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gst-libs/gst/audio/gstbaseaudiosink.*: Store private stuff in GstBaseAudioSinkPrivate.
Original commit message from CVS: * gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type), (gst_base_audio_sink_class_init), (gst_base_audio_sink_init), (gst_base_audio_sink_query), (gst_base_audio_sink_get_time), (gst_base_audio_sink_set_property), (gst_base_audio_sink_get_property), (gst_base_audio_sink_event), (clock_convert_external), (gst_base_audio_sink_resample_slaving), (gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render), (gst_base_audio_sink_async_play): * gst-libs/gst/audio/gstbaseaudiosink.h: Store private stuff in GstBaseAudioSinkPrivate. Add configurable clock slaving modes property. API:: GstBaseAudioSink::slave-method property Some more latency reporting tweaks. Added skew based clock slaving correction and make it the default until the resampling method is more robust.
This commit is contained in:
Wim Taymans
parent
293a9c09b8
commit
450030ebaf
3 changed files with 310 additions and 92 deletions
19
ChangeLog
19
ChangeLog
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@ -1,3 +1,22 @@
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2007-03-28 Wim Taymans <wim@fluendo.com>
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* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
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(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
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(gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
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(gst_base_audio_sink_set_property),
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(gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
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(clock_convert_external), (gst_base_audio_sink_resample_slaving),
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(gst_base_audio_sink_skew_slaving),
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(gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
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(gst_base_audio_sink_async_play):
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* gst-libs/gst/audio/gstbaseaudiosink.h:
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Store private stuff in GstBaseAudioSinkPrivate.
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Add configurable clock slaving modes property.
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API:: GstBaseAudioSink::slave-method property
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Some more latency reporting tweaks.
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Added skew based clock slaving correction and make it the default until
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the resampling method is more robust.
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2007-03-27 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/audioconvert/audioconvert.c:
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@ -39,6 +39,19 @@
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GST_DEBUG_CATEGORY_STATIC (gst_base_audio_sink_debug);
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#define GST_CAT_DEFAULT gst_base_audio_sink_debug
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#define GST_BASE_AUDIO_SINK_GET_PRIVATE(obj) \
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(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_AUDIO_SINK, GstBaseAudioSinkPrivate))
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struct _GstBaseAudioSinkPrivate
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{
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/* upstream latency */
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GstClockTime us_latency;
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/* the clock slaving algorithm in use */
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GstBaseAudioSinkSlaveMethod slave_method;
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/* running average of clock skew */
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GstClockTimeDiff avg_skew;
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};
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/* BaseAudioSink signals and args */
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enum
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{
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@ -59,6 +72,7 @@ enum
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#define DEFAULT_BUFFER_TIME ((200 * GST_MSECOND) / GST_USECOND)
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#define DEFAULT_LATENCY_TIME ((10 * GST_MSECOND) / GST_USECOND)
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#define DEFAULT_PROVIDE_CLOCK TRUE
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#define DEFAULT_SLAVE_METHOD GST_BASE_AUDIO_SINK_SLAVE_SKEW
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enum
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{
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@ -66,8 +80,29 @@ enum
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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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PROP_SLAVE_METHOD
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};
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#define GST_TYPE_SLAVE_METHOD (slave_method_get_type ())
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static GType
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slave_method_get_type (void)
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{
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static GType slave_method_type = 0;
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static const GEnumValue slave_method[] = {
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{GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE, "Resampling slaving", "resample"},
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{GST_BASE_AUDIO_SINK_SLAVE_SKEW, "Skew slaving", "skew"},
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{0, NULL, NULL},
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};
