BufferQueueDrainCallback(void *userdata, Uint8 *stream, int len)
{
/* this function always holds the mixer lock before being called. */
SDL_AudioDevice *device = (SDL_AudioDevice *) userdata;
Uint32 written;
size_t dequeued;
SDL_assert(device != NULL); /* this shouldn't ever happen, right?! */
SDL_assert(!device->iscapture); /* this shouldn't ever happen, right?! */
SDL_assert(len >= 0); /* this shouldn't ever happen, right?! */
written = dequeue_audio_from_device(device, stream, (Uint32) len);
stream += written;
len -= (int) written;
dequeued = SDL_ReadFromDataQueue(device->buffer_queue, stream, len);
stream += dequeued;
len -= (int) dequeued;
if (len > 0) { /* fill any remaining space in the stream with silence. */
SDL_assert(device->buffer_queue_head == NULL);
SDL_assert(SDL_CountDataQueue(device->buffer_queue) == 0);
SDL_memset(stream, device->spec.silence, len);
}
}
@@ -595,7 +547,7 @@ SDL_BufferQueueFillCallback(void *userdata, Uint8 *stream, int len)
/* note that if this needs to allocate more space and run out of memory,
we have no choice but to quietly drop the data and hope it works out
later, but you probably have bigger problems in this case anyhow. */
queue_audio_to_device(device, stream, (Uint32) len);
SDL_WriteToDataQueue(device->buffer_queue, stream, len);
}
int
@@ -608,13 +560,13 @@ SDL_QueueAudio(SDL_AudioDeviceID devid, const void *data, Uint32 len)
return -1; /* get_audio_device() will have set the error state */
} else if (device->iscapture) {
return SDL_SetError("This is a capture device, queueing not allowed");
} else if (device->spec.callback != SDL_BufferQueueDrainCallback) {
} else if (device->callbackspec.callback != SDL_BufferQueueDrainCallback) {
return SDL_SetError("Audio device has a callback, queueing not allowed");
}
if (len > 0) {
current_audio.impl.LockDevice(device);
rc = queue_audio_to_device(device, data, len);
rc = SDL_WriteToDataQueue(device->buffer_queue, data, len);
current_audio.impl.UnlockDevice(device);
}
@@ -630,12 +582,12 @@ SDL_DequeueAudio(SDL_AudioDeviceID devid, void *data, Uint32 len)
if ( (len == 0) || /* nothing to do? */
(!device) || /* called with bogus device id */
(!device->iscapture) || /* playback devices can't dequeue */
(device->spec.callback != SDL_BufferQueueFillCallback) ) { /* not set for queueing */
(device->callbackspec.callback != SDL_BufferQueueFillCallback) ) { /* not set for queueing */
return 0; /* just report zero bytes dequeued. */
}
current_audio.impl.LockDevice(device);
rc = dequeue_audio_from_device(device, data, len);
rc = (Uint32) SDL_ReadFromDataQueue(device->buffer_queue, data, len);
current_audio.impl.UnlockDevice(device);
return rc;
}
@@ -651,13 +603,13 @@ SDL_GetQueuedAudioSize(SDL_AudioDeviceID devid)
}
/* Nothing to do unless we're set up for queueing. */
if (device->spec.callback == SDL_BufferQueueDrainCallback) {
if (device->callbackspec.callback == SDL_BufferQueueDrainCallback) {
current_audio.impl.LockDevice(device);
retval = device->queued_bytes + current_audio.impl.GetPendingBytes(device);
retval = ((Uint32) SDL_CountDataQueue(device->buffer_queue)) + current_audio.impl.GetPendingBytes(device);
current_audio.impl.UnlockDevice(device);
} else if (device->spec.callback == SDL_BufferQueueFillCallback) {
} else if (device->callbackspec.callback == SDL_BufferQueueFillCallback) {
current_audio.impl.LockDevice(device);
retval = device->queued_bytes;
retval = (Uint32) SDL_CountDataQueue(device->buffer_queue);
current_audio.impl.UnlockDevice(device);
}
@@ -668,7 +620,6 @@ void
SDL_ClearQueuedAudio(SDL_AudioDeviceID devid)
{
SDL_AudioDevice *device = get_audio_device(devid);
SDL_AudioBufferQueue *packet;
if (!device) {
return; /* nothing to do. */
@@ -677,35 +628,10 @@ SDL_ClearQueuedAudio(SDL_AudioDeviceID devid)
/* Blank out the device and release the mutex. Free it afterwards. */
current_audio.impl.LockDevice(device);
/* merge the available pool and the current queue into one list. */
packet = device->buffer_queue_head;
if (packet) {
device->buffer_queue_tail->next = device->buffer_queue_pool;
} else {
packet = device->buffer_queue_pool;
}
/* Remove the queued packets from the device. */
device->buffer_queue_tail = NULL;
device->buffer_queue_head = NULL;
device->queued_bytes = 0;
device->buffer_queue_pool = packet;
/* Keep up to two packets in the pool to reduce future malloc pressure. */
if (packet) {
if (!packet->next) {
packet = NULL; /* one packet (the only one) for the pool. */
} else {
SDL_AudioBufferQueue *next = packet->next->next;
packet->next->next = NULL; /* two packets for the pool. */
packet = next; /* rest will be freed. */
}
}
SDL_ClearDataQueue(device->buffer_queue, SDL_AUDIOBUFFERQUEUE_PACKETLEN * 2);
current_audio.impl.UnlockDevice(device);
/* free any extra packets we didn't keep in the pool. */
free_audio_queue(packet);
}
@@ -714,12 +640,9 @@ static int SDLCALL
SDL_RunAudio(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;
void *udata = device->callbackspec.userdata;
SDL_AudioCallback callback = device->callbackspec.callback;
Uint8 *data;
SDL_assert(!device->iscapture);
@@ -732,51 +655,63 @@ SDL_RunAudio(void *devicep)
/* Loop, filling the audio buffers */
while (!SDL_AtomicGet(&device->shutdown)) {
const int data_len = device->callbackspec.size;
/* Fill the current buffer with sound */
if (device->convert.needed) {
stream = device->convert.buf;
} else if (SDL_AtomicGet(&device->enabled)) {
stream = current_audio.impl.GetDeviceBuf(device);
if (!device->stream && SDL_AtomicGet(&device->enabled)) {
SDL_assert(data_len == device->spec.size);
data = current_audio.impl.GetDeviceBuf(device);
} else {
/* if the device isn't enabled, we still write to the
fake_stream, so the app's callback will fire with
work_buffer, so the app's callback will fire with
a regular frequency, in case they depend on that
for timing or progress. They can use hotplug
now to know if the device failed. */
stream = NULL;
now to know if the device failed.
Streaming playback uses work_buffer, too. */
data = NULL;
}
if (stream == NULL) {
stream = device->fake_stream;
if (data == NULL) {
data = device->work_buffer;
}
/* !!! FIXME: this should be LockDevice. */
if ( SDL_AtomicGet(&device->enabled) ) {
SDL_LockMutex(device->mixer_lock);
if (SDL_AtomicGet(&device->paused)) {
SDL_memset(stream, silence, stream_len);
} else {
(*callback) (udata, stream, stream_len);
}
SDL_UnlockMutex(device->mixer_lock);
}
/* Convert the audio if necessary */
if (device->convert.needed && SDL_AtomicGet(&device->enabled)) {
SDL_ConvertAudio(&device->convert);
stream = current_audio.impl.GetDeviceBuf(device);
if (stream == NULL) {
stream = device->fake_stream;
} else {
SDL_memcpy(stream, device->convert.buf,
device->convert.len_cvt);
}
}
/* Ready current buffer for play and change current buffer */
if (stream == device->fake_stream) {
SDL_Delay(delay);
SDL_LockMutex(device->mixer_lock);
if (SDL_AtomicGet(&device->paused)) {
SDL_memset(data, device->spec.silence, data_len);
} else {
callback(udata, data, data_len);
}
SDL_UnlockMutex(device->mixer_lock);
if (device->stream) {
/* Stream available audio to device, converting/resampling. */
/* if this fails...oh well. We'll play silence here. */
SDL_AudioStreamPut(device->stream, data, data_len);
while (SDL_AudioStreamAvailable(device->stream) >= ((int) device->spec.size)) {
int got;
data = SDL_AtomicGet(&device->enabled) ? current_audio.impl.GetDeviceBuf(device) : NULL;
got = SDL_AudioStreamGet(device->stream, data ? data : device->work_buffer, device->spec.size);
SDL_assert((got < 0) || (got == device->spec.size));
if (data == NULL) { /* device is having issues... */
const Uint32 delay = ((device->spec.samples * 1000) / device->spec.freq);
SDL_Delay(delay); /* wait for as long as this buffer would have played. Maybe device recovers later? */
} else {
if (got != device->spec.size) {
SDL_memset(data, device->spec.silence, device->spec.size);
}
current_audio.impl.PlayDevice(device);
current_audio.impl.WaitDevice(device);
}
}
} else if (data == device->work_buffer) {
/* nothing to do; pause like we queued a buffer to play. */
const Uint32 delay = ((device->spec.samples * 1000) / device->spec.freq);
SDL_Delay(delay);
} else { /* writing directly to the device. */
/* queue this buffer and wait for it to finish playing. */
current_audio.impl.PlayDevice(device);
current_audio.impl.WaitDevice(device);
}
@@ -787,9 +722,12 @@ SDL_RunAudio(void *devicep)
/* Wait for the audio to drain. */
SDL_Delay(((device->spec.samples * 1000) / device->spec.freq) * 2);
current_audio.impl.ThreadDeinit(device);
return 0;
}
/* !!! FIXME: this needs to deal with device spec changes. */
/* The general capture thread function */
static int SDLCALL
SDL_CaptureAudio(void *devicep)
@@ -797,10 +735,10 @@ 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;
const int data_len = device->spec.size;
Uint8 *data;
void *udata = device->callbackspec.userdata;
SDL_AudioCallback callback = device->callbackspec.callback;
SDL_assert(device->iscapture);
@@ -816,34 +754,41 @@ SDL_CaptureAudio(void *devicep)
int still_need;
Uint8 *ptr;
if (!SDL_AtomicGet(&device->enabled) || SDL_AtomicGet(&device->paused)) {
if (SDL_AtomicGet(&device->paused)) {
SDL_Delay(delay); /* just so we don't cook the CPU. */
if (device->stream) {
SDL_AudioStreamClear(device->stream);
}
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;
}
still_need = data_len;
/* Use the work_buffer to hold data read from the device. */
data = device->work_buffer;
SDL_assert(data != NULL);
ptr = data;
/* 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 <= 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 (!SDL_AtomicGet(&device->enabled)) {
SDL_Delay(delay); /* try to keep callback firing at normal pace. */
} else {
while (still_need > 0) {
const int rc = current_audio.impl.CaptureFromDevice(device, ptr, still_need);
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;
}
}
}
@@ -852,22 +797,38 @@ SDL_CaptureAudio(void *devicep)
SDL_memset(ptr, silence, still_need);
}
if (device->convert.needed) {
SDL_ConvertAudio(&device->convert);
}
if (device->stream) {
/* if this fails...oh well. */
SDL_AudioStreamPut(device->stream, data, data_len);
/* !!! FIXME: this should be LockDevice. */
SDL_LockMutex(device->mixer_lock);
if (SDL_AtomicGet(&device->paused)) {
current_audio.impl.FlushCapture(device); /* one snuck in! */
} else {
(*callback)(udata, stream, stream_len);
while (SDL_AudioStreamAvailable(device->stream) >= ((int) device->callbackspec.size)) {
const int got = SDL_AudioStreamGet(device->stream, device->work_buffer, device->callbackspec.size);
SDL_assert((got < 0) || (got == device->callbackspec.size));
if (got != device->callbackspec.size) {
SDL_memset(device->work_buffer, device->spec.silence, device->callbackspec.size);
}
/* !!! FIXME: this should be LockDevice. */
SDL_LockMutex(device->mixer_lock);
if (!SDL_AtomicGet(&device->paused)) {
callback(udata, device->work_buffer, device->callbackspec.size);
}
SDL_UnlockMutex(device->mixer_lock);
}
} else { /* feeding user callback directly without streaming. */
/* !!! FIXME: this should be LockDevice. */
SDL_LockMutex(device->mixer_lock);
if (!SDL_AtomicGet(&device->paused)) {
callback(udata, data, device->callbackspec.size);
}
SDL_UnlockMutex(device->mixer_lock);
}
SDL_UnlockMutex(device->mixer_lock);
}
current_audio.impl.FlushCapture(device);
current_audio.impl.ThreadDeinit(device);
return 0;
}
@@ -968,6 +929,10 @@ SDL_AudioInit(const char *driver_name)
/* Make sure we have a list of devices available at startup. */
current_audio.impl.DetectDevices();
#ifdef HAVE_LIBSAMPLERATE_H
LoadLibSampleRate();
#endif
return 0;
}
@@ -1098,16 +1063,15 @@ close_audio_device(SDL_AudioDevice * device)
if (device->mixer_lock != NULL) {
SDL_DestroyMutex(device->mixer_lock);
}
SDL_free(device->fake_stream);
if (device->convert.needed) {
SDL_free(device->convert.buf);
}
SDL_free(device->work_buffer);
SDL_FreeAudioStream(device->stream);
if (device->hidden != NULL) {
current_audio.impl.CloseDevice(device);
}
free_audio_queue(device->buffer_queue_head);
free_audio_queue(device->buffer_queue_pool);
SDL_FreeDataQueue(device->buffer_queue);
SDL_free(device);
}
@@ -1180,11 +1144,11 @@ open_audio_device(const char *devname, int iscapture,
const SDL_AudioSpec * desired, SDL_AudioSpec * obtained,
int allowed_changes, int min_id)
{
const SDL_bool is_internal_thread = (desired->callback != NULL);
const SDL_bool is_internal_thread = (desired->callback == NULL);
SDL_AudioDeviceID id = 0;
SDL_AudioSpec _obtained;
SDL_AudioDevice *device;
SDL_bool build_cvt;
SDL_bool build_stream;
void *handle = NULL;
int i = 0;
@@ -1319,84 +1283,87 @@ open_audio_device(const char *devname, int iscapture,
SDL_assert(device->hidden != NULL);
/* See if we need to do any conversion */
build_cvt = SDL_FALSE;
build_stream = SDL_FALSE;
if (obtained->freq != device->spec.freq) {
if (allowed_changes & SDL_AUDIO_ALLOW_FREQUENCY_CHANGE) {
obtained->freq = device->spec.freq;
} else {
build_cvt = SDL_TRUE;
build_stream = SDL_TRUE;
}
}
if (obtained->format != device->spec.format) {
if (allowed_changes & SDL_AUDIO_ALLOW_FORMAT_CHANGE) {
obtained->format = device->spec.format;
} else {
build_cvt = SDL_TRUE;
build_stream = SDL_TRUE;
}
}
if (obtained->channels != device->spec.channels) {
if (allowed_changes & SDL_AUDIO_ALLOW_CHANNELS_CHANGE) {
obtained->channels = device->spec.channels;
} else {
build_cvt = SDL_TRUE;
build_stream = SDL_TRUE;
}
}
/* If the audio driver changes the buffer size, accept it.
This needs to be done after the format is modified above,
otherwise it might not have the correct buffer size.
/* !!! FIXME in 2.1: add SDL_AUDIO_ALLOW_SAMPLES_CHANGE flag?
As of 2.0.6, we will build a stream to buffer the difference between
what the app wants to feed and the device wants to eat, so everyone
gets their way. In prior releases, SDL would force the callback to
feed at the rate the device requested, adjusted for resampling.
*/
if (device->spec.samples != obtained->samples) {
obtained->samples = device->spec.samples;
SDL_CalculateAudioSpec(obtained);
build_stream = SDL_TRUE;
}
if (build_cvt) {
/* Build an audio conversion block */
if (SDL_BuildAudioCVT(&device->convert,
obtained->format, obtained->channels,
obtained->freq,
device->spec.format, device->spec.channels,
device->spec.freq) < 0) {
SDL_CalculateAudioSpec(obtained); /* recalc after possible changes. */
device->callbackspec = *obtained;
if (build_stream) {
if (iscapture) {
device->stream = SDL_NewAudioStream(device->spec.format,
device->spec.channels, device->spec.freq,
obtained->format, obtained->channels, obtained->freq);
} else {
device->stream = SDL_NewAudioStream(obtained->format, obtained->channels,
obtained->freq, device->spec.format,
device->spec.channels, device->spec.freq);
}
if (!device->stream) {
close_audio_device(device);
return 0;
}
if (device->convert.needed) {
device->convert.len = (int) (((double) device->spec.size) /
device->convert.len_ratio);
device->convert.buf =
(Uint8 *) SDL_malloc(device->convert.len *
device->convert.len_mult);
if (device->convert.buf == NULL) {
close_audio_device(device);
SDL_OutOfMemory();
return 0;
}
}
}
if (device->spec.callback == NULL) { /* use buffer queueing? */
/* pool a few packets to start. Enough for two callbacks. */
const int packetlen = SDL_AUDIOBUFFERQUEUE_PACKETLEN;
const int wantbytes = ((device->convert.needed) ? device->convert.len : device->spec.size) * 2;
const int wantpackets = (wantbytes / packetlen) + ((wantbytes % packetlen) ? packetlen : 0);
for (i = 0; i < wantpackets; i++) {
SDL_AudioBufferQueue *packet = (SDL_AudioBufferQueue *) SDL_malloc(sizeof (SDL_AudioBufferQueue));
if (packet) { /* don't care if this fails, we'll deal later. */
packet->datalen = 0;
packet->startpos = 0;
packet->next = device->buffer_queue_pool;
device->buffer_queue_pool = packet;
}
device->buffer_queue = SDL_NewDataQueue(SDL_AUDIOBUFFERQUEUE_PACKETLEN, obtained->size * 2);
if (!device->buffer_queue) {
close_audio_device(device);
SDL_SetError("Couldn't create audio buffer queue");
return 0;
}
device->spec.callback = iscapture ? SDL_BufferQueueFillCallback : SDL_BufferQueueDrainCallback;
device->spec.userdata = device;
device->callbackspec.callback = iscapture ? SDL_BufferQueueFillCallback : SDL_BufferQueueDrainCallback;
device->callbackspec.userdata = device;
}
/* add it to our list of open devices. */
open_devices[id] = device;
/* Allocate a scratch audio buffer */
device->work_buffer_len = build_stream ? device->callbackspec.size : 0;
if (device->spec.size > device->work_buffer_len) {
device->work_buffer_len = device->spec.size;
}
SDL_assert(device->work_buffer_len > 0);
device->work_buffer = (Uint8 *) SDL_malloc(device->work_buffer_len);
if (device->work_buffer == NULL) {
close_audio_device(device);
SDL_OutOfMemory();
return 0;
}
open_devices[id] = device; /* add it to our list of open devices. */
/* Start the audio thread if necessary */
if (!current_audio.impl.ProvidesOwnCallbackThread) {
@@ -1406,21 +1373,7 @@ open_audio_device(const char *devname, int iscapture,
const size_t stacksize = is_internal_thread ? 64 * 1024 : 0;
char threadname[64];
/* Allocate a fake audio buffer; only used by our internal threads. */
Uint32 stream_len = (device->convert.needed) ? device->convert.len_cvt : 0;
if (device->spec.size > stream_len) {
stream_len = device->spec.size;
}
SDL_assert(stream_len > 0);
device->fake_stream = (Uint8 *) SDL_malloc(stream_len);
if (device->fake_stream == NULL) {
close_audio_device(device);
SDL_OutOfMemory();
return 0;
}
SDL_snprintf(threadname, sizeof (threadname), "SDLAudioDev%d", (int) device->id);
SDL_snprintf(threadname, sizeof (threadname), "SDLAudio%c%d", (iscapture) ? 'C' : 'P', (int) device->id);
device->thread = SDL_CreateThreadInternal(iscapture ? SDL_CaptureAudio : SDL_RunAudio, threadname, stacksize, device);
if (device->thread == NULL) {
@@ -1456,7 +1409,14 @@ SDL_OpenAudio(SDL_AudioSpec * desired, SDL_AudioSpec * obtained)
id = open_audio_device(NULL, 0, desired, obtained,
SDL_AUDIO_ALLOW_ANY_CHANGE, 1);
} else {
id = open_audio_device(NULL, 0, desired, NULL, 0, 1);
SDL_AudioSpec _obtained;
SDL_zero(_obtained);
id = open_audio_device(NULL, 0, desired, &_obtained, 0, 1);
/* On successful open, copy calculated values into 'desired'. */
if (id > 0) {
desired->size = _obtained.size;
desired->silence = _obtained.silence;
}
}
SDL_assert((id == 0) || (id == 1));
@@ -1579,6 +1539,12 @@ SDL_AudioQuit(void)
SDL_zero(current_audio);
SDL_zero(open_devices);
#ifdef HAVE_LIBSAMPLERATE_H
UnloadLibSampleRate();
#endif
SDL_FreeResampleFilter();
}
#define NUM_FORMATS 10
@@ -1655,13 +1621,7 @@ SDL_MixAudio(Uint8 * dst, const Uint8 * src, Uint32 len, int volume)
/* Mix the user-level audio format */
SDL_AudioDevice *device = get_audio_device(1);
if (device != NULL) {
SDL_AudioFormat format;
if (device->convert.needed) {
format = device->convert.src_format;
} else {
format = device->spec.format;
}
SDL_MixAudioFormat(dst, src, format, len, volume);
SDL_MixAudioFormat(dst, src, device->callbackspec.format, len, volume);
}
}
libs/SDL2/src/audio/SDL_audio_c.h
+46 -19
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -18,10 +18,35 @@
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#ifndef SDL_audio_c_h_
#define SDL_audio_c_h_
#include "../SDL_internal.h"
#ifndef DEBUG_CONVERT
#define DEBUG_CONVERT 0
#endif
#if DEBUG_CONVERT
#define LOG_DEBUG_CONVERT(from, to) fprintf(stderr, "Converting %s to %s.\n", from, to);
#else
#define LOG_DEBUG_CONVERT(from, to)
#endif
/* Functions and variables exported from SDL_audio.c for SDL_sysaudio.c */
#ifdef HAVE_LIBSAMPLERATE_H
#include "samplerate.h"
extern SDL_bool SRC_available;
extern int SRC_converter;
extern SRC_STATE* (*SRC_src_new)(int converter_type, int channels, int *error);
extern int (*SRC_src_process)(SRC_STATE *state, SRC_DATA *data);
extern int (*SRC_src_reset)(SRC_STATE *state);
extern SRC_STATE* (*SRC_src_delete)(SRC_STATE *state);
extern const char* (*SRC_src_strerror)(int error);
#endif
/* Functions to get a list of "close" audio formats */
extern SDL_AudioFormat SDL_FirstAudioFormat(SDL_AudioFormat format);
extern SDL_AudioFormat SDL_NextAudioFormat(void);
@@ -29,24 +54,26 @@ extern SDL_AudioFormat SDL_NextAudioFormat(void);
/* Function to calculate the size and silence for a SDL_AudioSpec */
extern void SDL_CalculateAudioSpec(SDL_AudioSpec * spec);
/* this is used internally to access some autogenerated code. */
typedef struct
{
SDL_AudioFormat src_fmt;
SDL_AudioFormat dst_fmt;
SDL_AudioFilter filter;
} SDL_AudioTypeFilters;
extern const SDL_AudioTypeFilters sdl_audio_type_filters[];
/* Choose the audio filter functions below */
extern void SDL_ChooseAudioConverters(void);
/* this is used internally to access some autogenerated code. */
typedef struct
{
SDL_AudioFormat fmt;
int channels;
int upsample;
int multiple;
SDL_AudioFilter filter;
} SDL_AudioRateFilters;
extern const SDL_AudioRateFilters sdl_audio_rate_filters[];
/* These pointers get set during SDL_ChooseAudioConverters() to various SIMD implementations. */
extern SDL_AudioFilter SDL_Convert_S8_to_F32;
extern SDL_AudioFilter SDL_Convert_U8_to_F32;
extern SDL_AudioFilter SDL_Convert_S16_to_F32;
extern SDL_AudioFilter SDL_Convert_U16_to_F32;
extern SDL_AudioFilter SDL_Convert_S32_to_F32;
extern SDL_AudioFilter SDL_Convert_F32_to_S8;
extern SDL_AudioFilter SDL_Convert_F32_to_U8;
extern SDL_AudioFilter SDL_Convert_F32_to_S16;
extern SDL_AudioFilter SDL_Convert_F32_to_U16;
extern SDL_AudioFilter SDL_Convert_F32_to_S32;
/* You need to call SDL_PrepareResampleFilter() before using the internal resampler.
