/*

Simple DirectMedia Layer

Copyright (C) 1997-2014 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.

*/

/*

The PulseAudio target for SDL 1.3 is based on the 1.3 arts target, with

the appropriate parts replaced with the 1.2 PulseAudio target code. This

was the cleanest way to move it to 1.3. The 1.2 target was written by

Stéphan Kochen: stephan .a.t. kochen.nl

*/

#include "../../SDL_internal.h"

#include "SDL_assert.h"

#if SDL_AUDIO_DRIVER_PULSEAUDIO

/* Allow access to a raw mixing buffer */

#ifdef HAVE_SIGNAL_H

#include <signal.h>

#endif

#include <unistd.h>

#include <sys/types.h>

#include <errno.h>

#include <pulse/pulseaudio.h>

#include "SDL_timer.h"

#include "SDL_audio.h"

#include "../SDL_audiomem.h"

#include "../SDL_audio_c.h"

#include "SDL_pulseaudio.h"

#include "SDL_loadso.h"

#if (PA_API_VERSION < 12)

/** Return non-zero if the passed state is one of the connected states */

static SDL_INLINE int PA_CONTEXT_IS_GOOD(pa_context_state_t x) {

return

x == PA_CONTEXT_CONNECTING ||

x == PA_CONTEXT_AUTHORIZING ||

x == PA_CONTEXT_SETTING_NAME ||

x == PA_CONTEXT_READY;

}

/** Return non-zero if the passed state is one of the connected states */

static SDL_INLINE int PA_STREAM_IS_GOOD(pa_stream_state_t x) {

return

x == PA_STREAM_CREATING ||

x == PA_STREAM_READY;

}

#endif /* pulseaudio <= 0.9.10 */

static const char *(*PULSEAUDIO_pa_get_library_version) (void);

static pa_channel_map *(*PULSEAUDIO_pa_channel_map_init_auto) (

pa_channel_map *, unsigned, pa_channel_map_def_t);

static const char * (*PULSEAUDIO_pa_strerror) (int);

static pa_mainloop * (*PULSEAUDIO_pa_mainloop_new) (void);

static pa_mainloop_api * (*PULSEAUDIO_pa_mainloop_get_api) (pa_mainloop *);

static int (*PULSEAUDIO_pa_mainloop_iterate) (pa_mainloop *, int, int *);

static int (*PULSEAUDIO_pa_mainloop_run) (pa_mainloop *, int *);

static void (*PULSEAUDIO_pa_mainloop_quit) (pa_mainloop *, int);

static void (*PULSEAUDIO_pa_mainloop_free) (pa_mainloop *);

static pa_operation_state_t (*PULSEAUDIO_pa_operation_get_state) (

pa_operation *);

static void (*PULSEAUDIO_pa_operation_cancel) (pa_operation *);

static void (*PULSEAUDIO_pa_operation_unref) (pa_operation *);

static pa_context * (*PULSEAUDIO_pa_context_new) (pa_mainloop_api *,

const char *);

static int (*PULSEAUDIO_pa_context_connect) (pa_context *, const char *,

pa_context_flags_t, const pa_spawn_api *);

static pa_operation * (*PULSEAUDIO_pa_context_get_sink_info_list) (pa_context *, pa_sink_info_cb_t, void *);

static pa_operation * (*PULSEAUDIO_pa_context_get_source_info_list) (pa_context *, pa_source_info_cb_t, void *);

static pa_operation * (*PULSEAUDIO_pa_context_get_sink_info_by_index) (pa_context *, uint32_t, pa_sink_info_cb_t, void *);

static pa_operation * (*PULSEAUDIO_pa_context_get_source_info_by_index) (pa_context *, uint32_t, pa_source_info_cb_t, void *);

static pa_context_state_t (*PULSEAUDIO_pa_context_get_state) (pa_context *);

static pa_operation * (*PULSEAUDIO_pa_context_subscribe) (pa_context *, pa_subscription_mask_t, pa_context_success_cb_t, void *);

static void (*PULSEAUDIO_pa_context_set_subscribe_callback) (pa_context *, pa_context_subscribe_cb_t, void *);

static void (*PULSEAUDIO_pa_context_disconnect) (pa_context *);

static void (*PULSEAUDIO_pa_context_unref) (pa_context *);

static pa_stream * (*PULSEAUDIO_pa_stream_new) (pa_context *, const char *,

const pa_sample_spec *, const pa_channel_map *);