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if (!slave_method_type) {
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slave_method_type =
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g_enum_register_static ("GstBaseAudioSinkSlaveMethod", slave_method);
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}
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return slave_method_type;
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}
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#define _do_init(bla) \
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GST_DEBUG_CATEGORY_INIT (gst_base_audio_sink_debug, "baseaudiosink", 0, "baseaudiosink element");
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@ -126,6 +161,8 @@ gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass)
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gstelement_class = (GstElementClass *) klass;
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gstbasesink_class = (GstBaseSinkClass *) klass;
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g_type_class_add_private (klass, sizeof (GstBaseAudioSinkPrivate));
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gobject_class->set_property =
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GST_DEBUG_FUNCPTR (gst_base_audio_sink_set_property);
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gobject_class->get_property =
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@ -147,6 +184,11 @@ gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass)
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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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g_object_class_install_property (gobject_class, PROP_SLAVE_METHOD,
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g_param_spec_enum ("slave-method", "Slave Method",
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"Algorithm to use to match the rate of the masterclock",
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GST_TYPE_SLAVE_METHOD, DEFAULT_SLAVE_METHOD, 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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gstelement_class->provide_clock =
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@ -170,9 +212,12 @@ static void
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gst_base_audio_sink_init (GstBaseAudioSink * baseaudiosink,
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GstBaseAudioSinkClass * g_class)
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{
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baseaudiosink->priv = GST_BASE_AUDIO_SINK_GET_PRIVATE (baseaudiosink);
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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->priv->slave_method = DEFAULT_SLAVE_METHOD;
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baseaudiosink->provided_clock = gst_audio_clock_new ("GstAudioSinkClock",
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(GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time, baseaudiosink);
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@ -276,14 +321,17 @@ gst_base_audio_sink_query (GstElement * element, GstQuery * query)
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spec = &basesink->ringbuffer->spec;
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basesink->priv->us_latency = min_l;
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min_latency =
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gst_util_uint64_scale_int (spec->segtotal * spec->segsize,
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GST_SECOND, spec->rate * spec->bytes_per_sample);
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/* we cannot go lower than the buffer size */
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min_latency = MAX (min_latency, min_l);
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/* we cannot go lower than the buffer size and the min peer latency */
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min_latency = min_latency + min_l;
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/* the max latency is the max of the peer, we can delay an infinite
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* amount of time. */
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max_latency = max_l;
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max_latency = min_latency + (max_l == -1 ? 0 : max_l);
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GST_DEBUG_OBJECT (basesink,
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"peer min %" GST_TIME_FORMAT ", our min latency: %"
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@ -314,7 +362,7 @@ gst_base_audio_sink_get_time (GstClock * clock, GstBaseAudioSink * sink)
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{
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guint64 raw, samples;
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guint delay;
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GstClockTime result;
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GstClockTime result, us_latency;
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if (sink->ringbuffer == NULL || sink->ringbuffer->spec.rate == 0)
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return GST_CLOCK_TIME_NONE;
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@ -334,9 +382,15 @@ gst_base_audio_sink_get_time (GstClock * clock, GstBaseAudioSink * sink)
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result = gst_util_uint64_scale_int (samples, GST_SECOND,
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sink->ringbuffer->spec.rate);
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/* latency before starting the clock */
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us_latency = sink->priv->us_latency;
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result += us_latency;
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GST_DEBUG_OBJECT (sink,
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"processed samples: raw %llu, delay %u, real %llu, time %"
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GST_TIME_FORMAT, raw, delay, samples, GST_TIME_ARGS (result));