SDL_AudioQuit() calls SDL_FreeResamplerFilter(), you should never call it yourself. */
extern int SDL_PrepareResampleFilter(void);
extern void SDL_FreeResampleFilter(void);
#endif /* SDL_audio_c_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/SDL_audiocvt.c
+1430 -878
View File
File diff suppressed because it is too large Load Diff
libs/SDL2/src/audio/SDL_audiodev.c
+4 -3
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -22,7 +22,7 @@
/* Get the name of the audio device we use for output */
#if SDL_AUDIO_DRIVER_BSD || SDL_AUDIO_DRIVER_OSS || SDL_AUDIO_DRIVER_SUNAUDIO
#if SDL_AUDIO_DRIVER_NETBSD || SDL_AUDIO_DRIVER_OSS || SDL_AUDIO_DRIVER_SUNAUDIO
#include <fcntl.h>
#include <sys/types.h>
@@ -103,9 +103,10 @@ SDL_EnumUnixAudioDevices_Internal(const int iscapture, const int classic, int (*
if (SDL_strlen(audiodev) < (sizeof(audiopath) - 3)) {
int instance = 0;
while (instance++ <= 64) {
while (instance <= 64) {
SDL_snprintf(audiopath, SDL_arraysize(audiopath),
"%s%d", audiodev, instance);
instance++;
test_device(iscapture, audiopath, flags, test);
}
}
libs/SDL2/src/audio/SDL_audiodev_c.h
+1 -1
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
libs/SDL2/src/audio/SDL_audiotypecvt.c
+610 -15806
View File
File diff suppressed because it is too large Load Diff
libs/SDL2/src/audio/SDL_mixer.c
+1 -1
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
libs/SDL2/src/audio/SDL_sysaudio.h
+46 -40
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -20,11 +20,13 @@
*/
#include "../SDL_internal.h"
#ifndef _SDL_sysaudio_h
#define _SDL_sysaudio_h
#ifndef SDL_sysaudio_h_
#define SDL_sysaudio_h_
#include "SDL_mutex.h"
#include "SDL_thread.h"
#include "../SDL_dataqueue.h"
#include "./SDL_audio_c.h"
/* !!! FIXME: These are wordy and unlocalized... */
#define DEFAULT_OUTPUT_DEVNAME "System audio output device"
@@ -49,7 +51,6 @@ extern void SDL_RemoveAudioDevice(const int iscapture, void *handle);
as appropriate so SDL's list of devices is accurate. */
extern void SDL_OpenedAudioDeviceDisconnected(SDL_AudioDevice *device);
/* This is the size of a packet when using SDL_QueueAudio(). We allocate
these as necessary and pool them, under the assumption that we'll
eventually end up with a handful that keep recycling, meeting whatever
@@ -61,20 +62,12 @@ extern void SDL_OpenedAudioDeviceDisconnected(SDL_AudioDevice *device);
The system preallocates enough packets for 2 callbacks' worth of data. */
#define SDL_AUDIOBUFFERQUEUE_PACKETLEN (8 * 1024)
/* Used by apps that queue audio instead of using the callback. */
typedef struct SDL_AudioBufferQueue
{
Uint8 data[SDL_AUDIOBUFFERQUEUE_PACKETLEN]; /* packet data. */
Uint32 datalen; /* bytes currently in use in this packet. */
Uint32 startpos; /* bytes currently consumed in this packet. */
struct SDL_AudioBufferQueue *next; /* next item in linked list. */
} SDL_AudioBufferQueue;
typedef struct SDL_AudioDriverImpl
{
void (*DetectDevices) (void);
int (*OpenDevice) (_THIS, void *handle, const char *devname, int iscapture);
void (*ThreadInit) (_THIS); /* Called by audio thread at start */
void (*ThreadDeinit) (_THIS); /* Called by audio thread at end */
void (*WaitDevice) (_THIS);
void (*PlayDevice) (_THIS);
int (*GetPendingBytes) (_THIS);
@@ -105,11 +98,7 @@ typedef struct SDL_AudioDeviceItem
{
void *handle;
struct SDL_AudioDeviceItem *next;
#if (defined(__GNUC__) && (__GNUC__ <= 2))
char name[1]; /* actually variable length. */
#else
char name[];
#endif
char name[SDL_VARIABLE_LENGTH_ARRAY];
} SDL_AudioDeviceItem;
@@ -136,15 +125,6 @@ typedef struct SDL_AudioDriver
} SDL_AudioDriver;
/* Streamer */
typedef struct
{
Uint8 *buffer;
int max_len; /* the maximum length in bytes */
int read_pos, write_pos; /* the position of the write and read heads in bytes */
} SDL_AudioStreamer;
/* Define the SDL audio driver structure */
struct SDL_AudioDevice
{
@@ -152,15 +132,14 @@ struct SDL_AudioDevice
/* Data common to all devices */
SDL_AudioDeviceID id;
/* The current audio specification (shared with audio thread) */
/* The device's current audio specification */
SDL_AudioSpec spec;
/* An audio conversion block for audio format emulation */
SDL_AudioCVT convert;
/* The callback's expected audio specification (converted vs device's spec). */
SDL_AudioSpec callbackspec;
/* The streamer, if sample rate conversion necessitates it */
int use_streamer;
SDL_AudioStreamer streamer;
/* Stream that converts and resamples. NULL if not needed. */
SDL_AudioStream *stream;
/* Current state flags */
SDL_atomic_t shutdown; /* true if we are signaling the play thread to end. */
@@ -168,8 +147,11 @@ struct SDL_AudioDevice
SDL_atomic_t paused;
SDL_bool iscapture;
/* Fake audio buffer for when the audio hardware is busy */
Uint8 *fake_stream;
/* Scratch buffer used in the bridge between SDL and the user callback. */
Uint8 *work_buffer;
/* Size, in bytes, of work_buffer. */
Uint32 work_buffer_len;
/* A mutex for locking the mixing buffers */
SDL_mutex *mixer_lock;
@@ -179,10 +161,7 @@ struct SDL_AudioDevice
SDL_threadID threadid;
/* Queued buffers (if app not using callback). */
SDL_AudioBufferQueue *buffer_queue_head; /* device fed from here. */
SDL_AudioBufferQueue *buffer_queue_tail; /* queue fills to here. */
SDL_AudioBufferQueue *buffer_queue_pool; /* these are unused packets. */
Uint32 queued_bytes; /* number of bytes of audio data in the queue. */
SDL_DataQueue *buffer_queue;
/* * * */
/* Data private to this driver */
@@ -200,6 +179,33 @@ typedef struct AudioBootStrap
int demand_only; /* 1==request explicitly, or it won't be available. */
} AudioBootStrap;
#endif /* _SDL_sysaudio_h */
/* Not all of these are available in a given build. Use #ifdefs, etc. */
extern AudioBootStrap PULSEAUDIO_bootstrap;
extern AudioBootStrap ALSA_bootstrap;
extern AudioBootStrap JACK_bootstrap;
extern AudioBootStrap SNDIO_bootstrap;
extern AudioBootStrap NETBSDAUDIO_bootstrap;
extern AudioBootStrap DSP_bootstrap;
extern AudioBootStrap QSAAUDIO_bootstrap;
extern AudioBootStrap SUNAUDIO_bootstrap;
extern AudioBootStrap ARTS_bootstrap;
extern AudioBootStrap ESD_bootstrap;
extern AudioBootStrap NACLAUDIO_bootstrap;
extern AudioBootStrap NAS_bootstrap;
extern AudioBootStrap WASAPI_bootstrap;
extern AudioBootStrap XAUDIO2_bootstrap;
extern AudioBootStrap DSOUND_bootstrap;
extern AudioBootStrap WINMM_bootstrap;
extern AudioBootStrap PAUDIO_bootstrap;
extern AudioBootStrap HAIKUAUDIO_bootstrap;
extern AudioBootStrap COREAUDIO_bootstrap;
extern AudioBootStrap DISKAUDIO_bootstrap;
extern AudioBootStrap DUMMYAUDIO_bootstrap;
extern AudioBootStrap FUSIONSOUND_bootstrap;
extern AudioBootStrap ANDROIDAUDIO_bootstrap;
extern AudioBootStrap PSPAUDIO_bootstrap;
extern AudioBootStrap EMSCRIPTENAUDIO_bootstrap;
#endif /* SDL_sysaudio_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/SDL_wave.c
+71 -1
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -403,6 +403,47 @@ IMA_ADPCM_decode(Uint8 ** audio_buf, Uint32 * audio_len)
return (0);
}
static int
ConvertSint24ToSint32(Uint8 ** audio_buf, Uint32 * audio_len)
{
const double DIVBY8388608 = 0.00000011920928955078125;
const Uint32 original_len = *audio_len;
const Uint32 samples = original_len / 3;
const Uint32 expanded_len = samples * sizeof (Uint32);
Uint8 *ptr = (Uint8 *) SDL_realloc(*audio_buf, expanded_len);
const Uint8 *src;
Uint32 *dst;
Uint32 i;
if (!ptr) {
return SDL_OutOfMemory();
}
*audio_buf = ptr;
*audio_len = expanded_len;
/* work from end to start, since we're expanding in-place. */
src = (ptr + original_len) - 3;
dst = ((Uint32 *) (ptr + expanded_len)) - 1;
for (i = 0; i < samples; i++) {
/* There's probably a faster way to do all this. */
const Sint32 converted = ((Sint32) ( (((Uint32) src[2]) << 24) |
(((Uint32) src[1]) << 16) |
(((Uint32) src[0]) << 8) )) >> 8;
const double scaled = (((double) converted) * DIVBY8388608);
src -= 3;
*(dst--) = (Sint32) (scaled * 2147483647.0);
}
return 0;
}
/* GUIDs that are used by WAVE_FORMAT_EXTENSIBLE */
static const Uint8 extensible_pcm_guid[16] = { 1, 0, 0, 0, 0, 0, 16, 0, 128, 0, 0, 170, 0, 56, 155, 113 };
static const Uint8 extensible_ieee_guid[16] = { 3, 0, 0, 0, 0, 0, 16, 0, 128, 0, 0, 170, 0, 56, 155, 113 };
SDL_AudioSpec *
SDL_LoadWAV_RW(SDL_RWops * src, int freesrc,
SDL_AudioSpec * spec, Uint8 ** audio_buf, Uint32 * audio_len)
@@ -421,6 +462,7 @@ SDL_LoadWAV_RW(SDL_RWops * src, int freesrc,
/* FMT chunk */
WaveFMT *format = NULL;
WaveExtensibleFMT *ext = NULL;
SDL_zero(chunk);
@@ -494,6 +536,24 @@ SDL_LoadWAV_RW(SDL_RWops * src, int freesrc,
}
IMA_ADPCM_encoded = 1;
break;
case EXTENSIBLE_CODE:
/* note that this ignores channel masks, smaller valid bit counts
inside a larger container, and most subtypes. This is just enough
to get things that didn't really _need_ WAVE_FORMAT_EXTENSIBLE
to be useful working when they use this format flag. */
ext = (WaveExtensibleFMT *) format;
if (SDL_SwapLE16(ext->size) < 22) {
SDL_SetError("bogus extended .wav header");
was_error = 1;
goto done;
}
if (SDL_memcmp(ext->subformat, extensible_pcm_guid, 16) == 0) {
break; /* cool. */
} else if (SDL_memcmp(ext->subformat, extensible_ieee_guid, 16) == 0) {
IEEE_float_encoded = 1;
break;
}
break;
case MP3_CODE:
SDL_SetError("MPEG Layer 3 data not supported");
was_error = 1;
@@ -528,6 +588,9 @@ SDL_LoadWAV_RW(SDL_RWops * src, int freesrc,
case 16:
spec->format = AUDIO_S16;
break;
case 24: /* convert this. */
spec->format = AUDIO_S32;
break;
case 32:
spec->format = AUDIO_S32;
break;
@@ -575,6 +638,13 @@ SDL_LoadWAV_RW(SDL_RWops * src, int freesrc,
}
}
if (SDL_SwapLE16(format->bitspersample) == 24) {
if (ConvertSint24ToSint32(audio_buf, audio_len) < 0) {
was_error = 1;
goto done;
}
}
/* Don't return a buffer that isn't a multiple of samplesize */
samplesize = ((SDL_AUDIO_BITSIZE(spec->format)) / 8) * spec->channels;
*audio_len &= ~(samplesize - 1);
libs/SDL2/src/audio/SDL_wave.h
+11 -1
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -38,6 +38,7 @@
#define IEEE_FLOAT_CODE 0x0003
#define IMA_ADPCM_CODE 0x0011
#define MP3_CODE 0x0055
#define EXTENSIBLE_CODE 0xFFFE
#define WAVE_MONO 1
#define WAVE_STEREO 2
@@ -64,4 +65,13 @@ typedef struct Chunk
Uint8 *data;
} Chunk;
typedef struct WaveExtensibleFMT
{
WaveFMT format;
Uint16 size;
Uint16 validbits;
Uint32 channelmask;
Uint8 subformat[16]; /* a GUID. */
} WaveExtensibleFMT;
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/alsa/SDL_alsa_audio.c
+53 -41
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -26,7 +26,6 @@
#include <sys/types.h>
#include <signal.h> /* For kill() */
#include <errno.h>
#include <string.h>
#include "SDL_assert.h"
@@ -310,12 +309,14 @@ swizzle_alsa_channels(_THIS, void *buffer, Uint32 bufferlen)
/* !!! FIXME: update this for 7.1 if needed, later. */
}
#ifdef SND_CHMAP_API_VERSION
/* Some devices have the right channel map, no swizzling necessary */
static void
no_swizzle(_THIS, void *buffer, Uint32 bufferlen)
{
return;
}
#endif /* SND_CHMAP_API_VERSION */
static void
@@ -329,20 +330,7 @@ ALSA_PlayDevice(_THIS)
this->hidden->swizzle_func(this, this->hidden->mixbuf, frames_left);
while ( frames_left > 0 && SDL_AtomicGet(&this->enabled) ) {
int status;
/* This wait is a work-around for a hang when USB devices are
unplugged. Normally it should not result in any waiting,
but in the case of a USB unplug, it serves as a way to
join the playback thread after the timeout occurs */
status = ALSA_snd_pcm_wait(this->hidden->pcm_handle, 1000);
if (status == 0) {
/*fprintf(stderr, "ALSA timeout waiting for available buffer space\n");*/
SDL_OpenedAudioDeviceDisconnected(this);
return;
}
status = ALSA_snd_pcm_writei(this->hidden->pcm_handle,
int status = ALSA_snd_pcm_writei(this->hidden->pcm_handle,
sample_buf, frames_left);
if (status < 0) {
@@ -362,6 +350,13 @@ ALSA_PlayDevice(_THIS)
}
continue;
}
else if (status == 0) {
/* No frames were written (no available space in pcm device).
Allow other threads to catch up. */
Uint32 delay = (frames_left / 2 * 1000) / this->spec.freq;
SDL_Delay(delay);
}
sample_buf += status * frame_size;
frames_left -= status;
}
@@ -381,23 +376,22 @@ ALSA_CaptureFromDevice(_THIS, void *buffer, int buflen)
this->spec.channels;
const int total_frames = buflen / frame_size;
snd_pcm_uframes_t frames_left = total_frames;
snd_pcm_uframes_t wait_time = frame_size / 2;
SDL_assert((buflen % frame_size) == 0);
while ( frames_left > 0 && SDL_AtomicGet(&this->enabled) ) {
/* !!! FIXME: This works, but needs more testing before going live */
/* ALSA_snd_pcm_wait(this->hidden->pcm_handle, -1); */
int status = ALSA_snd_pcm_readi(this->hidden->pcm_handle,
int status;
status = ALSA_snd_pcm_readi(this->hidden->pcm_handle,
sample_buf, frames_left);
if (status < 0) {
if (status == -EAGAIN) {
ALSA_snd_pcm_wait(this->hidden->pcm_handle, wait_time);
status = 0;
}
else if (status < 0) {
/*printf("ALSA: capture error %d\n", status);*/
if (status == -EAGAIN) {
/* Apparently snd_pcm_recover() doesn't handle this case -
does it assume snd_pcm_wait() above? */
SDL_Delay(1);
continue;
}
status = ALSA_snd_pcm_recover(this->hidden->pcm_handle, status, 0);
if (status < 0) {
/* Hmm, not much we can do - abort */
@@ -743,8 +737,9 @@ ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->hidden->mixlen);
}
/* Switch to blocking mode for playback */
ALSA_snd_pcm_nonblock(pcm_handle, 0);
if (!iscapture) {
ALSA_snd_pcm_nonblock(pcm_handle, 0);
}
/* We're ready to rock and roll. :-) */
return 0;
@@ -761,18 +756,28 @@ static void
add_device(const int iscapture, const char *name, void *hint, ALSA_Device **pSeen)
{
ALSA_Device *dev = SDL_malloc(sizeof (ALSA_Device));
char *desc = ALSA_snd_device_name_get_hint(hint, "DESC");
char *desc;
char *handle = NULL;
char *ptr;
if (!desc) {
SDL_free(dev);
return;
} else if (!dev) {
free(desc);
if (!dev) {
return;
}
/* Not all alsa devices are enumerable via snd_device_name_get_hint
(i.e. bluetooth devices). Therefore if hint is passed in to this
function as NULL, assume name contains desc.
Make sure not to free the storage associated with desc in this case */
if (hint) {
desc = ALSA_snd_device_name_get_hint(hint, "DESC");
if (!desc) {
SDL_free(dev);
return;
}
} else {
desc = (char *) name;
}
SDL_assert(name != NULL);
/* some strings have newlines, like "HDA NVidia, HDMI 0\nHDMI Audio Output".
@@ -786,14 +791,16 @@ add_device(const int iscapture, const char *name, void *hint, ALSA_Device **pSee
handle = SDL_strdup(name);
if (!handle) {
free(desc);
if (hint) {
free(desc);
}
SDL_free(dev);
return;
}
SDL_AddAudioDevice(iscapture, desc, handle);
free(desc);
if (hint)
free(desc);
dev->name = handle;
dev->iscapture = iscapture;
dev->next = *pSeen;
@@ -813,12 +820,15 @@ ALSA_HotplugThread(void *arg)
ALSA_Device *dev;
Uint32 ticks;
SDL_SetThreadPriority(SDL_THREAD_PRIORITY_LOW);
while (!SDL_AtomicGet(&ALSA_hotplug_shutdown)) {
void **hints = NULL;
ALSA_Device *unseen;
ALSA_Device *seen;
ALSA_Device *prev;
if (ALSA_snd_device_name_hint(-1, "pcm", &hints) != -1) {
ALSA_Device *unseen = devices;
ALSA_Device *seen = NULL;
ALSA_Device *prev;
int i, j;
const char *match = NULL;
int bestmatch = 0xFFFF;
@@ -828,6 +838,8 @@ ALSA_HotplugThread(void *arg)
"hw:", "sysdefault:", "default:", NULL
};
unseen = devices;
seen = NULL;
/* Apparently there are several different ways that ALSA lists
actual hardware. It could be prefixed with "hw:" or "default:"
or "sysdefault:" and maybe others. Go through the list and see
@@ -922,7 +934,7 @@ ALSA_HotplugThread(void *arg)
/* report anything still in unseen as removed. */
for (dev = unseen; dev; dev = next) {
/*printf("ALSA: removing %s device '%s'\n", dev->iscapture ? "capture" : "output", dev->name);*/
/*printf("ALSA: removing usb %s device '%s'\n", dev->iscapture ? "capture" : "output", dev->name);*/
next = dev->next;
SDL_RemoveAudioDevice(dev->iscapture, dev->name);
SDL_free(dev->name);
libs/SDL2/src/audio/alsa/SDL_alsa_audio.h
+4 -4
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -20,8 +20,8 @@
*/
#include "../../SDL_internal.h"
#ifndef _SDL_ALSA_audio_h
#define _SDL_ALSA_audio_h
#ifndef SDL_ALSA_audio_h_
#define SDL_ALSA_audio_h_
#include <alsa/asoundlib.h>
@@ -43,6 +43,6 @@ struct SDL_PrivateAudioData
void (*swizzle_func)(_THIS, void *buffer, Uint32 bufferlen);
};
#endif /* _SDL_ALSA_audio_h */
#endif /* SDL_ALSA_audio_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/android/SDL_androidaudio.c
+5 -1
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -215,6 +215,10 @@ void ANDROIDAUDIO_ResumeDevices(void)
}
}
#else
void ANDROIDAUDIO_ResumeDevices(void) {}
void ANDROIDAUDIO_PauseDevices(void) {}
#endif /* SDL_AUDIO_DRIVER_ANDROID */
libs/SDL2/src/audio/android/SDL_androidaudio.h
+7 -4
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -20,8 +20,8 @@
*/
#include "../../SDL_internal.h"
#ifndef _SDL_androidaudio_h
#define _SDL_androidaudio_h
#ifndef SDL_androidaudio_h_
#define SDL_androidaudio_h_
#include "../SDL_sysaudio.h"
@@ -34,6 +34,9 @@ struct SDL_PrivateAudioData
int resume;
};
#endif /* _SDL_androidaudio_h */
void ANDROIDAUDIO_ResumeDevices(void);
void ANDROIDAUDIO_PauseDevices(void);
#endif /* SDL_androidaudio_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/arts/SDL_artsaudio.c
+1 -1
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
libs/SDL2/src/audio/arts/SDL_artsaudio.h
+6 -5
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -20,8 +20,8 @@
*/
#include "../../SDL_internal.h"
#ifndef _SDL_artscaudio_h
#define _SDL_artscaudio_h
#ifndef SDL_artsaudio_h_
#define SDL_artsaudio_h_
#include <artsc.h>
@@ -42,11 +42,12 @@ struct SDL_PrivateAudioData
Uint8 *mixbuf;
int mixlen;
/* Support for audio timing using a timer, in addition to select() */
/* Support for audio timing using a timer, in addition to SDL_IOReady() */
float frame_ticks;
float next_frame;
};
#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */
#endif /* _SDL_artscaudio_h */
#endif /* SDL_artsaudio_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/coreaudio/SDL_coreaudio.h
+6 -5
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -20,8 +20,8 @@
*/
#include "../../SDL_internal.h"
#ifndef _SDL_coreaudio_h
#define _SDL_coreaudio_h
#ifndef SDL_coreaudio_h_
#define SDL_coreaudio_h_
#include "../SDL_sysaudio.h"
@@ -47,7 +47,7 @@ struct SDL_PrivateAudioData
{
SDL_Thread *thread;
AudioQueueRef audioQueue;
AudioQueueBufferRef audioBuffer[2];
AudioQueueBufferRef *audioBuffer;
void *buffer;
UInt32 bufferOffset;
UInt32 bufferSize;
@@ -63,5 +63,6 @@ struct SDL_PrivateAudioData
#endif
};
#endif /* _SDL_coreaudio_h */
#endif /* SDL_coreaudio_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/coreaudio/SDL_coreaudio.m
+75 -33
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -25,6 +25,7 @@
/* !!! FIXME: clean out some of the macro salsa in here. */
#include "SDL_audio.h"
#include "SDL_hints.h"
#include "../SDL_audio_c.h"
#include "../SDL_sysaudio.h"
#include "SDL_coreaudio.h"
@@ -285,9 +286,9 @@ static void interruption_begin(_THIS)
static void interruption_end(_THIS)
{
if (this != NULL && this->hidden != NULL && this->hidden->audioQueue != NULL
&& this->hidden->interrupted) {
&& this->hidden->interrupted
&& AudioQueueStart(this->hidden->audioQueue, NULL) == AVAudioSessionErrorCodeNone) {
this->hidden->interrupted = SDL_FALSE;
AudioQueueStart(this->hidden->audioQueue, NULL);
}
}
@@ -325,7 +326,8 @@ static BOOL update_audio_session(_THIS, SDL_bool open)
@autoreleasepool {
AVAudioSession *session = [AVAudioSession sharedInstance];
NSNotificationCenter *center = [NSNotificationCenter defaultCenter];
NSString *category;
/* Set category to ambient by default so that other music continues playing. */
NSString *category = AVAudioSessionCategoryAmbient;
NSError *err = nil;
if (open_playback_devices && open_capture_devices) {
@@ -333,10 +335,17 @@ static BOOL update_audio_session(_THIS, SDL_bool open)
} else if (open_capture_devices) {
category = AVAudioSessionCategoryRecord;
} else {
/* Set category to ambient so that other music continues playing.
