static void
gst_wasapi_src_init (GstWasapiSrc * self, GstWasapiSrcClass * gclass)
{
GstBaseSrc *basesrc = GST_BASE_SRC (self);
gst_base_src_set_format (basesrc, GST_FORMAT_TIME);
gst_base_src_set_live (basesrc, TRUE);
self->rate = 8000;
self->buffer_time = 20 * GST_MSECOND;
self->period_time = 20 * GST_MSECOND;
self->latency = GST_CLOCK_TIME_NONE;
self->samples_per_buffer = self->rate / (GST_SECOND / self->period_time);
self->start_time = GST_CLOCK_TIME_NONE;
self->next_time = GST_CLOCK_TIME_NONE;
#if GST_CHECK_VERSION(0, 10, 31) || (GST_CHECK_VERSION(0, 10, 30) && GST_VERSION_NANO > 0)
self->clock = gst_audio_clock_new_full ("GstWasapiSrcClock",
gst_wasapi_src_get_time, gst_object_ref (self),
(GDestroyNotify) gst_object_unref);
#else
self->clock = gst_audio_clock_new ("GstWasapiSrcClock",
gst_wasapi_src_get_time, self);
#endif
CoInitialize (NULL);
}
static void
gst_dasf_src_init (GstDasfSrc* self, GstDasfSrcClass* klass)
{
GST_DEBUG ("");
GstBaseAudioSrc *baseaudiosrc;
baseaudiosrc = GST_BASE_AUDIO_SRC (self);
GstDasfSrcPrivate* priv = GST_DASF_SRC_GET_PRIVATE (self);
priv->num_output_buffers = NUM_OUTPUT_BUFFERS_DEFAULT;
priv->incount = 0;
priv->outcount = 0;
priv->volume = 100;
priv->mute = FALSE;
baseaudiosrc->clock = gst_audio_clock_new ("GstDasfSrcClock", (GstAudioClockGetTimeFunc) gst_dasf_src_get_time, self);
self->component = NULL;
self->tracks = NULL;
GstMixerTrack* track = g_object_new (GST_TYPE_MIXER_TRACK, NULL);
track->label = g_strdup ("DASF Volume");
track->num_channels = 1;
track->min_volume = 0;
track->max_volume = 100;
track->flags = GST_MIXER_TRACK_MASTER | GST_MIXER_TRACK_MUTE;
self->tracks = g_slist_append (self->tracks, track);
return;
}
static void
alsaspdifsink_init (AlsaSPDIFSink * sink, AlsaSPDIFSinkClass * g_class)
{
/* Create the provided clock. */
sink->clock = gst_audio_clock_new ("clock", alsaspdifsink_get_time, sink);
sink->card = 0;
sink->device = g_strdup ("default");
}
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);
}
static void
gst_base_audio_src_init (GstBaseAudioSrc * baseaudiosrc,
GstBaseAudioSrcClass * g_class)
{
baseaudiosrc->priv = GST_BASE_AUDIO_SRC_GET_PRIVATE (baseaudiosrc);
baseaudiosrc->buffer_time = DEFAULT_BUFFER_TIME;
baseaudiosrc->latency_time = DEFAULT_LATENCY_TIME;
baseaudiosrc->priv->provide_clock = DEFAULT_PROVIDE_CLOCK;
baseaudiosrc->priv->slave_method = DEFAULT_SLAVE_METHOD;
/* reset blocksize we use latency time to calculate a more useful
* value based on negotiated format. */
GST_BASE_SRC (baseaudiosrc)->blocksize = 0;
baseaudiosrc->clock = gst_audio_clock_new ("GstAudioSrcClock",
(GstAudioClockGetTimeFunc) gst_base_audio_src_get_time, baseaudiosrc);
/* we are always a live source */
gst_base_src_set_live (GST_BASE_SRC (baseaudiosrc), TRUE);
/* we operate in time */
gst_base_src_set_format (GST_BASE_SRC (baseaudiosrc), GST_FORMAT_TIME);
}