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GST_TIME_FORMAT ", upstream latency %" GST_TIME_FORMAT, raw, delay,
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samples, GST_TIME_ARGS (result), GST_TIME_ARGS (us_latency));
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return result;
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}
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@ -361,6 +415,9 @@ gst_base_audio_sink_set_property (GObject * object, guint prop_id,
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sink->provide_clock = g_value_get_boolean (value);
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GST_OBJECT_UNLOCK (sink);
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break;
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case PROP_SLAVE_METHOD:
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sink->priv->slave_method = g_value_get_enum (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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@ -387,6 +444,9 @@ gst_base_audio_sink_get_property (GObject * object, guint prop_id,
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g_value_set_boolean (value, sink->provide_clock);
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GST_OBJECT_UNLOCK (sink);
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break;
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case PROP_SLAVE_METHOD:
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g_value_set_enum (value, sink->priv->slave_method);
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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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@ -542,6 +602,7 @@ gst_base_audio_sink_event (GstBaseSink * bsink, GstEvent * event)
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break;
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case GST_EVENT_FLUSH_STOP:
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/* always resync on sample after a flush */
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sink->priv->avg_skew = -1;
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sink->next_sample = -1;
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if (sink->ringbuffer)
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gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
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@ -621,6 +682,163 @@ gst_base_audio_sink_get_offset (GstBaseAudioSink * sink)
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return sample;
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}
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static GstClockTime
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clock_convert_external (GstClockTime external, GstClockTime cinternal,
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GstClockTime cexternal, GstClockTime crate_num, GstClockTime crate_denom,
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GstClockTime us_latency)
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{
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/* adjust for rate and speed */
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if (external >= cexternal) {
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external =
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gst_util_uint64_scale (external - cexternal, crate_denom, crate_num);
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external += cinternal;
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} else {
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external = gst_util_uint64_scale (cexternal - external,
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crate_denom, crate_num);
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if (cinternal > external)
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external = cinternal - external;
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else
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external = 0;
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}
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/* adjust for offset when slaving started */
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if (external > us_latency)
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external -= us_latency;
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else
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external = 0;
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return external;
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}
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/* algorithm to calculate sample positions that will result in resampling to
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* match the clock rate of the master */
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static void
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gst_base_audio_sink_resample_slaving (GstBaseAudioSink * sink,
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GstClockTime render_start, GstClockTime render_stop,
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GstClockTime * srender_start, GstClockTime * srender_stop)
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{
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GstClockTime cinternal, cexternal;
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GstClockTime crate_num, crate_denom;
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/* get calibration parameters to compensate for speed and offset differences
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* when we are slaved */
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gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
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&crate_num, &crate_denom);
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GST_DEBUG_OBJECT (sink, "internal %" GST_TIME_FORMAT " external %"
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GST_TIME_FORMAT " %" G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT " = %f",
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GST_TIME_ARGS (cinternal), GST_TIME_ARGS (cexternal), crate_num,
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crate_denom, gst_guint64_to_gdouble (crate_num) /