You can change this at runtime in your own code if you need different
behavior. If this is common, we can add an SDL hint for this. */
category = AVAudioSessionCategoryAmbient;
const char *hint = SDL_GetHint(SDL_HINT_AUDIO_CATEGORY);
if (hint) {
if (SDL_strcasecmp(hint, "AVAudioSessionCategoryAmbient") == 0) {
category = AVAudioSessionCategoryAmbient;
} else if (SDL_strcasecmp(hint, "AVAudioSessionCategorySoloAmbient") == 0) {
category = AVAudioSessionCategorySoloAmbient;
} else if (SDL_strcasecmp(hint, "AVAudioSessionCategoryPlayback") == 0 ||
SDL_strcasecmp(hint, "playback") == 0) {
category = AVAudioSessionCategoryPlayback;
}
}
}
if (![session setCategory:category error:&err]) {
@@ -400,8 +409,12 @@ static void
outputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer)
{
SDL_AudioDevice *this = (SDL_AudioDevice *) inUserData;
if (SDL_AtomicGet(&this->hidden->shutdown)) {
return; /* don't do anything. */
}
if (!SDL_AtomicGet(&this->enabled) || SDL_AtomicGet(&this->paused)) {
/* Supply silence if audio is enabled and not paused */
/* Supply silence if audio is not enabled or paused */
SDL_memset(inBuffer->mAudioData, this->spec.silence, inBuffer->mAudioDataBytesCapacity);
} else {
UInt32 remaining = inBuffer->mAudioDataBytesCapacity;
@@ -412,7 +425,7 @@ outputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffe
if (this->hidden->bufferOffset >= this->hidden->bufferSize) {
/* Generate the data */
SDL_LockMutex(this->mixer_lock);
(*this->spec.callback)(this->spec.userdata,
(*this->callbackspec.callback)(this->callbackspec.userdata,
this->hidden->buffer, this->hidden->bufferSize);
SDL_UnlockMutex(this->mixer_lock);
this->hidden->bufferOffset = 0;
@@ -430,9 +443,7 @@ outputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffe
}
}
if (!SDL_AtomicGet(&this->hidden->shutdown)) {
AudioQueueEnqueueBuffer(this->hidden->audioQueue, inBuffer, 0, NULL);
}
AudioQueueEnqueueBuffer(this->hidden->audioQueue, inBuffer, 0, NULL);
inBuffer->mAudioDataByteSize = inBuffer->mAudioDataBytesCapacity;
}
@@ -443,7 +454,13 @@ inputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer
const AudioStreamPacketDescription *inPacketDescs )
{
SDL_AudioDevice *this = (SDL_AudioDevice *) inUserData;
if (SDL_AtomicGet(&this->enabled) && !SDL_AtomicGet(&this->paused)) { /* ignore unless we're active. */
if (SDL_AtomicGet(&this->shutdown)) {
return; /* don't do anything. */
}
/* ignore unless we're active. */
if (!SDL_AtomicGet(&this->paused) && SDL_AtomicGet(&this->enabled) && !SDL_AtomicGet(&this->paused)) {
const Uint8 *ptr = (const Uint8 *) inBuffer->mAudioData;
UInt32 remaining = inBuffer->mAudioDataByteSize;
while (remaining > 0) {
@@ -459,16 +476,14 @@ inputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer
if (this->hidden->bufferOffset >= this->hidden->bufferSize) {
SDL_LockMutex(this->mixer_lock);
(*this->spec.callback)(this->spec.userdata, this->hidden->buffer, this->hidden->bufferSize);
(*this->callbackspec.callback)(this->callbackspec.userdata, this->hidden->buffer, this->hidden->bufferSize);
SDL_UnlockMutex(this->mixer_lock);
this->hidden->bufferOffset = 0;
}
}
}
if (!SDL_AtomicGet(&this->hidden->shutdown)) {
AudioQueueEnqueueBuffer(this->hidden->audioQueue, inBuffer, 0, NULL);
}
AudioQueueEnqueueBuffer(this->hidden->audioQueue, inBuffer, 0, NULL);
}
@@ -514,7 +529,6 @@ static void
COREAUDIO_CloseDevice(_THIS)
{
const SDL_bool iscapture = this->iscapture;
int i;
/* !!! FIXME: what does iOS do when a bluetooth audio device vanishes? Headphones unplugged? */
/* !!! FIXME: (we only do a "default" device on iOS right now...can we do more?) */
@@ -527,24 +541,24 @@ COREAUDIO_CloseDevice(_THIS)
update_audio_session(this, SDL_FALSE);
#endif
/* if callback fires again, feed silence; don't call into the app. */
SDL_AtomicSet(&this->paused, 1);
if (this->hidden->audioQueue) {
AudioQueueDispose(this->hidden->audioQueue, 1);
}
if (this->hidden->thread) {
SDL_AtomicSet(&this->hidden->shutdown, 1);
SDL_WaitThread(this->hidden->thread, NULL);
}
if (this->hidden->audioQueue) {
for (i = 0; i < SDL_arraysize(this->hidden->audioBuffer); i++) {
if (this->hidden->audioBuffer[i]) {
AudioQueueFreeBuffer(this->hidden->audioQueue, this->hidden->audioBuffer[i]);
}
}
AudioQueueDispose(this->hidden->audioQueue, 1);
}
if (this->hidden->ready_semaphore) {
SDL_DestroySemaphore(this->hidden->ready_semaphore);
}
/* AudioQueueDispose() frees the actual buffer objects. */
SDL_free(this->hidden->audioBuffer);
SDL_free(this->hidden->thread_error);
SDL_free(this->hidden->buffer);
SDL_free(this->hidden);
@@ -663,7 +677,31 @@ prepare_audioqueue(_THIS)
return 0;
}
for (i = 0; i < SDL_arraysize(this->hidden->audioBuffer); i++) {
/* Make sure we can feed the device a minimum amount of time */
double MINIMUM_AUDIO_BUFFER_TIME_MS = 15.0;
#if defined(__IPHONEOS__)
if (floor(NSFoundationVersionNumber) <= NSFoundationVersionNumber_iOS_7_1) {
/* Older iOS hardware, use 40 ms as a minimum time */
MINIMUM_AUDIO_BUFFER_TIME_MS = 40.0;
}
#endif
const double msecs = (this->spec.samples / ((double) this->spec.freq)) * 1000.0;
int numAudioBuffers = 2;
if (msecs < MINIMUM_AUDIO_BUFFER_TIME_MS) { /* use more buffers if we have a VERY small sample set. */
numAudioBuffers = ((int)SDL_ceil(MINIMUM_AUDIO_BUFFER_TIME_MS / msecs) * 2);
}
this->hidden->audioBuffer = SDL_calloc(1, sizeof (AudioQueueBufferRef) * numAudioBuffers);
if (this->hidden->audioBuffer == NULL) {
SDL_OutOfMemory();
return 0;
}
#if DEBUG_COREAUDIO
printf("COREAUDIO: numAudioBuffers == %d\n", numAudioBuffers);
#endif
for (i = 0; i < numAudioBuffers; i++) {
result = AudioQueueAllocateBuffer(this->hidden->audioQueue, this->spec.size, &this->hidden->audioBuffer[i]);
CHECK_RESULT("AudioQueueAllocateBuffer");
SDL_memset(this->hidden->audioBuffer[i]->mAudioData, this->spec.silence, this->hidden->audioBuffer[i]->mAudioDataBytesCapacity);
@@ -696,10 +734,7 @@ audioqueue_thread(void *arg)
CFRunLoopRunInMode(kCFRunLoopDefaultMode, 0.10, 1);
}
if (this->iscapture) { /* just stop immediately for capture devices. */
AudioQueueStop(this->hidden->audioQueue, 1);
} else { /* Drain off any pending playback. */
AudioQueueStop(this->hidden->audioQueue, 0);
if (!this->iscapture) { /* Drain off any pending playback. */
const CFTimeInterval secs = (((this->spec.size / (SDL_AUDIO_BITSIZE(this->spec.format) / 8)) / this->spec.channels) / ((CFTimeInterval) this->spec.freq)) * 2.0;
CFRunLoopRunInMode(kCFRunLoopDefaultMode, secs, 0);
}
@@ -734,6 +769,13 @@ COREAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
if (!update_audio_session(this, SDL_TRUE)) {
return -1;
}
/* Stop CoreAudio from doing expensive audio rate conversion */
@autoreleasepool {
AVAudioSession* session = [AVAudioSession sharedInstance];
[session setPreferredSampleRate:this->spec.freq error:nil];
this->spec.freq = (int)session.sampleRate;
}
#endif
/* Setup a AudioStreamBasicDescription with the requested format */
libs/SDL2/src/audio/directsound/SDL_directsound.c
+3 -4
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -434,7 +434,6 @@ CreateCaptureBuffer(_THIS, const DWORD bufsize, WAVEFORMATEX *wfmt)
LPDIRECTSOUNDCAPTURE capture = this->hidden->capture;
LPDIRECTSOUNDCAPTUREBUFFER *capturebuf = &this->hidden->capturebuf;
DSCBUFFERDESC format;
// DWORD junk, cursor;
HRESULT result;
SDL_zero(format);
@@ -523,8 +522,8 @@ DSOUND_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
bufsize = numchunks * this->spec.size;
if ((bufsize < DSBSIZE_MIN) || (bufsize > DSBSIZE_MAX)) {
SDL_SetError("Sound buffer size must be between %d and %d",
(DSBSIZE_MIN < numchunks) ? 1 : DSBSIZE_MIN / numchunks,
DSBSIZE_MAX / numchunks);
(int) ((DSBSIZE_MIN < numchunks) ? 1 : DSBSIZE_MIN / numchunks),
(int) (DSBSIZE_MAX / numchunks));
} else {
int rc;
WAVEFORMATEX wfmt;
libs/SDL2/src/audio/directsound/SDL_directsound.h
+4 -4
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -20,8 +20,8 @@
*/
#include "../../SDL_internal.h"
#ifndef _SDL_directsound_h
#define _SDL_directsound_h
#ifndef SDL_directsound_h_
#define SDL_directsound_h_
#include "../../core/windows/SDL_directx.h"
@@ -42,6 +42,6 @@ struct SDL_PrivateAudioData
Uint8 *locked_buf;
};
#endif /* _SDL_directsound_h */
#endif /* SDL_directsound_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/disk/SDL_diskaudio.c
+7 -7
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -33,6 +33,7 @@
#include "SDL_audio.h"
#include "../SDL_audio_c.h"
#include "SDL_diskaudio.h"
#include "SDL_log.h"
/* !!! FIXME: these should be SDL hints, not environment variables. */
/* environment variables and defaults. */
@@ -160,12 +161,11 @@ DISKAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
}
#if HAVE_STDIO_H
fprintf(stderr,
"WARNING: You are using the SDL disk i/o audio driver!\n"
" %s file [%s].\n", iscapture ? "Reading from" : "Writing to",
fname);
#endif
SDL_LogCritical(SDL_LOG_CATEGORY_AUDIO,
"You are using the SDL disk i/o audio driver!\n");
SDL_LogCritical(SDL_LOG_CATEGORY_AUDIO,
" %s file [%s].\n", iscapture ? "Reading from" : "Writing to",
fname);
/* We're ready to rock and roll. :-) */
return 0;
libs/SDL2/src/audio/disk/SDL_diskaudio.h
+4 -4
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -20,8 +20,8 @@
*/
#include "../../SDL_internal.h"
#ifndef _SDL_diskaudio_h
#define _SDL_diskaudio_h
#ifndef SDL_diskaudio_h_
#define SDL_diskaudio_h_
#include "SDL_rwops.h"
#include "../SDL_sysaudio.h"
@@ -37,5 +37,5 @@ struct SDL_PrivateAudioData
Uint8 *mixbuf;
};
#endif /* _SDL_diskaudio_h */
#endif /* SDL_diskaudio_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/dsp/SDL_dspaudio.c
+1 -1
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
libs/SDL2/src/audio/dsp/SDL_dspaudio.h
+4 -4
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -20,8 +20,8 @@
*/
#include "../../SDL_internal.h"
#ifndef _SDL_dspaudio_h
#define _SDL_dspaudio_h
#ifndef SDL_dspaudio_h_
#define SDL_dspaudio_h_
#include "../SDL_sysaudio.h"
@@ -39,5 +39,5 @@ struct SDL_PrivateAudioData
};
#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */
#endif /* _SDL_dspaudio_h */
#endif /* SDL_dspaudio_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/dummy/SDL_dummyaudio.c
+1 -1
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
libs/SDL2/src/audio/dummy/SDL_dummyaudio.h
+4 -4
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -20,8 +20,8 @@
*/
#include "../../SDL_internal.h"
#ifndef _SDL_dummyaudio_h
#define _SDL_dummyaudio_h
#ifndef SDL_dummyaudio_h_
#define SDL_dummyaudio_h_
#include "../SDL_sysaudio.h"
@@ -37,5 +37,5 @@ struct SDL_PrivateAudioData
Uint32 initial_calls;
};
#endif /* _SDL_dummyaudio_h */
#endif /* SDL_dummyaudio_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/emscripten/SDL_emscriptenaudio.c
+74 -138
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -26,175 +26,121 @@
#include "SDL_log.h"
#include "../SDL_audio_c.h"
#include "SDL_emscriptenaudio.h"
#include "SDL_assert.h"
#include <emscripten/emscripten.h>
static int
copyData(_THIS)
static void
FeedAudioDevice(_THIS, const void *buf, const int buflen)
{
int byte_len;
const int framelen = (SDL_AUDIO_BITSIZE(this->spec.format) / 8) * this->spec.channels;
EM_ASM_ARGS({
var numChannels = SDL2.audio.currentOutputBuffer['numberOfChannels'];
for (var c = 0; c < numChannels; ++c) {
var channelData = SDL2.audio.currentOutputBuffer['getChannelData'](c);
if (channelData.length != $1) {
throw 'Web Audio output buffer length mismatch! Destination size: ' + channelData.length + ' samples vs expected ' + $1 + ' samples!';
}
if (this->hidden->write_off + this->convert.len_cvt > this->hidden->mixlen) {
if (this->hidden->write_off > this->hidden->read_off) {
SDL_memmove(this->hidden->mixbuf,
this->hidden->mixbuf + this->hidden->read_off,
this->hidden->mixlen - this->hidden->read_off);
this->hidden->write_off = this->hidden->write_off - this->hidden->read_off;
} else {
this->hidden->write_off = 0;
for (var j = 0; j < $1; ++j) {
channelData[j] = HEAPF32[$0 + ((j*numChannels + c) << 2) >> 2]; /* !!! FIXME: why are these shifts here? */
}
}
this->hidden->read_off = 0;
}
SDL_memcpy(this->hidden->mixbuf + this->hidden->write_off,
this->convert.buf,
this->convert.len_cvt);
this->hidden->write_off += this->convert.len_cvt;
byte_len = this->hidden->write_off - this->hidden->read_off;
return byte_len;
}, buf, buflen / framelen);
}
static void
HandleAudioProcess(_THIS)
{
Uint8 *buf = NULL;
int byte_len = 0;
int bytes = SDL_AUDIO_BITSIZE(this->spec.format) / 8;
SDL_AudioCallback callback = this->callbackspec.callback;
const int stream_len = this->callbackspec.size;
/* Only do something if audio is enabled */
if (!SDL_AtomicGet(&this->enabled) || SDL_AtomicGet(&this->paused)) {
if (this->stream) {
SDL_AudioStreamClear(this->stream);
}
return;
}
if (this->convert.needed) {
const int bytes_in = SDL_AUDIO_BITSIZE(this->convert.src_format) / 8;
if (this->hidden->conv_in_len != 0) {
this->convert.len = this->hidden->conv_in_len * bytes_in * this->spec.channels;
}
(*this->spec.callback) (this->spec.userdata,
this->convert.buf,
this->convert.len);
SDL_ConvertAudio(&this->convert);
buf = this->convert.buf;
byte_len = this->convert.len_cvt;
/* size mismatch*/
if (byte_len != this->spec.size) {
if (!this->hidden->mixbuf) {
this->hidden->mixlen = this->spec.size > byte_len ? this->spec.size * 2 : byte_len * 2;
this->hidden->mixbuf = SDL_malloc(this->hidden->mixlen);
if (this->stream == NULL) { /* no conversion necessary. */
SDL_assert(this->spec.size == stream_len);
callback(this->callbackspec.userdata, this->work_buffer, stream_len);
} else { /* streaming/converting */
int got;
while (SDL_AudioStreamAvailable(this->stream) < ((int) this->spec.size)) {
callback(this->callbackspec.userdata, this->work_buffer, stream_len);
if (SDL_AudioStreamPut(this->stream, this->work_buffer, stream_len) == -1) {
SDL_AudioStreamClear(this->stream);
SDL_AtomicSet(&this->enabled, 0);
break;
}
/* copy existing data */
byte_len = copyData(this);
/* read more data*/
while (byte_len < this->spec.size) {
(*this->spec.callback) (this->spec.userdata,
this->convert.buf,
this->convert.len);
SDL_ConvertAudio(&this->convert);
byte_len = copyData(this);
}
byte_len = this->spec.size;
buf = this->hidden->mixbuf + this->hidden->read_off;
this->hidden->read_off += byte_len;
}
} else {
if (!this->hidden->mixbuf) {
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = SDL_malloc(this->hidden->mixlen);
got = SDL_AudioStreamGet(this->stream, this->work_buffer, this->spec.size);
SDL_assert((got < 0) || (got == this->spec.size));
if (got != this->spec.size) {
SDL_memset(this->work_buffer, this->spec.silence, this->spec.size);
}
(*this->spec.callback) (this->spec.userdata,
this->hidden->mixbuf,
this->hidden->mixlen);
buf = this->hidden->mixbuf;
byte_len = this->hidden->mixlen;
}
if (buf) {
EM_ASM_ARGS({
var numChannels = SDL2.audio.currentOutputBuffer['numberOfChannels'];
for (var c = 0; c < numChannels; ++c) {
var channelData = SDL2.audio.currentOutputBuffer['getChannelData'](c);
if (channelData.length != $1) {
throw 'Web Audio output buffer length mismatch! Destination size: ' + channelData.length + ' samples vs expected ' + $1 + ' samples!';
}
for (var j = 0; j < $1; ++j) {
channelData[j] = HEAPF32[$0 + ((j*numChannels + c) << 2) >> 2];
}
}
}, buf, byte_len / bytes / this->spec.channels);
}
FeedAudioDevice(this, this->work_buffer, this->spec.size);
}
static void
HandleCaptureProcess(_THIS)
{
Uint8 *buf;
int buflen;
SDL_AudioCallback callback = this->callbackspec.callback;
const int stream_len = this->callbackspec.size;
/* Only do something if audio is enabled */
if (!SDL_AtomicGet(&this->enabled) || SDL_AtomicGet(&this->paused)) {
SDL_AudioStreamClear(this->stream);
return;
}
if (this->convert.needed) {
buf = this->convert.buf;
buflen = this->convert.len_cvt;
} else {
if (!this->hidden->mixbuf) {
this->hidden->mixbuf = (Uint8 *) SDL_malloc(this->spec.size);
if (!this->hidden->mixbuf) {
return; /* oh well. */
}
}
buf = this->hidden->mixbuf;
buflen = this->spec.size;
}
EM_ASM_ARGS({
var numChannels = SDL2.capture.currentCaptureBuffer.numberOfChannels;
if (numChannels == 1) { /* fastpath this a little for the common (mono) case. */
var channelData = SDL2.capture.currentCaptureBuffer.getChannelData(0);
for (var c = 0; c < numChannels; ++c) {
var channelData = SDL2.capture.currentCaptureBuffer.getChannelData(c);
if (channelData.length != $1) {
throw 'Web Audio capture buffer length mismatch! Destination size: ' + channelData.length + ' samples vs expected ' + $1 + ' samples!';
}
for (var j = 0; j < $1; ++j) {
setValue($0 + (j * 4), channelData[j], 'float');
}
} else {
for (var c = 0; c < numChannels; ++c) {
var channelData = SDL2.capture.currentCaptureBuffer.getChannelData(c);
if (channelData.length != $1) {
throw 'Web Audio capture buffer length mismatch! Destination size: ' + channelData.length + ' samples vs expected ' + $1 + ' samples!';
}
if (numChannels == 1) { /* fastpath this a little for the common (mono) case. */
for (var j = 0; j < $1; ++j) {
setValue($0 + (j * 4), channelData[j], 'float');
}
} else {
for (var j = 0; j < $1; ++j) {
setValue($0 + (((j * numChannels) + c) * 4), channelData[j], 'float');
}
}
}
}, buf, (this->spec.size / sizeof (float)) / this->spec.channels);
}, this->work_buffer, (this->spec.size / sizeof (float)) / this->spec.channels);
/* okay, we've got an interleaved float32 array in C now. */
if (this->convert.needed) {
SDL_ConvertAudio(&this->convert);
if (this->stream == NULL) { /* no conversion necessary. */
SDL_assert(this->spec.size == stream_len);
callback(this->callbackspec.userdata, this->work_buffer, stream_len);
} else { /* streaming/converting */
if (SDL_AudioStreamPut(this->stream, this->work_buffer, this->spec.size) == -1) {
SDL_AtomicSet(&this->enabled, 0);
}
while (SDL_AudioStreamAvailable(this->stream) >= stream_len) {
const int got = SDL_AudioStreamGet(this->stream, this->work_buffer, stream_len);
SDL_assert((got < 0) || (got == stream_len));
if (got != stream_len) {
SDL_memset(this->work_buffer, this->callbackspec.silence, stream_len);
}
callback(this->callbackspec.userdata, this->work_buffer, stream_len); /* Send it to the app. */
}
}
/* Send it to the app. */
(*this->spec.callback) (this->spec.userdata, buf, buflen);
}
static void
EMSCRIPTENAUDIO_CloseDevice(_THIS)
{
@@ -236,8 +182,9 @@ EMSCRIPTENAUDIO_CloseDevice(_THIS)
}
}, this->iscapture);
SDL_free(this->hidden->mixbuf);
#if 0 /* !!! FIXME: currently not used. Can we move some stuff off the SDL2 namespace? --ryan. */
SDL_free(this->hidden);
#endif
}
static int
@@ -245,8 +192,6 @@ EMSCRIPTENAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscaptu
{
SDL_bool valid_format = SDL_FALSE;
SDL_AudioFormat test_format;
int i;
float f;
int result;
/* based on parts of library_sdl.js */
@@ -293,29 +238,17 @@ EMSCRIPTENAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscaptu
}
/* Initialize all variables that we clean on shutdown */
#if 0 /* !!! FIXME: currently not used. Can we move some stuff off the SDL2 namespace? --ryan. */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_zerop(this->hidden);
#endif
/* limit to native freq */
const int sampleRate = EM_ASM_INT_V({
return SDL2.audioContext.sampleRate;
});
if(this->spec.freq != sampleRate) {
for (i = this->spec.samples; i > 0; i--) {
f = (float)i / (float)sampleRate * (float)this->spec.freq;
if (SDL_floor(f) == f) {
this->hidden->conv_in_len = SDL_floor(f);
break;
}
}
this->spec.freq = sampleRate;
}
this->spec.freq = EM_ASM_INT_V({ return SDL2.audioContext.sampleRate; });
SDL_CalculateAudioSpec(&this->spec);
@@ -395,6 +328,9 @@ EMSCRIPTENAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscaptu
static int
EMSCRIPTENAUDIO_Init(SDL_AudioDriverImpl * impl)
{
int available;
int capture_available;
/* Set the function pointers */
impl->OpenDevice = EMSCRIPTENAUDIO_OpenDevice;
impl->CloseDevice = EMSCRIPTENAUDIO_CloseDevice;
@@ -406,7 +342,7 @@ EMSCRIPTENAUDIO_Init(SDL_AudioDriverImpl * impl)
impl->ProvidesOwnCallbackThread = 1;
/* check availability */
const int available = EM_ASM_INT_V({
available = EM_ASM_INT_V({
if (typeof(AudioContext) !== 'undefined') {
return 1;
} else if (typeof(webkitAudioContext) !== 'undefined') {
@@ -419,7 +355,7 @@ EMSCRIPTENAUDIO_Init(SDL_AudioDriverImpl * impl)
SDL_SetError("No audio context available");
}
const int capture_available = available && EM_ASM_INT_V({
capture_available = available && EM_ASM_INT_V({
