*initHEADmaster

1 files changed, 782 insertions, 0 deletions

diff --git a/source/3rd-party/SDL2/src/audio/pulseaudio/SDL_pulseaudio.c b/source/3rd-party/SDL2/src/audio/pulseaudio/SDL_pulseaudio.c
new file mode 100644
index 0000000..053a1c3
--- /dev/null
+++ b/source/3rd-party/SDL2/src/audio/pulseaudio/SDL_pulseaudio.c
@@ -0,0 +1,782 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2018 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 <pulse/pulseaudio.h>
+
+#include "SDL_timer.h"
+#include "SDL_audio.h"
+#include "../SDL_audio_c.h"
+#include "SDL_pulseaudio.h"
+#include "SDL_loadso.h"
+#include "../../thread/SDL_systhread.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 int (*PULSEAUDIO_pa_stream_connect_record) (pa_stream *, const char *,
+    const pa_buffer_attr *, pa_stream_flags_t);
+static pa_stream_state_t (*PULSEAUDIO_pa_stream_get_state) (pa_stream *);
+static size_t (*PULSEAUDIO_pa_stream_writable_size) (pa_stream *);
+static size_t (*PULSEAUDIO_pa_stream_readable_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_peek) (pa_stream *, const void **, size_t *);
+static int (*PULSEAUDIO_pa_stream_drop) (pa_stream *);
+static pa_operation * (*PULSEAUDIO_pa_stream_flush) (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_connect_record);
+    SDL_PULSEAUDIO_SYM(pa_stream_get_state);
+    SDL_PULSEAUDIO_SYM(pa_stream_writable_size);
+    SDL_PULSEAUDIO_SYM(pa_stream_readable_size);
+    SDL_PULSEAUDIO_SYM(pa_stream_write);
+    SDL_PULSEAUDIO_SYM(pa_stream_drain);
+    SDL_PULSEAUDIO_SYM(pa_stream_disconnect);
+    SDL_PULSEAUDIO_SYM(pa_stream_peek);
+    SDL_PULSEAUDIO_SYM(pa_stream_drop);
+    SDL_PULSEAUDIO_SYM(pa_stream_flush);
+    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 (SDL_AtomicGet(&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 (SDL_AtomicGet(&this->enabled)) {
+        if (PULSEAUDIO_pa_stream_write(h->stream, h->mixbuf, h->mixlen, NULL, 0LL, PA_SEEK_RELATIVE) < 0) {
+            SDL_OpenedAudioDeviceDisconnected(this);
+        }
+    }
+}
+
+static Uint8 *
+PULSEAUDIO_GetDeviceBuf(_THIS)
+{
+    return (this->hidden->mixbuf);
+}
+
+
+static int
+PULSEAUDIO_CaptureFromDevice(_THIS, void *buffer, int buflen)
+{
+    struct SDL_PrivateAudioData *h = this->hidden;
+    const void *data = NULL;
+    size_t nbytes = 0;
+
+    while (SDL_AtomicGet(&this->enabled)) {
+        if (h->capturebuf != NULL) {
+            const int cpy = SDL_min(buflen, h->capturelen);
+            SDL_memcpy(buffer, h->capturebuf, cpy);
+            /*printf("PULSEAUDIO: fed %d captured bytes\n", cpy);*/
+            h->capturebuf += cpy;
+            h->capturelen -= cpy;
+            if (h->capturelen == 0) {
+                h->capturebuf = NULL;
+                PULSEAUDIO_pa_stream_drop(h->stream);  /* done with this fragment. */
+            }
+            return cpy;  /* new data, return it. */
+        }
+
+        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 -1;  /* uhoh, pulse failed! */
+        }
+
+        if (PULSEAUDIO_pa_stream_readable_size(h->stream) == 0) {
+            continue;  /* no data available yet. */
+        }
+
+        /* a new fragment is available! */
+        PULSEAUDIO_pa_stream_peek(h->stream, &data, &nbytes);
+        SDL_assert(nbytes > 0);
+        if (data == NULL) {  /* NULL==buffer had a hole. Ignore that. */
+            PULSEAUDIO_pa_stream_drop(h->stream);  /* drop this fragment. */
+        } else {
+            /* store this fragment's data, start feeding it to SDL. */
+            /*printf("PULSEAUDIO: captured %d new bytes\n", (int) nbytes);*/
+            h->capturebuf = (const Uint8 *) data;
+            h->capturelen = nbytes;
+        }
+    }
+
+    return -1;  /* not enabled? */
+}
+
+static void
+PULSEAUDIO_FlushCapture(_THIS)
+{