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gst_guint64_to_gdouble (crate_denom));
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if (crate_num == 0)
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crate_denom = crate_num = 1;
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/* bring external time to internal time */
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render_start = clock_convert_external (render_start, cinternal, cexternal,
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crate_num, crate_denom, sink->priv->us_latency);
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render_stop = clock_convert_external (render_stop, cinternal, cexternal,
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crate_num, crate_denom, sink->priv->us_latency);
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GST_DEBUG_OBJECT (sink,
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"after slaving: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
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GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
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*srender_start = render_start;
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*srender_stop = render_stop;
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}
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/* algorithm to calculate sample positions that will result in changing the
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* playout pointer to match the clock rate of the master */
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static void
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gst_base_audio_sink_skew_slaving (GstBaseAudioSink * sink,
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GstClockTime render_start, GstClockTime render_stop,
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GstClockTime * srender_start, GstClockTime * srender_stop)
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{
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GstClockTime cinternal, cexternal, crate_num, crate_denom;
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GstClockTime etime, itime;
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GstClockTimeDiff skew, segtime;
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/* get calibration parameters to compensate for offsets */
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gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
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&crate_num, &crate_denom);
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/* sample clocks and figure out clock skew */
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etime = gst_clock_get_time (GST_ELEMENT_CLOCK (sink));
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itime = gst_clock_get_internal_time (sink->provided_clock);
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etime -= cexternal;
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itime -= cinternal;
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skew = GST_CLOCK_DIFF (etime, itime);
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if (sink->priv->avg_skew == -1) {
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/* first observation */
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sink->priv->avg_skew = skew;
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} else {
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/* next observations use a moving average */
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sink->priv->avg_skew = (31 * sink->priv->avg_skew + skew) / 32;
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}
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GST_DEBUG_OBJECT (sink, "internal %" GST_TIME_FORMAT " external %"
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GST_TIME_FORMAT " skew %" G_GINT64_FORMAT " avg %" G_GINT64_FORMAT,
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GST_TIME_ARGS (itime), GST_TIME_ARGS (etime), skew, sink->priv->avg_skew);
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/* the max drift we allow is the length of a segment */
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segtime = sink->ringbuffer->spec.latency_time * 1000;
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/* adjust playout pointer based on skew */
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if (sink->priv->avg_skew > segtime) {
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/* master is running slower */
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GST_WARNING_OBJECT (sink,
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"correct clock skew %" G_GINT64_FORMAT " > %" G_GINT64_FORMAT,
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sink->priv->avg_skew, segtime);
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cinternal += segtime;
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sink->priv->avg_skew -= segtime;
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sink->next_sample = -1;
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gst_clock_set_calibration (sink->provided_clock, cinternal, cexternal,
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crate_num, crate_denom);
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} else if (sink->priv->avg_skew < -segtime) {
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/* master is running faster */
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GST_WARNING_OBJECT (sink,
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"correct clock skew %" G_GINT64_FORMAT " < %" G_GINT64_FORMAT,
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sink->priv->avg_skew, -segtime);
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cinternal -= segtime;
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sink->priv->avg_skew += segtime;
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sink->next_sample = -1;
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gst_clock_set_calibration (sink->provided_clock, cinternal, cexternal,
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crate_num, crate_denom);
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}