if ((typeof(navigator.mediaDevices) !== 'undefined') && (typeof(navigator.mediaDevices.getUserMedia) !== 'undefined')) {
return 1;
} else if (typeof(navigator.webkitGetUserMedia) !== 'undefined') {
libs/SDL2/src/audio/emscripten/SDL_emscriptenaudio.h
+6 -10
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -20,8 +20,8 @@
*/
#include "../../SDL_internal.h"
#ifndef _SDL_emscriptenaudio_h
#define _SDL_emscriptenaudio_h
#ifndef SDL_emscriptenaudio_h_
#define SDL_emscriptenaudio_h_
#include "../SDL_sysaudio.h"
@@ -30,13 +30,9 @@
struct SDL_PrivateAudioData
{
Uint8 *mixbuf;
Uint32 mixlen;
Uint32 conv_in_len;
Uint32 write_off, read_off;
int unused;
};
#endif /* _SDL_emscriptenaudio_h */
#endif /* SDL_emscriptenaudio_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/esd/SDL_esdaudio.c
+1 -1
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
libs/SDL2/src/audio/esd/SDL_esdaudio.h
+5 -4
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -20,8 +20,8 @@
*/
#include "../../SDL_internal.h"
#ifndef _SDL_esdaudio_h
#define _SDL_esdaudio_h
#ifndef SDL_esdaudio_h_
#define SDL_esdaudio_h_
#include "../SDL_sysaudio.h"
@@ -46,5 +46,6 @@ struct SDL_PrivateAudioData
};
#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */
#endif /* _SDL_esdaudio_h */
#endif /* SDL_esdaudio_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/fusionsound/SDL_fsaudio.c
+1 -1
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
libs/SDL2/src/audio/fusionsound/SDL_fsaudio.h
+5 -4
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -20,8 +20,8 @@
*/
#include "../../SDL_internal.h"
#ifndef _SDL_fsaudio_h
#define _SDL_fsaudio_h
#ifndef SDL_fsaudio_h_
#define SDL_fsaudio_h_
#include <fusionsound/fusionsound.h>
@@ -45,5 +45,6 @@ struct SDL_PrivateAudioData
};
#endif /* _SDL_fsaudio_h */
#endif /* SDL_fsaudio_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/haiku/SDL_haikuaudio.cc
+31 -17
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -36,6 +36,7 @@ extern "C"
#include "../SDL_audio_c.h"
#include "../SDL_sysaudio.h"
#include "SDL_haikuaudio.h"
#include "SDL_assert.h"
}
@@ -47,26 +48,39 @@ FillSound(void *device, void *stream, size_t len,
const media_raw_audio_format & format)
{
SDL_AudioDevice *audio = (SDL_AudioDevice *) device;
SDL_AudioCallback callback = audio->callbackspec.callback;
/* Only do soemthing if audio is enabled */
if (!SDL_AtomicGet(&audio->enabled)) {
/* Only do something if audio is enabled */
if (!SDL_AtomicGet(&audio->enabled) || SDL_AtomicGet(&audio->paused)) {
if (audio->stream) {
SDL_AudioStreamClear(audio->stream);
}
SDL_memset(stream, audio->spec.silence, len);
return;
}
if (!SDL_AtomicGet(&audio->paused)) {
if (audio->convert.needed) {
SDL_LockMutex(audio->mixer_lock);
(*audio->spec.callback) (audio->spec.userdata,
(Uint8 *) audio->convert.buf,
audio->convert.len);
SDL_UnlockMutex(audio->mixer_lock);
SDL_ConvertAudio(&audio->convert);
SDL_memcpy(stream, audio->convert.buf, audio->convert.len_cvt);
} else {
SDL_LockMutex(audio->mixer_lock);
(*audio->spec.callback) (audio->spec.userdata,
(Uint8 *) stream, len);
SDL_UnlockMutex(audio->mixer_lock);
SDL_assert(audio->spec.size == len);
if (audio->stream == NULL) { /* no conversion necessary. */
SDL_LockMutex(audio->mixer_lock);
callback(audio->callbackspec.userdata, (Uint8 *) stream, len);
SDL_UnlockMutex(audio->mixer_lock);
} else { /* streaming/converting */
const int stream_len = audio->callbackspec.size;
const int ilen = (int) len;
while (SDL_AudioStreamAvailable(audio->stream) < ilen) {
callback(audio->callbackspec.userdata, audio->work_buffer, stream_len);
if (SDL_AudioStreamPut(audio->stream, audio->work_buffer, stream_len) == -1) {
SDL_AudioStreamClear(audio->stream);
SDL_AtomicSet(&audio->enabled, 0);
break;
}
}
const int got = SDL_AudioStreamGet(audio->stream, stream, ilen);
SDL_assert((got < 0) || (got == ilen));
if (got != ilen) {
SDL_memset(stream, audio->spec.silence, len);
}
}
}
libs/SDL2/src/audio/haiku/SDL_haikuaudio.h
+4 -4
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -20,8 +20,8 @@
*/
#include "../../SDL_internal.h"
#ifndef _SDL_beaudio_h
#define _SDL_beaudio_h
#ifndef SDL_haikuaudio_h_
#define SDL_haikuaudio_h_
#include "../SDL_sysaudio.h"
@@ -33,6 +33,6 @@ struct SDL_PrivateAudioData
BSoundPlayer *audio_obj;
};
#endif /* _SDL_beaudio_h */
#endif /* SDL_haikuaudio_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/jack/SDL_jackaudio.c
+429
View File
@@ -0,0 +1,429 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#include "../../SDL_internal.h"
#if SDL_AUDIO_DRIVER_JACK
#include "SDL_assert.h"
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audio_c.h"
#include "SDL_jackaudio.h"
#include "SDL_loadso.h"
#include "../../thread/SDL_systhread.h"
static jack_client_t * (*JACK_jack_client_open) (const char *, jack_options_t, jack_status_t *, ...);
static int (*JACK_jack_client_close) (jack_client_t *);
static void (*JACK_jack_on_shutdown) (jack_client_t *, JackShutdownCallback, void *);
static int (*JACK_jack_activate) (jack_client_t *);
static int (*JACK_jack_deactivate) (jack_client_t *);
static void * (*JACK_jack_port_get_buffer) (jack_port_t *, jack_nframes_t);
static int (*JACK_jack_port_unregister) (jack_client_t *, jack_port_t *);
static void (*JACK_jack_free) (void *);
static const char ** (*JACK_jack_get_ports) (jack_client_t *, const char *, const char *, unsigned long);
static jack_nframes_t (*JACK_jack_get_sample_rate) (jack_client_t *);
static jack_nframes_t (*JACK_jack_get_buffer_size) (jack_client_t *);
static jack_port_t * (*JACK_jack_port_register) (jack_client_t *, const char *, const char *, unsigned long, unsigned long);
static const char * (*JACK_jack_port_name) (const jack_port_t *);
static int (*JACK_jack_connect) (jack_client_t *, const char *, const char *);
static int (*JACK_jack_set_process_callback) (jack_client_t *, JackProcessCallback, void *);
static int load_jack_syms(void);
#ifdef SDL_AUDIO_DRIVER_JACK_DYNAMIC
static const char *jack_library = SDL_AUDIO_DRIVER_JACK_DYNAMIC;
static void *jack_handle = NULL;
/* !!! FIXME: this is copy/pasted in several places now */
static int
load_jack_sym(const char *fn, void **addr)
{
*addr = SDL_LoadFunction(jack_handle, fn);
if (*addr == NULL) {
/* Don't call SDL_SetError(): SDL_LoadFunction already did. */
return 0;
}
return 1;
}
/* cast funcs to char* first, to please GCC's strict aliasing rules. */
#define SDL_JACK_SYM(x) \
if (!load_jack_sym(#x, (void **) (char *) &JACK_##x)) return -1
static void
UnloadJackLibrary(void)
{
if (jack_handle != NULL) {
SDL_UnloadObject(jack_handle);
jack_handle = NULL;
}
}
static int
LoadJackLibrary(void)
{
int retval = 0;
if (jack_handle == NULL) {
jack_handle = SDL_LoadObject(jack_library);
if (jack_handle == NULL) {
retval = -1;
/* Don't call SDL_SetError(): SDL_LoadObject already did. */
} else {
retval = load_jack_syms();
if (retval < 0) {
UnloadJackLibrary();
}
}
}
return retval;
}
#else
#define SDL_JACK_SYM(x) JACK_##x = x
static void
UnloadJackLibrary(void)
{
}
static int
LoadJackLibrary(void)
{
load_jack_syms();
return 0;
}
#endif /* SDL_AUDIO_DRIVER_JACK_DYNAMIC */
static int
load_jack_syms(void)
{
SDL_JACK_SYM(jack_client_open);
SDL_JACK_SYM(jack_client_close);
SDL_JACK_SYM(jack_on_shutdown);
SDL_JACK_SYM(jack_activate);
SDL_JACK_SYM(jack_deactivate);
SDL_JACK_SYM(jack_port_get_buffer);
SDL_JACK_SYM(jack_port_unregister);
SDL_JACK_SYM(jack_free);
SDL_JACK_SYM(jack_get_ports);
SDL_JACK_SYM(jack_get_sample_rate);
SDL_JACK_SYM(jack_get_buffer_size);
SDL_JACK_SYM(jack_port_register);
SDL_JACK_SYM(jack_port_name);
SDL_JACK_SYM(jack_connect);
SDL_JACK_SYM(jack_set_process_callback);
return 0;
}
static void
jackShutdownCallback(void *arg) /* JACK went away; device is lost. */
{
SDL_AudioDevice *this = (SDL_AudioDevice *) arg;
SDL_OpenedAudioDeviceDisconnected(this);
SDL_SemPost(this->hidden->iosem); /* unblock the SDL thread. */
}
// !!! FIXME: implement and register these!
//typedef int(* JackSampleRateCallback)(jack_nframes_t nframes, void *arg)
//typedef int(* JackBufferSizeCallback)(jack_nframes_t nframes, void *arg)
static int
jackProcessPlaybackCallback(jack_nframes_t nframes, void *arg)
{
SDL_AudioDevice *this = (SDL_AudioDevice *) arg;
jack_port_t **ports = this->hidden->sdlports;
const int total_channels = this->spec.channels;
const int total_frames = this->spec.samples;
int channelsi;
if (!SDL_AtomicGet(&this->enabled)) {
/* silence the buffer to avoid repeats and corruption. */
SDL_memset(this->hidden->iobuffer, '\0', this->spec.size);
}
for (channelsi = 0; channelsi < total_channels; channelsi++) {
float *dst = (float *) JACK_jack_port_get_buffer(ports[channelsi], nframes);
if (dst) {
const float *src = ((float *) this->hidden->iobuffer) + channelsi;
int framesi;
for (framesi = 0; framesi < total_frames; framesi++) {
*(dst++) = *src;
src += total_channels;
}
}
}
SDL_SemPost(this->hidden->iosem); /* tell SDL thread we're done; refill the buffer. */
return 0; /* success */
}
/* This function waits until it is possible to write a full sound buffer */
static void
JACK_WaitDevice(_THIS)
{
if (SDL_AtomicGet(&this->enabled)) {
if (SDL_SemWait(this->hidden->iosem) == -1) {
SDL_OpenedAudioDeviceDisconnected(this);
}
}
}
static Uint8 *
JACK_GetDeviceBuf(_THIS)
{
return (Uint8 *) this->hidden->iobuffer;
}
static int
jackProcessCaptureCallback(jack_nframes_t nframes, void *arg)
{
SDL_AudioDevice *this = (SDL_AudioDevice *) arg;
if (SDL_AtomicGet(&this->enabled)) {
jack_port_t **ports = this->hidden->sdlports;
const int total_channels = this->spec.channels;
const int total_frames = this->spec.samples;
int channelsi;
for (channelsi = 0; channelsi < total_channels; channelsi++) {
const float *src = (const float *) JACK_jack_port_get_buffer(ports[channelsi], nframes);
if (src) {
float *dst = ((float *) this->hidden->iobuffer) + channelsi;
int framesi;
for (framesi = 0; framesi < total_frames; framesi++) {
*dst = *(src++);
dst += total_channels;
}
}
}
}
SDL_SemPost(this->hidden->iosem); /* tell SDL thread we're done; new buffer is ready! */
return 0; /* success */
}
static int
JACK_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
SDL_assert(buflen == this->spec.size); /* we always fill a full buffer. */
/* Wait for JACK to fill the iobuffer */
if (SDL_SemWait(this->hidden->iosem) == -1) {
return -1;
}
SDL_memcpy(buffer, this->hidden->iobuffer, buflen);
return buflen;
}
static void
JACK_FlushCapture(_THIS)
{
SDL_SemWait(this->hidden->iosem);
}
static void
JACK_CloseDevice(_THIS)
{
if (this->hidden->client) {
JACK_jack_deactivate(this->hidden->client);
if (this->hidden->sdlports) {
const int channels = this->spec.channels;
int i;
for (i = 0; i < channels; i++) {
JACK_jack_port_unregister(this->hidden->client, this->hidden->sdlports[i]);
}
SDL_free(this->hidden->sdlports);
}
JACK_jack_client_close(this->hidden->client);
}
if (this->hidden->iosem) {
SDL_DestroySemaphore(this->hidden->iosem);
}
if (this->hidden->devports) {
JACK_jack_free(this->hidden->devports);
}
SDL_free(this->hidden->iobuffer);
}
static int
JACK_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
/* Note that JACK uses "output" for capture devices (they output audio
data to us) and "input" for playback (we input audio data to them).
Likewise, SDL's playback port will be "output" (we write data out)
and capture will be "input" (we read data in). */
const unsigned long sysportflags = iscapture ? JackPortIsOutput : JackPortIsInput;
const unsigned long sdlportflags = iscapture ? JackPortIsInput : JackPortIsOutput;
const JackProcessCallback callback = iscapture ? jackProcessCaptureCallback : jackProcessPlaybackCallback;
const char *sdlportstr = iscapture ? "input" : "output";
const char **devports = NULL;
jack_client_t *client = NULL;
jack_status_t status;
int channels = 0;
int i;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, sizeof (*this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
/* !!! FIXME: we _still_ need an API to specify an app name */
client = JACK_jack_client_open("SDL", JackNoStartServer, &status, NULL);
this->hidden->client = client;
if (client == NULL) {
return SDL_SetError("Can't open JACK client");
}
devports = JACK_jack_get_ports(client, NULL, NULL, JackPortIsPhysical | sysportflags);
this->hidden->devports = devports;
if (!devports || !devports[0]) {
return SDL_SetError("No physical JACK ports available");
}
while (devports[++channels]) {
/* spin to count devports */
}
/* !!! FIXME: docs say about buffer size: "This size may change, clients that depend on it must register a bufsize_callback so they will be notified if it does." */
/* Jack pretty much demands what it wants. */
this->spec.format = AUDIO_F32SYS;
this->spec.freq = JACK_jack_get_sample_rate(client);
this->spec.channels = channels;
this->spec.samples = JACK_jack_get_buffer_size(client);
SDL_CalculateAudioSpec(&this->spec);
this->hidden->iosem = SDL_CreateSemaphore(0);
if (!this->hidden->iosem) {
return -1; /* error was set by SDL_CreateSemaphore */
}
this->hidden->iobuffer = (float *) SDL_calloc(1, this->spec.size);
if (!this->hidden->iobuffer) {
return SDL_OutOfMemory();
}
/* Build SDL's ports, which we will connect to the device ports. */
this->hidden->sdlports = (jack_port_t **) SDL_calloc(channels, sizeof (jack_port_t *));
if (this->hidden->sdlports == NULL) {
return SDL_OutOfMemory();
}
for (i = 0; i < channels; i++) {
char portname[32];
SDL_snprintf(portname, sizeof (portname), "sdl_jack_%s_%d", sdlportstr, i);
this->hidden->sdlports[i] = JACK_jack_port_register(client, portname, JACK_DEFAULT_AUDIO_TYPE, sdlportflags, 0);
if (this->hidden->sdlports[i] == NULL) {
return SDL_SetError("jack_port_register failed");
}
}
if (JACK_jack_set_process_callback(client, callback, this) != 0) {
return SDL_SetError("JACK: Couldn't set process callback");
}
JACK_jack_on_shutdown(client, jackShutdownCallback, this);
if (JACK_jack_activate(client) != 0) {
return SDL_SetError("Failed to activate JACK client");
}
/* once activated, we can connect all the ports. */
for (i = 0; i < channels; i++) {
const char *sdlport = JACK_jack_port_name(this->hidden->sdlports[i]);
const char *srcport = iscapture ? devports[i] : sdlport;
const char *dstport = iscapture ? sdlport : devports[i];
if (JACK_jack_connect(client, srcport, dstport) != 0) {
return SDL_SetError("Couldn't connect JACK ports: %s => %s", srcport, dstport);
}
}
/* don't need these anymore. */
this->hidden->devports = NULL;
JACK_jack_free(devports);
/* We're ready to rock and roll. :-) */
return 0;
}
static void
JACK_Deinitialize(void)
{
UnloadJackLibrary();
}
static int
JACK_Init(SDL_AudioDriverImpl * impl)
{
if (LoadJackLibrary() < 0) {
return 0;
} else {
/* Make sure a JACK server is running and available. */
jack_status_t status;
jack_client_t *client = JACK_jack_client_open("SDL", JackNoStartServer, &status, NULL);
if (client == NULL) {
UnloadJackLibrary();
return 0;
}
JACK_jack_client_close(client);
}
/* Set the function pointers */
impl->OpenDevice = JACK_OpenDevice;
impl->WaitDevice = JACK_WaitDevice;
impl->GetDeviceBuf = JACK_GetDeviceBuf;
impl->CloseDevice = JACK_CloseDevice;
impl->Deinitialize = JACK_Deinitialize;
impl->CaptureFromDevice = JACK_CaptureFromDevice;
impl->FlushCapture = JACK_FlushCapture;
impl->OnlyHasDefaultOutputDevice = SDL_TRUE;
impl->OnlyHasDefaultCaptureDevice = SDL_TRUE;
impl->HasCaptureSupport = SDL_TRUE;
return 1; /* this audio target is available. */
}
AudioBootStrap JACK_bootstrap = {
"jack", "JACK Audio Connection Kit", JACK_Init, 0
};
#endif /* SDL_AUDIO_DRIVER_JACK */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/jack/SDL_jackaudio.h
+42
View File
@@ -0,0 +1,42 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#ifndef SDL_jackaudio_h_
#define SDL_jackaudio_h_
#include <jack/jack.h>
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
jack_client_t *client;
SDL_sem *iosem;
float *iobuffer;
const char **devports;
jack_port_t **sdlports;
};
#endif /* SDL_jackaudio_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/nacl/SDL_naclaudio.c
+37 -21
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -45,29 +45,46 @@
static void nacl_audio_callback(void* samples, uint32_t buffer_size, PP_TimeDelta latency, void* data);
/* FIXME: Make use of latency if needed */
static void nacl_audio_callback(void* samples, uint32_t buffer_size, PP_TimeDelta latency, void* data) {
static void nacl_audio_callback(void* stream, uint32_t buffer_size, PP_TimeDelta latency, void* data) {
const int len = (int) buffer_size;
SDL_AudioDevice* _this = (SDL_AudioDevice*) data;
SDL_AudioCallback callback = _this->callbackspec.callback;
SDL_LockMutex(private->mutex); /* !!! FIXME: is this mutex necessary? */
if (SDL_AtomicGet(&_this->enabled) && !SDL_AtomicGet(&_this->paused)) {
if (_this->convert.needed) {
SDL_LockMutex(_this->mixer_lock);
(*_this->spec.callback) (_this->spec.userdata,
(Uint8 *) _this->convert.buf,
_this->convert.len);
SDL_UnlockMutex(_this->mixer_lock);
SDL_ConvertAudio(&_this->convert);
SDL_memcpy(samples, _this->convert.buf, _this->convert.len_cvt);
} else {
SDL_LockMutex(_this->mixer_lock);
(*_this->spec.callback) (_this->spec.userdata, (Uint8 *) samples, buffer_size);
SDL_UnlockMutex(_this->mixer_lock);
/* Only do something if audio is enabled */
if (!SDL_AtomicGet(&_this->enabled) || SDL_AtomicGet(&_this->paused)) {
if (_this->stream) {
SDL_AudioStreamClear(_this->stream);
}
} else {
SDL_memset(samples, _this->spec.silence, buffer_size);
SDL_memset(stream, _this->spec.silence, len);
return;
}
SDL_assert(_this->spec.size == len);
if (_this->stream == NULL) { /* no conversion necessary. */
SDL_LockMutex(_this->mixer_lock);
callback(_this->callbackspec.userdata, stream, len);
SDL_UnlockMutex(_this->mixer_lock);
} else { /* streaming/converting */
const int stream_len = _this->callbackspec.size;
while (SDL_AudioStreamAvailable(_this->stream) < len) {
callback(_this->callbackspec.userdata, _this->work_buffer, stream_len);
if (SDL_AudioStreamPut(_this->stream, _this->work_buffer, stream_len) == -1) {
SDL_AudioStreamClear(_this->stream);
SDL_AtomicSet(&_this->enabled, 0);
break;
}
}
const int got = SDL_AudioStreamGet(_this->stream, stream, len);
SDL_assert((got < 0) || (got == len));
if (got != len) {
SDL_memset(stream, _this->spec.silence, len);
}
}
SDL_UnlockMutex(private->mutex);
}
@@ -89,8 +106,7 @@ NACLAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture) {
private = (SDL_PrivateAudioData *) SDL_calloc(1, (sizeof *private));
if (private == NULL) {
SDL_OutOfMemory();
return 0;
return SDL_OutOfMemory();
}
private->mutex = SDL_CreateMutex();
@@ -114,7 +130,7 @@ NACLAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture) {
/* Start audio playback while we are still on the main thread. */
ppb_audio->StartPlayback(private->audio);
return 1;
return 0;
}
static int
libs/SDL2/src/audio/nacl/SDL_naclaudio.h
+6 -4
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -21,8 +21,8 @@
#include "../../SDL_internal.h"
#ifndef _SDL_naclaudio_h
#define _SDL_naclaudio_h
#ifndef SDL_naclaudio_h_
#define SDL_naclaudio_h_
#include "SDL_audio.h"
#include "../SDL_sysaudio.h"
@@ -38,4 +38,6 @@ typedef struct SDL_PrivateAudioData {
PP_Resource audio;
} SDL_PrivateAudioData;
#endif /* _SDL_naclaudio_h */
#endif /* SDL_naclaudio_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/nas/SDL_nasaudio.c
+2 -2
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -116,7 +116,7 @@ LoadNASLibrary(void)
char *err = (char *) alloca(len);
SDL_strlcpy(err, origerr, len);
retval = -1;
SDL_SetError("NAS: SDL_LoadObject('%s') failed: %s\n",
SDL_SetError("NAS: SDL_LoadObject('%s') failed: %s",
nas_library, err);
} else {
retval = load_nas_syms();
libs/SDL2/src/audio/nas/SDL_nasaudio.h
+4 -4
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -20,8 +20,8 @@
*/
#include "../../SDL_internal.h"
#ifndef _SDL_nasaudio_h
#define _SDL_nasaudio_h
#ifndef SDL_nasaudio_h_
#define SDL_nasaudio_h_
#ifdef __sgi
#include <nas/audiolib.h>
@@ -51,6 +51,6 @@ struct SDL_PrivateAudioData
struct timeval last_tv;
int buf_free;
};
#endif /* _SDL_nasaudio_h */
#endif /* SDL_nasaudio_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/bsd/SDL_bsdaudio.c → libs/SDL2/src/audio/netbsd/SDL_netbsdaudio.c
+30 -36
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -20,10 +20,10 @@
*/
#include "../../SDL_internal.h"
#if SDL_AUDIO_DRIVER_BSD
#if SDL_AUDIO_DRIVER_NETBSD
/*
* Driver for native OpenBSD/NetBSD audio(4).