static int (*PULSEAUDIO_pa_stream_connect_playback) (pa_stream *, const char *,

const pa_buffer_attr *, pa_stream_flags_t, pa_cvolume *, pa_stream *);

static pa_stream_state_t (*PULSEAUDIO_pa_stream_get_state) (pa_stream *);

static size_t (*PULSEAUDIO_pa_stream_writable_size) (pa_stream *);

static int (*PULSEAUDIO_pa_stream_write) (pa_stream *, const void *, size_t,

pa_free_cb_t, int64_t, pa_seek_mode_t);

static pa_operation * (*PULSEAUDIO_pa_stream_drain) (pa_stream *,

pa_stream_success_cb_t, void *);

static int (*PULSEAUDIO_pa_stream_disconnect) (pa_stream *);

static void (*PULSEAUDIO_pa_stream_unref) (pa_stream *);

static int load_pulseaudio_syms(void);

#ifdef SDL_AUDIO_DRIVER_PULSEAUDIO_DYNAMIC

static const char *pulseaudio_library = SDL_AUDIO_DRIVER_PULSEAUDIO_DYNAMIC;

static void *pulseaudio_handle = NULL;

static int

load_pulseaudio_sym(const char *fn, void **addr)

{

*addr = SDL_LoadFunction(pulseaudio_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_PULSEAUDIO_SYM(x) \

if (!load_pulseaudio_sym(#x, (void **) (char *) &PULSEAUDIO_##x)) return -1

static void

UnloadPulseAudioLibrary(void)

{

if (pulseaudio_handle != NULL) {

SDL_UnloadObject(pulseaudio_handle);

pulseaudio_handle = NULL;

}

}

static int

LoadPulseAudioLibrary(void)

{

int retval = 0;

if (pulseaudio_handle == NULL) {

pulseaudio_handle = SDL_LoadObject(pulseaudio_library);

if (pulseaudio_handle == NULL) {

retval = -1;

/* Don't call SDL_SetError(): SDL_LoadObject already did. */

} else {

retval = load_pulseaudio_syms();

if (retval < 0) {

UnloadPulseAudioLibrary();

}

}

}

return retval;

}

#else

#define SDL_PULSEAUDIO_SYM(x) PULSEAUDIO_##x = x

static void

UnloadPulseAudioLibrary(void)

{

}

static int

LoadPulseAudioLibrary(void)

{

load_pulseaudio_syms();

return 0;

}

#endif /* SDL_AUDIO_DRIVER_PULSEAUDIO_DYNAMIC */

static int

load_pulseaudio_syms(void)

{

SDL_PULSEAUDIO_SYM(pa_get_library_version);

SDL_PULSEAUDIO_SYM(pa_mainloop_new);

SDL_PULSEAUDIO_SYM(pa_mainloop_get_api);

SDL_PULSEAUDIO_SYM(pa_mainloop_iterate);

SDL_PULSEAUDIO_SYM(pa_mainloop_run);

SDL_PULSEAUDIO_SYM(pa_mainloop_quit);

SDL_PULSEAUDIO_SYM(pa_mainloop_free);

SDL_PULSEAUDIO_SYM(pa_operation_get_state);

SDL_PULSEAUDIO_SYM(pa_operation_cancel);

SDL_PULSEAUDIO_SYM(pa_operation_unref);

SDL_PULSEAUDIO_SYM(pa_context_new);

SDL_PULSEAUDIO_SYM(pa_context_connect);

SDL_PULSEAUDIO_SYM(pa_context_get_sink_info_list);

SDL_PULSEAUDIO_SYM(pa_context_get_source_info_list);

SDL_PULSEAUDIO_SYM(pa_context_get_sink_info_by_index);

SDL_PULSEAUDIO_SYM(pa_context_get_source_info_by_index);

SDL_PULSEAUDIO_SYM(pa_context_get_state);

SDL_PULSEAUDIO_SYM(pa_context_subscribe);

SDL_PULSEAUDIO_SYM(pa_context_set_subscribe_callback);

SDL_PULSEAUDIO_SYM(pa_context_disconnect);

SDL_PULSEAUDIO_SYM(pa_context_unref);

SDL_PULSEAUDIO_SYM(pa_stream_new);

SDL_PULSEAUDIO_SYM(pa_stream_connect_playback);

SDL_PULSEAUDIO_SYM(pa_stream_get_state);

SDL_PULSEAUDIO_SYM(pa_stream_writable_size);

SDL_PULSEAUDIO_SYM(pa_stream_write);

SDL_PULSEAUDIO_SYM(pa_stream_drain);