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/* convert, ignoring speed */
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render_start = clock_convert_external (render_start, cinternal, cexternal,
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crate_num, crate_denom, sink->priv->us_latency);
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render_stop = clock_convert_external (render_stop, cinternal, cexternal,
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crate_num, crate_denom, sink->priv->us_latency);
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*srender_start = render_start;
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*srender_stop = render_stop;
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}
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/* converts render_start and render_stop to their slaved values */
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static void
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gst_base_audio_sink_handle_slaving (GstBaseAudioSink * sink,
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GstClockTime render_start, GstClockTime render_stop,
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GstClockTime * srender_start, GstClockTime * srender_stop)
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{
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switch (sink->priv->slave_method) {
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case GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE:
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gst_base_audio_sink_resample_slaving (sink, render_start, render_stop,
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srender_start, srender_stop);
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break;
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case GST_BASE_AUDIO_SINK_SLAVE_SKEW:
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gst_base_audio_sink_skew_slaving (sink, render_start, render_stop,
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srender_start, srender_stop);
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break;
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default:
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g_warning ("unknown slaving method %d", sink->priv->slave_method);
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break;
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}
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}
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static GstFlowReturn
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gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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{
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@ -634,10 +852,8 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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guint samples, written;
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gint bps;
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gint accum;
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GstClockTime crate_num;
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GstClockTime crate_denom;
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gint out_samples;
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GstClockTime base_time, cinternal, cexternal, latency;
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GstClockTime base_time, latency;
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GstClock *clock;
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gboolean sync, slaved;
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@ -751,64 +967,21 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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GST_DEBUG_OBJECT (sink,
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"compensating for latency %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
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/* add latency to get the timestamp to sync against the pipeline clock */
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render_start += latency;
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render_stop += latency;
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GST_DEBUG_OBJECT (sink,
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"after latency: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
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GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
|
||||
|
||||
slaved = clock != sink->provided_clock;
|
||||
if (slaved) {
|
||||
/* get calibration parameters to compensate for speed and offset differences
|
||||
* when we are slaved */
|
||||
gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
|
||||
&crate_num, &crate_denom);
|
||||
|
||||
cinternal += latency;
|
||||
|
||||
GST_DEBUG_OBJECT (sink, "internal %" GST_TIME_FORMAT " external %"
|
||||
GST_TIME_FORMAT " %" G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT " = %f",
|
||||
GST_TIME_ARGS (cinternal), GST_TIME_ARGS (cexternal), crate_num,
|
||||
crate_denom, gst_guint64_to_gdouble (crate_num / crate_denom));
|
||||
|
||||
if (crate_num == 0)
|
||||
crate_denom = crate_num = 1;
|
||||
|
||||
/* bring to our slaved clock time */
|
||||
if (render_start >= cexternal) {
|
||||
render_start =
|
||||
gst_util_uint64_scale (render_start - cexternal, crate_denom,
|
||||
crate_num);
|
||||
render_start += cinternal;
|
||||
} else {
|
||||
render_start = gst_util_uint64_scale (cexternal - render_start,
|
||||
crate_denom, crate_num);
|
||||
if (cinternal > render_start)
|
||||
render_start = cinternal - render_start;
|
||||
else
|
||||
render_start = 0;
|
||||
}
|
||||
|
||||
if (render_stop >= cexternal) {
|
||||
render_stop =
|
||||
gst_util_uint64_scale (render_stop - cexternal, crate_denom,
|
||||
crate_num);
|
||||
render_stop += cinternal;
|
||||
} else {
|
||||
render_stop = gst_util_uint64_scale (cexternal - render_stop,
|
||||
crate_denom, crate_num);
|
||||
if (cinternal > render_stop)
|
||||
render_stop = cinternal - render_stop;
|
||||
else
|
||||
render_stop = 0;
|
||||
}
|
||||
|
||||
GST_DEBUG_OBJECT (sink,
|
||||
"after slaving: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
|
||||
} else {
|
