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gst-libs/gst/audio/gstbaseaudiosink.h: Clarify some docs.

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Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.h:
Clarify some docs.
* gst-libs/gst/audio/gstbaseaudiosrc.c: (slave_method_get_type),
(gst_base_audio_src_class_init), (gst_base_audio_src_init),
(gst_base_audio_src_set_slave_method),
(gst_base_audio_src_get_slave_method),
(gst_base_audio_src_set_property),
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
Add property and methods for selecting the clock slave method in the
source, like in the sink.
We only implement "none" and "re-timestamp" for now.
API: gst_base_audio_src_set_slave_method()
API: gst_base_audio_src_get_slave_method()
This commit is contained in:
Wim Taymans 2008-04-28 08:51:38 +00:00
parent 0c5d01b59f
commit 7916e386ca
4 changed files with 162 additions and 18 deletions
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18
ChangeLog
View file

@ -1,3 +1,21 @@
2008-04-28 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.h:
Clarify some docs.
* gst-libs/gst/audio/gstbaseaudiosrc.c: (slave_method_get_type),
(gst_base_audio_src_class_init), (gst_base_audio_src_init),
(gst_base_audio_src_set_slave_method),
(gst_base_audio_src_get_slave_method),
(gst_base_audio_src_set_property),
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
Add property and methods for selecting the clock slave method in the
source, like in the sink.
We only implement "none" and "re-timestamp" for now.
API: gst_base_audio_src_set_slave_method()
API: gst_base_audio_src_get_slave_method()
2008-04-25 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstrtpbuffer.c:

3
gst-libs/gst/audio/gstbaseaudiosink.h
View file

@ -85,7 +85,8 @@ G_BEGIN_DECLS
* drifts too much.
* @GST_BASE_AUDIO_SINK_SLAVE_NONE: No adjustment is done.
*
* Different possible clock slaving algorithms
* Different possible clock slaving algorithms used when the internal audio
* clock is not selected as the pipeline master clock.
*/
typedef enum
{

133
gst-libs/gst/audio/gstbaseaudiosrc.c
View file

@ -45,12 +45,36 @@
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 */
@ -63,13 +87,16 @@ enum
#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
@ -157,6 +184,12 @@ gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass)
"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 =
@ -187,6 +220,7 @@ gst_base_audio_src_init (GstBaseAudioSrc * 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;
@ -341,6 +375,50 @@ gst_base_audio_src_get_provide_clock (GstBaseAudioSrc * src)
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)
@ -359,6 +437,9 @@ gst_base_audio_src_set_property (GObject * object, guint prop_id,
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;
@ -383,6 +464,9 @@ gst_base_audio_src_get_property (GObject * object, guint prop_id,
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;
@ -682,25 +766,40 @@ gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
GST_OBJECT_LOCK (src);
clock = GST_ELEMENT_CLOCK (src);
if (clock != NULL && clock != src->clock) {
GstClockTime base_time, latency;
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 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;
/* 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;
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;
/* 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);

26
gst-libs/gst/audio/gstbaseaudiosrc.h
View file

@ -59,6 +59,26 @@ typedef struct _GstBaseAudioSrc GstBaseAudioSrc;
typedef struct _GstBaseAudioSrcClass GstBaseAudioSrcClass;
typedef struct _GstBaseAudioSrcPrivate GstBaseAudioSrcPrivate;
/**
* GstBaseAudioSrcSlaveMethod:
* @GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE: Resample to match the master clock.
* @GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP: Retimestamp output buffers with master
* clock time.
* @GST_BASE_AUDIO_SRC_SLAVE_SKEW: Adjust capture pointer when master clock
* drifts too much.
* @GST_BASE_AUDIO_SRC_SLAVE_NONE: No adjustment is done.
*
* Different possible clock slaving algorithms when the internal audio clock was
* not selected as the pipeline clock.
*/
typedef enum
{
GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE,
GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP,
GST_BASE_AUDIO_SRC_SLAVE_SKEW,
GST_BASE_AUDIO_SRC_SLAVE_NONE
} GstBaseAudioSrcSlaveMethod;
/**
* GstBaseAudioSrc:
*
@ -112,6 +132,12 @@ GstRingBuffer *gst_base_audio_src_create_ringbuffer (GstBaseAudioSrc *src);
void gst_base_audio_src_set_provide_clock (GstBaseAudioSrc *src, gboolean provide);
gboolean gst_base_audio_src_get_provide_clock (GstBaseAudioSrc *src);
void gst_base_audio_src_set_slave_method (GstBaseAudioSrc *src,
GstBaseAudioSrcSlaveMethod method);
GstBaseAudioSrcSlaveMethod
gst_base_audio_src_get_slave_method (GstBaseAudioSrc *src);
G_END_DECLS
#endif /* __GST_BASE_AUDIO_SRC_H__ */