SDL_PULSEAUDIO_SYM(pa_stream_disconnect);

SDL_PULSEAUDIO_SYM(pa_stream_unref);

SDL_PULSEAUDIO_SYM(pa_channel_map_init_auto);

SDL_PULSEAUDIO_SYM(pa_strerror);

return 0;

}

static SDL_INLINE int

squashVersion(const int major, const int minor, const int patch)

{

return ((major & 0xFF) << 16) | ((minor & 0xFF) << 8) | (patch & 0xFF);

}

/* Workaround for older pulse: pa_context_new() must have non-NULL appname */

static const char *

getAppName(void)

{

const char *verstr = PULSEAUDIO_pa_get_library_version();

if (verstr != NULL) {

int maj, min, patch;

if (SDL_sscanf(verstr, "%d.%d.%d", &maj, &min, &patch) == 3) {

if (squashVersion(maj, min, patch) >= squashVersion(0, 9, 15)) {

return NULL; /* 0.9.15+ handles NULL correctly. */

}

}

}

return "SDL Application"; /* oh well. */

}

static void

WaitForPulseOperation(pa_mainloop *mainloop, pa_operation *o)

{

/* This checks for NO errors currently. Either fix that, check results elsewhere, or do things you don't care about. */

if (mainloop && o) {

SDL_bool okay = SDL_TRUE;

while (okay && (PULSEAUDIO_pa_operation_get_state(o) == PA_OPERATION_RUNNING)) {

okay = (PULSEAUDIO_pa_mainloop_iterate(mainloop, 1, NULL) >= 0);

}

PULSEAUDIO_pa_operation_unref(o);

}

}

static void

DisconnectFromPulseServer(pa_mainloop *mainloop, pa_context *context)

{

if (context) {

PULSEAUDIO_pa_context_disconnect(context);

PULSEAUDIO_pa_context_unref(context);

}

if (mainloop != NULL) {

PULSEAUDIO_pa_mainloop_free(mainloop);

}

}

static int

ConnectToPulseServer_Internal(pa_mainloop **_mainloop, pa_context **_context)

{

pa_mainloop *mainloop = NULL;

pa_context *context = NULL;

pa_mainloop_api *mainloop_api = NULL;

int state = 0;

*_mainloop = NULL;

*_context = NULL;

/* Set up a new main loop */

if (!(mainloop = PULSEAUDIO_pa_mainloop_new())) {

return SDL_SetError("pa_mainloop_new() failed");

}

*_mainloop = mainloop;

mainloop_api = PULSEAUDIO_pa_mainloop_get_api(mainloop);

SDL_assert(mainloop_api); /* this never fails, right? */

context = PULSEAUDIO_pa_context_new(mainloop_api, getAppName());

if (!context) {

return SDL_SetError("pa_context_new() failed");

}

*_context = context;

/* Connect to the PulseAudio server */

if (PULSEAUDIO_pa_context_connect(context, NULL, 0, NULL) < 0) {

return SDL_SetError("Could not setup connection to PulseAudio");

}

do {

if (PULSEAUDIO_pa_mainloop_iterate(mainloop, 1, NULL) < 0) {

return SDL_SetError("pa_mainloop_iterate() failed");

}

state = PULSEAUDIO_pa_context_get_state(context);

if (!PA_CONTEXT_IS_GOOD(state)) {

return SDL_SetError("Could not connect to PulseAudio");

}

} while (state != PA_CONTEXT_READY);

return 0; /* connected and ready! */

}

static int

ConnectToPulseServer(pa_mainloop **_mainloop, pa_context **_context)

{

const int retval = ConnectToPulseServer_Internal(_mainloop, _context);

if (retval < 0) {

DisconnectFromPulseServer(*_mainloop, *_context);

}

return retval;

}

/* This function waits until it is possible to write a full sound buffer */

static void

PULSEAUDIO_WaitDevice(_THIS)

{

struct SDL_PrivateAudioData *h = this->hidden;

while (this->enabled) {

if (PULSEAUDIO_pa_context_get_state(h->context) != PA_CONTEXT_READY ||

PULSEAUDIO_pa_stream_get_state(h->stream) != PA_STREAM_READY ||

PULSEAUDIO_pa_mainloop_iterate(h->mainloop, 1, NULL) < 0) {

SDL_OpenedAudioDeviceDisconnected(this);

return;

}

if (PULSEAUDIO_pa_stream_writable_size(h->stream) >= h->mixlen) {

return;

}

}

}

static void

PULSEAUDIO_PlayDevice(_THIS)