* Driver for native NetBSD audio(4).
* vedge@vedge.com.ar.
*/
@@ -38,9 +38,10 @@
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../../core/unix/SDL_poll.h"
#include "../SDL_audio_c.h"
#include "../SDL_audiodev_c.h"
#include "SDL_bsdaudio.h"
#include "SDL_netbsdaudio.h"
/* Use timer for synchronization */
/* #define USE_TIMER_SYNC */
@@ -50,14 +51,14 @@
static void
BSDAUDIO_DetectDevices(void)
NETBSDAUDIO_DetectDevices(void)
{
SDL_EnumUnixAudioDevices(0, NULL);
}
static void
BSDAUDIO_Status(_THIS)
NETBSDAUDIO_Status(_THIS)
{
#ifdef DEBUG_AUDIO
/* *INDENT-OFF* */
@@ -121,7 +122,7 @@ BSDAUDIO_Status(_THIS)
/* This function waits until it is possible to write a full sound buffer */
static void
BSDAUDIO_WaitDevice(_THIS)
NETBSDAUDIO_WaitDevice(_THIS)
{
#ifndef USE_BLOCKING_WRITES /* Not necessary when using blocking writes */
/* See if we need to use timed audio synchronization */
@@ -134,18 +135,11 @@ BSDAUDIO_WaitDevice(_THIS)
SDL_Delay(ticks);
}
} else {
/* Use select() for audio synchronization */
fd_set fdset;
struct timeval timeout;
FD_ZERO(&fdset);
FD_SET(this->hidden->audio_fd, &fdset);
timeout.tv_sec = 10;
timeout.tv_usec = 0;
/* Use SDL_IOReady() for audio synchronization */
#ifdef DEBUG_AUDIO
fprintf(stderr, "Waiting for audio to get ready\n");
#endif
if (select(this->hidden->audio_fd + 1, NULL, &fdset, NULL, &timeout)
if (SDL_IOReady(this->hidden->audio_fd, SDL_TRUE, 10 * 1000)
<= 0) {
const char *message =
"Audio timeout - buggy audio driver? (disabled)";
@@ -169,7 +163,7 @@ BSDAUDIO_WaitDevice(_THIS)
}
static void
BSDAUDIO_PlayDevice(_THIS)
NETBSDAUDIO_PlayDevice(_THIS)
{
int written, p = 0;
@@ -208,19 +202,19 @@ BSDAUDIO_PlayDevice(_THIS)
}
static Uint8 *
BSDAUDIO_GetDeviceBuf(_THIS)
NETBSDAUDIO_GetDeviceBuf(_THIS)
{
return (this->hidden->mixbuf);
}
static int
BSDAUDIO_CaptureFromDevice(_THIS, void *_buffer, int buflen)
NETBSDAUDIO_CaptureFromDevice(_THIS, void *_buffer, int buflen)
{
Uint8 *buffer = (Uint8 *) _buffer;
int br, p = 0;
/* Write the audio data, checking for EAGAIN on broken audio drivers */
/* Capture the audio data, checking for EAGAIN on broken audio drivers */
do {
br = read(this->hidden->audio_fd, buffer + p, buflen - p);
if (br > 0)
@@ -243,7 +237,7 @@ BSDAUDIO_CaptureFromDevice(_THIS, void *_buffer, int buflen)
}
static void
BSDAUDIO_FlushCapture(_THIS)
NETBSDAUDIO_FlushCapture(_THIS)
{
audio_info_t info;
size_t remain;
@@ -265,7 +259,7 @@ BSDAUDIO_FlushCapture(_THIS)
}
static void
BSDAUDIO_CloseDevice(_THIS)
NETBSDAUDIO_CloseDevice(_THIS)
{
if (this->hidden->audio_fd >= 0) {
close(this->hidden->audio_fd);
@@ -275,7 +269,7 @@ BSDAUDIO_CloseDevice(_THIS)
}
static int
BSDAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
NETBSDAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
const int flags = iscapture ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT;
SDL_AudioFormat format = 0;
@@ -383,24 +377,24 @@ BSDAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
}
BSDAUDIO_Status(this);
NETBSDAUDIO_Status(this);
/* We're ready to rock and roll. :-) */
return 0;
}
static int
BSDAUDIO_Init(SDL_AudioDriverImpl * impl)
NETBSDAUDIO_Init(SDL_AudioDriverImpl * impl)
{
/* Set the function pointers */
impl->DetectDevices = BSDAUDIO_DetectDevices;
impl->OpenDevice = BSDAUDIO_OpenDevice;
impl->PlayDevice = BSDAUDIO_PlayDevice;
impl->WaitDevice = BSDAUDIO_WaitDevice;
impl->GetDeviceBuf = BSDAUDIO_GetDeviceBuf;
impl->CloseDevice = BSDAUDIO_CloseDevice;
impl->CaptureFromDevice = BSDAUDIO_CaptureFromDevice;
impl->FlushCapture = BSDAUDIO_FlushCapture;
impl->DetectDevices = NETBSDAUDIO_DetectDevices;
impl->OpenDevice = NETBSDAUDIO_OpenDevice;
impl->PlayDevice = NETBSDAUDIO_PlayDevice;
impl->WaitDevice = NETBSDAUDIO_WaitDevice;
impl->GetDeviceBuf = NETBSDAUDIO_GetDeviceBuf;
impl->CloseDevice = NETBSDAUDIO_CloseDevice;
impl->CaptureFromDevice = NETBSDAUDIO_CaptureFromDevice;
impl->FlushCapture = NETBSDAUDIO_FlushCapture;
impl->HasCaptureSupport = SDL_TRUE;
impl->AllowsArbitraryDeviceNames = 1;
@@ -409,10 +403,10 @@ BSDAUDIO_Init(SDL_AudioDriverImpl * impl)
}
AudioBootStrap BSD_AUDIO_bootstrap = {
"bsd", "BSD audio", BSDAUDIO_Init, 0
AudioBootStrap NETBSDAUDIO_bootstrap = {
"netbsd", "NetBSD audio", NETBSDAUDIO_Init, 0
};
#endif /* SDL_AUDIO_DRIVER_BSD */
#endif /* SDL_AUDIO_DRIVER_NETBSD */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/bsd/SDL_bsdaudio.h → libs/SDL2/src/audio/netbsd/SDL_netbsdaudio.h
+5 -8
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -20,8 +20,8 @@
*/
#include "../../SDL_internal.h"
#ifndef _SDL_bsdaudio_h
#define _SDL_bsdaudio_h
#ifndef SDL_netbsdaudio_h_
#define SDL_netbsdaudio_h_
#include "../SDL_sysaudio.h"
@@ -32,20 +32,17 @@ struct SDL_PrivateAudioData
/* The file descriptor for the audio device */
int audio_fd;
/* The parent process id, to detect when application quits */
pid_t parent;
/* Raw mixing buffer */
Uint8 *mixbuf;
int mixlen;
/* Support for audio timing using a timer, in addition to select() */
/* Support for audio timing using a timer, in addition to SDL_IOReady() */
float frame_ticks;
float next_frame;
};
#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */
#endif /* _SDL_bsdaudio_h */
#endif /* SDL_netbsdaudio_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/paudio/SDL_paudio.c
+15 -28
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -36,9 +36,10 @@
#include "SDL_audio.h"
#include "SDL_stdinc.h"
#include "../SDL_audio_c.h"
#include "../../core/unix/SDL_poll.h"
#include "SDL_paudio.h"
#define DEBUG_AUDIO 0
/* #define DEBUG_AUDIO */
/* A conflict within AIX 4.3.3 <sys/> headers and probably others as well.
* I guess nobody ever uses audio... Shame over AIX header files. */
@@ -137,44 +138,31 @@ PAUDIO_WaitDevice(_THIS)
SDL_Delay(ticks);
}
} else {
int timeoutMS;
audio_buffer paud_bufinfo;
/* Use select() for audio synchronization */
struct timeval timeout;
FD_ZERO(&fdset);
FD_SET(this->hidden->audio_fd, &fdset);
if (ioctl(this->hidden->audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0) {
#ifdef DEBUG_AUDIO
fprintf(stderr, "Couldn't get audio buffer information\n");
#endif
timeout.tv_sec = 10;
timeout.tv_usec = 0;
timeoutMS = 10 * 1000;
} else {
long ms_in_buf = paud_bufinfo.write_buf_time;
timeout.tv_sec = ms_in_buf / 1000;
ms_in_buf = ms_in_buf - timeout.tv_sec * 1000;
timeout.tv_usec = ms_in_buf * 1000;
timeoutMS = paud_bufinfo.write_buf_time;
#ifdef DEBUG_AUDIO
fprintf(stderr,
"Waiting for write_buf_time=%ld,%ld\n",
timeout.tv_sec, timeout.tv_usec);
fprintf(stderr, "Waiting for write_buf_time=%d ms\n", timeoutMS);
#endif
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Waiting for audio to get ready\n");
#endif
if (select(this->hidden->audio_fd + 1, NULL, &fdset, NULL, &timeout)
<= 0) {
const char *message =
"Audio timeout - buggy audio driver? (disabled)";
if (SDL_IOReady(this->hidden->audio_fd, SDL_TRUE, timeoutMS) <= 0) {
/*
* In general we should never print to the screen,
* but in this case we have no other way of letting
* the user know what happened.
*/
fprintf(stderr, "SDL: %s - %s\n", strerror(errno), message);
fprintf(stderr, "SDL: %s - Audio timeout - buggy audio driver? (disabled)\n", strerror(errno));
SDL_OpenedAudioDeviceDisconnected(this);
/* Don't try to close - may hang */
this->hidden->audio_fd = -1;
@@ -245,7 +233,6 @@ PAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
SDL_AudioFormat test_format;
audio_init paud_init;
audio_buffer paud_bufinfo;
audio_status paud_status;
audio_control paud_control;
audio_change paud_change;
int fd = -1;
@@ -487,7 +474,7 @@ PAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
return SDL_SetError("Can't start audio play");
}
/* Check to see if we need to use select() workaround */
/* Check to see if we need to use SDL_IOReady() workaround */
if (workaround != NULL) {
this->hidden->frame_ticks = (float) (this->spec.samples * 1000) /
this->spec.freq;
@@ -510,11 +497,11 @@ PAUDIO_Init(SDL_AudioDriverImpl * impl)
close(fd);
/* Set the function pointers */
impl->OpenDevice = DSP_OpenDevice;
impl->PlayDevice = DSP_PlayDevice;
impl->PlayDevice = DSP_WaitDevice;
impl->GetDeviceBuf = DSP_GetDeviceBuf;
impl->CloseDevice = DSP_CloseDevice;
impl->OpenDevice = PAUDIO_OpenDevice;
impl->PlayDevice = PAUDIO_PlayDevice;
impl->PlayDevice = PAUDIO_WaitDevice;
impl->GetDeviceBuf = PAUDIO_GetDeviceBuf;
impl->CloseDevice = PAUDIO_CloseDevice;
impl->OnlyHasDefaultOutputDevice = 1; /* !!! FIXME: add device enum! */
return 1; /* this audio target is available. */
libs/SDL2/src/audio/paudio/SDL_paudio.h
+6 -5
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -20,8 +20,8 @@
*/
#include "../../SDL_internal.h"
#ifndef _SDL_paudaudio_h
#define _SDL_paudaudio_h
#ifndef SDL_paudio_h_
#define SDL_paudio_h_
#include "../SDL_sysaudio.h"
@@ -37,11 +37,12 @@ struct SDL_PrivateAudioData
Uint8 *mixbuf;
int mixlen;
/* Support for audio timing using a timer, in addition to select() */
/* Support for audio timing using a timer, in addition to SDL_IOReady() */
float frame_ticks;
float next_frame;
};
#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */
#endif /* _SDL_paudaudio_h */
#endif /* SDL_paudio_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/psp/SDL_pspaudio.c
+2 -1
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -80,6 +80,7 @@ PSPAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
if (this->spec.channels == 1) {
format = PSP_AUDIO_FORMAT_MONO;
} else {
this->spec.channels = 2;
format = PSP_AUDIO_FORMAT_STEREO;
}
this->hidden->channel = sceAudioChReserve(PSP_AUDIO_NEXT_CHANNEL, this->spec.samples, format);
libs/SDL2/src/audio/psp/SDL_pspaudio.h
+6 -6
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -19,8 +19,8 @@
3. This notice may not be removed or altered from any source distribution.
*/
#ifndef _SDL_pspaudio_h
#define _SDL_pspaudio_h
#ifndef SDL_pspaudio_h_
#define SDL_pspaudio_h_
#include "../SDL_sysaudio.h"
@@ -40,6 +40,6 @@ struct SDL_PrivateAudioData {
int next_buffer;
};
#endif /* _SDL_pspaudio_h */
/* vim: ts=4 sw=4
*/
#endif /* SDL_pspaudio_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/pulseaudio/SDL_pulseaudio.c
+1 -2
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -37,7 +37,6 @@
#endif
#include <unistd.h>
#include <sys/types.h>
#include <errno.h>
#include <pulse/pulseaudio.h>
#include "SDL_timer.h"
libs/SDL2/src/audio/pulseaudio/SDL_pulseaudio.h
+4 -4
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -20,8 +20,8 @@
*/
#include "../../SDL_internal.h"
#ifndef _SDL_pulseaudio_h
#define _SDL_pulseaudio_h
#ifndef SDL_pulseaudio_h_
#define SDL_pulseaudio_h_
#include <pulse/simple.h>
@@ -47,6 +47,6 @@ struct SDL_PrivateAudioData
int capturelen;
};
#endif /* _SDL_pulseaudio_h */
#endif /* SDL_pulseaudio_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/qsa/SDL_qsa_audio.c
+27 -132
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -45,6 +45,7 @@
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../../core/unix/SDL_poll.h"
#include "../SDL_audio_c.h"
#include "SDL_qsa_audio.h"
@@ -56,24 +57,6 @@
#define DEFAULT_CPARAMS_FRAGS_MIN 1
#define DEFAULT_CPARAMS_FRAGS_MAX 1
#define QSA_NO_WORKAROUNDS 0x00000000
#define QSA_MMAP_WORKAROUND 0x00000001
struct BuggyCards
{
char *cardname;
unsigned long bugtype;
};
#define QSA_WA_CARDS 3
#define QSA_MAX_CARD_NAME_LENGTH 33
struct BuggyCards buggycards[QSA_WA_CARDS] = {
{"Sound Blaster Live!", QSA_MMAP_WORKAROUND},
{"Vortex 8820", QSA_MMAP_WORKAROUND},
{"Vortex 8830", QSA_MMAP_WORKAROUND},
};
/* List of found devices */
#define QSA_MAX_DEVICES 32
#define QSA_MAX_NAME_LENGTH 81+16 /* Hardcoded in QSA, can't be changed */
@@ -97,40 +80,16 @@ QSA_SetError(const char *fn, int status)
return SDL_SetError("QSA: %s() failed: %s", fn, snd_strerror(status));
}
/* card names check to apply the workarounds */
static int
QSA_CheckBuggyCards(_THIS, unsigned long checkfor)
{
char scardname[QSA_MAX_CARD_NAME_LENGTH];
int it;
if (snd_card_get_name
(this->hidden->cardno, scardname, QSA_MAX_CARD_NAME_LENGTH - 1) < 0) {
return 0;
}
for (it = 0; it < QSA_WA_CARDS; it++) {
if (SDL_strcmp(buggycards[it].cardname, scardname) == 0) {
if (buggycards[it].bugtype == checkfor) {
return 1;
}
}
}
return 0;
}
/* !!! FIXME: does this need to be here? Does the SDL version not work? */
static void
QSA_ThreadInit(_THIS)
{
struct sched_param param;
int status;
/* Increase default 10 priority to 25 to avoid jerky sound */
status = SchedGet(0, 0, &param);
param.sched_priority = param.sched_curpriority + 15;
status = SchedSet(0, 0, SCHED_NOCHANGE, &param);
struct sched_param param;
if (SchedGet(0, 0, &param) != -1) {
param.sched_priority = param.sched_curpriority + 15;
SchedSet(0, 0, SCHED_NOCHANGE, &param);
}
}
/* PCM channel parameters initialize function */
@@ -155,67 +114,25 @@ QSA_InitAudioParams(snd_pcm_channel_params_t * cpars)
static void
QSA_WaitDevice(_THIS)
{
fd_set wfds;
fd_set rfds;
int selectret;
struct timeval timeout;
int result;
if (!this->hidden->iscapture) {
FD_ZERO(&wfds);
FD_SET(this->hidden->audio_fd, &wfds);
} else {
FD_ZERO(&rfds);
FD_SET(this->hidden->audio_fd, &rfds);
}
do {
/* Setup timeout for playing one fragment equal to 2 seconds */
/* If timeout occured than something wrong with hardware or driver */
/* For example, Vortex 8820 audio driver stucks on second DAC because */
/* it doesn't exist ! */
timeout.tv_sec = 2;
timeout.tv_usec = 0;
/* Setup timeout for playing one fragment equal to 2 seconds */
/* If timeout occured than something wrong with hardware or driver */
/* For example, Vortex 8820 audio driver stucks on second DAC because */
/* it doesn't exist ! */
result = SDL_IOReady(this->hidden->audio_fd, !this->hidden->iscapture, 2 * 1000);
switch (result) {
case -1:
SDL_SetError("QSA: SDL_IOReady() failed: %s", strerror(errno));
break;
case 0:
SDL_SetError("QSA: timeout on buffer waiting occured");
this->hidden->timeout_on_wait = 1;
break;
default:
this->hidden->timeout_on_wait = 0;
if (!this->hidden->iscapture) {
selectret =
select(this->hidden->audio_fd + 1, NULL, &wfds, NULL,
&timeout);
} else {
selectret =
select(this->hidden->audio_fd + 1, &rfds, NULL, NULL,
&timeout);
}
switch (selectret) {
case -1:
{
SDL_SetError("QSA: select() failed: %s", strerror(errno));
return;
}
break;
case 0:
{
SDL_SetError("QSA: timeout on buffer waiting occured");
this->hidden->timeout_on_wait = 1;
return;
}
break;
default:
{
if (!this->hidden->iscapture) {
if (FD_ISSET(this->hidden->audio_fd, &wfds)) {
return;
}
} else {
if (FD_ISSET(this->hidden->audio_fd, &rfds)) {
return;
}
}
}
break;
}
} while (1);
break;
}
}
static void
@@ -357,7 +274,6 @@ QSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_zerop(this->hidden);
/* Initialize channel transfer parameters to default */
QSA_InitAudioParams(&cparams);
@@ -371,13 +287,13 @@ QSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
this->hidden->cardno = device->cardno;
status = snd_pcm_open(&this->hidden->audio_handle,
device->cardno, device->deviceno,
iscapture ? SND_PCM_OPEN_PLAYBACK : SND_PCM_OPEN_CAPTURE);
iscapture ? SND_PCM_OPEN_CAPTURE : SND_PCM_OPEN_PLAYBACK);
} else {
/* Open system default audio device */
status = snd_pcm_open_preferred(&this->hidden->audio_handle,
&this->hidden->cardno,
&this->hidden->deviceno,
iscapture ? SND_PCM_OPEN_PLAYBACK : SND_PCM_OPEN_CAPTURE);
iscapture ? SND_PCM_OPEN_CAPTURE : SND_PCM_OPEN_PLAYBACK);
}
/* Check if requested device is opened */
@@ -386,16 +302,6 @@ QSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
return QSA_SetError("snd_pcm_open", status);
}
if (!QSA_CheckBuggyCards(this, QSA_MMAP_WORKAROUND)) {
/* Disable QSA MMAP plugin for buggy audio drivers */
status =
snd_pcm_plugin_set_disable(this->hidden->audio_handle,
PLUGIN_DISABLE_MMAP);
if (status < 0) {
return QSA_SetError("snd_pcm_plugin_set_disable", status);
}
}
/* Try for a closest match on audio format */
format = 0;
/* can't use format as SND_PCM_SFMT_U8 = 0 in qsa */
@@ -494,7 +400,7 @@ QSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
/* Setup the transfer parameters according to cparams */
status = snd_pcm_plugin_params(this->hidden->audio_handle, &cparams);
if (status < 0) {
return QSA_SetError("snd_pcm_channel_params", status);
return QSA_SetError("snd_pcm_plugin_params", status);
}
/* Make sure channel is setup right one last time */
@@ -722,9 +628,6 @@ QSA_Deinitialize(void)
static int
QSA_Init(SDL_AudioDriverImpl * impl)
{
snd_pcm_t *handle = NULL;
int32_t status = 0;
/* Clear devices array */
SDL_zero(qsa_playback_device);
SDL_zero(qsa_capture_device);
@@ -745,20 +648,12 @@ QSA_Init(SDL_AudioDriverImpl * impl)
impl->LockDevice = NULL;
impl->UnlockDevice = NULL;
impl->OnlyHasDefaultOutputDevice = 0;
impl->ProvidesOwnCallbackThread = 0;
impl->SkipMixerLock = 0;
impl->HasCaptureSupport = 1;
impl->OnlyHasDefaultOutputDevice = 0;
impl->OnlyHasDefaultCaptureDevice = 0;
/* Check if io-audio manager is running or not */
status = snd_cards();
if (status == 0) {
/* if no, return immediately */
return 1;
}
return 1; /* this audio target is available. */
}
libs/SDL2/src/audio/qsa/SDL_qsa_audio.h
+1 -1
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
libs/SDL2/src/audio/sdlgenaudiocvt.pl
-761
View File
@@ -1,761 +0,0 @@
#!/usr/bin/perl -w
use warnings;
use strict;
my @audiotypes = qw(
U8
S8
U16LSB
S16LSB
U16MSB
S16MSB
S32LSB
S32MSB
F32LSB
F32MSB
);
my @channels = ( 1, 2, 4, 6, 8 );
my %funcs;
my $custom_converters = 0;
sub getTypeConvertHashId {
my ($from, $to) = @_;
return "TYPECONVERTER $from/$to";
}
sub getResamplerHashId {
my ($from, $channels, $upsample, $multiple) = @_;
return "RESAMPLER $from/$channels/$upsample/$multiple";
}
sub outputHeader {
print <<EOF;
/* DO NOT EDIT! This file is generated by sdlgenaudiocvt.pl */
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken\@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#include "../SDL_internal.h"
#include "SDL_audio.h"
#include "SDL_audio_c.h"
#ifndef DEBUG_CONVERT
#define DEBUG_CONVERT 0
#endif
/* If you can guarantee your data and need space, you can eliminate code... */
/* Just build the arbitrary resamplers if you're saving code space. */
#ifndef LESS_RESAMPLERS
#define LESS_RESAMPLERS 0
#endif
/* Don't build any resamplers if you're REALLY saving code space. */
#ifndef NO_RESAMPLERS
#define NO_RESAMPLERS 0
#endif
/* Don't build any type converters if you're saving code space. */
#ifndef NO_CONVERTERS
#define NO_CONVERTERS 0
#endif
/* *INDENT-OFF* */
EOF
my @vals = ( 127, 32767, 2147483647 );
foreach (@vals) {
my $val = $_;
my $fval = 1.0 / $val;
print("#define DIVBY${val} ${fval}f\n");
}
print("\n");
}
sub outputFooter {
print <<EOF;
/* $custom_converters converters generated. */
/* *INDENT-ON* */
/* vi: set ts=4 sw=4 expandtab: */
EOF
}
sub splittype {
my $t = shift;
my ($signed, $size, $endian) = $t =~ /([USF])(\d+)([LM]SB|)/;
my $float = ($signed eq 'F') ? 1 : 0;
$signed = (($float) or ($signed eq 'S')) ? 1 : 0;
$endian = 'NONE' if ($endian eq '');
my $ctype = '';
if ($float) {
$ctype = (($size == 32) ? 'float' : 'double');
} else {
$ctype = (($signed) ? 'S' : 'U') . "int${size}";
}
return ($signed, $float, $size, $endian, $ctype);
}
sub getSwapFunc {
my ($size, $signed, $float, $endian, $val) = @_;
my $BEorLE = (($endian eq 'MSB') ? 'BE' : 'LE');
my $code = '';
if ($float) {
$code = "SDL_SwapFloat${BEorLE}($val)";
} else {
if ($size > 8) {
$code = "SDL_Swap${BEorLE}${size}($val)";
} else {
$code = $val;
}
if (($signed) and (!$float)) {
$code = "((Sint${size}) $code)";
}
}
return "${code}";
}
sub maxIntVal {
my $size = shift;
if ($size == 8) {
return 0x7F;
} elsif ($size == 16) {
return 0x7FFF;
} elsif ($size == 32) {
return 0x7FFFFFFF;
}
die("bug in script.\n");
}
sub getFloatToIntMult {
my $size = shift;
my $val = maxIntVal($size) . '.0';
$val .= 'f' if ($size < 32);
return $val;
}
sub getIntToFloatDivBy {
my $size = shift;
return 'DIVBY' . maxIntVal($size);
}
sub getSignFlipVal {
my $size = shift;
if ($size == 8) {
return '0x80';
} elsif ($size == 16) {
return '0x8000';
} elsif ($size == 32) {
return '0x80000000';
}
die("bug in script.\n");
}
sub buildCvtFunc {
my ($from, $to) = @_;
my ($fsigned, $ffloat, $fsize, $fendian, $fctype) = splittype($from);
my ($tsigned, $tfloat, $tsize, $tendian, $tctype) = splittype($to);
my $diffs = 0;
$diffs++ if ($fsize != $tsize);
$diffs++ if ($fsigned != $tsigned);
$diffs++ if ($ffloat != $tfloat);
$diffs++ if ($fendian ne $tendian);
return if ($diffs == 0);
my $hashid = getTypeConvertHashId($from, $to);
if (1) { # !!! FIXME: if ($diffs > 1) {
my $sym = "SDL_Convert_${from}_to_${to}";
$funcs{$hashid} = $sym;
$custom_converters++;
