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gst-libs/gst/audio/gstaudioclock.c: This clock can be slaved to a master clock now.
Original commit message from CVS: * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init): This clock can be slaved to a master clock now. * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_class_init), (gst_base_audio_sink_init), (gst_base_audio_sink_dispose), (gst_base_audio_sink_provide_clock), (gst_base_audio_sink_set_clock), (gst_base_audio_sink_set_property), (gst_base_audio_sink_get_property), (gst_base_audio_sink_preroll), (gst_base_audio_sink_render), (gst_base_audio_sink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: Handle slaving the internal clock to the clock selected in the pipeline. Add property to make the basesink not provide a clock. * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_wait): * gst-libs/gst/rtp/gstbasertpdepayload.h: We can use the clock in GstElement, no need to store it ourselves.
This commit is contained in:
Wim Taymans
parent
c0759093eb
commit
a3cb4d4937
6 changed files with 78 additions and 42 deletions
23
ChangeLog
23
ChangeLog
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@ -1,3 +1,26 @@
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2005-11-22 Wim Taymans <wim@fluendo.com>
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* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init):
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This clock can be slaved to a master clock now.
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
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(gst_base_audio_sink_dispose), (gst_base_audio_sink_provide_clock),
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(gst_base_audio_sink_set_clock),
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(gst_base_audio_sink_set_property),
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(gst_base_audio_sink_get_property), (gst_base_audio_sink_preroll),
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(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
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* gst-libs/gst/audio/gstbaseaudiosink.h:
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Handle slaving the internal clock to the clock selected in the
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pipeline.
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Add property to make the basesink not provide a clock.
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* gst-libs/gst/rtp/gstbasertpdepayload.c:
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(gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
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(gst_base_rtp_depayload_wait):
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* gst-libs/gst/rtp/gstbasertpdepayload.h:
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We can use the clock in GstElement, no need to store it ourselves.
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2005-11-22 Thomas Vander Stichele <thomas at apestaart dot org>
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* docs/libs/tmpl/gstaudio.sgml:
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@ -83,6 +83,7 @@ gst_audio_clock_init (GstAudioClock * clock)
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gst_object_set_name (GST_OBJECT (clock), "GstAudioClock");
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clock->last_time = 0;
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GST_OBJECT_FLAG_SET (clock, GST_CLOCK_FLAG_CAN_SET_MASTER);
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}
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GstClock *
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@ -41,11 +41,15 @@ enum
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#define DEFAULT_BUFFER_TIME 500 * GST_USECOND
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#define DEFAULT_LATENCY_TIME 10 * GST_USECOND
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//#define DEFAULT_PROVIDE_CLOCK TRUE
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#define DEFAULT_PROVIDE_CLOCK FALSE
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enum
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{
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PROP_0,
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PROP_BUFFER_TIME,
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PROP_LATENCY_TIME,
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PROP_PROVIDE_CLOCK,
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};
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#define _do_init(bla) \
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@ -65,7 +69,8 @@ static GstStateChangeReturn gst_base_audio_sink_change_state (GstElement *
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element, GstStateChange transition);
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static GstClock *gst_base_audio_sink_provide_clock (GstElement * elem);
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static void gst_base_audio_sink_set_clock (GstElement * elem, GstClock * clock);
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static gboolean gst_base_audio_sink_set_clock (GstElement * elem,
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GstClock * clock);
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static GstClockTime gst_base_audio_sink_get_time (GstClock * clock,
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GstBaseAudioSink * sink);
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static void gst_base_audio_sink_callback (GstRingBuffer * rbuf, guint8 * data,
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@ -114,6 +119,10 @@ gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass)
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g_param_spec_int64 ("latency-time", "Latency Time",
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"Audio latency in milliseconds (-1 = default)",
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-1, G_MAXINT64, DEFAULT_LATENCY_TIME, G_PARAM_READWRITE));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PROVIDE_CLOCK,
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g_param_spec_boolean ("provide-clock", "Provide Clock",
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"Provide a clock to be used as the global pipeline clock",
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DEFAULT_PROVIDE_CLOCK, G_PARAM_READWRITE));
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_base_audio_sink_change_state);
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@ -136,8 +145,9 @@ gst_base_audio_sink_init (GstBaseAudioSink * baseaudiosink,
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{
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baseaudiosink->buffer_time = DEFAULT_BUFFER_TIME;
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baseaudiosink->latency_time = DEFAULT_LATENCY_TIME;
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baseaudiosink->provide_clock = DEFAULT_PROVIDE_CLOCK;
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baseaudiosink->clock = gst_audio_clock_new ("clock",
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baseaudiosink->provided_clock = gst_audio_clock_new ("clock",
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(GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time, baseaudiosink);
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}
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@ -148,9 +158,9 @@ gst_base_audio_sink_dispose (GObject * object)