{

/* Write the audio data */

struct SDL_PrivateAudioData *h = this->hidden;

if (this->enabled) {

if (PULSEAUDIO_pa_stream_write(h->stream, h->mixbuf, h->mixlen, NULL, 0LL, PA_SEEK_RELATIVE) < 0) {

SDL_OpenedAudioDeviceDisconnected(this);

}

}

}

static void

stream_drain_complete(pa_stream *s, int success, void *userdata)

{

/* no-op for pa_stream_drain() to use for callback. */

}

static void

PULSEAUDIO_WaitDone(_THIS)

{

if (this->enabled) {

struct SDL_PrivateAudioData *h = this->hidden;

pa_operation *o = PULSEAUDIO_pa_stream_drain(h->stream, stream_drain_complete, NULL);

if (o) {

while (PULSEAUDIO_pa_operation_get_state(o) != PA_OPERATION_DONE) {

if (PULSEAUDIO_pa_context_get_state(h->context) != PA_CONTEXT_READY ||

PULSEAUDIO_pa_stream_get_state(h->stream) != PA_STREAM_READY ||

PULSEAUDIO_pa_mainloop_iterate(h->mainloop, 1, NULL) < 0) {

PULSEAUDIO_pa_operation_cancel(o);

break;

}

}

PULSEAUDIO_pa_operation_unref(o);

}

}

}

static Uint8 *

PULSEAUDIO_GetDeviceBuf(_THIS)

{

return (this->hidden->mixbuf);

}

static void

PULSEAUDIO_CloseDevice(_THIS)

{

if (this->hidden != NULL) {

SDL_FreeAudioMem(this->hidden->mixbuf);

SDL_free(this->hidden->device_name);

if (this->hidden->stream) {

PULSEAUDIO_pa_stream_disconnect(this->hidden->stream);

PULSEAUDIO_pa_stream_unref(this->hidden->stream);

}

DisconnectFromPulseServer(this->hidden->mainloop, this->hidden->context);

SDL_free(this->hidden);

this->hidden = NULL;

}

}

static void

DeviceNameCallback(pa_context *c, const pa_sink_info *i, int is_last, void *data)

{

if (i) {

char **devname = (char **) data;

*devname = SDL_strdup(i->name);

}

}

static SDL_bool

FindDeviceName(struct SDL_PrivateAudioData *h, void *handle)

{

const uint32_t idx = ((uint32_t) ((size_t) handle)) - 1;

if (handle == NULL) { /* NULL == default device. */

return SDL_TRUE;

}

WaitForPulseOperation(h->mainloop, PULSEAUDIO_pa_context_get_sink_info_by_index(h->context, idx, DeviceNameCallback, &h->device_name));

return (h->device_name != NULL);

}

static int

PULSEAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)

{

struct SDL_PrivateAudioData *h = NULL;

Uint16 test_format = 0;

pa_sample_spec paspec;

pa_buffer_attr paattr;

pa_channel_map pacmap;

pa_stream_flags_t flags = 0;

int state = 0;

/* 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_memset(this->hidden, 0, (sizeof *this->hidden));

h = this->hidden;

paspec.format = PA_SAMPLE_INVALID;

/* Try for a closest match on audio format */

for (test_format = SDL_FirstAudioFormat(this->spec.format);

(paspec.format == PA_SAMPLE_INVALID) && test_format;) {

#ifdef DEBUG_AUDIO

fprintf(stderr, "Trying format 0x%4.4x\n", test_format);

#endif

switch (test_format) {

case AUDIO_U8:

paspec.format = PA_SAMPLE_U8;

break;

case AUDIO_S16LSB:

paspec.format = PA_SAMPLE_S16LE;

break;

case AUDIO_S16MSB:

paspec.format = PA_SAMPLE_S16BE;

break;

case AUDIO_S32LSB:

paspec.format = PA_SAMPLE_S32LE;

break;

case AUDIO_S32MSB:

paspec.format = PA_SAMPLE_S32BE;

break;

case AUDIO_F32LSB:

paspec.format = PA_SAMPLE_FLOAT32LE;

break;

case AUDIO_F32MSB:

paspec.format = PA_SAMPLE_FLOAT32BE;

break;

default:

paspec.format = PA_SAMPLE_INVALID;

break;

}

if (paspec.format == PA_SAMPLE_INVALID) {

test_format = SDL_NextAudioFormat();

}

}

if (paspec.format == PA_SAMPLE_INVALID) {

PULSEAUDIO_CloseDevice(this);

return SDL_SetError("Couldn't find any hardware audio formats");