# Always unsigned for ints, for possible byteswaps.
my $srctype = (($ffloat) ? 'float' : "Uint${fsize}");
print <<EOF;
static void SDLCALL
${sym}(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
int i;
const $srctype *src;
$tctype *dst;
#if DEBUG_CONVERT
fprintf(stderr, "Converting AUDIO_${from} to AUDIO_${to}.\\n");
#endif
EOF
if ($fsize < $tsize) {
my $mult = $tsize / $fsize;
print <<EOF;
src = ((const $srctype *) (cvt->buf + cvt->len_cvt)) - 1;
dst = (($tctype *) (cvt->buf + cvt->len_cvt * $mult)) - 1;
for (i = cvt->len_cvt / sizeof ($srctype); i; --i, --src, --dst) {
EOF
} else {
print <<EOF;
src = (const $srctype *) cvt->buf;
dst = ($tctype *) cvt->buf;
for (i = cvt->len_cvt / sizeof ($srctype); i; --i, ++src, ++dst) {
EOF
}
# Have to convert to/from float/int.
# !!! FIXME: cast through double for int32<->float?
my $code = getSwapFunc($fsize, $fsigned, $ffloat, $fendian, '*src');
if ($ffloat != $tfloat) {
if ($ffloat) {
my $mult = getFloatToIntMult($tsize);
if (!$tsigned) { # bump from -1.0f/1.0f to 0.0f/2.0f
$code = "($code + 1.0f)";
}
$code = "(($tctype) ($code * $mult))";
} else {
# $divby will be the reciprocal, to avoid pipeline stalls
# from floating point division...so multiply it.
my $divby = getIntToFloatDivBy($fsize);
$code = "(((float) $code) * $divby)";
if (!$fsigned) { # bump from 0.0f/2.0f to -1.0f/1.0f.
$code = "($code - 1.0f)";
}
}
} else {
# All integer conversions here.
if ($fsigned != $tsigned) {
my $signflipval = getSignFlipVal($fsize);
$code = "(($code) ^ $signflipval)";
}
my $shiftval = abs($fsize - $tsize);
if ($fsize < $tsize) {
$code = "((($tctype) $code) << $shiftval)";
} elsif ($fsize > $tsize) {
$code = "(($tctype) ($code >> $shiftval))";
}
}
my $swap = getSwapFunc($tsize, $tsigned, $tfloat, $tendian, 'val');
print <<EOF;
const $tctype val = $code;
*dst = ${swap};
}
EOF
if ($fsize > $tsize) {
my $divby = $fsize / $tsize;
print(" cvt->len_cvt /= $divby;\n");
} elsif ($fsize < $tsize) {
my $mult = $tsize / $fsize;
print(" cvt->len_cvt *= $mult;\n");
}
print <<EOF;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, AUDIO_$to);
}
}
EOF
} else {
if ($fsigned != $tsigned) {
$funcs{$hashid} = 'SDL_ConvertSigned';
} elsif ($ffloat != $tfloat) {
$funcs{$hashid} = 'SDL_ConvertFloat';
} elsif ($fsize != $tsize) {
$funcs{$hashid} = 'SDL_ConvertSize';
} elsif ($fendian ne $tendian) {
$funcs{$hashid} = 'SDL_ConvertEndian';
} else {
die("error in script.\n");
}
}
}
sub buildTypeConverters {
print "#if !NO_CONVERTERS\n\n";
foreach (@audiotypes) {
my $from = $_;
foreach (@audiotypes) {
my $to = $_;
buildCvtFunc($from, $to);
}
}
print "#endif /* !NO_CONVERTERS */\n\n\n";
print "const SDL_AudioTypeFilters sdl_audio_type_filters[] =\n{\n";
print "#if !NO_CONVERTERS\n";
foreach (@audiotypes) {
my $from = $_;
foreach (@audiotypes) {
my $to = $_;
if ($from ne $to) {
my $hashid = getTypeConvertHashId($from, $to);
my $sym = $funcs{$hashid};
print(" { AUDIO_$from, AUDIO_$to, $sym },\n");
}
}
}
print "#endif /* !NO_CONVERTERS */\n";
print(" { 0, 0, NULL }\n");
print "};\n\n\n";
}
sub getBiggerCtype {
my ($isfloat, $size) = @_;
if ($isfloat) {
if ($size == 32) {
return 'double';
}
die("bug in script.\n");
}
if ($size == 8) {
return 'Sint16';
} elsif ($size == 16) {
return 'Sint32'
} elsif ($size == 32) {
return 'Sint64'
}
die("bug in script.\n");
}
# These handle arbitrary resamples...44100Hz to 48000Hz, for example.
# Man, this code is skanky.
sub buildArbitraryResampleFunc {
# !!! FIXME: we do a lot of unnecessary and ugly casting in here, due to getSwapFunc().
my ($from, $channels, $upsample) = @_;
my ($fsigned, $ffloat, $fsize, $fendian, $fctype) = splittype($from);
my $bigger = getBiggerCtype($ffloat, $fsize);
my $interp = ($ffloat) ? '* 0.5' : '>> 1';
my $resample = ($upsample) ? 'Upsample' : 'Downsample';
my $hashid = getResamplerHashId($from, $channels, $upsample, 0);
my $sym = "SDL_${resample}_${from}_${channels}c";
$funcs{$hashid} = $sym;
$custom_converters++;
my $fudge = $fsize * $channels * 2; # !!! FIXME
my $eps_adjust = ($upsample) ? 'dstsize' : 'srcsize';
my $incr = '';
my $incr2 = '';
my $block_align = $channels * $fsize/8;
# !!! FIXME: DEBUG_CONVERT should report frequencies.
print <<EOF;
static void SDLCALL
${sym}(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
#if DEBUG_CONVERT
fprintf(stderr, "$resample arbitrary (x%f) AUDIO_${from}, ${channels} channels.\\n", cvt->rate_incr);
#endif
const int srcsize = cvt->len_cvt - $fudge;
const int dstsize = (int) (((double)(cvt->len_cvt/${block_align})) * cvt->rate_incr) * ${block_align};
register int eps = 0;
EOF
my $endcomparison = '!=';
# Upsampling (growing the buffer) needs to work backwards, since we
# overwrite the buffer as we go.
if ($upsample) {
$endcomparison = '>='; # dst > target
print <<EOF;
$fctype *dst = (($fctype *) (cvt->buf + dstsize)) - $channels;
const $fctype *src = (($fctype *) (cvt->buf + cvt->len_cvt)) - $channels;
const $fctype *target = ((const $fctype *) cvt->buf);
EOF
} else {
$endcomparison = '<'; # dst < target
print <<EOF;
$fctype *dst = ($fctype *) cvt->buf;
const $fctype *src = ($fctype *) cvt->buf;
const $fctype *target = (const $fctype *) (cvt->buf + dstsize);
EOF
}
for (my $i = 0; $i < $channels; $i++) {
my $idx = ($upsample) ? (($channels - $i) - 1) : $i;
my $val = getSwapFunc($fsize, $fsigned, $ffloat, $fendian, "src[$idx]");
print <<EOF;
$fctype sample${idx} = $val;
EOF
}
for (my $i = 0; $i < $channels; $i++) {
my $idx = ($upsample) ? (($channels - $i) - 1) : $i;
print <<EOF;
$fctype last_sample${idx} = sample${idx};
EOF
}
print <<EOF;
while (dst $endcomparison target) {
EOF
if ($upsample) {
for (my $i = 0; $i < $channels; $i++) {
# !!! FIXME: don't do this swap every write, just when the samples change.
my $idx = (($channels - $i) - 1);
my $val = getSwapFunc($fsize, $fsigned, $ffloat, $fendian, "sample${idx}");
print <<EOF;
dst[$idx] = $val;
EOF
}
$incr = ($channels == 1) ? 'dst--' : "dst -= $channels";
$incr2 = ($channels == 1) ? 'src--' : "src -= $channels";
print <<EOF;
$incr;
eps += srcsize;
if ((eps << 1) >= dstsize) {
$incr2;
EOF
} else { # downsample.
$incr = ($channels == 1) ? 'src++' : "src += $channels";
print <<EOF;
$incr;
eps += dstsize;
if ((eps << 1) >= srcsize) {
EOF
for (my $i = 0; $i < $channels; $i++) {
my $val = getSwapFunc($fsize, $fsigned, $ffloat, $fendian, "sample${i}");
print <<EOF;
dst[$i] = $val;
EOF
}
$incr = ($channels == 1) ? 'dst++' : "dst += $channels";
print <<EOF;
$incr;
EOF
}
for (my $i = 0; $i < $channels; $i++) {
my $idx = ($upsample) ? (($channels - $i) - 1) : $i;
my $swapped = getSwapFunc($fsize, $fsigned, $ffloat, $fendian, "src[$idx]");
print <<EOF;
sample${idx} = ($fctype) (((($bigger) $swapped) + (($bigger) last_sample${idx})) $interp);
EOF
}
for (my $i = 0; $i < $channels; $i++) {
my $idx = ($upsample) ? (($channels - $i) - 1) : $i;
print <<EOF;
last_sample${idx} = sample${idx};
EOF
}
print <<EOF;
eps -= $eps_adjust;
}
}
EOF
print <<EOF;
cvt->len_cvt = dstsize;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
EOF
}
# These handle clean resamples...doubling and quadrupling the sample rate, etc.
sub buildMultipleResampleFunc {
# !!! FIXME: we do a lot of unnecessary and ugly casting in here, due to getSwapFunc().
my ($from, $channels, $upsample, $multiple) = @_;
my ($fsigned, $ffloat, $fsize, $fendian, $fctype) = splittype($from);
my $bigger = getBiggerCtype($ffloat, $fsize);
my $interp = ($ffloat) ? '* 0.5' : '>> 1';
my $interp2 = ($ffloat) ? '* 0.25' : '>> 2';
my $mult3 = ($ffloat) ? '3.0' : '3';
my $lencvtop = ($upsample) ? '*' : '/';
my $resample = ($upsample) ? 'Upsample' : 'Downsample';
my $hashid = getResamplerHashId($from, $channels, $upsample, $multiple);
my $sym = "SDL_${resample}_${from}_${channels}c_x${multiple}";
$funcs{$hashid} = $sym;
$custom_converters++;
# !!! FIXME: DEBUG_CONVERT should report frequencies.
print <<EOF;
static void SDLCALL
${sym}(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
#if DEBUG_CONVERT
fprintf(stderr, "$resample (x${multiple}) AUDIO_${from}, ${channels} channels.\\n");
#endif
const int dstsize = cvt->len_cvt $lencvtop $multiple;
EOF
my $endcomparison = '!=';
# Upsampling (growing the buffer) needs to work backwards, since we
# overwrite the buffer as we go.
if ($upsample) {
$endcomparison = '>='; # dst > target
print <<EOF;
$fctype *dst = (($fctype *) (cvt->buf + dstsize)) - $channels * $multiple;
const $fctype *src = (($fctype *) (cvt->buf + cvt->len_cvt)) - $channels;
const $fctype *target = ((const $fctype *) cvt->buf);
EOF
} else {
$endcomparison = '<'; # dst < target
print <<EOF;
$fctype *dst = ($fctype *) cvt->buf;
const $fctype *src = ($fctype *) cvt->buf;
const $fctype *target = (const $fctype *) (cvt->buf + dstsize);
EOF
}
for (my $i = 0; $i < $channels; $i++) {
my $idx = ($upsample) ? (($channels - $i) - 1) : $i;
my $val = getSwapFunc($fsize, $fsigned, $ffloat, $fendian, "src[$idx]");
print <<EOF;
$bigger last_sample${idx} = ($bigger) $val;
EOF
}
print <<EOF;
while (dst $endcomparison target) {
EOF
for (my $i = 0; $i < $channels; $i++) {
my $idx = ($upsample) ? (($channels - $i) - 1) : $i;
my $val = getSwapFunc($fsize, $fsigned, $ffloat, $fendian, "src[$idx]");
print <<EOF;
const $bigger sample${idx} = ($bigger) $val;
EOF
}
my $incr = '';
if ($upsample) {
$incr = ($channels == 1) ? 'src--' : "src -= $channels";
} else {
my $amount = $channels * $multiple;
$incr = "src += $amount"; # can't ever be 1, so no "++" version.
}
print <<EOF;
$incr;
EOF
# !!! FIXME: This really begs for some Altivec or SSE, etc.
if ($upsample) {
if ($multiple == 2) {
for (my $i = $channels-1; $i >= 0; $i--) {
my $dsti = $i + $channels;
print <<EOF;
dst[$dsti] = ($fctype) ((sample${i} + last_sample${i}) $interp);
EOF
}
for (my $i = $channels-1; $i >= 0; $i--) {
my $dsti = $i;
print <<EOF;
dst[$dsti] = ($fctype) sample${i};
EOF
}
} elsif ($multiple == 4) {
for (my $i = $channels-1; $i >= 0; $i--) {
my $dsti = $i + ($channels * 3);
print <<EOF;
dst[$dsti] = ($fctype) ((sample${i} + ($mult3 * last_sample${i})) $interp2);
EOF
}
for (my $i = $channels-1; $i >= 0; $i--) {
my $dsti = $i + ($channels * 2);
print <<EOF;
dst[$dsti] = ($fctype) ((sample${i} + last_sample${i}) $interp);
EOF
}
for (my $i = $channels-1; $i >= 0; $i--) {
my $dsti = $i + ($channels * 1);
print <<EOF;
dst[$dsti] = ($fctype) ((($mult3 * sample${i}) + last_sample${i}) $interp2);
EOF
}
for (my $i = $channels-1; $i >= 0; $i--) {
my $dsti = $i + ($channels * 0);
print <<EOF;
dst[$dsti] = ($fctype) sample${i};
EOF
}
} else {
die('bug in program.'); # we only handle x2 and x4.
}
} else { # downsample.
if ($multiple == 2) {
for (my $i = 0; $i < $channels; $i++) {
print <<EOF;
dst[$i] = ($fctype) ((sample${i} + last_sample${i}) $interp);
EOF
}
} elsif ($multiple == 4) {
# !!! FIXME: interpolate all 4 samples?
for (my $i = 0; $i < $channels; $i++) {
print <<EOF;
dst[$i] = ($fctype) ((sample${i} + last_sample${i}) $interp);
EOF
}
} else {
die('bug in program.'); # we only handle x2 and x4.
}
}
for (my $i = 0; $i < $channels; $i++) {
my $idx = ($upsample) ? (($channels - $i) - 1) : $i;
print <<EOF;
last_sample${idx} = sample${idx};
EOF
}
if ($upsample) {
my $amount = $channels * $multiple;
$incr = "dst -= $amount"; # can't ever be 1, so no "--" version.
} else {
$incr = ($channels == 1) ? 'dst++' : "dst += $channels";
}
print <<EOF;
$incr;
}
cvt->len_cvt = dstsize;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
EOF
}
sub buildResamplers {
print "#if !NO_RESAMPLERS\n\n";
foreach (@audiotypes) {
my $from = $_;
foreach (@channels) {
my $channel = $_;
buildArbitraryResampleFunc($from, $channel, 1);
buildArbitraryResampleFunc($from, $channel, 0);
}
}
print "\n#if !LESS_RESAMPLERS\n\n";
foreach (@audiotypes) {
my $from = $_;
foreach (@channels) {
my $channel = $_;
for (my $multiple = 2; $multiple <= 4; $multiple += 2) {
buildMultipleResampleFunc($from, $channel, 1, $multiple);
buildMultipleResampleFunc($from, $channel, 0, $multiple);
}
}
}
print "#endif /* !LESS_RESAMPLERS */\n";
print "#endif /* !NO_RESAMPLERS */\n\n\n";
print "const SDL_AudioRateFilters sdl_audio_rate_filters[] =\n{\n";
print "#if !NO_RESAMPLERS\n";
foreach (@audiotypes) {
my $from = $_;
foreach (@channels) {
my $channel = $_;
for (my $upsample = 0; $upsample <= 1; $upsample++) {
my $hashid = getResamplerHashId($from, $channel, $upsample, 0);
my $sym = $funcs{$hashid};
print(" { AUDIO_$from, $channel, $upsample, 0, $sym },\n");
}
}
}
print "#if !LESS_RESAMPLERS\n";
foreach (@audiotypes) {
my $from = $_;
foreach (@channels) {
my $channel = $_;
for (my $multiple = 2; $multiple <= 4; $multiple += 2) {
for (my $upsample = 0; $upsample <= 1; $upsample++) {
my $hashid = getResamplerHashId($from, $channel, $upsample, $multiple);
my $sym = $funcs{$hashid};
print(" { AUDIO_$from, $channel, $upsample, $multiple, $sym },\n");
}
}
}
}
print "#endif /* !LESS_RESAMPLERS */\n";
print "#endif /* !NO_RESAMPLERS */\n";
print(" { 0, 0, 0, 0, NULL }\n");
print "};\n\n";
}
# mainline ...
outputHeader();
buildTypeConverters();
buildResamplers();
outputFooter();
exit 0;