+    struct SDL_PrivateAudioData *h = this->hidden;
+    const void *data = NULL;
+    size_t nbytes = 0;
+
+    if (h->capturebuf != NULL) {
+        PULSEAUDIO_pa_stream_drop(h->stream);
+        h->capturebuf = NULL;
+        h->capturelen = 0;
+    }
+
+    while (SDL_TRUE) {
+        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;  /* uhoh, pulse failed! */
+        }
+
+        if (PULSEAUDIO_pa_stream_readable_size(h->stream) == 0) {
+            break;  /* no data available, so we're done. */
+        }
+
+        /* a new fragment is available! Just dump it. */
+        PULSEAUDIO_pa_stream_peek(h->stream, &data, &nbytes);
+        PULSEAUDIO_pa_stream_drop(h->stream);  /* drop this fragment. */
+    }
+}
+
+static void
+PULSEAUDIO_CloseDevice(_THIS)
+{
+    if (this->hidden->stream) {
+        if (this->hidden->capturebuf != NULL) {
+            PULSEAUDIO_pa_stream_drop(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->mixbuf);
+    SDL_free(this->hidden->device_name);
+    SDL_free(this->hidden);
+}
+
+static void
+SinkDeviceNameCallback(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 void
+SourceDeviceNameCallback(pa_context *c, const pa_source_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, const int iscapture, void *handle)
+{
+    const uint32_t idx = ((uint32_t) ((size_t) handle)) - 1;
+
+    if (handle == NULL) {  /* NULL == default device. */
+        return SDL_TRUE;
+    }
+
+    if (iscapture) {
+        WaitForPulseOperation(h->mainloop,
+            PULSEAUDIO_pa_context_get_source_info_by_index(h->context, idx,
+                SourceDeviceNameCallback, &h->device_name));
+    } else {
+        WaitForPulseOperation(h->mainloop,
+            PULSEAUDIO_pa_context_get_sink_info_by_index(h->context, idx,
+                SinkDeviceNameCallback, &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;
+    int rc = 0;
+
+    /* Initialize all variables that we clean on shutdown */
+    h = this->hidden = (struct SDL_PrivateAudioData *)
+        SDL_malloc((sizeof *this->hidden));
+    if (this->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+    SDL_zerop(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) {
+        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 */
+    if (!iscapture) {
+        h->mixlen = this->spec.size;
+        h->mixbuf = (Uint8 *) SDL_malloc(h->mixlen);
+        if (h->mixbuf == NULL) {
+            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) {
+        return SDL_SetError("Could not connect to PulseAudio server");
+    }
+
+    if (!FindDeviceName(h, iscapture, handle)) {
+        return SDL_SetError("Requested PulseAudio sink/source 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) {
+        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 (iscapture) {
+        rc = PULSEAUDIO_pa_stream_connect_record(h->stream, h->device_name, &paattr, flags);
+    } else {
+        rc = PULSEAUDIO_pa_stream_connect_playback(h->stream, h->device_name, &paattr, flags, NULL, NULL);
+    }
+
+    if (rc < 0) {
+        return SDL_SetError("Could not connect PulseAudio stream");
+    }
+
+    do {
+        if (PULSEAUDIO_pa_mainloop_iterate(h->mainloop, 1, NULL) < 0) {
+            return SDL_SetError("pa_mainloop_iterate() failed");
+        }
+        state = PULSEAUDIO_pa_stream_get_state(h->stream);
+        if (!PA_STREAM_IS_GOOD(state)) {
+            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_CreateThreadInternal(HotplugThread, "PulseHotplug", 256 * 1024, 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->Deinitialize = PULSEAUDIO_Deinitialize;
+    impl->CaptureFromDevice = PULSEAUDIO_CaptureFromDevice;
+    impl->FlushCapture = PULSEAUDIO_FlushCapture;
+
+    impl->HasCaptureSupport = SDL_TRUE;
+
+    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: */