}

this->spec.format = test_format;

/* Calculate the final parameters for this audio specification */

#ifdef PA_STREAM_ADJUST_LATENCY

this->spec.samples /= 2; /* Mix in smaller chunck to avoid underruns */

#endif

SDL_CalculateAudioSpec(&this->spec);

/* Allocate mixing buffer */

h->mixlen = this->spec.size;

h->mixbuf = (Uint8 *) SDL_AllocAudioMem(h->mixlen);

if (h->mixbuf == NULL) {

PULSEAUDIO_CloseDevice(this);

return SDL_OutOfMemory();

}

SDL_memset(h->mixbuf, this->spec.silence, this->spec.size);

paspec.channels = this->spec.channels;

paspec.rate = this->spec.freq;

/* Reduced prebuffering compared to the defaults. */

#ifdef PA_STREAM_ADJUST_LATENCY

/* 2x original requested bufsize */

paattr.tlength = h->mixlen * 4;

paattr.prebuf = -1;

paattr.maxlength = -1;

/* -1 can lead to pa_stream_writable_size() >= mixlen never being true */

paattr.minreq = h->mixlen;

flags = PA_STREAM_ADJUST_LATENCY;

#else

paattr.tlength = h->mixlen*2;

paattr.prebuf = h->mixlen*2;

paattr.maxlength = h->mixlen*2;

paattr.minreq = h->mixlen;

#endif

if (ConnectToPulseServer(&h->mainloop, &h->context) < 0) {

PULSEAUDIO_CloseDevice(this);

return SDL_SetError("Could not connect to PulseAudio server");

}

if (!FindDeviceName(h, handle)) {

PULSEAUDIO_CloseDevice(this);

return SDL_SetError("Requested PulseAudio sink missing?");

}

/* The SDL ALSA output hints us that we use Windows' channel mapping */

/* http://bugzilla.libsdl.org/show_bug.cgi?id=110 */

PULSEAUDIO_pa_channel_map_init_auto(&pacmap, this->spec.channels,

PA_CHANNEL_MAP_WAVEEX);

h->stream = PULSEAUDIO_pa_stream_new(

h->context,

"Simple DirectMedia Layer", /* stream description */

&paspec, /* sample format spec */

&pacmap /* channel map */

);

if (h->stream == NULL) {

PULSEAUDIO_CloseDevice(this);

return SDL_SetError("Could not set up PulseAudio stream");

}

/* now that we have multi-device support, don't move a stream from

a device that was unplugged to something else, unless we're default. */

if (h->device_name != NULL) {

flags |= PA_STREAM_DONT_MOVE;

}

if (PULSEAUDIO_pa_stream_connect_playback(h->stream, h->device_name, &paattr, flags,

NULL, NULL) < 0) {

PULSEAUDIO_CloseDevice(this);

return SDL_SetError("Could not connect PulseAudio stream");

}

do {

if (PULSEAUDIO_pa_mainloop_iterate(h->mainloop, 1, NULL) < 0) {

PULSEAUDIO_CloseDevice(this);

return SDL_SetError("pa_mainloop_iterate() failed");

}

state = PULSEAUDIO_pa_stream_get_state(h->stream);

if (!PA_STREAM_IS_GOOD(state)) {

PULSEAUDIO_CloseDevice(this);

return SDL_SetError("Could not connect PulseAudio stream");

}

} while (state != PA_STREAM_READY);

/* We're ready to rock and roll. :-) */

return 0;

}

static pa_mainloop *hotplug_mainloop = NULL;

static pa_context *hotplug_context = NULL;

static SDL_Thread *hotplug_thread = NULL;

/* device handles are device index + 1, cast to void*, so we never pass a NULL. */

/* This is called when PulseAudio adds an output ("sink") device. */

static void

SinkInfoCallback(pa_context *c, const pa_sink_info *i, int is_last, void *data)

{

if (i) {

SDL_AddAudioDevice(SDL_FALSE, i->description, (void *) ((size_t) i->index+1));

}

}

/* This is called when PulseAudio adds a capture ("source") device. */

static void

SourceInfoCallback(pa_context *c, const pa_source_info *i, int is_last, void *data)

{

if (i) {

/* Skip "monitor" sources. These are just output from other sinks. */

if (i->monitor_of_sink == PA_INVALID_INDEX) {

SDL_AddAudioDevice(SDL_TRUE, i->description, (void *) ((size_t) i->index+1));