# end of sdlgenaudiocvt.pl ...
libs/SDL2/src/audio/sndio/SDL_sndioaudio.c
+71 -4
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -33,6 +33,7 @@
#include <signal.h>
#endif
#include <poll.h>
#include <unistd.h>
#include "SDL_audio.h"
@@ -43,6 +44,14 @@
#include "SDL_loadso.h"
#endif
#ifndef INFTIM
#define INFTIM -1
#endif
#ifndef SIO_DEVANY
#define SIO_DEVANY "default"
#endif
static struct sio_hdl * (*SNDIO_sio_open)(const char *, unsigned int, int);
static void (*SNDIO_sio_close)(struct sio_hdl *);
static int (*SNDIO_sio_setpar)(struct sio_hdl *, struct sio_par *);
@@ -51,6 +60,10 @@ static int (*SNDIO_sio_start)(struct sio_hdl *);
static int (*SNDIO_sio_stop)(struct sio_hdl *);
static size_t (*SNDIO_sio_read)(struct sio_hdl *, void *, size_t);
static size_t (*SNDIO_sio_write)(struct sio_hdl *, const void *, size_t);
static int (*SNDIO_sio_nfds)(struct sio_hdl *);
static int (*SNDIO_sio_pollfd)(struct sio_hdl *, struct pollfd *, int);
static int (*SNDIO_sio_revents)(struct sio_hdl *, struct pollfd *);
static int (*SNDIO_sio_eof)(struct sio_hdl *);
static void (*SNDIO_sio_initpar)(struct sio_par *);
#ifdef SDL_AUDIO_DRIVER_SNDIO_DYNAMIC
@@ -87,6 +100,10 @@ load_sndio_syms(void)
SDL_SNDIO_SYM(sio_stop);
SDL_SNDIO_SYM(sio_read);
SDL_SNDIO_SYM(sio_write);
SDL_SNDIO_SYM(sio_nfds);
SDL_SNDIO_SYM(sio_pollfd);
SDL_SNDIO_SYM(sio_revents);
SDL_SNDIO_SYM(sio_eof);
SDL_SNDIO_SYM(sio_initpar);
return 0;
}
@@ -164,6 +181,41 @@ SNDIO_PlayDevice(_THIS)
#endif
}
static int
SNDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
size_t r;
int revents;
int nfds;
/* Emulate a blocking read */
r = SNDIO_sio_read(this->hidden->dev, buffer, buflen);
while (r == 0 && !SNDIO_sio_eof(this->hidden->dev)) {
if ((nfds = SNDIO_sio_pollfd(this->hidden->dev, this->hidden->pfd, POLLIN)) <= 0
|| poll(this->hidden->pfd, nfds, INFTIM) < 0) {
return -1;
}
revents = SNDIO_sio_revents(this->hidden->dev, this->hidden->pfd);
if (revents & POLLIN) {
r = SNDIO_sio_read(this->hidden->dev, buffer, buflen);
}
if (revents & POLLHUP) {
break;
}
}
return (int) r;
}
static void
SNDIO_FlushCapture(_THIS)
{
char buf[512];
while (SNDIO_sio_read(this->hidden->dev, buf, sizeof(buf)) != 0) {
/* do nothing */;
}
}
static Uint8 *
SNDIO_GetDeviceBuf(_THIS)
{
@@ -173,6 +225,9 @@ SNDIO_GetDeviceBuf(_THIS)
static void
SNDIO_CloseDevice(_THIS)
{
if ( this->hidden->pfd != NULL ) {
SDL_free(this->hidden->pfd);
}
if ( this->hidden->dev != NULL ) {
SNDIO_sio_stop(this->hidden->dev);
SNDIO_sio_close(this->hidden->dev);
@@ -197,11 +252,19 @@ SNDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
this->hidden->mixlen = this->spec.size;
/* !!! FIXME: SIO_DEVANY can be a specific device... */
if ((this->hidden->dev = SNDIO_sio_open(SIO_DEVANY, SIO_PLAY, 0)) == NULL) {
/* Capture devices must be non-blocking for SNDIO_FlushCapture */
if ((this->hidden->dev =
SNDIO_sio_open(devname != NULL ? devname : SIO_DEVANY,
iscapture ? SIO_REC : SIO_PLAY, iscapture)) == NULL) {
return SDL_SetError("sio_open() failed");
}
/* Allocate the pollfd array for capture devices */
if (iscapture && (this->hidden->pfd =
SDL_malloc(sizeof(struct pollfd) * SNDIO_sio_nfds(this->hidden->dev))) == NULL) {
return SDL_OutOfMemory();
}
SNDIO_sio_initpar(&par);
par.rate = this->spec.freq;
@@ -300,8 +363,12 @@ SNDIO_Init(SDL_AudioDriverImpl * impl)
impl->PlayDevice = SNDIO_PlayDevice;
impl->GetDeviceBuf = SNDIO_GetDeviceBuf;
impl->CloseDevice = SNDIO_CloseDevice;
impl->CaptureFromDevice = SNDIO_CaptureFromDevice;
impl->FlushCapture = SNDIO_FlushCapture;
impl->Deinitialize = SNDIO_Deinitialize;
impl->OnlyHasDefaultOutputDevice = 1; /* !!! FIXME: sndio can handle multiple devices. */
impl->AllowsArbitraryDeviceNames = 1;
impl->HasCaptureSupport = SDL_TRUE;
return 1; /* this audio target is available. */
}
libs/SDL2/src/audio/sndio/SDL_sndioaudio.h
+8 -4
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -20,9 +20,10 @@
*/
#include "../../SDL_internal.h"
#ifndef _SDL_sndioaudio_h
#define _SDL_sndioaudio_h
#ifndef SDL_sndioaudio_h_
#define SDL_sndioaudio_h_
#include <poll.h>
#include <sndio.h>
#include "../SDL_sysaudio.h"
@@ -38,8 +39,11 @@ struct SDL_PrivateAudioData
/* Raw mixing buffer */
Uint8 *mixbuf;
int mixlen;
/* Polling structures for non-blocking sndio devices */
struct pollfd *pfd;
};
#endif /* _SDL_sndioaudio_h */
#endif /* SDL_sndioaudio_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/sun/SDL_sunaudio.c
+8 -10
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -40,6 +40,7 @@
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../../core/unix/SDL_poll.h"
#include "../SDL_audio_c.h"
#include "../SDL_audiodev_c.h"
#include "SDL_sunaudio.h"
@@ -97,11 +98,7 @@ SUNAUDIO_WaitDevice(_THIS)
}
}
#else
fd_set fdset;
FD_ZERO(&fdset);
FD_SET(this->hidden->audio_fd, &fdset);
select(this->hidden->audio_fd + 1, NULL, &fdset, NULL, NULL);
SDL_IOReady(this->hidden->audio_fd, SDL_TRUE, -1);
#endif
}
@@ -193,6 +190,10 @@ SUNAUDIO_CloseDevice(_THIS)
static int
SUNAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
#ifdef AUDIO_SETINFO
int enc;
#endif
int desired_freq = 0;
const int flags = ((iscapture) ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT);
SDL_AudioFormat format = 0;
audio_info_t info;
@@ -220,10 +221,7 @@ SUNAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
return SDL_SetError("Couldn't open %s: %s", devname, strerror(errno));
}
#ifdef AUDIO_SETINFO
int enc;
#endif
int desired_freq = this->spec.freq;
desired_freq = this->spec.freq;
/* Determine the audio parameters from the AudioSpec */
switch (SDL_AUDIO_BITSIZE(this->spec.format)) {
libs/SDL2/src/audio/sun/SDL_sunaudio.h
+4 -4
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -20,8 +20,8 @@
*/
#include "../../SDL_internal.h"
#ifndef _SDL_sunaudio_h
#define _SDL_sunaudio_h
#ifndef SDL_sunaudio_h_
#define SDL_sunaudio_h_
#include "../SDL_sysaudio.h"
@@ -42,6 +42,6 @@ struct SDL_PrivateAudioData
int frequency; /* The audio frequency in KHz */
};
#endif /* _SDL_sunaudio_h */
#endif /* SDL_sunaudio_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/wasapi/SDL_wasapi.c
+964
View File
@@ -0,0 +1,964 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#include "../../SDL_internal.h"
#if SDL_AUDIO_DRIVER_WASAPI
#include "../../core/windows/SDL_windows.h"
#include "SDL_audio.h"
#include "SDL_timer.h"
#include "../SDL_audio_c.h"
#include "../SDL_sysaudio.h"
#include "SDL_assert.h"
#include "SDL_log.h"
#define COBJMACROS
#include <mmdeviceapi.h>
#include <audioclient.h>
#include "SDL_wasapi.h"
static const ERole SDL_WASAPI_role = eConsole; /* !!! FIXME: should this be eMultimedia? Should be a hint? */
/* This is global to the WASAPI target, to handle hotplug and default device lookup. */
static IMMDeviceEnumerator *enumerator = NULL;
/* these increment as default devices change. Opened default devices pick up changes in their threads. */
static SDL_atomic_t default_playback_generation;
static SDL_atomic_t default_capture_generation;
/* This is a list of device id strings we have inflight, so we have consistent pointers to the same device. */
typedef struct DevIdList
{
WCHAR *str;
struct DevIdList *next;
} DevIdList;
static DevIdList *deviceid_list = NULL;
/* handle to Avrt.dll--Vista and later!--for flagging the callback thread as "Pro Audio" (low latency). */
#ifndef __WINRT__
static HMODULE libavrt = NULL;
#endif
typedef HANDLE (WINAPI *pfnAvSetMmThreadCharacteristicsW)(LPWSTR,LPDWORD);
typedef BOOL (WINAPI *pfnAvRevertMmThreadCharacteristics)(HANDLE);
static pfnAvSetMmThreadCharacteristicsW pAvSetMmThreadCharacteristicsW = NULL;
static pfnAvRevertMmThreadCharacteristics pAvRevertMmThreadCharacteristics = NULL;
/* Some GUIDs we need to know without linking to libraries that aren't available before Vista. */
static const CLSID SDL_CLSID_MMDeviceEnumerator = { 0xbcde0395, 0xe52f, 0x467c, { 0x8e, 0x3d, 0xc4, 0x57, 0x92, 0x91, 0x69, 0x2e } };
static const IID SDL_IID_IMMDeviceEnumerator = { 0xa95664d2, 0x9614, 0x4f35, { 0xa7, 0x46, 0xde, 0x8d, 0xb6, 0x36, 0x17, 0xe6 } };
static const IID SDL_IID_IMMNotificationClient = { 0x7991eec9, 0x7e89, 0x4d85, { 0x83, 0x90, 0x6c, 0x70, 0x3c, 0xec, 0x60, 0xc0 } };
static const IID SDL_IID_IMMEndpoint = { 0x1be09788, 0x6894, 0x4089, { 0x85, 0x86, 0x9a, 0x2a, 0x6c, 0x26, 0x5a, 0xc5 } };
static const IID SDL_IID_IAudioClient = { 0x1cb9ad4c, 0xdbfa, 0x4c32, { 0xb1, 0x78, 0xc2, 0xf5, 0x68, 0xa7, 0x03, 0xb2 } };
static const IID SDL_IID_IAudioRenderClient = { 0xf294acfc, 0x3146, 0x4483, { 0xa7, 0xbf, 0xad, 0xdc, 0xa7, 0xc2, 0x60, 0xe2 } };
static const IID SDL_IID_IAudioCaptureClient = { 0xc8adbd64, 0xe71e, 0x48a0, { 0xa4, 0xde, 0x18, 0x5c, 0x39, 0x5c, 0xd3, 0x17 } };
static const GUID SDL_KSDATAFORMAT_SUBTYPE_PCM = { 0x00000001, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 } };
static const GUID SDL_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT = { 0x00000003, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 } };
static const PROPERTYKEY SDL_PKEY_Device_FriendlyName = { { 0xa45c254e, 0xdf1c, 0x4efd, { 0x80, 0x20, 0x67, 0xd1, 0x46, 0xa8, 0x50, 0xe0, } }, 14 };
/* PropVariantInit() is an inline function/macro in PropIdl.h that calls the C runtime's memset() directly. Use ours instead, to avoid dependency. */
#ifdef PropVariantInit
#undef PropVariantInit
#endif
#define PropVariantInit(p) SDL_zerop(p)
static void AddWASAPIDevice(const SDL_bool iscapture, IMMDevice *device, LPCWSTR devid);
static void RemoveWASAPIDevice(const SDL_bool iscapture, LPCWSTR devid);
/* We need a COM subclass of IMMNotificationClient for hotplug support, which is
easy in C++, but we have to tapdance more to make work in C.
Thanks to this page for coaching on how to make this work:
https://www.codeproject.com/Articles/13601/COM-in-plain-C */
typedef struct SDLMMNotificationClient
{
const IMMNotificationClientVtbl *lpVtbl;
SDL_atomic_t refcount;
} SDLMMNotificationClient;
static HRESULT STDMETHODCALLTYPE
SDLMMNotificationClient_QueryInterface(IMMNotificationClient *this, REFIID iid, void **ppv)
{
if ((WIN_IsEqualIID(iid, &IID_IUnknown)) || (WIN_IsEqualIID(iid, &SDL_IID_IMMNotificationClient)))
{
*ppv = this;
this->lpVtbl->AddRef(this);
return S_OK;
}
*ppv = NULL;
return E_NOINTERFACE;
}
static ULONG STDMETHODCALLTYPE
SDLMMNotificationClient_AddRef(IMMNotificationClient *ithis)
{
SDLMMNotificationClient *this = (SDLMMNotificationClient *) ithis;
return (ULONG) (SDL_AtomicIncRef(&this->refcount) + 1);
}
static ULONG STDMETHODCALLTYPE
SDLMMNotificationClient_Release(IMMNotificationClient *ithis)
{
/* this is a static object; we don't ever free it. */
SDLMMNotificationClient *this = (SDLMMNotificationClient *) ithis;
const ULONG retval = SDL_AtomicDecRef(&this->refcount);
if (retval == 0) {
SDL_AtomicSet(&this->refcount, 0); /* uhh... */
return 0;
}
return retval - 1;
}
/* These are the entry points called when WASAPI device endpoints change. */
static HRESULT STDMETHODCALLTYPE
SDLMMNotificationClient_OnDefaultDeviceChanged(IMMNotificationClient *ithis, EDataFlow flow, ERole role, LPCWSTR pwstrDeviceId)
{
if (role != SDL_WASAPI_role) {
return S_OK; /* ignore it. */
}
/* Increment the "generation," so opened devices will pick this up in their threads. */
switch (flow) {
case eRender:
SDL_AtomicAdd(&default_playback_generation, 1);
break;
case eCapture:
SDL_AtomicAdd(&default_capture_generation, 1);
break;
case eAll:
SDL_AtomicAdd(&default_playback_generation, 1);
SDL_AtomicAdd(&default_capture_generation, 1);
break;
default:
SDL_assert(!"uhoh, unexpected OnDefaultDeviceChange flow!");
break;
}
return S_OK;
}
static HRESULT STDMETHODCALLTYPE
SDLMMNotificationClient_OnDeviceAdded(IMMNotificationClient *ithis, LPCWSTR pwstrDeviceId)
{
/* we ignore this; devices added here then progress to ACTIVE, if appropriate, in
OnDeviceStateChange, making that a better place to deal with device adds. More
importantly: the first time you plug in a USB audio device, this callback will
fire, but when you unplug it, it isn't removed (it's state changes to NOTPRESENT).
Plugging it back in won't fire this callback again. */
return S_OK;
}
static HRESULT STDMETHODCALLTYPE
SDLMMNotificationClient_OnDeviceRemoved(IMMNotificationClient *ithis, LPCWSTR pwstrDeviceId)
{
/* See notes in OnDeviceAdded handler about why we ignore this. */
return S_OK;
}
static HRESULT STDMETHODCALLTYPE
SDLMMNotificationClient_OnDeviceStateChanged(IMMNotificationClient *ithis, LPCWSTR pwstrDeviceId, DWORD dwNewState)
{
IMMDevice *device = NULL;
if (SUCCEEDED(IMMDeviceEnumerator_GetDevice(enumerator, pwstrDeviceId, &device))) {
IMMEndpoint *endpoint = NULL;
if (SUCCEEDED(IMMDevice_QueryInterface(device, &SDL_IID_IMMEndpoint, (void **) &endpoint))) {
EDataFlow flow;
if (SUCCEEDED(IMMEndpoint_GetDataFlow(endpoint, &flow))) {
const SDL_bool iscapture = (flow == eCapture);
if (dwNewState == DEVICE_STATE_ACTIVE) {
AddWASAPIDevice(iscapture, device, pwstrDeviceId);
} else {
RemoveWASAPIDevice(iscapture, pwstrDeviceId);
}
}
IMMEndpoint_Release(endpoint);
}
IMMDevice_Release(device);
}
return S_OK;
}
static HRESULT STDMETHODCALLTYPE
SDLMMNotificationClient_OnPropertyValueChanged(IMMNotificationClient *this, LPCWSTR pwstrDeviceId, const PROPERTYKEY key)
{
return S_OK; /* we don't care about these. */
}
static const IMMNotificationClientVtbl notification_client_vtbl = {
SDLMMNotificationClient_QueryInterface,
SDLMMNotificationClient_AddRef,
SDLMMNotificationClient_Release,
SDLMMNotificationClient_OnDeviceStateChanged,
SDLMMNotificationClient_OnDeviceAdded,
SDLMMNotificationClient_OnDeviceRemoved,
SDLMMNotificationClient_OnDefaultDeviceChanged,
SDLMMNotificationClient_OnPropertyValueChanged
};
static SDLMMNotificationClient notification_client = { &notification_client_vtbl, { 1 } };
static SDL_bool
WStrEqual(const WCHAR *a, const WCHAR *b)
{
while (*a) {
if (*a != *b) {
return SDL_FALSE;
}
a++;
b++;
}
return *b == 0;
}
static WCHAR *
WStrDupe(const WCHAR *wstr)
{
const int len = (lstrlenW(wstr) + 1) * sizeof (WCHAR);
WCHAR *retval = (WCHAR *) SDL_malloc(len);
if (retval) {
SDL_memcpy(retval, wstr, len);
}
return retval;
}
static void
RemoveWASAPIDevice(const SDL_bool iscapture, LPCWSTR devid)
{
DevIdList *i;
DevIdList *next;
DevIdList *prev = NULL;
for (i = deviceid_list; i; i = next) {
next = i->next;
if (WStrEqual(i->str, devid)) {
if (prev) {
prev->next = next;
} else {
deviceid_list = next;
}
SDL_RemoveAudioDevice(iscapture, i->str);
SDL_free(i->str);
SDL_free(i);
}
prev = i;
}
}
static void
AddWASAPIDevice(const SDL_bool iscapture, IMMDevice *device, LPCWSTR devid)
{
IPropertyStore *props = NULL;
char *utf8dev = NULL;
DevIdList *devidlist;
PROPVARIANT var;
/* You can have multiple endpoints on a device that are mutually exclusive ("Speakers" vs "Line Out" or whatever).