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sink = GST_BASE_AUDIO_SINK (object);
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if (sink->clock)
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gst_object_unref (sink->clock);
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sink->clock = NULL;
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if (sink->provided_clock)
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gst_object_unref (sink->provided_clock);
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sink->provided_clock = NULL;
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if (sink->ringbuffer)
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gst_object_unref (sink->ringbuffer);
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@ -167,25 +177,31 @@ gst_base_audio_sink_provide_clock (GstElement * elem)
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sink = GST_BASE_AUDIO_SINK (elem);
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clock = GST_CLOCK_CAST (gst_object_ref (sink->clock));
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if (sink->provide_clock)
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clock = GST_CLOCK_CAST (gst_object_ref (sink->provided_clock));
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else
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clock = NULL;
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return clock;
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}
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static void
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static gboolean
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gst_base_audio_sink_set_clock (GstElement * elem, GstClock * clock)
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{
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GstBaseAudioSink *sink;
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gboolean ret;
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sink = GST_BASE_AUDIO_SINK (elem);
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GST_OBJECT_LOCK (sink);
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if (clock != sink->clock) {
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gst_clock_set_master (sink->clock, clock);
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if (clock != sink->provided_clock) {
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ret = gst_clock_set_master (sink->provided_clock, clock);
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} else {
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gst_clock_set_master (sink->clock, NULL);
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ret = gst_clock_set_master (sink->provided_clock, NULL);
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}
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GST_OBJECT_UNLOCK (sink);
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return ret;
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}
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static GstClockTime
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@ -220,6 +236,9 @@ gst_base_audio_sink_set_property (GObject * object, guint prop_id,
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case PROP_LATENCY_TIME:
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sink->latency_time = g_value_get_int64 (value);
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break;
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case PROP_PROVIDE_CLOCK:
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sink->provide_clock = g_value_get_boolean (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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@ -241,6 +260,9 @@ gst_base_audio_sink_get_property (GObject * object, guint prop_id,
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case PROP_LATENCY_TIME:
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g_value_set_int64 (value, sink->latency_time);
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break;
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case PROP_PROVIDE_CLOCK:
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g_value_set_boolean (value, sink->provide_clock);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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@ -350,8 +372,8 @@ wrong_state:
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{
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GST_DEBUG ("ringbuffer in wrong state");
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GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
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("sink not negotiated."), (NULL));
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return GST_FLOW_ERROR;
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("sink not negotiated."), ("sink not negotiated."));
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return GST_FLOW_NOT_NEGOTIATED;
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}
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}
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@ -441,7 +463,8 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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if (render_diff < 0)
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goto out_of_segment;
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gst_clock_get_calibration (sink->clock, &cinternal, &cexternal, &crate);
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gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
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&crate);
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GST_DEBUG_OBJECT (sink,
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"internal %" G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT ", rate %g",
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cinternal, cexternal, crate);
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@ -567,18 +590,23 @@ gst_base_audio_sink_change_state (GstElement * element,
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break;
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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{
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GstClockTime time;
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gdouble rate;
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/* if we are slaved to a clock, we need to set the initial
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* calibration */
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/* FIXME, this is not yet accurate enough for smooth playback */
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if (gst_clock_get_master (sink->provided_clock)) {
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GstClockTime time;
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gdouble rate;
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time = gst_clock_get_internal_time (sink->clock);
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time = gst_clock_get_internal_time (sink->provided_clock);
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GST_DEBUG_OBJECT (sink, "time: %" GST_TIME_FORMAT, GST_TIME_ARGS (time));
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GST_DEBUG_OBJECT (sink, "time: %" GST_TIME_FORMAT,
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GST_TIME_ARGS (time));
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gst_clock_get_calibration (sink->clock, NULL, NULL, &rate);
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/* Does not work yet.
