GST_DEBUG_CATEGORY_STATIC (gst_base_audio_src_debug);
#define GST_CAT_DEFAULT gst_base_audio_src_debug
+#define GST_TYPE_SLAVE_METHOD (slave_method_get_type ())
+
+static GType
+slave_method_get_type (void)
+{
+ static GType slave_method_type = 0;
+ static const GEnumValue slave_method[] = {
+ {GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE, "Resampling slaving", "resample"},
+ {GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP, "Re-timestamp", "re-timestamp"},
+ {GST_BASE_AUDIO_SRC_SLAVE_SKEW, "Skew", "skew"},
+ {GST_BASE_AUDIO_SRC_SLAVE_NONE, "No slaving", "none"},
+ {0, NULL, NULL},
+ };
+
+ if (!slave_method_type) {
+ slave_method_type =
+ g_enum_register_static ("GstBaseAudioSrcSlaveMethod", slave_method);
+ }
+ return slave_method_type;
+}
+
#define GST_BASE_AUDIO_SRC_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_AUDIO_SRC, GstBaseAudioSrcPrivate))
struct _GstBaseAudioSrcPrivate
{
gboolean provide_clock;
+
+ /* the clock slaving algorithm in use */
+ GstBaseAudioSrcSlaveMethod slave_method;
};
/* BaseAudioSrc signals and args */
#define DEFAULT_BUFFER_TIME ((200 * GST_MSECOND) / GST_USECOND)
#define DEFAULT_LATENCY_TIME ((10 * GST_MSECOND) / GST_USECOND)
#define DEFAULT_PROVIDE_CLOCK TRUE
+#define DEFAULT_SLAVE_METHOD GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP
enum
{
PROP_0,
PROP_BUFFER_TIME,
PROP_LATENCY_TIME,
- PROP_PROVIDE_CLOCK
+ PROP_PROVIDE_CLOCK,
+ PROP_SLAVE_METHOD,
+ PROP_LAST
};
static void
"Provide a clock to be used as the global pipeline clock",
DEFAULT_PROVIDE_CLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_SLAVE_METHOD,
+ g_param_spec_enum ("slave-method", "Slave Method",
+ "Algorithm to use to match the rate of the masterclock",
+ GST_TYPE_SLAVE_METHOD, DEFAULT_SLAVE_METHOD,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_base_audio_src_change_state);
gstelement_class->provide_clock =
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;
return result;
}
+/**
+ * gst_base_audio_src_set_slave_method:
+ * @src: a #GstBaseAudioSrc
+ * @method: the new slave method
+ *
+ * Controls how clock slaving will be performed in @src.
+ *
+ * Since: 0.10.20
+ */
+void
+gst_base_audio_src_set_slave_method (GstBaseAudioSrc * src,
+ GstBaseAudioSrcSlaveMethod method)
+{
+ g_return_if_fail (GST_IS_BASE_AUDIO_SRC (src));
+
+ GST_OBJECT_LOCK (src);
+ src->priv->slave_method = method;
+ GST_OBJECT_UNLOCK (src);
+}
+
+/**
+ * gst_base_audio_src_get_slave_method:
+ * @src: a #GstBaseAudioSrc
+ *
+ * Get the current slave method used by @src.
+ *
+ * Returns: The current slave method used by @src.
+ *
+ * Since: 0.10.20
+ */
+GstBaseAudioSrcSlaveMethod
+gst_base_audio_src_get_slave_method (GstBaseAudioSrc * src)
+{
+ GstBaseAudioSrcSlaveMethod result;
+
+ g_return_val_if_fail (GST_IS_BASE_AUDIO_SRC (src), -1);
+
+ GST_OBJECT_LOCK (src);
+ result = src->priv->slave_method;
+ GST_OBJECT_UNLOCK (src);
+
+ return result;
+}
+
static void
gst_base_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
case PROP_PROVIDE_CLOCK:
gst_base_audio_src_set_provide_clock (src, g_value_get_boolean (value));
break;
+ case PROP_SLAVE_METHOD:
+ gst_base_audio_src_set_slave_method (src, g_value_get_enum (value));
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
case PROP_PROVIDE_CLOCK:
g_value_set_boolean (value, gst_base_audio_src_get_provide_clock (src));
break;
+ case PROP_SLAVE_METHOD:
+ g_value_set_enum (value, gst_base_audio_src_get_slave_method (src));
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
GST_OBJECT_LOCK (src);
clock = GST_ELEMENT_CLOCK (src);
if (clock != NULL && clock != src->clock) {
- GstClockTime base_time, latency;
-
- /* We are slaved to another clock, take running time of the clock and just
- * timestamp against it. Somebody else in the pipeline should figure out the
- * clock drift, for now. We keep the duration we calculated above. */
- timestamp = gst_clock_get_time (clock);
- base_time = GST_ELEMENT_CAST (src)->base_time;
-
- if (timestamp > base_time)
- timestamp -= base_time;
- else
- timestamp = 0;
-
- /* subtract latency */
- latency = gst_util_uint64_scale_int (total_samples, GST_SECOND, spec->rate);
- if (timestamp > latency)
- timestamp -= latency;
- else
- timestamp = 0;
+ switch (src->priv->slave_method) {
+ case GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE:
+ /* not implemented, use retimestamp algorithm. This algorithm should
+ * work on the readout pointer and produces more or less samples based
+ * on the clock drift */
+ case GST_BASE_AUDIO_SRC_SLAVE_SKEW:
+ /* not implemented, use retimestamp algorithm. This algortihm should work
+ * on the readout pointer above and creates small jumps when needed. */
+ case GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP:
+ {
+ GstClockTime base_time, latency;
+
+ /* We are slaved to another clock, take running time of the pipeline clock and
+ * timestamp against it. Somebody else in the pipeline should figure out the
+ * clock drift. We keep the duration we calculated above. */
+ timestamp = gst_clock_get_time (clock);
+ base_time = GST_ELEMENT_CAST (src)->base_time;
+
+ if (timestamp > base_time)
+ timestamp -= base_time;
+ else
+ timestamp = 0;
+
+ /* subtract latency */
+ latency =
+ gst_util_uint64_scale_int (total_samples, GST_SECOND, spec->rate);
+ if (timestamp > latency)
+ timestamp -= latency;
+ else
+ timestamp = 0;
+ }
+ case GST_BASE_AUDIO_SRC_SLAVE_NONE:
+ break;
+ }
}
GST_OBJECT_UNLOCK (src);
projects / platform / upstream / gst-plugins-base.git / commitdiff