}

}

}

/* This is called when PulseAudio has a device connected/removed/changed. */

static void

HotplugCallback(pa_context *c, pa_subscription_event_type_t t, uint32_t idx, void *data)

{

const SDL_bool added = ((t & PA_SUBSCRIPTION_EVENT_TYPE_MASK) == PA_SUBSCRIPTION_EVENT_NEW);

const SDL_bool removed = ((t & PA_SUBSCRIPTION_EVENT_TYPE_MASK) == PA_SUBSCRIPTION_EVENT_REMOVE);

if (added || removed) { /* we only care about add/remove events. */

const SDL_bool sink = ((t & PA_SUBSCRIPTION_EVENT_FACILITY_MASK) == PA_SUBSCRIPTION_EVENT_SINK);

const SDL_bool source = ((t & PA_SUBSCRIPTION_EVENT_FACILITY_MASK) == PA_SUBSCRIPTION_EVENT_SOURCE);

/* adds need sink details from the PulseAudio server. Another callback... */

if (added && sink) {

PULSEAUDIO_pa_context_get_sink_info_by_index(hotplug_context, idx, SinkInfoCallback, NULL);

} else if (added && source) {

PULSEAUDIO_pa_context_get_source_info_by_index(hotplug_context, idx, SourceInfoCallback, NULL);

} else if (removed && (sink || source)) {

/* removes we can handle just with the device index. */

SDL_RemoveAudioDevice(source != 0, (void *) ((size_t) idx+1));

}

}

}

/* this runs as a thread while the Pulse target is initialized to catch hotplug events. */

static int SDLCALL

HotplugThread(void *data)

{

pa_operation *o;

SDL_SetThreadPriority(SDL_THREAD_PRIORITY_LOW);

PULSEAUDIO_pa_context_set_subscribe_callback(hotplug_context, HotplugCallback, NULL);

o = PULSEAUDIO_pa_context_subscribe(hotplug_context, PA_SUBSCRIPTION_MASK_SINK | PA_SUBSCRIPTION_MASK_SOURCE, NULL, NULL);

PULSEAUDIO_pa_operation_unref(o); /* don't wait for it, just do our thing. */

PULSEAUDIO_pa_mainloop_run(hotplug_mainloop, NULL);

return 0;

}

static void

PULSEAUDIO_DetectDevices()

{

WaitForPulseOperation(hotplug_mainloop, PULSEAUDIO_pa_context_get_sink_info_list(hotplug_context, SinkInfoCallback, NULL));

WaitForPulseOperation(hotplug_mainloop, PULSEAUDIO_pa_context_get_source_info_list(hotplug_context, SourceInfoCallback, NULL));

/* ok, we have a sane list, let's set up hotplug notifications now... */

hotplug_thread = SDL_CreateThread(HotplugThread, "PulseHotplug", NULL);

}

static void

PULSEAUDIO_Deinitialize(void)

{

if (hotplug_thread) {

PULSEAUDIO_pa_mainloop_quit(hotplug_mainloop, 0);

SDL_WaitThread(hotplug_thread, NULL);

hotplug_thread = NULL;

}

DisconnectFromPulseServer(hotplug_mainloop, hotplug_context);

hotplug_mainloop = NULL;

hotplug_context = NULL;

UnloadPulseAudioLibrary();

}

static int

PULSEAUDIO_Init(SDL_AudioDriverImpl * impl)

{

if (LoadPulseAudioLibrary() < 0) {

return 0;

}

if (ConnectToPulseServer(&hotplug_mainloop, &hotplug_context) < 0) {

UnloadPulseAudioLibrary();

return 0;

}

/* Set the function pointers */

impl->DetectDevices = PULSEAUDIO_DetectDevices;

impl->OpenDevice = PULSEAUDIO_OpenDevice;

impl->PlayDevice = PULSEAUDIO_PlayDevice;

impl->WaitDevice = PULSEAUDIO_WaitDevice;

impl->GetDeviceBuf = PULSEAUDIO_GetDeviceBuf;

impl->CloseDevice = PULSEAUDIO_CloseDevice;

impl->WaitDone = PULSEAUDIO_WaitDone;

impl->Deinitialize = PULSEAUDIO_Deinitialize;

return 1; /* this audio target is available. */

}

AudioBootStrap PULSEAUDIO_bootstrap = {

"pulseaudio", "PulseAudio", PULSEAUDIO_Init, 0

};

#endif /* SDL_AUDIO_DRIVER_PULSEAUDIO */

/* vi: set ts=4 sw=4 expandtab: */