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gst_clock_set_calibration (sink->clock,
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time, element->base_time, rate);
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*/
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gst_clock_get_calibration (sink->provided_clock, NULL, NULL, &rate);
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/* Does not work yet. */
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gst_clock_set_calibration (sink->provided_clock,
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time, element->base_time, rate);
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}
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break;
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}
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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@ -84,7 +84,8 @@ struct _GstBaseAudioSink {
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guint64 next_sample;
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/* clock */
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GstClock *clock;
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gboolean provide_clock;
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GstClock *provided_clock;
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING];
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@ -80,8 +80,6 @@ static gboolean gst_base_rtp_depayload_setcaps (GstPad * pad, GstCaps * caps);
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static GstFlowReturn gst_base_rtp_depayload_chain (GstPad * pad,
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GstBuffer * in);
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static void gst_base_rtp_depayload_set_clock (GstElement * element,
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GstClock * clock);
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static GstStateChangeReturn gst_base_rtp_depayload_change_state (GstElement *
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element, GstStateChange transition);
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static GstFlowReturn gst_base_rtp_depayload_add_to_queue (GstBaseRTPDepayload *
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@ -118,8 +116,6 @@ gst_base_rtp_depayload_class_init (GstBaseRTPDepayloadClass * klass)
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gobject_class->finalize = gst_base_rtp_depayload_finalize;
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gstelement_class->set_clock =
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GST_DEBUG_FUNCPTR (gst_base_rtp_depayload_set_clock);
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gstelement_class->change_state = gst_base_rtp_depayload_change_state;
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klass->add_to_queue = gst_base_rtp_depayload_add_to_queue;
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@ -158,7 +154,6 @@ gst_base_rtp_depayload_init (GstBaseRTPDepayload * filter, gpointer g_class)
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/* this one needs to be overwritten by child */
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filter->clock_rate = 0;
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filter->clock = NULL;
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}
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static void
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@ -399,27 +394,17 @@ gst_base_rtp_depayload_wait (GstBaseRTPDepayload * filter, GstClockTime time)
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GstClockID id;
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g_return_if_fail (GST_CLOCK_TIME_IS_VALID (time));
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if (filter->clock == NULL) {
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if (GST_ELEMENT_CLOCK (filter) == NULL) {
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GST_DEBUG_OBJECT (filter, "No clock given yet");
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return;
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}
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id = gst_clock_new_single_shot_id (filter->clock, time);
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id = gst_clock_new_single_shot_id (GST_ELEMENT_CLOCK (filter), time);
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gst_clock_id_wait (id, NULL);
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gst_clock_id_unref (id);
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}
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static void
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gst_base_rtp_depayload_set_clock (GstElement * element, GstClock * clock)
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{
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GstBaseRTPDepayload *sink;
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sink = GST_BASE_RTP_DEPAYLOAD (element);
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sink->clock = clock;
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}
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static GstStateChangeReturn
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gst_base_rtp_depayload_change_state (GstElement * element,
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GstStateChange transition)
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@ -78,8 +78,6 @@ struct _GstBaseRTPDepayload
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*/
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GQueue *queue;
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GstClock *clock;
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/*< private >*/
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union {
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gboolean need_newsegment;
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