static void
gst_pulsesrc_init (GstPulseSrc * pulsesrc, GstPulseSrcClass * klass)
{
pulsesrc->server = NULL;
pulsesrc->device = NULL;
pulsesrc->device_description = NULL;
pulsesrc->context = NULL;
pulsesrc->stream = NULL;
pulsesrc->read_buffer = NULL;
pulsesrc->read_buffer_length = 0;
#ifdef HAVE_PULSE_0_9_13
pa_sample_spec_init (&pulsesrc->sample_spec);
#else
pulsesrc->sample_spec.format = PA_SAMPLE_INVALID;
pulsesrc->sample_spec.rate = 0;
pulsesrc->sample_spec.channels = 0;
#endif
pulsesrc->operation_success = FALSE;
pulsesrc->paused = FALSE;
pulsesrc->in_read = FALSE;
pulsesrc->mixer = NULL;
pulsesrc->probe = gst_pulseprobe_new (G_OBJECT (pulsesrc), G_OBJECT_GET_CLASS (pulsesrc), PROP_DEVICE, pulsesrc->server, FALSE, TRUE); /* FALSE for sinks, TRUE for sources */
/* this should be the default but it isn't yet */
gst_base_audio_src_set_slave_method (GST_BASE_AUDIO_SRC (pulsesrc),
GST_BASE_AUDIO_SRC_SLAVE_SKEW);
}
int pa_scache_add_item(
pa_core *c,
const char *name,
const pa_sample_spec *ss,
const pa_channel_map *map,
const pa_memchunk *chunk,
pa_proplist *p,
uint32_t *idx) {
pa_scache_entry *e;
char st[PA_SAMPLE_SPEC_SNPRINT_MAX];
pa_channel_map tmap;
pa_assert(c);
pa_assert(name);
pa_assert(!ss || pa_sample_spec_valid(ss));
pa_assert(!map || (pa_channel_map_valid(map) && ss && pa_channel_map_compatible(map, ss)));
if (ss && !map) {
pa_channel_map_init_extend(&tmap, ss->channels, PA_CHANNEL_MAP_DEFAULT);
map = &tmap;
}
if (chunk && chunk->length > PA_SCACHE_ENTRY_SIZE_MAX)
return -1;
if (!(e = scache_add_item(c, name)))
return -1;
pa_sample_spec_init(&e->sample_spec);
pa_channel_map_init(&e->channel_map);
pa_cvolume_init(&e->volume);
e->volume_is_set = FALSE;
if (ss) {
e->sample_spec = *ss;
pa_cvolume_reset(&e->volume, ss->channels);
}
if (map)
e->channel_map = *map;
if (chunk) {
e->memchunk = *chunk;
pa_memblock_ref(e->memchunk.memblock);
}
if (p)
pa_proplist_update(e->proplist, PA_UPDATE_REPLACE, p);
if (idx)
*idx = e->index;
pa_log_debug("Created sample \"%s\" (#%d), %lu bytes with sample spec %s",
name, e->index, (unsigned long) e->memchunk.length,
pa_sample_spec_snprint(st, sizeof(st), &e->sample_spec));
return 0;
}
static pa_scache_entry* scache_add_item(pa_core *c, const char *name) {
pa_scache_entry *e;
pa_assert(c);
pa_assert(name);
if ((e = pa_namereg_get(c, name, PA_NAMEREG_SAMPLE))) {
if (e->memchunk.memblock)
pa_memblock_unref(e->memchunk.memblock);
pa_xfree(e->filename);
pa_proplist_clear(e->proplist);
pa_assert(e->core == c);
pa_subscription_post(c, PA_SUBSCRIPTION_EVENT_SAMPLE_CACHE|PA_SUBSCRIPTION_EVENT_CHANGE, e->index);
} else {
e = pa_xnew(pa_scache_entry, 1);
if (!pa_namereg_register(c, name, PA_NAMEREG_SAMPLE, e, TRUE)) {
pa_xfree(e);
return NULL;
}
e->name = pa_xstrdup(name);
e->core = c;
e->proplist = pa_proplist_new();
pa_idxset_put(c->scache, e, &e->index);
pa_subscription_post(c, PA_SUBSCRIPTION_EVENT_SAMPLE_CACHE|PA_SUBSCRIPTION_EVENT_NEW, e->index);
}
e->last_used_time = 0;
pa_memchunk_reset(&e->memchunk);
e->filename = NULL;
e->lazy = FALSE;
e->last_used_time = 0;
pa_sample_spec_init(&e->sample_spec);
pa_channel_map_init(&e->channel_map);
pa_cvolume_init(&e->volume);
e->volume_is_set = FALSE;
pa_proplist_sets(e->proplist, PA_PROP_MEDIA_ROLE, "event");
return e;
}
static void
gst_pulsesrc_init (GstPulseSrc * pulsesrc)
{
pulsesrc->server = NULL;
pulsesrc->device = NULL;
pulsesrc->client_name = gst_pulse_client_name ();
pulsesrc->device_description = NULL;
pulsesrc->context = NULL;
pulsesrc->stream = NULL;
pulsesrc->stream_connected = FALSE;
pulsesrc->source_output_idx = PA_INVALID_INDEX;
pulsesrc->read_buffer = NULL;
pulsesrc->read_buffer_length = 0;
pa_sample_spec_init (&pulsesrc->sample_spec);
pulsesrc->operation_success = FALSE;
pulsesrc->paused = TRUE;
pulsesrc->in_read = FALSE;
pulsesrc->volume = DEFAULT_VOLUME;
pulsesrc->volume_set = FALSE;
pulsesrc->mute = DEFAULT_MUTE;
pulsesrc->mute_set = FALSE;
pulsesrc->notify = 0;
pulsesrc->properties = NULL;
pulsesrc->proplist = NULL;
pulsesrc->probe = gst_pulseprobe_new (G_OBJECT (pulsesrc), G_OBJECT_GET_CLASS (pulsesrc), PROP_DEVICE, pulsesrc->server, FALSE, TRUE); /* FALSE for sinks, TRUE for sources */
/* this should be the default but it isn't yet */
gst_audio_base_src_set_slave_method (GST_AUDIO_BASE_SRC (pulsesrc),
GST_AUDIO_BASE_SRC_SLAVE_SKEW);
/* override with a custom clock */
if (GST_AUDIO_BASE_SRC (pulsesrc)->clock)
gst_object_unref (GST_AUDIO_BASE_SRC (pulsesrc)->clock);
GST_AUDIO_BASE_SRC (pulsesrc)->clock =
gst_audio_clock_new ("GstPulseSrcClock",
(GstAudioClockGetTimeFunc) gst_pulsesrc_get_time, pulsesrc, NULL);
}