||||
render_start += latency;
|
||||
render_stop += latency;
|
||||
GST_DEBUG_OBJECT (sink,
|
||||
"after latency: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
|
||||
/* handle clock slaving */
|
||||
gst_base_audio_sink_handle_slaving (sink, render_start, render_stop,
|
||||
&render_start, &render_stop);
|
||||
}
|
||||
|
||||
|
||||
/* and bring the time to the rate corrected offset in the buffer */
|
||||
render_start = gst_util_uint64_scale_int (render_start,
|
||||
ringbuf->spec.rate, GST_SECOND);
|
||||
|
|
@ -827,6 +1000,8 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
|
|||
goto no_align;
|
||||
}
|
||||
|
||||
/* positive playback rate, first sample is render_start, negative rate, first
|
||||
* sample is render_stop */
|
||||
if (bsink->segment.rate >= 1.0)
|
||||
sample_offset = render_start;
|
||||
else
|
||||
|
|
@ -864,8 +1039,8 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
|
|||
/* apply alignment */
|
||||
render_start += align;
|
||||
|
||||
/* only align stop if we are not slaved */
|
||||
if (slaved) {
|
||||
/* only align stop if we are not slaved to resample */
|
||||
if (slaved && sink->priv->slave_method == GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE) {
|
||||
GST_DEBUG_OBJECT (sink, "no stop time align needed: we are slaved");
|
||||
goto no_align;
|
||||
}
|
||||
|
|
@ -876,15 +1051,15 @@ no_align:
|
|||
out_samples = render_stop - render_start;
|
||||
|
||||
no_sync:
|
||||
GST_DEBUG_OBJECT (sink, "rendering at %" G_GUINT64_FORMAT " %d/%d",
|
||||
sink->next_sample, samples, out_samples);
|
||||
|
||||
/* we render the first or last sample first, depending on the rate */
|
||||
if (bsink->segment.rate >= 1.0)
|
||||
sample_offset = render_start;
|
||||
else
|
||||
sample_offset = render_stop;
|
||||
|
||||
GST_DEBUG_OBJECT (sink, "rendering at %" G_GUINT64_FORMAT " %d/%d",
|
||||
sample_offset, samples, out_samples);
|
||||
|
||||
/* we need to accumulate over different runs for when we get interrupted */
|
||||
accum = 0;
|
||||
do {
|
||||
|
|
@ -1038,8 +1213,9 @@ static GstStateChangeReturn
|
|||
gst_base_audio_sink_async_play (GstBaseSink * basesink)
|
||||
{
|
||||
GstClock *clock;
|
||||
GstClockTime itime, etime;
|
||||
GstBaseAudioSink *sink;
|
||||
GstClockTime itime, etime;
|
||||
GstClockTime rate_num, rate_denom;
|
||||
|
||||
sink = GST_BASE_AUDIO_SINK (basesink);
|
||||
|
||||
|
|
@ -1048,34 +1224,43 @@ gst_base_audio_sink_async_play (GstBaseSink * basesink)
|
|||
|
||||
clock = GST_ELEMENT_CLOCK (sink);
|
||||
if (clock == NULL)
|
||||
goto no_clock;
|
||||
goto done;
|
||||
|
||||
/* we provided the global clock, don't need to do anything special */
|
||||
if (clock == sink->provided_clock)
|
||||
goto done;
|
||||
|
||||
/* FIXME, only start slaving when we really start the ringbuffer */
|
||||
/* if we are slaved to a clock, we need to set the initial
|
||||
* calibration */
|
||||
if (clock != sink->provided_clock) {
|
||||
GstClockTime rate_num, rate_denom;
|
||||
/* get external and internal time to set as calibration params */
|
||||
etime = gst_clock_get_time (clock);
|
||||
itime = gst_clock_get_internal_time (sink->provided_clock);
|
||||
|
||||
etime = gst_clock_get_time (clock);
|
||||
itime = gst_clock_get_internal_time (sink->provided_clock);
|
||||
sink->priv->avg_skew = -1;
|
||||
|
||||
GST_DEBUG_OBJECT (sink,
|
||||
"internal time: %" GST_TIME_FORMAT " external time: %" GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (itime), GST_TIME_ARGS (etime));
|
||||
GST_DEBUG_OBJECT (sink,
|
||||
"internal time: %" GST_TIME_FORMAT " external time: %" GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (itime), GST_TIME_ARGS (etime));
|
||||
|
||||
/* FIXME, this is not yet accurate enough for smooth playback */
|
||||
gst_clock_get_calibration (sink->provided_clock, NULL, NULL, &rate_num,
|
||||
&rate_denom);
|
||||
/* Does not work yet. */
|
||||
gst_clock_set_calibration (sink->provided_clock, itime, etime,
|
||||
rate_num, rate_denom);
|
||||
gst_clock_get_calibration (sink->provided_clock, NULL, NULL, &rate_num,
|
||||
&rate_denom);
|
||||
gst_clock_set_calibration (sink->provided_clock, itime, etime,
|
||||
rate_num, rate_denom);
|
||||
|
||||
gst_clock_set_master (sink->provided_clock, clock);
|
||||
|
||||
/* start ringbuffer so we can start slaving right away */
|
||||
gst_ring_buffer_start (sink->ringbuffer);
|
||||
switch (sink->priv->slave_method) {
|
||||
case GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE:
|
||||
/* only set as master if we need to resample */
|
||||
GST_DEBUG_OBJECT (sink, "Setting clock as master");
|
||||
gst_clock_set_master (sink->provided_clock, clock);
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
no_clock:
|
||||
|
||||
/* start ringbuffer so we can start slaving right away when we need to */
|
||||
gst_ring_buffer_start (sink->ringbuffer);
|
||||
|
||||
done:
|
||||
return GST_STATE_CHANGE_SUCCESS;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -78,8 +78,23 @@ G_BEGIN_DECLS
|
|||
*/
|
||||
#define GST_BASE_AUDIO_SINK_PAD(obj) (GST_BASE_SINK (obj)->sinkpad)
|
||||
|
||||
/**
|
||||
* GstBaseAudioSinkSlaveMethod:
|
||||
* @GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE: Resample to match the master clock
|
||||
* @GST_BASE_AUDIO_SINK_SLAVE_SKEW: Adjust playout pointer when master clock
|
||||
* drifts too much.
|
||||
*
|
||||
* Different possible clock slaving algorithms
|
||||
*/
|
||||
typedef enum
|
||||
{
|
||||
GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE,
|
||||
GST_BASE_AUDIO_SINK_SLAVE_SKEW,
|
||||
} GstBaseAudioSinkSlaveMethod;
|
||||
|
||||
typedef struct _GstBaseAudioSink GstBaseAudioSink;
|
||||
typedef struct _GstBaseAudioSinkClass GstBaseAudioSinkClass;
|
||||
typedef struct _GstBaseAudioSinkPrivate GstBaseAudioSinkPrivate;
|
||||
|
||||
/**
|
||||
* GstBaseAudioSink:
|
||||
|
|
@ -105,10 +120,9 @@ struct _GstBaseAudioSink {
|
|||
GstClock *provided_clock;
|
||||
|
||||
/*< private >*/
|
||||
union {
|
||||
/* adding + 0 to mark ABI change to be undone later */
|
||||
gpointer _gst_reserved[GST_PADDING + 0];
|
||||
} abidata;
|
||||
GstBaseAudioSinkPrivate *priv;
|
||||
|
||||
gpointer _gst_reserved[GST_PADDING - 1];
|
||||
};
|
||||
|
||||
/**
|
||||
|
|
|
|||
Loading…
Reference in a new issue