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gst-libs/gst/audio/: update strings, values are in microseconds change the default sink buffer time to something that...

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Original commit message from CVS:

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
update strings, values are in microseconds
change the default sink buffer time to something that is smaller
(to help software volume mixing have a slightly lower delay) but
still be acceptable on Wim's laptop
This commit is contained in:
Thomas Vander Stichele 2005-12-20 12:00:26 +00:00
parent ffc33e35d9
commit b4b2b62a74
3 changed files with 41 additions and 13 deletions
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11
ChangeLog
View file

@ -1,3 +1,14 @@
2005-12-20 Thomas Vander Stichele <thomas at apestaart dot org>
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
update strings, values are in microseconds
change the default sink buffer time to something that is smaller
(to help software volume mixing have a slightly lower delay) but
still be acceptable on Wim's laptop
2005-12-20 Edward Hervey <edward@fluendo.com> 2005-12-20 Edward Hervey <edward@fluendo.com>
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_template_caps): * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_template_caps):

23
gst-libs/gst/audio/gstbaseaudiosink.c
View file

@ -39,7 +39,7 @@ enum
* and sample offset position. */ * and sample offset position. */
#define DIFF_TOLERANCE 10 #define DIFF_TOLERANCE 10
#define DEFAULT_BUFFER_TIME 500 * GST_USECOND #define DEFAULT_BUFFER_TIME 200 * GST_USECOND
#define DEFAULT_LATENCY_TIME 10 * GST_USECOND #define DEFAULT_LATENCY_TIME 10 * GST_USECOND
#define DEFAULT_PROVIDE_CLOCK TRUE #define DEFAULT_PROVIDE_CLOCK TRUE
@ -94,6 +94,8 @@ gst_base_audio_sink_base_init (gpointer g_class)
static void static void
gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass) gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass)
{ {
gchar *longdesc;
GObjectClass *gobject_class; GObjectClass *gobject_class;
GstElementClass *gstelement_class; GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class; GstBaseSinkClass *gstbasesink_class;
@ -108,14 +110,21 @@ gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass)
GST_DEBUG_FUNCPTR (gst_base_audio_sink_get_property); GST_DEBUG_FUNCPTR (gst_base_audio_sink_get_property);
gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_base_audio_sink_dispose); gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_base_audio_sink_dispose);
longdesc =
g_strdup_printf
("Size of audio buffer in microseconds (use -1 for default of %"
G_GUINT64_FORMAT " µs)", DEFAULT_BUFFER_TIME / GST_USECOND);
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BUFFER_TIME, g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BUFFER_TIME,
g_param_spec_int64 ("buffer-time", "Buffer Time", g_param_spec_int64 ("buffer-time", "Buffer Time", longdesc, -1,
"Size of audio buffer in milliseconds (-1 = default)", G_MAXINT64, DEFAULT_BUFFER_TIME, G_PARAM_READWRITE));
-1, G_MAXINT64, DEFAULT_BUFFER_TIME, G_PARAM_READWRITE)); g_free (longdesc);
longdesc =
g_strdup_printf ("Audio latency in microseconds (use -1 for default of %"
G_GUINT64_FORMAT " µs)", DEFAULT_LATENCY_TIME / GST_USECOND);
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_LATENCY_TIME, g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_LATENCY_TIME,
g_param_spec_int64 ("latency-time", "Latency Time", g_param_spec_int64 ("latency-time", "Latency Time", longdesc, -1,
"Audio latency in milliseconds (-1 = default)", G_MAXINT64, DEFAULT_LATENCY_TIME, G_PARAM_READWRITE));
-1, G_MAXINT64, DEFAULT_LATENCY_TIME, G_PARAM_READWRITE)); g_free (longdesc);
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PROVIDE_CLOCK, g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PROVIDE_CLOCK,
g_param_spec_boolean ("provide-clock", "Provide Clock", g_param_spec_boolean ("provide-clock", "Provide Clock",
"Provide a clock to be used as the global pipeline clock", "Provide a clock to be used as the global pipeline clock",

20
gst-libs/gst/audio/gstbaseaudiosrc.c
View file

@ -81,6 +81,7 @@ gst_base_audio_src_base_init (gpointer g_class)
static void static void
gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass) gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass)
{ {
gchar *longdesc;
GObjectClass *gobject_class; GObjectClass *gobject_class;
GstElementClass *gstelement_class; GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class; GstBaseSrcClass *gstbasesrc_class;
@ -96,14 +97,21 @@ gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass)
gobject_class->get_property = gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_base_audio_src_get_property); GST_DEBUG_FUNCPTR (gst_base_audio_src_get_property);
longdesc =
g_strdup_printf
("Size of audio buffer in microseconds (use -1 for default of %"
G_GUINT64_FORMAT " µs)", DEFAULT_BUFFER_TIME / GST_USECOND);
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BUFFER_TIME, g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BUFFER_TIME,
g_param_spec_int64 ("buffer-time", "Buffer Time", g_param_spec_int64 ("buffer-time", "Buffer Time", longdesc, -1,
"Size of audio buffer in milliseconds (-1 = default)", G_MAXINT64, DEFAULT_BUFFER_TIME, G_PARAM_READWRITE));
-1, G_MAXINT64, DEFAULT_BUFFER_TIME, G_PARAM_READWRITE)); g_free (longdesc);
longdesc =
g_strdup_printf ("Audio latency in microseconds (use -1 for default of %"
G_GUINT64_FORMAT " µs)", DEFAULT_LATENCY_TIME / GST_USECOND);
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_LATENCY_TIME, g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_LATENCY_TIME,
g_param_spec_int64 ("latency-time", "Latency Time", g_param_spec_int64 ("latency-time", "Latency Time", longdesc, -1,
"Audio latency in milliseconds (-1 = default)", G_MAXINT64, DEFAULT_LATENCY_TIME, G_PARAM_READWRITE));
-1, G_MAXINT64, DEFAULT_LATENCY_TIME, G_PARAM_READWRITE)); g_free (longdesc);
gstelement_class->change_state = gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_base_audio_src_change_state); GST_DEBUG_FUNCPTR (gst_base_audio_src_change_state);