In a perfect world, things that are unplugged won't be in this collection. The only gotcha is probably for
phones and tablets, where you might have an internal speaker and a headphone jack and expect both to be
available and switch automatically. (!!! FIXME...?) */
/* PKEY_Device_FriendlyName gives you "Speakers (SoundBlaster Pro)" which drives me nuts. I'd rather it be
"SoundBlaster Pro (Speakers)" but I guess that's developers vs users. Windows uses the FriendlyName in
its own UIs, like Volume Control, etc. */
/* see if we already have this one. */
for (devidlist = deviceid_list; devidlist; devidlist = devidlist->next) {
if (WStrEqual(devidlist->str, devid)) {
return; /* we already have this. */
}
}
devidlist = (DevIdList *) SDL_malloc(sizeof (*devidlist));
if (!devidlist) {
return; /* oh well. */
}
devid = WStrDupe(devid);
if (!devid) {
SDL_free(devidlist);
return; /* oh well. */
}
devidlist->str = (WCHAR *) devid;
devidlist->next = deviceid_list;
deviceid_list = devidlist;
if (SUCCEEDED(IMMDevice_OpenPropertyStore(device, STGM_READ, &props))) {
PropVariantInit(&var);
if (SUCCEEDED(IPropertyStore_GetValue(props, &SDL_PKEY_Device_FriendlyName, &var))) {
utf8dev = WIN_StringToUTF8(var.pwszVal);
if (utf8dev) {
SDL_AddAudioDevice(iscapture, utf8dev, (void *) devid);
SDL_free(utf8dev);
}
}
PropVariantClear(&var);
IPropertyStore_Release(props);
}
}
static void
EnumerateEndpoints(const SDL_bool iscapture)
{
IMMDeviceCollection *collection = NULL;
UINT i, total;
/* Note that WASAPI separates "adapter devices" from "audio endpoint devices"
...one adapter device ("SoundBlaster Pro") might have multiple endpoint devices ("Speakers", "Line-Out"). */
if (FAILED(IMMDeviceEnumerator_EnumAudioEndpoints(enumerator, iscapture ? eCapture : eRender, DEVICE_STATE_ACTIVE, &collection))) {
return;
}
if (FAILED(IMMDeviceCollection_GetCount(collection, &total))) {
IMMDeviceCollection_Release(collection);
return;
}
for (i = 0; i < total; i++) {
IMMDevice *device = NULL;
if (SUCCEEDED(IMMDeviceCollection_Item(collection, i, &device))) {
LPWSTR devid = NULL;
if (SUCCEEDED(IMMDevice_GetId(device, &devid))) {
AddWASAPIDevice(iscapture, device, devid);
CoTaskMemFree(devid);
}
IMMDevice_Release(device);
}
}
IMMDeviceCollection_Release(collection);
}
static void
WASAPI_DetectDevices(void)
{
EnumerateEndpoints(SDL_FALSE); /* playback */
EnumerateEndpoints(SDL_TRUE); /* capture */
/* if this fails, we just won't get hotplug events. Carry on anyhow. */
IMMDeviceEnumerator_RegisterEndpointNotificationCallback(enumerator, (IMMNotificationClient *) &notification_client);
}
static int
WASAPI_GetPendingBytes(_THIS)
{
UINT32 frames = 0;
/* it's okay to fail here; we'll deal with failures in the audio thread. */
/* FIXME: need a lock around checking this->hidden->client */
if (!this->hidden->client || FAILED(IAudioClient_GetCurrentPadding(this->hidden->client, &frames))) {
return 0; /* oh well. */
}
return ((int) frames) * this->hidden->framesize;
}
static SDL_INLINE SDL_bool
WasapiFailed(_THIS, const HRESULT err)
{
if (err == S_OK) {
return SDL_FALSE;
}
if (err == AUDCLNT_E_DEVICE_INVALIDATED) {
this->hidden->device_lost = SDL_TRUE;
} else if (SDL_AtomicGet(&this->enabled)) {
IAudioClient_Stop(this->hidden->client);
SDL_OpenedAudioDeviceDisconnected(this);
SDL_assert(!SDL_AtomicGet(&this->enabled));
}
return SDL_TRUE;
}
static int PrepWasapiDevice(_THIS, const int iscapture, IMMDevice *device);
static void ReleaseWasapiDevice(_THIS);
static SDL_bool
RecoverWasapiDevice(_THIS)
{
const SDL_AudioSpec oldspec = this->spec;
IMMDevice *device = NULL;
HRESULT ret = S_OK;
if (this->hidden->default_device_generation) {
const EDataFlow dataflow = this->iscapture ? eCapture : eRender;
ReleaseWasapiDevice(this); /* dump the lost device's handles. */
this->hidden->default_device_generation = SDL_AtomicGet(this->iscapture ? &default_capture_generation : &default_playback_generation);
ret = IMMDeviceEnumerator_GetDefaultAudioEndpoint(enumerator, dataflow, SDL_WASAPI_role, &device);
if (FAILED(ret)) {
return SDL_FALSE; /* can't find a new default device! */
}
} else {
device = this->hidden->device;
this->hidden->device = NULL; /* don't release this in ReleaseWasapiDevice(). */
ReleaseWasapiDevice(this); /* dump the lost device's handles. */
}
SDL_assert(device != NULL);
/* this can fail for lots of reasons, but the most likely is we had a
non-default device that was disconnected, so we can't recover. Default
devices try to reinitialize whatever the new default is, so it's more
likely to carry on here, but this handles a non-default device that
simply had its format changed in the Windows Control Panel. */
if (PrepWasapiDevice(this, this->iscapture, device) == -1) {
return SDL_FALSE;
}
/* Since WASAPI requires us to handle all audio conversion, and our
device format might have changed, we might have to add/remove/change
the audio stream that the higher level uses to convert data, so
SDL keeps firing the callback as if nothing happened here. */
if ( (this->callbackspec.channels == this->spec.channels) &&
(this->callbackspec.format == this->spec.format) &&
(this->callbackspec.freq == this->spec.freq) &&
(this->callbackspec.samples == this->spec.samples) ) {
/* no need to buffer/convert in an AudioStream! */
SDL_FreeAudioStream(this->stream);
this->stream = NULL;
} else if ( (oldspec.channels == this->spec.channels) &&
(oldspec.format == this->spec.format) &&
(oldspec.freq == this->spec.freq) ) {
/* The existing audio stream is okay to keep using. */
} else {
/* replace the audiostream for new format */
SDL_FreeAudioStream(this->stream);
if (this->iscapture) {
this->stream = SDL_NewAudioStream(this->spec.format,
this->spec.channels, this->spec.freq,
this->callbackspec.format,
this->callbackspec.channels,
this->callbackspec.freq);
} else {
this->stream = SDL_NewAudioStream(this->callbackspec.format,
this->callbackspec.channels,
this->callbackspec.freq, this->spec.format,
this->spec.channels, this->spec.freq);
}
if (!this->stream) {
return SDL_FALSE;
}
}
/* make sure our scratch buffer can cover the new device spec. */
if (this->spec.size > this->work_buffer_len) {
Uint8 *ptr = (Uint8 *) SDL_realloc(this->work_buffer, this->spec.size);
if (ptr == NULL) {
SDL_OutOfMemory();
return SDL_FALSE;
}
this->work_buffer = ptr;
this->work_buffer_len = this->spec.size;
}
this->hidden->device_lost = SDL_FALSE;
return SDL_TRUE; /* okay, carry on with new device details! */
}
static SDL_bool
RecoverWasapiIfLost(_THIS)
{
const int generation = this->hidden->default_device_generation;
SDL_bool lost = this->hidden->device_lost;
if (!SDL_AtomicGet(&this->enabled)) {
return SDL_FALSE; /* already failed. */
}
if (!lost && (generation > 0)) { /* is a default device? */
const int newgen = SDL_AtomicGet(this->iscapture ? &default_capture_generation : &default_playback_generation);
if (generation != newgen) { /* the desired default device was changed, jump over to it. */
lost = SDL_TRUE;
}
}
return lost ? RecoverWasapiDevice(this) : SDL_TRUE;
}
static Uint8 *
WASAPI_GetDeviceBuf(_THIS)
{
/* get an endpoint buffer from WASAPI. */
BYTE *buffer = NULL;
while (RecoverWasapiIfLost(this)) {
if (!WasapiFailed(this, IAudioRenderClient_GetBuffer(this->hidden->render, this->spec.samples, &buffer))) {
return (Uint8 *) buffer;
}
SDL_assert(buffer == NULL);
}
return (Uint8 *) buffer;
}
static void
WASAPI_PlayDevice(_THIS)
{
/* WasapiFailed() will mark the device for reacquisition or removal elsewhere. */
WasapiFailed(this, IAudioRenderClient_ReleaseBuffer(this->hidden->render, this->spec.samples, 0));
}
static void
WASAPI_WaitDevice(_THIS)
{
const UINT32 maxpadding = this->spec.samples;
while (RecoverWasapiIfLost(this)) {
UINT32 padding = 0;
if (!WasapiFailed(this, IAudioClient_GetCurrentPadding(this->hidden->client, &padding))) {
if (padding <= maxpadding) {
break;
}
/* Sleep long enough for half the buffer to be free. */
SDL_Delay(((padding - maxpadding) * 1000) / this->spec.freq);
}
}
}
static int
WASAPI_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
SDL_AudioStream *stream = this->hidden->capturestream;
const int avail = SDL_AudioStreamAvailable(stream);
if (avail > 0) {
const int cpy = SDL_min(buflen, avail);
SDL_AudioStreamGet(stream, buffer, cpy);
return cpy;
}
while (RecoverWasapiIfLost(this)) {
HRESULT ret;
BYTE *ptr = NULL;
UINT32 frames = 0;
DWORD flags = 0;
ret = IAudioCaptureClient_GetBuffer(this->hidden->capture, &ptr, &frames, &flags, NULL, NULL);
if (ret != AUDCLNT_S_BUFFER_EMPTY) {
WasapiFailed(this, ret); /* mark device lost/failed if necessary. */
}
if ((ret == AUDCLNT_S_BUFFER_EMPTY) || !frames) {
WASAPI_WaitDevice(this);
} else if (ret == S_OK) {
const int total = ((int) frames) * this->hidden->framesize;
const int cpy = SDL_min(buflen, total);
const int leftover = total - cpy;
const SDL_bool silent = (flags & AUDCLNT_BUFFERFLAGS_SILENT) ? SDL_TRUE : SDL_FALSE;
if (silent) {
SDL_memset(buffer, this->spec.silence, cpy);
} else {
SDL_memcpy(buffer, ptr, cpy);
}
if (leftover > 0) {
ptr += cpy;
if (silent) {
SDL_memset(ptr, this->spec.silence, leftover); /* I guess this is safe? */
}
if (SDL_AudioStreamPut(stream, ptr, leftover) == -1) {
return -1; /* uhoh, out of memory, etc. Kill device. :( */
}
}
ret = IAudioCaptureClient_ReleaseBuffer(this->hidden->capture, frames);
WasapiFailed(this, ret); /* mark device lost/failed if necessary. */
return cpy;
}
}
return -1; /* unrecoverable error. */
}
static void
WASAPI_FlushCapture(_THIS)
{
BYTE *ptr = NULL;
UINT32 frames = 0;
DWORD flags = 0;
/* just read until we stop getting packets, throwing them away. */
while (SDL_TRUE) {
const HRESULT ret = IAudioCaptureClient_GetBuffer(this->hidden->capture, &ptr, &frames, &flags, NULL, NULL);
if (ret == AUDCLNT_S_BUFFER_EMPTY) {
break; /* no more buffered data; we're done. */
} else if (WasapiFailed(this, ret)) {
break; /* failed for some other reason, abort. */
} else if (WasapiFailed(this, IAudioCaptureClient_ReleaseBuffer(this->hidden->capture, frames))) {
break; /* something broke. */
}
}
SDL_AudioStreamClear(this->hidden->capturestream);
}
static void
ReleaseWasapiDevice(_THIS)
{
if (this->hidden->client) {
IAudioClient_Stop(this->hidden->client);
this->hidden->client = NULL;
}
if (this->hidden->render) {
IAudioRenderClient_Release(this->hidden->render);
this->hidden->render = NULL;
}
if (this->hidden->capture) {
IAudioCaptureClient_Release(this->hidden->capture);
this->hidden->capture = NULL;
}
if (this->hidden->waveformat) {
CoTaskMemFree(this->hidden->waveformat);
this->hidden->waveformat = NULL;
}
if (this->hidden->device) {
IMMDevice_Release(this->hidden->device);
this->hidden->device = NULL;
}
if (this->hidden->capturestream) {
SDL_FreeAudioStream(this->hidden->capturestream);
this->hidden->capturestream = NULL;
}
}
static void
WASAPI_CloseDevice(_THIS)
{
/* don't touch this->hidden->task in here; it has to be reverted from
our callback thread. We do that in WASAPI_ThreadDeinit().
(likewise for this->hidden->coinitialized). */
ReleaseWasapiDevice(this);
SDL_free(this->hidden);
}
static int
PrepWasapiDevice(_THIS, const int iscapture, IMMDevice *device)
{
/* !!! FIXME: we could request an exclusive mode stream, which is lower latency;
!!! it will write into the kernel's audio buffer directly instead of
!!! shared memory that a user-mode mixer then writes to the kernel with
!!! everything else. Doing this means any other sound using this device will
!!! stop playing, including the user's MP3 player and system notification
!!! sounds. You'd probably need to release the device when the app isn't in
!!! the foreground, to be a good citizen of the system. It's doable, but it's
!!! more work and causes some annoyances, and I don't know what the latency
!!! wins actually look like. Maybe add a hint to force exclusive mode at
!!! some point. To be sure, defaulting to shared mode is the right thing to
!!! do in any case. */
const AUDCLNT_SHAREMODE sharemode = AUDCLNT_SHAREMODE_SHARED;
UINT32 bufsize = 0; /* this is in sample frames, not samples, not bytes. */
REFERENCE_TIME duration = 0;
IAudioClient *client = NULL;
IAudioRenderClient *render = NULL;
IAudioCaptureClient *capture = NULL;
WAVEFORMATEX *waveformat = NULL;
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
SDL_AudioFormat wasapi_format = 0;
SDL_bool valid_format = SDL_FALSE;
HRESULT ret = S_OK;
this->hidden->device = device;
ret = IMMDevice_Activate(device, &SDL_IID_IAudioClient, CLSCTX_ALL, NULL, (void **) &client);
if (FAILED(ret)) {
return WIN_SetErrorFromHRESULT("WASAPI can't activate audio endpoint", ret);
}
SDL_assert(client != NULL);
this->hidden->client = client;
ret = IAudioClient_GetMixFormat(client, &waveformat);
if (FAILED(ret)) {
return WIN_SetErrorFromHRESULT("WASAPI can't determine mix format", ret);
}
SDL_assert(waveformat != NULL);
this->hidden->waveformat = waveformat;
/* WASAPI will not do any conversion on our behalf. Force channels and sample rate. */
this->spec.channels = (Uint8) waveformat->nChannels;
this->spec.freq = waveformat->nSamplesPerSec;
/* Make sure we have a valid format that we can convert to whatever WASAPI wants. */
if ((waveformat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT) && (waveformat->wBitsPerSample == 32)) {
wasapi_format = AUDIO_F32SYS;
} else if ((waveformat->wFormatTag == WAVE_FORMAT_PCM) && (waveformat->wBitsPerSample == 16)) {
wasapi_format = AUDIO_S16SYS;
} else if ((waveformat->wFormatTag == WAVE_FORMAT_PCM) && (waveformat->wBitsPerSample == 32)) {
wasapi_format = AUDIO_S32SYS;
} else if (waveformat->wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
const WAVEFORMATEXTENSIBLE *ext = (const WAVEFORMATEXTENSIBLE *) waveformat;
if ((SDL_memcmp(&ext->SubFormat, &SDL_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, sizeof (GUID)) == 0) && (waveformat->wBitsPerSample == 32)) {
wasapi_format = AUDIO_F32SYS;
} else if ((SDL_memcmp(&ext->SubFormat, &SDL_KSDATAFORMAT_SUBTYPE_PCM, sizeof (GUID)) == 0) && (waveformat->wBitsPerSample == 16)) {
wasapi_format = AUDIO_S16SYS;
} else if ((SDL_memcmp(&ext->SubFormat, &SDL_KSDATAFORMAT_SUBTYPE_PCM, sizeof (GUID)) == 0) && (waveformat->wBitsPerSample == 32)) {
wasapi_format = AUDIO_S32SYS;
}
}
while ((!valid_format) && (test_format)) {
if (test_format == wasapi_format) {
this->spec.format = test_format;
valid_format = SDL_TRUE;
break;
}
test_format = SDL_NextAudioFormat();
}
if (!valid_format) {
return SDL_SetError("WASAPI: Unsupported audio format");
}
ret = IAudioClient_GetDevicePeriod(client, NULL, &duration);
if (FAILED(ret)) {
return WIN_SetErrorFromHRESULT("WASAPI can't determine minimum device period", ret);
}
ret = IAudioClient_Initialize(client, sharemode, 0, duration, sharemode == AUDCLNT_SHAREMODE_SHARED ? 0 : duration, waveformat, NULL);
if (FAILED(ret)) {
return WIN_SetErrorFromHRESULT("WASAPI can't initialize audio client", ret);
}
ret = IAudioClient_GetBufferSize(client, &bufsize);
if (FAILED(ret)) {
return WIN_SetErrorFromHRESULT("WASAPI can't determine buffer size", ret);
}
this->spec.samples = (Uint16) bufsize;
if (!iscapture) {
this->spec.samples /= 2; /* fill half of the DMA buffer on each run. */
}
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&this->spec);
this->hidden->framesize = (SDL_AUDIO_BITSIZE(this->spec.format) / 8) * this->spec.channels;
if (iscapture) {
this->hidden->capturestream = SDL_NewAudioStream(this->spec.format, this->spec.channels, this->spec.freq, this->spec.format, this->spec.channels, this->spec.freq);
if (!this->hidden->capturestream) {
return -1; /* already set SDL_Error */
}
ret = IAudioClient_GetService(client, &SDL_IID_IAudioCaptureClient, (void**) &capture);
if (FAILED(ret)) {
return WIN_SetErrorFromHRESULT("WASAPI can't get capture client service", ret);
}
SDL_assert(capture != NULL);
this->hidden->capture = capture;
ret = IAudioClient_Start(client);
if (FAILED(ret)) {
return WIN_SetErrorFromHRESULT("WASAPI can't start capture", ret);
}
WASAPI_FlushCapture(this); /* MSDN says you should flush capture endpoint right after startup. */
} else {
ret = IAudioClient_GetService(client, &SDL_IID_IAudioRenderClient, (void**) &render);
if (FAILED(ret)) {
return WIN_SetErrorFromHRESULT("WASAPI can't get render client service", ret);
}
SDL_assert(render != NULL);
this->hidden->render = render;
ret = IAudioClient_Start(client);
if (FAILED(ret)) {
return WIN_SetErrorFromHRESULT("WASAPI can't start playback", ret);
}
}
return 0; /* good to go. */
}
static int
WASAPI_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
const SDL_bool is_default_device = (handle == NULL);
IMMDevice *device = NULL;
HRESULT ret = S_OK;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_zerop(this->hidden);
if (is_default_device) {
const EDataFlow dataflow = iscapture ? eCapture : eRender;
this->hidden->default_device_generation = SDL_AtomicGet(iscapture ? &default_capture_generation : &default_playback_generation);
ret = IMMDeviceEnumerator_GetDefaultAudioEndpoint(enumerator, dataflow, SDL_WASAPI_role, &device);
} else {
ret = IMMDeviceEnumerator_GetDevice(enumerator, (LPCWSTR) handle, &device);
}
if (FAILED(ret)) {
return WIN_SetErrorFromHRESULT("WASAPI can't find requested audio endpoint", ret);
}
SDL_assert(device != NULL);
return PrepWasapiDevice(this, iscapture, device);
}
static void
WASAPI_ThreadInit(_THIS)
{
/* this thread uses COM. */
if (SUCCEEDED(WIN_CoInitialize())) { /* can't report errors, hope it worked! */
this->hidden->coinitialized = SDL_TRUE;
}
/* Set this thread to very high "Pro Audio" priority. */
if (pAvSetMmThreadCharacteristicsW) {
DWORD idx = 0;
this->hidden->task = pAvSetMmThreadCharacteristicsW(TEXT("Pro Audio"), &idx);
}
}
static void
WASAPI_ThreadDeinit(_THIS)
{
/* Set this thread to very high "Pro Audio" priority. */
if (this->hidden->task && pAvRevertMmThreadCharacteristics) {
pAvRevertMmThreadCharacteristics(this->hidden->task);
this->hidden->task = NULL;
}
if (this->hidden->coinitialized) {
WIN_CoUninitialize();
}
}
static void
WASAPI_Deinitialize(void)
{
DevIdList *devidlist;
DevIdList *next;
if (enumerator) {
IMMDeviceEnumerator_UnregisterEndpointNotificationCallback(enumerator, (IMMNotificationClient *) &notification_client);
IMMDeviceEnumerator_Release(enumerator);
enumerator = NULL;
}
#ifndef __WINRT__
if (libavrt) {
FreeLibrary(libavrt);
libavrt = NULL;
}
#endif
pAvSetMmThreadCharacteristicsW = NULL;
pAvRevertMmThreadCharacteristics = NULL;
for (devidlist = deviceid_list; devidlist; devidlist = next) {
next = devidlist->next;
SDL_free(devidlist->str);
SDL_free(devidlist);
}
deviceid_list = NULL;
WIN_CoUninitialize();
}
static int
WASAPI_Init(SDL_AudioDriverImpl * impl)
{
HRESULT ret;
/* just skip the discussion with COM here. */
if (!WIN_IsWindowsVistaOrGreater()) {
SDL_SetError("WASAPI support requires Windows Vista or later");
return 0;
}
SDL_AtomicSet(&default_playback_generation, 1);
SDL_AtomicSet(&default_capture_generation, 1);
if (FAILED(WIN_CoInitialize())) {
SDL_SetError("WASAPI: CoInitialize() failed");
return 0;
}
ret = CoCreateInstance(&SDL_CLSID_MMDeviceEnumerator, NULL, CLSCTX_INPROC_SERVER, &SDL_IID_IMMDeviceEnumerator, (LPVOID) &enumerator);
if (FAILED(ret)) {
WIN_CoUninitialize();
WIN_SetErrorFromHRESULT("WASAPI CoCreateInstance(MMDeviceEnumerator)", ret);
return 0; /* oh well. */
}
#ifdef __WINRT__
pAvSetMmThreadCharacteristicsW = AvSetMmThreadCharacteristicsW;
pAvRevertMmThreadCharacteristics = AvRevertMmThreadCharacteristics;
#else
libavrt = LoadLibraryW(L"avrt.dll"); /* this library is available in Vista and later. No WinXP, so have to LoadLibrary to use it for now! */
if (libavrt) {
pAvSetMmThreadCharacteristicsW = (pfnAvSetMmThreadCharacteristicsW) GetProcAddress(libavrt, "AvSetMmThreadCharacteristicsW");
pAvRevertMmThreadCharacteristics = (pfnAvRevertMmThreadCharacteristics) GetProcAddress(libavrt, "AvRevertMmThreadCharacteristics");
}
#endif
/* Set the function pointers */
impl->DetectDevices = WASAPI_DetectDevices;
impl->ThreadInit = WASAPI_ThreadInit;
impl->ThreadDeinit = WASAPI_ThreadDeinit;
impl->OpenDevice = WASAPI_OpenDevice;
impl->PlayDevice = WASAPI_PlayDevice;
impl->WaitDevice = WASAPI_WaitDevice;
impl->GetPendingBytes = WASAPI_GetPendingBytes;
impl->GetDeviceBuf = WASAPI_GetDeviceBuf;
impl->CaptureFromDevice = WASAPI_CaptureFromDevice;
impl->FlushCapture = WASAPI_FlushCapture;
impl->CloseDevice = WASAPI_CloseDevice;
impl->Deinitialize = WASAPI_Deinitialize;
impl->HasCaptureSupport = 1;
return 1; /* this audio target is available. */
}
AudioBootStrap WASAPI_bootstrap = {
"wasapi", "WASAPI", WASAPI_Init, 0
};
#endif /* SDL_AUDIO_DRIVER_WASAPI */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/wasapi/SDL_wasapi.h
+48
View File
@@ -0,0 +1,48 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#include "../../SDL_internal.h"
#ifndef SDL_wasapi_h_
#define SDL_wasapi_h_
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
IMMDevice *device;
WAVEFORMATEX *waveformat;
IAudioClient *client;
IAudioRenderClient *render;
IAudioCaptureClient *capture;
SDL_AudioStream *capturestream;
HANDLE task;
SDL_bool coinitialized;
int framesize;
int default_device_generation;
SDL_bool device_lost;
};
#endif /* SDL_wasapi_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/winmm/SDL_winmm.c
+35 -1
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -33,6 +33,40 @@
#include "../SDL_audio_c.h"
#include "SDL_winmm.h"
/* MinGW32 mmsystem.h doesn't include these structures */
#if defined(__MINGW32__) && defined(_MMSYSTEM_H)
typedef struct tagWAVEINCAPS2W
{
WORD wMid;
WORD wPid;
MMVERSION vDriverVersion;
WCHAR szPname[MAXPNAMELEN];
DWORD dwFormats;
WORD wChannels;
WORD wReserved1;
GUID ManufacturerGuid;
GUID ProductGuid;
GUID NameGuid;
} WAVEINCAPS2W,*PWAVEINCAPS2W,*NPWAVEINCAPS2W,*LPWAVEINCAPS2W;
typedef struct tagWAVEOUTCAPS2W
{
WORD wMid;
WORD wPid;
MMVERSION vDriverVersion;
WCHAR szPname[MAXPNAMELEN];
DWORD dwFormats;
WORD wChannels;
WORD wReserved1;
DWORD dwSupport;
GUID ManufacturerGuid;
GUID ProductGuid;
GUID NameGuid;
} WAVEOUTCAPS2W,*PWAVEOUTCAPS2W,*NPWAVEOUTCAPS2W,*LPWAVEOUTCAPS2W;
#endif /* defined(__MINGW32__) && defined(_MMSYSTEM_H) */
#ifndef WAVE_FORMAT_IEEE_FLOAT
#define WAVE_FORMAT_IEEE_FLOAT 0x0003
#endif
libs/SDL2/src/audio/winmm/SDL_winmm.h
+4 -4
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -20,8 +20,8 @@
*/
#include "../../SDL_internal.h"
#ifndef _SDL_winmm_h
#define _SDL_winmm_h
#ifndef SDL_winmm_h_
#define SDL_winmm_h_
#include "../SDL_sysaudio.h"
@@ -40,6 +40,6 @@ struct SDL_PrivateAudioData
int next_buffer;
};
#endif /* _SDL_winmm_h */
#endif /* SDL_winmm_h_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/xaudio2/SDL_xaudio2.c
+7 -5
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -78,8 +78,8 @@
#else
# define SDL_XAUDIO2_HAS_SDK 1
#endif
#endif
#endif /* 0 */
#endif /* __GNUC__ */
#ifdef SDL_XAUDIO2_HAS_SDK
@@ -90,7 +90,7 @@
#endif
#endif
#if !defined(_SDL_XAUDIO2_H)
#if !defined(SDL_XAUDIO2_H_)
#define INITGUID 1
#include <xaudio2.h>
#endif
@@ -453,6 +453,7 @@ XAUDIO2_Init(SDL_AudioDriverImpl * impl)
#else
/* XAudio2Create() is a macro that uses COM; we don't load the .dll */
IXAudio2 *ixa2 = NULL;
HRESULT hr = S_FALSE;
#if defined(__WIN32__)
// TODO, WinRT: Investigate using CoInitializeEx here
if (FAILED(WIN_CoInitialize())) {
@@ -461,11 +462,12 @@ XAUDIO2_Init(SDL_AudioDriverImpl * impl)
}
#endif
if (XAudio2Create(&ixa2, 0, XAUDIO2_DEFAULT_PROCESSOR) != S_OK) {
hr = XAudio2Create( &ixa2, 0, XAUDIO2_DEFAULT_PROCESSOR );
if ( hr != S_OK) {
#if defined(__WIN32__)
WIN_CoUninitialize();
#endif
SDL_SetError("XAudio2: XAudio2Create() failed at initialization");
SDL_SetError("XAudio2: XAudio2Create() failed at initialization: 0x%.8x", hr );
return 0; /* not available. */
}
IXAudio2_Release(ixa2);
libs/SDL2/src/audio/xaudio2/SDL_xaudio2.h
+4 -4
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@@ -19,8 +19,8 @@
3. This notice may not be removed or altered from any source distribution.
*/
#ifndef _SDL_XAUDIO2_H
#define _SDL_XAUDIO2_H
#ifndef SDL_XAUDIO2_H_
#define SDL_XAUDIO2_H_
#include <windows.h>
#include <mmreg.h>
@@ -381,6 +381,6 @@ const struct IXAudio2VoiceCallbackVtbl
#pragma pack(pop) /* Undo pragma push */
#endif /* _SDL_XAUDIO2_H */
#endif /* SDL_XAUDIO2_H_ */
/* vi: set ts=4 sw=4 expandtab: */
libs/SDL2/src/audio/xaudio2/SDL_xaudio2_winrthelpers.cpp
+1 -1
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
libs/SDL2/src/audio/xaudio2/SDL_xaudio2_winrthelpers.h
+1 -1
View File
@@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages