From ee7072fe7ed0356a7c9b8fcc0f1d1d0feb36e724 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Fri, 11 Nov 2011 11:52:47 +0100 Subject: [PATCH] rename GstBaseAudio* ->GstAudioBase* --- docs/libs/gst-plugins-base-libs-sections.txt | 84 ++-- docs/libs/gst-plugins-base-libs.types | 4 +- ext/alsa/gstalsasrc.c | 2 +- gst-libs/gst/audio/gstaudioclock.c | 4 +- gst-libs/gst/audio/gstaudioringbuffer.c | 2 +- gst-libs/gst/audio/gstaudiosink.c | 14 +- gst-libs/gst/audio/gstaudiosink.h | 4 +- gst-libs/gst/audio/gstaudiosrc.c | 14 +- gst-libs/gst/audio/gstaudiosrc.h | 4 +- gst-libs/gst/audio/gstbaseaudiosink.c | 380 +++++++++---------- gst-libs/gst/audio/gstbaseaudiosink.h | 102 ++--- gst-libs/gst/audio/gstbaseaudiosrc.c | 228 +++++------ gst-libs/gst/audio/gstbaseaudiosrc.h | 92 ++--- 13 files changed, 467 insertions(+), 467 deletions(-) diff --git a/docs/libs/gst-plugins-base-libs-sections.txt b/docs/libs/gst-plugins-base-libs-sections.txt index 8305059a64..e3398285c2 100644 --- a/docs/libs/gst-plugins-base-libs-sections.txt +++ b/docs/libs/gst-plugins-base-libs-sections.txt @@ -288,57 +288,57 @@ GST_AUDIO_SRC_GET_CLASS
gstbaseaudiosink gst/audio/gstbaseaudiosink.h -GstBaseAudioSink -GstBaseAudioSinkClass -GstBaseAudioSinkSlaveMethod +GstAudioBaseSink +GstAudioBaseSinkClass +GstAudioBaseSinkSlaveMethod -GST_BASE_AUDIO_SINK_CLOCK -GST_BASE_AUDIO_SINK_PAD -gst_base_audio_sink_create_ringbuffer -gst_base_audio_sink_set_provide_clock -gst_base_audio_sink_get_provide_clock -gst_base_audio_sink_set_slave_method -gst_base_audio_sink_get_slave_method -gst_base_audio_sink_get_drift_tolerance -gst_base_audio_sink_set_drift_tolerance +GST_AUDIO_BASE_SINK_CLOCK +GST_AUDIO_BASE_SINK_PAD +gst_audio_base_sink_create_ringbuffer +gst_audio_base_sink_set_provide_clock +gst_audio_base_sink_get_provide_clock +gst_audio_base_sink_set_slave_method +gst_audio_base_sink_get_slave_method +gst_audio_base_sink_get_drift_tolerance +gst_audio_base_sink_set_drift_tolerance -GST_BASE_AUDIO_SINK -GST_IS_BASE_AUDIO_SINK -GST_TYPE_BASE_AUDIO_SINK -gst_base_audio_sink_get_type -GST_BASE_AUDIO_SINK_CLASS -GST_IS_BASE_AUDIO_SINK_CLASS -GST_BASE_AUDIO_SINK_GET_CLASS -GstBaseAudioSinkPrivate -gst_base_audio_sink_slave_method_get_type -GST_TYPE_BASE_AUDIO_SINK_SLAVE_METHOD +GST_AUDIO_BASE_SINK +GST_IS_AUDIO_BASE_SINK +GST_TYPE_AUDIO_BASE_SINK +gst_audio_base_sink_get_type +GST_AUDIO_BASE_SINK_CLASS +GST_IS_AUDIO_BASE_SINK_CLASS +GST_AUDIO_BASE_SINK_GET_CLASS +GstAudioBaseSinkPrivate +gst_audio_base_sink_slave_method_get_type +GST_TYPE_AUDIO_BASE_SINK_SLAVE_METHOD
gstbaseaudiosrc gst/audio/gstbaseaudiosrc.h -GstBaseAudioSrc -GstBaseAudioSrcClass -GstBaseAudioSrcSlaveMethod +GstAudioBaseSrc +GstAudioBaseSrcClass +GstAudioBaseSrcSlaveMethod -GST_BASE_AUDIO_SRC_CLOCK -GST_BASE_AUDIO_SRC_PAD -gst_base_audio_src_create_ringbuffer -gst_base_audio_src_set_provide_clock -gst_base_audio_src_get_provide_clock -gst_base_audio_src_get_slave_method -gst_base_audio_src_set_slave_method +GST_AUDIO_BASE_SRC_CLOCK +GST_AUDIO_BASE_SRC_PAD +gst_audio_base_src_create_ringbuffer +gst_audio_base_src_set_provide_clock +gst_audio_base_src_get_provide_clock +gst_audio_base_src_get_slave_method +gst_audio_base_src_set_slave_method -GstBaseAudioSrcPrivate -GST_BASE_AUDIO_SRC -GST_IS_BASE_AUDIO_SRC -GST_TYPE_BASE_AUDIO_SRC -gst_base_audio_src_get_type -GST_BASE_AUDIO_SRC_CLASS -GST_IS_BASE_AUDIO_SRC_CLASS -GST_BASE_AUDIO_SRC_GET_CLASS -gst_base_audio_src_slave_method_get_type -GST_TYPE_BASE_AUDIO_SRC_SLAVE_METHOD +GstAudioBaseSrcPrivate +GST_AUDIO_BASE_SRC +GST_IS_AUDIO_BASE_SRC +GST_TYPE_AUDIO_BASE_SRC +gst_audio_base_src_get_type +GST_AUDIO_BASE_SRC_CLASS +GST_IS_AUDIO_BASE_SRC_CLASS +GST_AUDIO_BASE_SRC_GET_CLASS +gst_audio_base_src_slave_method_get_type +GST_TYPE_AUDIO_BASE_SRC_SLAVE_METHOD
diff --git a/docs/libs/gst-plugins-base-libs.types b/docs/libs/gst-plugins-base-libs.types index 2bec332e49..48cd414612 100644 --- a/docs/libs/gst-plugins-base-libs.types +++ b/docs/libs/gst-plugins-base-libs.types @@ -14,9 +14,9 @@ gst_audio_sink_get_type #include gst_audio_src_get_type #include -gst_base_audio_sink_get_type +gst_audio_base_sink_get_type #include -gst_base_audio_src_get_type +gst_audio_base_src_get_type #include gst_audio_ring_buffer_get_type diff --git a/ext/alsa/gstalsasrc.c b/ext/alsa/gstalsasrc.c index 64e9319f2f..7c8242ebd4 100644 --- a/ext/alsa/gstalsasrc.c +++ b/ext/alsa/gstalsasrc.c @@ -316,7 +316,7 @@ static GstStateChangeReturn gst_alsasrc_change_state (GstElement * element, GstStateChange transition) { GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; - GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (element); + GstAudioBaseSrc *src = GST_AUDIO_BASE_SRC (element); GstAlsaSrc *asrc = GST_ALSA_SRC (element); GstClock *clk; diff --git a/gst-libs/gst/audio/gstaudioclock.c b/gst-libs/gst/audio/gstaudioclock.c index 2e9ffb1849..bbf36b071f 100644 --- a/gst-libs/gst/audio/gstaudioclock.c +++ b/gst-libs/gst/audio/gstaudioclock.c @@ -23,12 +23,12 @@ /** * SECTION:gstaudioclock * @short_description: Helper object for implementing audio clocks - * @see_also: #GstBaseAudioSink, #GstSystemClock + * @see_also: #GstAudioBaseSink, #GstSystemClock * * #GstAudioClock makes it easy for elements to implement a #GstClock, they * simply need to provide a function that returns the current clock time. * - * This object is internally used to implement the clock in #GstBaseAudioSink. + * This object is internally used to implement the clock in #GstAudioBaseSink. * * Last reviewed on 2006-09-27 (0.10.12) */ diff --git a/gst-libs/gst/audio/gstaudioringbuffer.c b/gst-libs/gst/audio/gstaudioringbuffer.c index d540d1fc33..a91488f7d3 100644 --- a/gst-libs/gst/audio/gstaudioringbuffer.c +++ b/gst-libs/gst/audio/gstaudioringbuffer.c @@ -20,7 +20,7 @@ /** * SECTION:gstaudioringbuffer * @short_description: Base class for audio ringbuffer implementations - * @see_also: #GstBaseAudioSink, #GstAudioSink + * @see_also: #GstAudioBaseSink, #GstAudioSink * * * diff --git a/gst-libs/gst/audio/gstaudiosink.c b/gst-libs/gst/audio/gstaudiosink.c index 7b6fce5f45..1fa8320cbb 100644 --- a/gst-libs/gst/audio/gstaudiosink.c +++ b/gst-libs/gst/audio/gstaudiosink.c @@ -23,7 +23,7 @@ /** * SECTION:gstaudiosink * @short_description: Simple base class for audio sinks - * @see_also: #GstBaseAudioSink, #GstAudioRingBuffer, #GstAudioSink. + * @see_also: #GstAudioBaseSink, #GstAudioRingBuffer, #GstAudioSink. * * This is the most simple base class for audio sinks that only requires * subclasses to implement a set of simple functions: @@ -61,7 +61,7 @@ * * * All scheduling of samples and timestamps is done in this base class - * together with #GstBaseAudioSink using a default implementation of a + * together with #GstAudioBaseSink using a default implementation of a * #GstAudioRingBuffer that uses threads. * * Last reviewed on 2006-09-27 (0.10.12) @@ -592,17 +592,17 @@ enum GST_DEBUG_CATEGORY_INIT (gst_audio_sink_debug, "audiosink", 0, "audiosink element"); #define gst_audio_sink_parent_class parent_class G_DEFINE_TYPE_WITH_CODE (GstAudioSink, gst_audio_sink, - GST_TYPE_BASE_AUDIO_SINK, _do_init); + GST_TYPE_AUDIO_BASE_SINK, _do_init); -static GstAudioRingBuffer *gst_audio_sink_create_ringbuffer (GstBaseAudioSink * +static GstAudioRingBuffer *gst_audio_sink_create_ringbuffer (GstAudioBaseSink * sink); static void gst_audio_sink_class_init (GstAudioSinkClass * klass) { - GstBaseAudioSinkClass *gstbaseaudiosink_class; + GstAudioBaseSinkClass *gstbaseaudiosink_class; - gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass; + gstbaseaudiosink_class = (GstAudioBaseSinkClass *) klass; gstbaseaudiosink_class->create_ringbuffer = GST_DEBUG_FUNCPTR (gst_audio_sink_create_ringbuffer); @@ -616,7 +616,7 @@ gst_audio_sink_init (GstAudioSink * audiosink) } static GstAudioRingBuffer * -gst_audio_sink_create_ringbuffer (GstBaseAudioSink * sink) +gst_audio_sink_create_ringbuffer (GstAudioBaseSink * sink) { GstAudioRingBuffer *buffer; diff --git a/gst-libs/gst/audio/gstaudiosink.h b/gst-libs/gst/audio/gstaudiosink.h index b513b5a1ff..ea8c5cbd6d 100644 --- a/gst-libs/gst/audio/gstaudiosink.h +++ b/gst-libs/gst/audio/gstaudiosink.h @@ -44,7 +44,7 @@ typedef struct _GstAudioSinkClass GstAudioSinkClass; * Opaque #GstAudioSink. */ struct _GstAudioSink { - GstBaseAudioSink element; + GstAudioBaseSink element; /*< private >*/ /* with LOCK */ GThread *thread; @@ -70,7 +70,7 @@ struct _GstAudioSink { * #GstAudioSink class. Override the vmethods to implement functionality. */ struct _GstAudioSinkClass { - GstBaseAudioSinkClass parent_class; + GstAudioBaseSinkClass parent_class; /* vtable */ diff --git a/gst-libs/gst/audio/gstaudiosrc.c b/gst-libs/gst/audio/gstaudiosrc.c index 6806f7fbfe..dc7b49ee67 100644 --- a/gst-libs/gst/audio/gstaudiosrc.c +++ b/gst-libs/gst/audio/gstaudiosrc.c @@ -23,7 +23,7 @@ /** * SECTION:gstaudiosrc * @short_description: Simple base class for audio sources - * @see_also: #GstBaseAudioSrc, #GstAudioRingBuffer, #GstAudioSrc. + * @see_also: #GstAudioBaseSrc, #GstAudioRingBuffer, #GstAudioSrc. * * This is the most simple base class for audio sources that only requires * subclasses to implement a set of simple functions: @@ -61,7 +61,7 @@ * * * All scheduling of samples and timestamps is done in this base class - * together with #GstBaseAudioSrc using a default implementation of a + * together with #GstAudioBaseSrc using a default implementation of a * #GstAudioRingBuffer that uses threads. * * Last reviewed on 2006-09-27 (0.10.12) @@ -505,17 +505,17 @@ enum GST_DEBUG_CATEGORY_INIT (gst_audio_src_debug, "audiosrc", 0, "audiosrc element"); #define gst_audio_src_parent_class parent_class G_DEFINE_TYPE_WITH_CODE (GstAudioSrc, gst_audio_src, - GST_TYPE_BASE_AUDIO_SRC, _do_init); + GST_TYPE_AUDIO_BASE_SRC, _do_init); -static GstAudioRingBuffer *gst_audio_src_create_ringbuffer (GstBaseAudioSrc * +static GstAudioRingBuffer *gst_audio_src_create_ringbuffer (GstAudioBaseSrc * src); static void gst_audio_src_class_init (GstAudioSrcClass * klass) { - GstBaseAudioSrcClass *gstbaseaudiosrc_class; + GstAudioBaseSrcClass *gstbaseaudiosrc_class; - gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass; + gstbaseaudiosrc_class = (GstAudioBaseSrcClass *) klass; gstbaseaudiosrc_class->create_ringbuffer = GST_DEBUG_FUNCPTR (gst_audio_src_create_ringbuffer); @@ -529,7 +529,7 @@ gst_audio_src_init (GstAudioSrc * audiosrc) } static GstAudioRingBuffer * -gst_audio_src_create_ringbuffer (GstBaseAudioSrc * src) +gst_audio_src_create_ringbuffer (GstAudioBaseSrc * src) { GstAudioRingBuffer *buffer; diff --git a/gst-libs/gst/audio/gstaudiosrc.h b/gst-libs/gst/audio/gstaudiosrc.h index 8e21ee1eb4..410f724eb1 100644 --- a/gst-libs/gst/audio/gstaudiosrc.h +++ b/gst-libs/gst/audio/gstaudiosrc.h @@ -44,7 +44,7 @@ typedef struct _GstAudioSrcClass GstAudioSrcClass; * Base class for simple audio sources. */ struct _GstAudioSrc { - GstBaseAudioSrc element; + GstAudioBaseSrc element; /*< private >*/ /* with LOCK */ GThread *thread; @@ -68,7 +68,7 @@ struct _GstAudioSrc { * functionality. */ struct _GstAudioSrcClass { - GstBaseAudioSrcClass parent_class; + GstAudioBaseSrcClass parent_class; /* vtable */ diff --git a/gst-libs/gst/audio/gstbaseaudiosink.c b/gst-libs/gst/audio/gstbaseaudiosink.c index e0b9c96033..fcec55a92f 100644 --- a/gst-libs/gst/audio/gstbaseaudiosink.c +++ b/gst-libs/gst/audio/gstbaseaudiosink.c @@ -36,18 +36,18 @@ #include "gstbaseaudiosink.h" -GST_DEBUG_CATEGORY_STATIC (gst_base_audio_sink_debug); -#define GST_CAT_DEFAULT gst_base_audio_sink_debug +GST_DEBUG_CATEGORY_STATIC (gst_audio_base_sink_debug); +#define GST_CAT_DEFAULT gst_audio_base_sink_debug -#define GST_BASE_AUDIO_SINK_GET_PRIVATE(obj) \ - (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_AUDIO_SINK, GstBaseAudioSinkPrivate)) +#define GST_AUDIO_BASE_SINK_GET_PRIVATE(obj) \ + (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_AUDIO_BASE_SINK, GstAudioBaseSinkPrivate)) -struct _GstBaseAudioSinkPrivate +struct _GstAudioBaseSinkPrivate { /* upstream latency */ GstClockTime us_latency; /* the clock slaving algorithm in use */ - GstBaseAudioSinkSlaveMethod slave_method; + GstAudioBaseSinkSlaveMethod slave_method; /* running average of clock skew */ GstClockTimeDiff avg_skew; /* the number of samples we aligned last time */ @@ -83,7 +83,7 @@ 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_SINK_SLAVE_SKEW +#define DEFAULT_SLAVE_METHOD GST_AUDIO_BASE_SINK_SLAVE_SKEW /* FIXME, enable pull mode when clock slaving and trick modes are figured out */ #define DEFAULT_CAN_ACTIVATE_PULL FALSE @@ -117,20 +117,20 @@ enum }; GType -gst_base_audio_sink_slave_method_get_type (void) +gst_audio_base_sink_slave_method_get_type (void) { static volatile gsize slave_method_type = 0; static const GEnumValue slave_method[] = { - {GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE, "GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE", + {GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE, "GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE", "resample"}, - {GST_BASE_AUDIO_SINK_SLAVE_SKEW, "GST_BASE_AUDIO_SINK_SLAVE_SKEW", "skew"}, - {GST_BASE_AUDIO_SINK_SLAVE_NONE, "GST_BASE_AUDIO_SINK_SLAVE_NONE", "none"}, + {GST_AUDIO_BASE_SINK_SLAVE_SKEW, "GST_AUDIO_BASE_SINK_SLAVE_SKEW", "skew"}, + {GST_AUDIO_BASE_SINK_SLAVE_NONE, "GST_AUDIO_BASE_SINK_SLAVE_NONE", "none"}, {0, NULL, NULL}, }; if (g_once_init_enter (&slave_method_type)) { GType tmp = - g_enum_register_static ("GstBaseAudioSinkSlaveMethod", slave_method); + g_enum_register_static ("GstAudioBaseSinkSlaveMethod", slave_method); g_once_init_leave (&slave_method_type, tmp); } @@ -139,55 +139,55 @@ gst_base_audio_sink_slave_method_get_type (void) #define _do_init \ - GST_DEBUG_CATEGORY_INIT (gst_base_audio_sink_debug, "baseaudiosink", 0, "baseaudiosink element"); -#define gst_base_audio_sink_parent_class parent_class -G_DEFINE_TYPE_WITH_CODE (GstBaseAudioSink, gst_base_audio_sink, + GST_DEBUG_CATEGORY_INIT (gst_audio_base_sink_debug, "baseaudiosink", 0, "baseaudiosink element"); +#define gst_audio_base_sink_parent_class parent_class +G_DEFINE_TYPE_WITH_CODE (GstAudioBaseSink, gst_audio_base_sink, GST_TYPE_BASE_SINK, _do_init); -static void gst_base_audio_sink_dispose (GObject * object); +static void gst_audio_base_sink_dispose (GObject * object); -static void gst_base_audio_sink_set_property (GObject * object, guint prop_id, +static void gst_audio_base_sink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); -static void gst_base_audio_sink_get_property (GObject * object, guint prop_id, +static void gst_audio_base_sink_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); #if 0 -static GstStateChangeReturn gst_base_audio_sink_async_play (GstBaseSink * +static GstStateChangeReturn gst_audio_base_sink_async_play (GstBaseSink * basesink); #endif -static GstStateChangeReturn gst_base_audio_sink_change_state (GstElement * +static GstStateChangeReturn gst_audio_base_sink_change_state (GstElement * element, GstStateChange transition); -static gboolean gst_base_audio_sink_activate_pull (GstBaseSink * basesink, +static gboolean gst_audio_base_sink_activate_pull (GstBaseSink * basesink, gboolean active); -static gboolean gst_base_audio_sink_query (GstElement * element, GstQuery * +static gboolean gst_audio_base_sink_query (GstElement * element, GstQuery * query); -static GstClock *gst_base_audio_sink_provide_clock (GstElement * elem); -static GstClockTime gst_base_audio_sink_get_time (GstClock * clock, - GstBaseAudioSink * sink); -static void gst_base_audio_sink_callback (GstAudioRingBuffer * rbuf, +static GstClock *gst_audio_base_sink_provide_clock (GstElement * elem); +static GstClockTime gst_audio_base_sink_get_time (GstClock * clock, + GstAudioBaseSink * sink); +static void gst_audio_base_sink_callback (GstAudioRingBuffer * rbuf, guint8 * data, guint len, gpointer user_data); -static GstFlowReturn gst_base_audio_sink_preroll (GstBaseSink * bsink, +static GstFlowReturn gst_audio_base_sink_preroll (GstBaseSink * bsink, GstBuffer * buffer); -static GstFlowReturn gst_base_audio_sink_render (GstBaseSink * bsink, +static GstFlowReturn gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buffer); -static gboolean gst_base_audio_sink_event (GstBaseSink * bsink, +static gboolean gst_audio_base_sink_event (GstBaseSink * bsink, GstEvent * event); -static void gst_base_audio_sink_get_times (GstBaseSink * bsink, +static void gst_audio_base_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer, GstClockTime * start, GstClockTime * end); -static gboolean gst_base_audio_sink_setcaps (GstBaseSink * bsink, +static gboolean gst_audio_base_sink_setcaps (GstBaseSink * bsink, GstCaps * caps); -static void gst_base_audio_sink_fixate (GstBaseSink * bsink, GstCaps * caps); +static void gst_audio_base_sink_fixate (GstBaseSink * bsink, GstCaps * caps); -static gboolean gst_base_audio_sink_query_pad (GstBaseSink * bsink, +static gboolean gst_audio_base_sink_query_pad (GstBaseSink * bsink, GstQuery * query); -/* static guint gst_base_audio_sink_signals[LAST_SIGNAL] = { 0 }; */ +/* static guint gst_audio_base_sink_signals[LAST_SIGNAL] = { 0 }; */ static void -gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass) +gst_audio_base_sink_class_init (GstAudioBaseSinkClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; @@ -197,11 +197,11 @@ gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass) gstelement_class = (GstElementClass *) klass; gstbasesink_class = (GstBaseSinkClass *) klass; - g_type_class_add_private (klass, sizeof (GstBaseAudioSinkPrivate)); + g_type_class_add_private (klass, sizeof (GstAudioBaseSinkPrivate)); - gobject_class->set_property = gst_base_audio_sink_set_property; - gobject_class->get_property = gst_base_audio_sink_get_property; - gobject_class->dispose = gst_base_audio_sink_dispose; + gobject_class->set_property = gst_audio_base_sink_set_property; + gobject_class->get_property = gst_audio_base_sink_get_property; + gobject_class->dispose = gst_audio_base_sink_dispose; g_object_class_install_property (gobject_class, PROP_BUFFER_TIME, g_param_spec_int64 ("buffer-time", "Buffer Time", @@ -223,7 +223,7 @@ gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass) 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_BASE_AUDIO_SINK_SLAVE_METHOD, DEFAULT_SLAVE_METHOD, + GST_TYPE_AUDIO_BASE_SINK_SLAVE_METHOD, DEFAULT_SLAVE_METHOD, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_CAN_ACTIVATE_PULL, @@ -231,7 +231,7 @@ gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass) "Allow pull-based scheduling", DEFAULT_CAN_ACTIVATE_PULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** - * GstBaseAudioSink:drift-tolerance + * GstAudioBaseSink:drift-tolerance * * Controls the amount of time in microseconds that clocks are allowed * to drift before resynchronisation happens. @@ -244,7 +244,7 @@ gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass) G_MAXINT64, DEFAULT_DRIFT_TOLERANCE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** - * GstBaseAudioSink:alignment_threshold + * GstAudioBaseSink:alignment_threshold * * Controls the amount of time in nanoseconds that timestamps are allowed * to drift from their ideal time before choosing not to align them. @@ -258,7 +258,7 @@ gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass) G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** - * GstBaseAudioSink:discont-wait + * GstAudioBaseSink:discont-wait * * A window of time in nanoseconds to wait before creating a discontinuity as * a result of breaching the drift-tolerance. @@ -273,21 +273,21 @@ gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass) G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gstelement_class->change_state = - GST_DEBUG_FUNCPTR (gst_base_audio_sink_change_state); + GST_DEBUG_FUNCPTR (gst_audio_base_sink_change_state); gstelement_class->provide_clock = - GST_DEBUG_FUNCPTR (gst_base_audio_sink_provide_clock); - gstelement_class->query = GST_DEBUG_FUNCPTR (gst_base_audio_sink_query); + GST_DEBUG_FUNCPTR (gst_audio_base_sink_provide_clock); + gstelement_class->query = GST_DEBUG_FUNCPTR (gst_audio_base_sink_query); - gstbasesink_class->fixate = GST_DEBUG_FUNCPTR (gst_base_audio_sink_fixate); - gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_base_audio_sink_setcaps); - gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_base_audio_sink_event); + gstbasesink_class->fixate = GST_DEBUG_FUNCPTR (gst_audio_base_sink_fixate); + gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_audio_base_sink_setcaps); + gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_audio_base_sink_event); gstbasesink_class->get_times = - GST_DEBUG_FUNCPTR (gst_base_audio_sink_get_times); - gstbasesink_class->preroll = GST_DEBUG_FUNCPTR (gst_base_audio_sink_preroll); - gstbasesink_class->render = GST_DEBUG_FUNCPTR (gst_base_audio_sink_render); - gstbasesink_class->query = GST_DEBUG_FUNCPTR (gst_base_audio_sink_query_pad); + GST_DEBUG_FUNCPTR (gst_audio_base_sink_get_times); + gstbasesink_class->preroll = GST_DEBUG_FUNCPTR (gst_audio_base_sink_preroll); + gstbasesink_class->render = GST_DEBUG_FUNCPTR (gst_audio_base_sink_render); + gstbasesink_class->query = GST_DEBUG_FUNCPTR (gst_audio_base_sink_query_pad); gstbasesink_class->activate_pull = - GST_DEBUG_FUNCPTR (gst_base_audio_sink_activate_pull); + GST_DEBUG_FUNCPTR (gst_audio_base_sink_activate_pull); /* ref class from a thread-safe context to work around missing bit of * thread-safety in GObject */ @@ -297,11 +297,11 @@ gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass) } static void -gst_base_audio_sink_init (GstBaseAudioSink * baseaudiosink) +gst_audio_base_sink_init (GstAudioBaseSink * baseaudiosink) { GstBaseSink *basesink; - baseaudiosink->priv = GST_BASE_AUDIO_SINK_GET_PRIVATE (baseaudiosink); + baseaudiosink->priv = GST_AUDIO_BASE_SINK_GET_PRIVATE (baseaudiosink); baseaudiosink->buffer_time = DEFAULT_BUFFER_TIME; baseaudiosink->latency_time = DEFAULT_LATENCY_TIME; @@ -312,7 +312,7 @@ gst_base_audio_sink_init (GstBaseAudioSink * baseaudiosink) baseaudiosink->priv->discont_wait = DEFAULT_DISCONT_WAIT; baseaudiosink->provided_clock = gst_audio_clock_new ("GstAudioSinkClock", - (GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time, baseaudiosink, + (GstAudioClockGetTimeFunc) gst_audio_base_sink_get_time, baseaudiosink, NULL); basesink = GST_BASE_SINK_CAST (baseaudiosink); @@ -323,11 +323,11 @@ gst_base_audio_sink_init (GstBaseAudioSink * baseaudiosink) } static void -gst_base_audio_sink_dispose (GObject * object) +gst_audio_base_sink_dispose (GObject * object) { - GstBaseAudioSink *sink; + GstAudioBaseSink *sink; - sink = GST_BASE_AUDIO_SINK (object); + sink = GST_AUDIO_BASE_SINK (object); if (sink->provided_clock) { gst_audio_clock_invalidate (sink->provided_clock); @@ -345,12 +345,12 @@ gst_base_audio_sink_dispose (GObject * object) static GstClock * -gst_base_audio_sink_provide_clock (GstElement * elem) +gst_audio_base_sink_provide_clock (GstElement * elem) { - GstBaseAudioSink *sink; + GstAudioBaseSink *sink; GstClock *clock; - sink = GST_BASE_AUDIO_SINK (elem); + sink = GST_AUDIO_BASE_SINK (elem); /* we have no ringbuffer (must be NULL state) */ if (sink->ringbuffer == NULL) @@ -383,12 +383,12 @@ clock_disabled: } static gboolean -gst_base_audio_sink_query_pad (GstBaseSink * bsink, GstQuery * query) +gst_audio_base_sink_query_pad (GstBaseSink * bsink, GstQuery * query) { gboolean res = FALSE; - GstBaseAudioSink *basesink; + GstAudioBaseSink *basesink; - basesink = GST_BASE_AUDIO_SINK (bsink); + basesink = GST_AUDIO_BASE_SINK (bsink); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_CONVERT: @@ -417,12 +417,12 @@ gst_base_audio_sink_query_pad (GstBaseSink * bsink, GstQuery * query) } static gboolean -gst_base_audio_sink_query (GstElement * element, GstQuery * query) +gst_audio_base_sink_query (GstElement * element, GstQuery * query) { gboolean res = FALSE; - GstBaseAudioSink *basesink; + GstAudioBaseSink *basesink; - basesink = GST_BASE_AUDIO_SINK (element); + basesink = GST_AUDIO_BASE_SINK (element); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_LATENCY: @@ -513,7 +513,7 @@ done: static GstClockTime -gst_base_audio_sink_get_time (GstClock * clock, GstBaseAudioSink * sink) +gst_audio_base_sink_get_time (GstClock * clock, GstAudioBaseSink * sink) { guint64 raw, samples; guint delay; @@ -546,8 +546,8 @@ gst_base_audio_sink_get_time (GstClock * clock, GstBaseAudioSink * sink) } /** - * gst_base_audio_sink_set_provide_clock: - * @sink: a #GstBaseAudioSink + * gst_audio_base_sink_set_provide_clock: + * @sink: a #GstAudioBaseSink * @provide: new state * * Controls whether @sink will provide a clock or not. If @provide is %TRUE, @@ -557,10 +557,10 @@ gst_base_audio_sink_get_time (GstClock * clock, GstBaseAudioSink * sink) * Since: 0.10.16 */ void -gst_base_audio_sink_set_provide_clock (GstBaseAudioSink * sink, +gst_audio_base_sink_set_provide_clock (GstAudioBaseSink * sink, gboolean provide) { - g_return_if_fail (GST_IS_BASE_AUDIO_SINK (sink)); + g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink)); GST_OBJECT_LOCK (sink); sink->provide_clock = provide; @@ -568,22 +568,22 @@ gst_base_audio_sink_set_provide_clock (GstBaseAudioSink * sink, } /** - * gst_base_audio_sink_get_provide_clock: - * @sink: a #GstBaseAudioSink + * gst_audio_base_sink_get_provide_clock: + * @sink: a #GstAudioBaseSink * * Queries whether @sink will provide a clock or not. See also - * gst_base_audio_sink_set_provide_clock. + * gst_audio_base_sink_set_provide_clock. * * Returns: %TRUE if @sink will provide a clock. * * Since: 0.10.16 */ gboolean -gst_base_audio_sink_get_provide_clock (GstBaseAudioSink * sink) +gst_audio_base_sink_get_provide_clock (GstAudioBaseSink * sink) { gboolean result; - g_return_val_if_fail (GST_IS_BASE_AUDIO_SINK (sink), FALSE); + g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), FALSE); GST_OBJECT_LOCK (sink); result = sink->provide_clock; @@ -593,8 +593,8 @@ gst_base_audio_sink_get_provide_clock (GstBaseAudioSink * sink) } /** - * gst_base_audio_sink_set_slave_method: - * @sink: a #GstBaseAudioSink + * gst_audio_base_sink_set_slave_method: + * @sink: a #GstAudioBaseSink * @method: the new slave method * * Controls how clock slaving will be performed in @sink. @@ -602,10 +602,10 @@ gst_base_audio_sink_get_provide_clock (GstBaseAudioSink * sink) * Since: 0.10.16 */ void -gst_base_audio_sink_set_slave_method (GstBaseAudioSink * sink, - GstBaseAudioSinkSlaveMethod method) +gst_audio_base_sink_set_slave_method (GstAudioBaseSink * sink, + GstAudioBaseSinkSlaveMethod method) { - g_return_if_fail (GST_IS_BASE_AUDIO_SINK (sink)); + g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink)); GST_OBJECT_LOCK (sink); sink->priv->slave_method = method; @@ -613,8 +613,8 @@ gst_base_audio_sink_set_slave_method (GstBaseAudioSink * sink, } /** - * gst_base_audio_sink_get_slave_method: - * @sink: a #GstBaseAudioSink + * gst_audio_base_sink_get_slave_method: + * @sink: a #GstAudioBaseSink * * Get the current slave method used by @sink. * @@ -622,12 +622,12 @@ gst_base_audio_sink_set_slave_method (GstBaseAudioSink * sink, * * Since: 0.10.16 */ -GstBaseAudioSinkSlaveMethod -gst_base_audio_sink_get_slave_method (GstBaseAudioSink * sink) +GstAudioBaseSinkSlaveMethod +gst_audio_base_sink_get_slave_method (GstAudioBaseSink * sink) { - GstBaseAudioSinkSlaveMethod result; + GstAudioBaseSinkSlaveMethod result; - g_return_val_if_fail (GST_IS_BASE_AUDIO_SINK (sink), -1); + g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), -1); GST_OBJECT_LOCK (sink); result = sink->priv->slave_method; @@ -638,8 +638,8 @@ gst_base_audio_sink_get_slave_method (GstBaseAudioSink * sink) /** - * gst_base_audio_sink_set_drift_tolerance: - * @sink: a #GstBaseAudioSink + * gst_audio_base_sink_set_drift_tolerance: + * @sink: a #GstAudioBaseSink * @drift_tolerance: the new drift tolerance in microseconds * * Controls the sink's drift tolerance. @@ -647,10 +647,10 @@ gst_base_audio_sink_get_slave_method (GstBaseAudioSink * sink) * Since: 0.10.31 */ void -gst_base_audio_sink_set_drift_tolerance (GstBaseAudioSink * sink, +gst_audio_base_sink_set_drift_tolerance (GstAudioBaseSink * sink, gint64 drift_tolerance) { - g_return_if_fail (GST_IS_BASE_AUDIO_SINK (sink)); + g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink)); GST_OBJECT_LOCK (sink); sink->priv->drift_tolerance = drift_tolerance; @@ -658,8 +658,8 @@ gst_base_audio_sink_set_drift_tolerance (GstBaseAudioSink * sink, } /** - * gst_base_audio_sink_get_drift_tolerance - * @sink: a #GstBaseAudioSink + * gst_audio_base_sink_get_drift_tolerance + * @sink: a #GstAudioBaseSink * * Get the current drift tolerance, in microseconds, used by @sink. * @@ -668,11 +668,11 @@ gst_base_audio_sink_set_drift_tolerance (GstBaseAudioSink * sink, * Since: 0.10.31 */ gint64 -gst_base_audio_sink_get_drift_tolerance (GstBaseAudioSink * sink) +gst_audio_base_sink_get_drift_tolerance (GstAudioBaseSink * sink) { gint64 result; - g_return_val_if_fail (GST_IS_BASE_AUDIO_SINK (sink), -1); + g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), -1); GST_OBJECT_LOCK (sink); result = sink->priv->drift_tolerance; @@ -682,8 +682,8 @@ gst_base_audio_sink_get_drift_tolerance (GstBaseAudioSink * sink) } /** - * gst_base_audio_sink_set_alignment_threshold: - * @sink: a #GstBaseAudioSink + * gst_audio_base_sink_set_alignment_threshold: + * @sink: a #GstAudioBaseSink * @alignment_threshold: the new alignment threshold in nanoseconds * * Controls the sink's alignment threshold. @@ -691,10 +691,10 @@ gst_base_audio_sink_get_drift_tolerance (GstBaseAudioSink * sink) * Since: 0.10.36 */ void -gst_base_audio_sink_set_alignment_threshold (GstBaseAudioSink * sink, +gst_audio_base_sink_set_alignment_threshold (GstAudioBaseSink * sink, GstClockTime alignment_threshold) { - g_return_if_fail (GST_IS_BASE_AUDIO_SINK (sink)); + g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink)); GST_OBJECT_LOCK (sink); sink->priv->alignment_threshold = alignment_threshold; @@ -702,8 +702,8 @@ gst_base_audio_sink_set_alignment_threshold (GstBaseAudioSink * sink, } /** - * gst_base_audio_sink_get_alignment_threshold - * @sink: a #GstBaseAudioSink + * gst_audio_base_sink_get_alignment_threshold + * @sink: a #GstAudioBaseSink * * Get the current alignment threshold, in nanoseconds, used by @sink. * @@ -712,11 +712,11 @@ gst_base_audio_sink_set_alignment_threshold (GstBaseAudioSink * sink, * Since: 0.10.36 */ GstClockTime -gst_base_audio_sink_get_alignment_threshold (GstBaseAudioSink * sink) +gst_audio_base_sink_get_alignment_threshold (GstAudioBaseSink * sink) { gint64 result; - g_return_val_if_fail (GST_IS_BASE_AUDIO_SINK (sink), -1); + g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), -1); GST_OBJECT_LOCK (sink); result = sink->priv->alignment_threshold; @@ -726,8 +726,8 @@ gst_base_audio_sink_get_alignment_threshold (GstBaseAudioSink * sink) } /** - * gst_base_audio_sink_set_discont_wait: - * @sink: a #GstBaseAudioSink + * gst_audio_base_sink_set_discont_wait: + * @sink: a #GstAudioBaseSink * @discont_wait: the new discont wait in nanoseconds * * Controls how long the sink will wait before creating a discontinuity. @@ -735,10 +735,10 @@ gst_base_audio_sink_get_alignment_threshold (GstBaseAudioSink * sink) * Since: 0.10.36 */ void -gst_base_audio_sink_set_discont_wait (GstBaseAudioSink * sink, +gst_audio_base_sink_set_discont_wait (GstAudioBaseSink * sink, GstClockTime discont_wait) { - g_return_if_fail (GST_IS_BASE_AUDIO_SINK (sink)); + g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink)); GST_OBJECT_LOCK (sink); sink->priv->discont_wait = discont_wait; @@ -746,8 +746,8 @@ gst_base_audio_sink_set_discont_wait (GstBaseAudioSink * sink, } /** - * gst_base_audio_sink_get_discont_wait - * @sink: a #GstBaseAudioSink + * gst_audio_base_sink_get_discont_wait + * @sink: a #GstAudioBaseSink * * Get the current discont wait, in nanoseconds, used by @sink. * @@ -756,11 +756,11 @@ gst_base_audio_sink_set_discont_wait (GstBaseAudioSink * sink, * Since: 0.10.36 */ GstClockTime -gst_base_audio_sink_get_discont_wait (GstBaseAudioSink * sink) +gst_audio_base_sink_get_discont_wait (GstAudioBaseSink * sink) { GstClockTime result; - g_return_val_if_fail (GST_IS_BASE_AUDIO_SINK (sink), -1); + g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), -1); GST_OBJECT_LOCK (sink); result = sink->priv->discont_wait; @@ -770,12 +770,12 @@ gst_base_audio_sink_get_discont_wait (GstBaseAudioSink * sink) } static void -gst_base_audio_sink_set_property (GObject * object, guint prop_id, +gst_audio_base_sink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { - GstBaseAudioSink *sink; + GstAudioBaseSink *sink; - sink = GST_BASE_AUDIO_SINK (object); + sink = GST_AUDIO_BASE_SINK (object); switch (prop_id) { case PROP_BUFFER_TIME: @@ -785,23 +785,23 @@ gst_base_audio_sink_set_property (GObject * object, guint prop_id, sink->latency_time = g_value_get_int64 (value); break; case PROP_PROVIDE_CLOCK: - gst_base_audio_sink_set_provide_clock (sink, g_value_get_boolean (value)); + gst_audio_base_sink_set_provide_clock (sink, g_value_get_boolean (value)); break; case PROP_SLAVE_METHOD: - gst_base_audio_sink_set_slave_method (sink, g_value_get_enum (value)); + gst_audio_base_sink_set_slave_method (sink, g_value_get_enum (value)); break; case PROP_CAN_ACTIVATE_PULL: GST_BASE_SINK (sink)->can_activate_pull = g_value_get_boolean (value); break; case PROP_DRIFT_TOLERANCE: - gst_base_audio_sink_set_drift_tolerance (sink, g_value_get_int64 (value)); + gst_audio_base_sink_set_drift_tolerance (sink, g_value_get_int64 (value)); break; case PROP_ALIGNMENT_THRESHOLD: - gst_base_audio_sink_set_alignment_threshold (sink, + gst_audio_base_sink_set_alignment_threshold (sink, g_value_get_uint64 (value)); break; case PROP_DISCONT_WAIT: - gst_base_audio_sink_set_discont_wait (sink, g_value_get_uint64 (value)); + gst_audio_base_sink_set_discont_wait (sink, g_value_get_uint64 (value)); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); @@ -810,12 +810,12 @@ gst_base_audio_sink_set_property (GObject * object, guint prop_id, } static void -gst_base_audio_sink_get_property (GObject * object, guint prop_id, +gst_audio_base_sink_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { - GstBaseAudioSink *sink; + GstAudioBaseSink *sink; - sink = GST_BASE_AUDIO_SINK (object); + sink = GST_AUDIO_BASE_SINK (object); switch (prop_id) { case PROP_BUFFER_TIME: @@ -825,23 +825,23 @@ gst_base_audio_sink_get_property (GObject * object, guint prop_id, g_value_set_int64 (value, sink->latency_time); break; case PROP_PROVIDE_CLOCK: - g_value_set_boolean (value, gst_base_audio_sink_get_provide_clock (sink)); + g_value_set_boolean (value, gst_audio_base_sink_get_provide_clock (sink)); break; case PROP_SLAVE_METHOD: - g_value_set_enum (value, gst_base_audio_sink_get_slave_method (sink)); + g_value_set_enum (value, gst_audio_base_sink_get_slave_method (sink)); break; case PROP_CAN_ACTIVATE_PULL: g_value_set_boolean (value, GST_BASE_SINK (sink)->can_activate_pull); break; case PROP_DRIFT_TOLERANCE: - g_value_set_int64 (value, gst_base_audio_sink_get_drift_tolerance (sink)); + g_value_set_int64 (value, gst_audio_base_sink_get_drift_tolerance (sink)); break; case PROP_ALIGNMENT_THRESHOLD: g_value_set_uint64 (value, - gst_base_audio_sink_get_alignment_threshold (sink)); + gst_audio_base_sink_get_alignment_threshold (sink)); break; case PROP_DISCONT_WAIT: - g_value_set_uint64 (value, gst_base_audio_sink_get_discont_wait (sink)); + g_value_set_uint64 (value, gst_audio_base_sink_get_discont_wait (sink)); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); @@ -850,9 +850,9 @@ gst_base_audio_sink_get_property (GObject * object, guint prop_id, } static gboolean -gst_base_audio_sink_setcaps (GstBaseSink * bsink, GstCaps * caps) +gst_audio_base_sink_setcaps (GstBaseSink * bsink, GstCaps * caps) { - GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink); + GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink); GstAudioRingBufferSpec *spec; GstClockTime now; GstClockTime crate_num, crate_denom; @@ -931,7 +931,7 @@ acquire_error: } static void -gst_base_audio_sink_fixate (GstBaseSink * bsink, GstCaps * caps) +gst_audio_base_sink_fixate (GstBaseSink * bsink, GstCaps * caps) { GstStructure *s; gint width, depth; @@ -959,7 +959,7 @@ gst_base_audio_sink_fixate (GstBaseSink * bsink, GstCaps * caps) } static void -gst_base_audio_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer, +gst_audio_base_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer, GstClockTime * start, GstClockTime * end) { /* our clock sync is a bit too much for the base class to handle so @@ -970,7 +970,7 @@ gst_base_audio_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer, /* This waits for the drain to happen and can be canceled */ static gboolean -gst_base_audio_sink_drain (GstBaseAudioSink * sink) +gst_audio_base_sink_drain (GstAudioBaseSink * sink) { if (!sink->ringbuffer) return TRUE; @@ -1003,9 +1003,9 @@ gst_base_audio_sink_drain (GstBaseAudioSink * sink) } static gboolean -gst_base_audio_sink_event (GstBaseSink * bsink, GstEvent * event) +gst_audio_base_sink_event (GstBaseSink * bsink, GstEvent * event) { - GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink); + GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_FLUSH_START: @@ -1023,7 +1023,7 @@ gst_base_audio_sink_event (GstBaseSink * bsink, GstEvent * event) break; case GST_EVENT_EOS: /* now wait till we played everything */ - gst_base_audio_sink_drain (sink); + gst_audio_base_sink_drain (sink); break; default: break; @@ -1032,9 +1032,9 @@ gst_base_audio_sink_event (GstBaseSink * bsink, GstEvent * event) } static GstFlowReturn -gst_base_audio_sink_preroll (GstBaseSink * bsink, GstBuffer * buffer) +gst_audio_base_sink_preroll (GstBaseSink * bsink, GstBuffer * buffer) { - GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink); + GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink); if (!gst_audio_ring_buffer_is_acquired (sink->ringbuffer)) goto wrong_state; @@ -1053,7 +1053,7 @@ wrong_state: } static guint64 -gst_base_audio_sink_get_offset (GstBaseAudioSink * sink) +gst_audio_base_sink_get_offset (GstAudioBaseSink * sink) { guint64 sample; gint writeseg, segdone, sps; @@ -1108,7 +1108,7 @@ clock_convert_external (GstClockTime external, GstClockTime cinternal, /* algorithm to calculate sample positions that will result in resampling to * match the clock rate of the master */ static void -gst_base_audio_sink_resample_slaving (GstBaseAudioSink * sink, +gst_audio_base_sink_resample_slaving (GstAudioBaseSink * sink, GstClockTime render_start, GstClockTime render_stop, GstClockTime * srender_start, GstClockTime * srender_stop) { @@ -1162,7 +1162,7 @@ gst_base_audio_sink_resample_slaving (GstBaseAudioSink * sink, /* algorithm to calculate sample positions that will result in changing the * playout pointer to match the clock rate of the master */ static void -gst_base_audio_sink_skew_slaving (GstBaseAudioSink * sink, +gst_audio_base_sink_skew_slaving (GstAudioBaseSink * sink, GstClockTime render_start, GstClockTime render_stop, GstClockTime * srender_start, GstClockTime * srender_stop) { @@ -1270,7 +1270,7 @@ gst_base_audio_sink_skew_slaving (GstBaseAudioSink * sink, /* apply the clock offset but do no slaving otherwise */ static void -gst_base_audio_sink_none_slaving (GstBaseAudioSink * sink, +gst_audio_base_sink_none_slaving (GstAudioBaseSink * sink, GstClockTime render_start, GstClockTime render_stop, GstClockTime * srender_start, GstClockTime * srender_stop) { @@ -1292,21 +1292,21 @@ gst_base_audio_sink_none_slaving (GstBaseAudioSink * sink, /* converts render_start and render_stop to their slaved values */ static void -gst_base_audio_sink_handle_slaving (GstBaseAudioSink * sink, +gst_audio_base_sink_handle_slaving (GstAudioBaseSink * sink, GstClockTime render_start, GstClockTime render_stop, GstClockTime * srender_start, GstClockTime * srender_stop) { switch (sink->priv->slave_method) { - case GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE: - gst_base_audio_sink_resample_slaving (sink, render_start, render_stop, + case GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE: + gst_audio_base_sink_resample_slaving (sink, render_start, render_stop, srender_start, srender_stop); break; - case GST_BASE_AUDIO_SINK_SLAVE_SKEW: - gst_base_audio_sink_skew_slaving (sink, render_start, render_stop, + case GST_AUDIO_BASE_SINK_SLAVE_SKEW: + gst_audio_base_sink_skew_slaving (sink, render_start, render_stop, srender_start, srender_stop); break; - case GST_BASE_AUDIO_SINK_SLAVE_NONE: - gst_base_audio_sink_none_slaving (sink, render_start, render_stop, + case GST_AUDIO_BASE_SINK_SLAVE_NONE: + gst_audio_base_sink_none_slaving (sink, render_start, render_stop, srender_start, srender_stop); break; default: @@ -1317,18 +1317,18 @@ gst_base_audio_sink_handle_slaving (GstBaseAudioSink * sink, /* must be called with LOCK */ static GstFlowReturn -gst_base_audio_sink_sync_latency (GstBaseSink * bsink, GstMiniObject * obj) +gst_audio_base_sink_sync_latency (GstBaseSink * bsink, GstMiniObject * obj) { GstClock *clock; GstClockReturn status; GstClockTime time, render_delay; GstFlowReturn ret; - GstBaseAudioSink *sink; + GstAudioBaseSink *sink; GstClockTime itime, etime; GstClockTime rate_num, rate_denom; GstClockTimeDiff jitter; - sink = GST_BASE_AUDIO_SINK (bsink); + sink = GST_AUDIO_BASE_SINK (bsink); clock = GST_ELEMENT_CLOCK (sink); if (G_UNLIKELY (clock == NULL)) @@ -1414,13 +1414,13 @@ gst_base_audio_sink_sync_latency (GstBaseSink * bsink, GstMiniObject * obj) rate_num, rate_denom); switch (sink->priv->slave_method) { - case GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE: + case GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE: /* only set as master when we are resampling */ GST_DEBUG_OBJECT (sink, "Setting clock as master"); gst_clock_set_master (sink->provided_clock, clock); break; - case GST_BASE_AUDIO_SINK_SLAVE_SKEW: - case GST_BASE_AUDIO_SINK_SLAVE_NONE: + case GST_AUDIO_BASE_SINK_SLAVE_SKEW: + case GST_AUDIO_BASE_SINK_SLAVE_NONE: default: break; } @@ -1452,7 +1452,7 @@ flushing: } static gint64 -gst_base_audio_sink_get_alignment (GstBaseAudioSink * sink, +gst_audio_base_sink_get_alignment (GstAudioBaseSink * sink, GstClockTime sample_offset) { GstAudioRingBuffer *ringbuf = sink->ringbuffer; @@ -1530,13 +1530,13 @@ gst_base_audio_sink_get_alignment (GstBaseAudioSink * sink, } static GstFlowReturn -gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf) +gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf) { guint64 in_offset; GstClockTime time, stop, render_start, render_stop, sample_offset; GstClockTimeDiff sync_offset, ts_offset; - GstBaseAudioSinkClass *bclass; - GstBaseAudioSink *sink; + GstAudioBaseSinkClass *bclass; + GstAudioBaseSink *sink; GstAudioRingBuffer *ringbuf; gint64 diff, align; guint64 ctime, cstop; @@ -1555,8 +1555,8 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf) gint64 time_offset; GstBuffer *out = NULL; - sink = GST_BASE_AUDIO_SINK (bsink); - bclass = GST_BASE_AUDIO_SINK_GET_CLASS (sink); + sink = GST_AUDIO_BASE_SINK (bsink); + bclass = GST_AUDIO_BASE_SINK_GET_CLASS (sink); ringbuf = sink->ringbuffer; @@ -1570,7 +1570,7 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf) GST_OBJECT_LOCK (sink); base_time = GST_ELEMENT_CAST (sink)->base_time; if (G_UNLIKELY (sink->priv->sync_latency)) { - ret = gst_base_audio_sink_sync_latency (bsink, GST_MINI_OBJECT_CAST (buf)); + ret = gst_audio_base_sink_sync_latency (bsink, GST_MINI_OBJECT_CAST (buf)); GST_OBJECT_UNLOCK (sink); if (G_UNLIKELY (ret != GST_FLOW_OK)) goto sync_latency_failed; @@ -1614,7 +1614,7 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf) /* if not valid timestamp or we can't clip or sync, try to play * sample ASAP */ if (!GST_CLOCK_TIME_IS_VALID (time)) { - render_start = gst_base_audio_sink_get_offset (sink); + render_start = gst_audio_base_sink_get_offset (sink); render_stop = render_start + samples; GST_DEBUG_OBJECT (sink, "Buffer of size %" G_GSIZE_FORMAT " has no time." " Using render_start=%" G_GUINT64_FORMAT, size, render_start); @@ -1691,7 +1691,7 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf) if (!sync) { /* no sync needed, play sample ASAP */ - render_start = gst_base_audio_sink_get_offset (sink); + render_start = gst_audio_base_sink_get_offset (sink); render_stop = render_start + samples; GST_DEBUG_OBJECT (sink, "no sync needed. Using render_start=%" G_GUINT64_FORMAT, render_start); @@ -1737,12 +1737,12 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf) if ((slaved = clock != sink->provided_clock)) { /* handle clock slaving */ - gst_base_audio_sink_handle_slaving (sink, render_start, render_stop, + gst_audio_base_sink_handle_slaving (sink, render_start, render_stop, &render_start, &render_stop); } else { /* no slaving needed but we need to adapt to the clock calibration * parameters */ - gst_base_audio_sink_none_slaving (sink, render_start, render_stop, + gst_audio_base_sink_none_slaving (sink, render_start, render_stop, &render_start, &render_stop); } @@ -1799,14 +1799,14 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf) goto no_align; } - align = gst_base_audio_sink_get_alignment (sink, sample_offset); + align = gst_audio_base_sink_get_alignment (sink, sample_offset); sink->priv->last_align = align; /* apply alignment */ render_start += align; /* only align stop if we are not slaved to resample */ - if (slaved && sink->priv->slave_method == GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE) { + if (slaved && sink->priv->slave_method == GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE) { GST_DEBUG_OBJECT (sink, "no stop time align needed: we are slaved"); goto no_align; } @@ -1935,8 +1935,8 @@ sync_latency_failed: } /** - * gst_base_audio_sink_create_ringbuffer: - * @sink: a #GstBaseAudioSink. + * gst_audio_base_sink_create_ringbuffer: + * @sink: a #GstAudioBaseSink. * * Create and return the #GstAudioRingBuffer for @sink. This function will call the * ::create_ringbuffer vmethod and will set @sink as the parent of the returned @@ -1945,12 +1945,12 @@ sync_latency_failed: * Returns: The new ringbuffer of @sink. */ GstAudioRingBuffer * -gst_base_audio_sink_create_ringbuffer (GstBaseAudioSink * sink) +gst_audio_base_sink_create_ringbuffer (GstAudioBaseSink * sink) { - GstBaseAudioSinkClass *bclass; + GstAudioBaseSinkClass *bclass; GstAudioRingBuffer *buffer = NULL; - bclass = GST_BASE_AUDIO_SINK_GET_CLASS (sink); + bclass = GST_AUDIO_BASE_SINK_GET_CLASS (sink); if (bclass->create_ringbuffer) buffer = bclass->create_ringbuffer (sink); @@ -1961,17 +1961,17 @@ gst_base_audio_sink_create_ringbuffer (GstBaseAudioSink * sink) } static void -gst_base_audio_sink_callback (GstAudioRingBuffer * rbuf, guint8 * data, +gst_audio_base_sink_callback (GstAudioRingBuffer * rbuf, guint8 * data, guint len, gpointer user_data) { GstBaseSink *basesink; - GstBaseAudioSink *sink; + GstAudioBaseSink *sink; GstBuffer *buf; GstFlowReturn ret; gsize size; basesink = GST_BASE_SINK (user_data); - sink = GST_BASE_AUDIO_SINK (user_data); + sink = GST_AUDIO_BASE_SINK (user_data); GST_PAD_STREAM_LOCK (basesink->sinkpad); @@ -2031,7 +2031,7 @@ eos: * the sink gets shut down; maybe we should set a flag somewhere, or * set segment.stop and segment.duration to the last sample or so */ GST_DEBUG_OBJECT (sink, "EOS"); - gst_base_audio_sink_drain (sink); + gst_audio_base_sink_drain (sink); gst_audio_ring_buffer_pause (rbuf); gst_element_post_message (GST_ELEMENT_CAST (sink), gst_message_new_eos (GST_OBJECT_CAST (sink))); @@ -2056,16 +2056,16 @@ preroll_error: } static gboolean -gst_base_audio_sink_activate_pull (GstBaseSink * basesink, gboolean active) +gst_audio_base_sink_activate_pull (GstBaseSink * basesink, gboolean active) { gboolean ret; - GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (basesink); + GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (basesink); if (active) { GST_DEBUG_OBJECT (basesink, "activating pull"); gst_audio_ring_buffer_set_callback (sink->ringbuffer, - gst_base_audio_sink_callback, sink); + gst_audio_base_sink_callback, sink); ret = gst_audio_ring_buffer_activate (sink->ringbuffer, TRUE); } else { @@ -2080,11 +2080,11 @@ gst_base_audio_sink_activate_pull (GstBaseSink * basesink, gboolean active) #if 0 /* should be called with the LOCK */ static GstStateChangeReturn -gst_base_audio_sink_async_play (GstBaseSink * basesink) +gst_audio_base_sink_async_play (GstBaseSink * basesink) { - GstBaseAudioSink *sink; + GstAudioBaseSink *sink; - sink = GST_BASE_AUDIO_SINK (basesink); + sink = GST_AUDIO_BASE_SINK (basesink); GST_DEBUG_OBJECT (sink, "ringbuffer may start now"); sink->priv->sync_latency = TRUE; @@ -2099,17 +2099,17 @@ gst_base_audio_sink_async_play (GstBaseSink * basesink) #endif static GstStateChangeReturn -gst_base_audio_sink_change_state (GstElement * element, +gst_audio_base_sink_change_state (GstElement * element, GstStateChange transition) { GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; - GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (element); + GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: if (sink->ringbuffer == NULL) { gst_audio_clock_reset (GST_AUDIO_CLOCK (sink->provided_clock), 0); - sink->ringbuffer = gst_base_audio_sink_create_ringbuffer (sink); + sink->ringbuffer = gst_audio_base_sink_create_ringbuffer (sink); } if (!gst_audio_ring_buffer_open_device (sink->ringbuffer)) goto open_failed; @@ -2127,7 +2127,7 @@ gst_base_audio_sink_change_state (GstElement * element, * should post this messages whenever necessary */ if (sink->provided_clock && GST_IS_AUDIO_CLOCK (sink->provided_clock) && GST_AUDIO_CLOCK_CAST (sink->provided_clock)->func == - (GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time) + (GstAudioClockGetTimeFunc) gst_audio_base_sink_get_time) gst_element_post_message (element, gst_message_new_clock_provide (GST_OBJECT_CAST (element), sink->provided_clock, TRUE)); @@ -2167,7 +2167,7 @@ gst_base_audio_sink_change_state (GstElement * element, * should post this messages whenever necessary */ if (sink->provided_clock && GST_IS_AUDIO_CLOCK (sink->provided_clock) && GST_AUDIO_CLOCK_CAST (sink->provided_clock)->func == - (GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time) + (GstAudioClockGetTimeFunc) gst_audio_base_sink_get_time) gst_element_post_message (element, gst_message_new_clock_lost (GST_OBJECT_CAST (element), sink->provided_clock)); diff --git a/gst-libs/gst/audio/gstbaseaudiosink.h b/gst-libs/gst/audio/gstbaseaudiosink.h index bc5da0df7b..32f0359ac5 100644 --- a/gst-libs/gst/audio/gstbaseaudiosink.h +++ b/gst-libs/gst/audio/gstbaseaudiosink.h @@ -46,8 +46,8 @@ * the methods in GstBaseSink and this class. */ -#ifndef __GST_BASE_AUDIO_SINK_H__ -#define __GST_BASE_AUDIO_SINK_H__ +#ifndef __GST_AUDIO_BASE_SINK_H__ +#define __GST_AUDIO_BASE_SINK_H__ #include #include @@ -56,57 +56,57 @@ G_BEGIN_DECLS -#define GST_TYPE_BASE_AUDIO_SINK (gst_base_audio_sink_get_type()) -#define GST_BASE_AUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_SINK,GstBaseAudioSink)) -#define GST_BASE_AUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_SINK,GstBaseAudioSinkClass)) -#define GST_BASE_AUDIO_SINK_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_BASE_AUDIO_SINK, GstBaseAudioSinkClass)) -#define GST_IS_BASE_AUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_SINK)) -#define GST_IS_BASE_AUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_SINK)) +#define GST_TYPE_AUDIO_BASE_SINK (gst_audio_base_sink_get_type()) +#define GST_AUDIO_BASE_SINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_BASE_SINK,GstAudioBaseSink)) +#define GST_AUDIO_BASE_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_BASE_SINK,GstAudioBaseSinkClass)) +#define GST_AUDIO_BASE_SINK_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIO_BASE_SINK, GstAudioBaseSinkClass)) +#define GST_IS_AUDIO_BASE_SINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_BASE_SINK)) +#define GST_IS_AUDIO_BASE_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_BASE_SINK)) /** - * GST_BASE_AUDIO_SINK_CLOCK: - * @obj: a #GstBaseAudioSink + * GST_AUDIO_BASE_SINK_CLOCK: + * @obj: a #GstAudioBaseSink * * Get the #GstClock of @obj. */ -#define GST_BASE_AUDIO_SINK_CLOCK(obj) (GST_BASE_AUDIO_SINK (obj)->clock) +#define GST_AUDIO_BASE_SINK_CLOCK(obj) (GST_AUDIO_BASE_SINK (obj)->clock) /** - * GST_BASE_AUDIO_SINK_PAD: - * @obj: a #GstBaseAudioSink + * GST_AUDIO_BASE_SINK_PAD: + * @obj: a #GstAudioBaseSink * * Get the sink #GstPad of @obj. */ -#define GST_BASE_AUDIO_SINK_PAD(obj) (GST_BASE_SINK (obj)->sinkpad) +#define GST_AUDIO_BASE_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 + * GstAudioBaseSinkSlaveMethod: + * @GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE: Resample to match the master clock + * @GST_AUDIO_BASE_SINK_SLAVE_SKEW: Adjust playout pointer when master clock * drifts too much. - * @GST_BASE_AUDIO_SINK_SLAVE_NONE: No adjustment is done. + * @GST_AUDIO_BASE_SINK_SLAVE_NONE: No adjustment is done. * * Different possible clock slaving algorithms used when the internal audio * clock is not selected as the pipeline master clock. */ typedef enum { - GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE, - GST_BASE_AUDIO_SINK_SLAVE_SKEW, - GST_BASE_AUDIO_SINK_SLAVE_NONE -} GstBaseAudioSinkSlaveMethod; + GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE, + GST_AUDIO_BASE_SINK_SLAVE_SKEW, + GST_AUDIO_BASE_SINK_SLAVE_NONE +} GstAudioBaseSinkSlaveMethod; -#define GST_TYPE_BASE_AUDIO_SINK_SLAVE_METHOD (gst_base_audio_sink_slave_method_get_type ()) +#define GST_TYPE_AUDIO_BASE_SINK_SLAVE_METHOD (gst_audio_base_sink_slave_method_get_type ()) -typedef struct _GstBaseAudioSink GstBaseAudioSink; -typedef struct _GstBaseAudioSinkClass GstBaseAudioSinkClass; -typedef struct _GstBaseAudioSinkPrivate GstBaseAudioSinkPrivate; +typedef struct _GstAudioBaseSink GstAudioBaseSink; +typedef struct _GstAudioBaseSinkClass GstAudioBaseSinkClass; +typedef struct _GstAudioBaseSinkPrivate GstAudioBaseSinkPrivate; /** - * GstBaseAudioSink: + * GstAudioBaseSink: * - * Opaque #GstBaseAudioSink. + * Opaque #GstAudioBaseSink. */ -struct _GstBaseAudioSink { +struct _GstAudioBaseSink { GstBaseSink element; /*< protected >*/ /* with LOCK */ @@ -128,13 +128,13 @@ struct _GstBaseAudioSink { gboolean eos_rendering; /*< private >*/ - GstBaseAudioSinkPrivate *priv; + GstAudioBaseSinkPrivate *priv; gpointer _gst_reserved[GST_PADDING]; }; /** - * GstBaseAudioSinkClass: + * GstAudioBaseSinkClass: * @parent_class: the parent class. * @create_ringbuffer: create and return a #GstAudioRingBuffer to write to. * @payload: payload data in a format suitable to write to the sink. If no @@ -142,50 +142,50 @@ struct _GstBaseAudioSink { * buffer, else returns the payloaded buffer with all other metadata * copied. (Since: 0.10.36) * - * #GstBaseAudioSink class. Override the vmethod to implement + * #GstAudioBaseSink class. Override the vmethod to implement * functionality. */ -struct _GstBaseAudioSinkClass { +struct _GstAudioBaseSinkClass { GstBaseSinkClass parent_class; /* subclass ringbuffer allocation */ - GstAudioRingBuffer* (*create_ringbuffer) (GstBaseAudioSink *sink); + GstAudioRingBuffer* (*create_ringbuffer) (GstAudioBaseSink *sink); /* subclass payloader */ - GstBuffer* (*payload) (GstBaseAudioSink *sink, + GstBuffer* (*payload) (GstAudioBaseSink *sink, GstBuffer *buffer); /*< private >*/ gpointer _gst_reserved[GST_PADDING]; }; -GType gst_base_audio_sink_get_type(void); -GType gst_base_audio_sink_slave_method_get_type (void); +GType gst_audio_base_sink_get_type(void); +GType gst_audio_base_sink_slave_method_get_type (void); GstAudioRingBuffer * - gst_base_audio_sink_create_ringbuffer (GstBaseAudioSink *sink); + gst_audio_base_sink_create_ringbuffer (GstAudioBaseSink *sink); -void gst_base_audio_sink_set_provide_clock (GstBaseAudioSink *sink, gboolean provide); -gboolean gst_base_audio_sink_get_provide_clock (GstBaseAudioSink *sink); +void gst_audio_base_sink_set_provide_clock (GstAudioBaseSink *sink, gboolean provide); +gboolean gst_audio_base_sink_get_provide_clock (GstAudioBaseSink *sink); -void gst_base_audio_sink_set_slave_method (GstBaseAudioSink *sink, - GstBaseAudioSinkSlaveMethod method); -GstBaseAudioSinkSlaveMethod - gst_base_audio_sink_get_slave_method (GstBaseAudioSink *sink); +void gst_audio_base_sink_set_slave_method (GstAudioBaseSink *sink, + GstAudioBaseSinkSlaveMethod method); +GstAudioBaseSinkSlaveMethod + gst_audio_base_sink_get_slave_method (GstAudioBaseSink *sink); -void gst_base_audio_sink_set_drift_tolerance (GstBaseAudioSink *sink, +void gst_audio_base_sink_set_drift_tolerance (GstAudioBaseSink *sink, gint64 drift_tolerance); -gint64 gst_base_audio_sink_get_drift_tolerance (GstBaseAudioSink *sink); +gint64 gst_audio_base_sink_get_drift_tolerance (GstAudioBaseSink *sink); -void gst_base_audio_sink_set_alignment_threshold (GstBaseAudioSink * sink, +void gst_audio_base_sink_set_alignment_threshold (GstAudioBaseSink * sink, GstClockTime alignment_threshold); GstClockTime - gst_base_audio_sink_get_alignment_threshold (GstBaseAudioSink * sink); + gst_audio_base_sink_get_alignment_threshold (GstAudioBaseSink * sink); -void gst_base_audio_sink_set_discont_wait (GstBaseAudioSink * sink, +void gst_audio_base_sink_set_discont_wait (GstAudioBaseSink * sink, GstClockTime discont_wait); GstClockTime - gst_base_audio_sink_get_discont_wait (GstBaseAudioSink * sink); + gst_audio_base_sink_get_discont_wait (GstAudioBaseSink * sink); G_END_DECLS -#endif /* __GST_BASE_AUDIO_SINK_H__ */ +#endif /* __GST_AUDIO_BASE_SINK_H__ */ diff --git a/gst-libs/gst/audio/gstbaseaudiosrc.c b/gst-libs/gst/audio/gstbaseaudiosrc.c index 1a6c19bc1f..2c5601e6b9 100644 --- a/gst-libs/gst/audio/gstbaseaudiosrc.c +++ b/gst-libs/gst/audio/gstbaseaudiosrc.c @@ -42,41 +42,41 @@ #include "gst/gst-i18n-plugin.h" -GST_DEBUG_CATEGORY_STATIC (gst_base_audio_src_debug); -#define GST_CAT_DEFAULT gst_base_audio_src_debug +GST_DEBUG_CATEGORY_STATIC (gst_audio_base_src_debug); +#define GST_CAT_DEFAULT gst_audio_base_src_debug GType -gst_base_audio_src_slave_method_get_type (void) +gst_audio_base_src_slave_method_get_type (void) { static volatile gsize slave_method_type = 0; /* FIXME 0.11: nick should be "retimestamp" not "re-timestamp" */ static const GEnumValue slave_method[] = { - {GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE, - "GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE", "resample"}, - {GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP, - "GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP", "re-timestamp"}, - {GST_BASE_AUDIO_SRC_SLAVE_SKEW, "GST_BASE_AUDIO_SRC_SLAVE_SKEW", "skew"}, - {GST_BASE_AUDIO_SRC_SLAVE_NONE, "GST_BASE_AUDIO_SRC_SLAVE_NONE", "none"}, + {GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE, + "GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE", "resample"}, + {GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP, + "GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP", "re-timestamp"}, + {GST_AUDIO_BASE_SRC_SLAVE_SKEW, "GST_AUDIO_BASE_SRC_SLAVE_SKEW", "skew"}, + {GST_AUDIO_BASE_SRC_SLAVE_NONE, "GST_AUDIO_BASE_SRC_SLAVE_NONE", "none"}, {0, NULL, NULL}, }; if (g_once_init_enter (&slave_method_type)) { GType tmp = - g_enum_register_static ("GstBaseAudioSrcSlaveMethod", slave_method); + g_enum_register_static ("GstAudioBaseSrcSlaveMethod", slave_method); g_once_init_leave (&slave_method_type, tmp); } return (GType) slave_method_type; } -#define GST_BASE_AUDIO_SRC_GET_PRIVATE(obj) \ - (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_AUDIO_SRC, GstBaseAudioSrcPrivate)) +#define GST_AUDIO_BASE_SRC_GET_PRIVATE(obj) \ + (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_AUDIO_BASE_SRC, GstAudioBaseSrcPrivate)) -struct _GstBaseAudioSrcPrivate +struct _GstAudioBaseSrcPrivate { gboolean provide_clock; /* the clock slaving algorithm in use */ - GstBaseAudioSrcSlaveMethod slave_method; + GstAudioBaseSrcSlaveMethod slave_method; }; /* BaseAudioSrc signals and args */ @@ -91,7 +91,7 @@ enum #define DEFAULT_ACTUAL_BUFFER_TIME -1 #define DEFAULT_ACTUAL_LATENCY_TIME -1 #define DEFAULT_PROVIDE_CLOCK TRUE -#define DEFAULT_SLAVE_METHOD GST_BASE_AUDIO_SRC_SLAVE_SKEW +#define DEFAULT_SLAVE_METHOD GST_AUDIO_BASE_SRC_SLAVE_SKEW enum { @@ -108,7 +108,7 @@ enum static void _do_init (GType type) { - GST_DEBUG_CATEGORY_INIT (gst_base_audio_src_debug, "baseaudiosrc", 0, + GST_DEBUG_CATEGORY_INIT (gst_audio_base_src_debug, "baseaudiosrc", 0, "baseaudiosrc element"); #ifdef ENABLE_NLS @@ -119,37 +119,37 @@ _do_init (GType type) #endif /* ENABLE_NLS */ } -#define gst_base_audio_src_parent_class parent_class -G_DEFINE_TYPE_WITH_CODE (GstBaseAudioSrc, gst_base_audio_src, GST_TYPE_PUSH_SRC, +#define gst_audio_base_src_parent_class parent_class +G_DEFINE_TYPE_WITH_CODE (GstAudioBaseSrc, gst_audio_base_src, GST_TYPE_PUSH_SRC, _do_init (g_define_type_id)); -static void gst_base_audio_src_set_property (GObject * object, guint prop_id, +static void gst_audio_base_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); -static void gst_base_audio_src_get_property (GObject * object, guint prop_id, +static void gst_audio_base_src_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); -static void gst_base_audio_src_dispose (GObject * object); +static void gst_audio_base_src_dispose (GObject * object); -static GstStateChangeReturn gst_base_audio_src_change_state (GstElement * +static GstStateChangeReturn gst_audio_base_src_change_state (GstElement * element, GstStateChange transition); -static GstClock *gst_base_audio_src_provide_clock (GstElement * elem); -static GstClockTime gst_base_audio_src_get_time (GstClock * clock, - GstBaseAudioSrc * src); +static GstClock *gst_audio_base_src_provide_clock (GstElement * elem); +static GstClockTime gst_audio_base_src_get_time (GstClock * clock, + GstAudioBaseSrc * src); -static GstFlowReturn gst_base_audio_src_create (GstBaseSrc * bsrc, +static GstFlowReturn gst_audio_base_src_create (GstBaseSrc * bsrc, guint64 offset, guint length, GstBuffer ** buf); -static gboolean gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event); -static void gst_base_audio_src_get_times (GstBaseSrc * bsrc, +static gboolean gst_audio_base_src_event (GstBaseSrc * bsrc, GstEvent * event); +static void gst_audio_base_src_get_times (GstBaseSrc * bsrc, GstBuffer * buffer, GstClockTime * start, GstClockTime * end); -static gboolean gst_base_audio_src_setcaps (GstBaseSrc * bsrc, GstCaps * caps); -static gboolean gst_base_audio_src_query (GstBaseSrc * bsrc, GstQuery * query); -static void gst_base_audio_src_fixate (GstBaseSrc * bsrc, GstCaps * caps); +static gboolean gst_audio_base_src_setcaps (GstBaseSrc * bsrc, GstCaps * caps); +static gboolean gst_audio_base_src_query (GstBaseSrc * bsrc, GstQuery * query); +static void gst_audio_base_src_fixate (GstBaseSrc * bsrc, GstCaps * caps); -/* static guint gst_base_audio_src_signals[LAST_SIGNAL] = { 0 }; */ +/* static guint gst_audio_base_src_signals[LAST_SIGNAL] = { 0 }; */ static void -gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass) +gst_audio_base_src_class_init (GstAudioBaseSrcClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; @@ -159,11 +159,11 @@ gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass) gstelement_class = (GstElementClass *) klass; gstbasesrc_class = (GstBaseSrcClass *) klass; - g_type_class_add_private (klass, sizeof (GstBaseAudioSrcPrivate)); + g_type_class_add_private (klass, sizeof (GstAudioBaseSrcPrivate)); - gobject_class->set_property = gst_base_audio_src_set_property; - gobject_class->get_property = gst_base_audio_src_get_property; - gobject_class->dispose = gst_base_audio_src_dispose; + gobject_class->set_property = gst_audio_base_src_set_property; + gobject_class->get_property = gst_audio_base_src_get_property; + gobject_class->dispose = gst_audio_base_src_dispose; g_object_class_install_property (gobject_class, PROP_BUFFER_TIME, g_param_spec_int64 ("buffer-time", "Buffer Time", @@ -178,7 +178,7 @@ gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass) G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** - * GstBaseAudioSrc:actual-buffer-time: + * GstAudioBaseSrc:actual-buffer-time: * * Actual configured size of audio buffer in microseconds. * @@ -191,7 +191,7 @@ gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass) G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); /** - * GstBaseAudioSrc:actual-latency-time: + * GstAudioBaseSrc:actual-latency-time: * * Actual configured audio latency in microseconds. * @@ -211,21 +211,21 @@ gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass) 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_BASE_AUDIO_SRC_SLAVE_METHOD, DEFAULT_SLAVE_METHOD, + GST_TYPE_AUDIO_BASE_SRC_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); + GST_DEBUG_FUNCPTR (gst_audio_base_src_change_state); gstelement_class->provide_clock = - GST_DEBUG_FUNCPTR (gst_base_audio_src_provide_clock); + GST_DEBUG_FUNCPTR (gst_audio_base_src_provide_clock); - gstbasesrc_class->set_caps = GST_DEBUG_FUNCPTR (gst_base_audio_src_setcaps); - gstbasesrc_class->event = GST_DEBUG_FUNCPTR (gst_base_audio_src_event); - gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_base_audio_src_query); + gstbasesrc_class->set_caps = GST_DEBUG_FUNCPTR (gst_audio_base_src_setcaps); + gstbasesrc_class->event = GST_DEBUG_FUNCPTR (gst_audio_base_src_event); + gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_audio_base_src_query); gstbasesrc_class->get_times = - GST_DEBUG_FUNCPTR (gst_base_audio_src_get_times); - gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_base_audio_src_create); - gstbasesrc_class->fixate = GST_DEBUG_FUNCPTR (gst_base_audio_src_fixate); + GST_DEBUG_FUNCPTR (gst_audio_base_src_get_times); + gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_audio_base_src_create); + gstbasesrc_class->fixate = GST_DEBUG_FUNCPTR (gst_audio_base_src_fixate); /* ref class from a thread-safe context to work around missing bit of * thread-safety in GObject */ @@ -234,9 +234,9 @@ gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass) } static void -gst_base_audio_src_init (GstBaseAudioSrc * baseaudiosrc) +gst_audio_base_src_init (GstAudioBaseSrc * baseaudiosrc) { - baseaudiosrc->priv = GST_BASE_AUDIO_SRC_GET_PRIVATE (baseaudiosrc); + baseaudiosrc->priv = GST_AUDIO_BASE_SRC_GET_PRIVATE (baseaudiosrc); baseaudiosrc->buffer_time = DEFAULT_BUFFER_TIME; baseaudiosrc->latency_time = DEFAULT_LATENCY_TIME; @@ -247,7 +247,7 @@ gst_base_audio_src_init (GstBaseAudioSrc * baseaudiosrc) GST_BASE_SRC (baseaudiosrc)->blocksize = 0; baseaudiosrc->clock = gst_audio_clock_new ("GstAudioSrcClock", - (GstAudioClockGetTimeFunc) gst_base_audio_src_get_time, baseaudiosrc, + (GstAudioClockGetTimeFunc) gst_audio_base_src_get_time, baseaudiosrc, NULL); /* we are always a live source */ @@ -257,11 +257,11 @@ gst_base_audio_src_init (GstBaseAudioSrc * baseaudiosrc) } static void -gst_base_audio_src_dispose (GObject * object) +gst_audio_base_src_dispose (GObject * object) { - GstBaseAudioSrc *src; + GstAudioBaseSrc *src; - src = GST_BASE_AUDIO_SRC (object); + src = GST_AUDIO_BASE_SRC (object); GST_OBJECT_LOCK (src); if (src->clock) { @@ -280,12 +280,12 @@ gst_base_audio_src_dispose (GObject * object) } static GstClock * -gst_base_audio_src_provide_clock (GstElement * elem) +gst_audio_base_src_provide_clock (GstElement * elem) { - GstBaseAudioSrc *src; + GstAudioBaseSrc *src; GstClock *clock; - src = GST_BASE_AUDIO_SRC (elem); + src = GST_AUDIO_BASE_SRC (elem); /* we have no ringbuffer (must be NULL state) */ if (src->ringbuffer == NULL) @@ -318,7 +318,7 @@ clock_disabled: } static GstClockTime -gst_base_audio_src_get_time (GstClock * clock, GstBaseAudioSrc * src) +gst_audio_base_src_get_time (GstClock * clock, GstAudioBaseSrc * src) { guint64 raw, samples; guint delay; @@ -348,8 +348,8 @@ gst_base_audio_src_get_time (GstClock * clock, GstBaseAudioSrc * src) } /** - * gst_base_audio_src_set_provide_clock: - * @src: a #GstBaseAudioSrc + * gst_audio_base_src_set_provide_clock: + * @src: a #GstAudioBaseSrc * @provide: new state * * Controls whether @src will provide a clock or not. If @provide is %TRUE, @@ -359,9 +359,9 @@ gst_base_audio_src_get_time (GstClock * clock, GstBaseAudioSrc * src) * Since: 0.10.16 */ void -gst_base_audio_src_set_provide_clock (GstBaseAudioSrc * src, gboolean provide) +gst_audio_base_src_set_provide_clock (GstAudioBaseSrc * src, gboolean provide) { - g_return_if_fail (GST_IS_BASE_AUDIO_SRC (src)); + g_return_if_fail (GST_IS_AUDIO_BASE_SRC (src)); GST_OBJECT_LOCK (src); src->priv->provide_clock = provide; @@ -369,22 +369,22 @@ gst_base_audio_src_set_provide_clock (GstBaseAudioSrc * src, gboolean provide) } /** - * gst_base_audio_src_get_provide_clock: - * @src: a #GstBaseAudioSrc + * gst_audio_base_src_get_provide_clock: + * @src: a #GstAudioBaseSrc * * Queries whether @src will provide a clock or not. See also - * gst_base_audio_src_set_provide_clock. + * gst_audio_base_src_set_provide_clock. * * Returns: %TRUE if @src will provide a clock. * * Since: 0.10.16 */ gboolean -gst_base_audio_src_get_provide_clock (GstBaseAudioSrc * src) +gst_audio_base_src_get_provide_clock (GstAudioBaseSrc * src) { gboolean result; - g_return_val_if_fail (GST_IS_BASE_AUDIO_SRC (src), FALSE); + g_return_val_if_fail (GST_IS_AUDIO_BASE_SRC (src), FALSE); GST_OBJECT_LOCK (src); result = src->priv->provide_clock; @@ -394,8 +394,8 @@ gst_base_audio_src_get_provide_clock (GstBaseAudioSrc * src) } /** - * gst_base_audio_src_set_slave_method: - * @src: a #GstBaseAudioSrc + * gst_audio_base_src_set_slave_method: + * @src: a #GstAudioBaseSrc * @method: the new slave method * * Controls how clock slaving will be performed in @src. @@ -403,10 +403,10 @@ gst_base_audio_src_get_provide_clock (GstBaseAudioSrc * src) * Since: 0.10.20 */ void -gst_base_audio_src_set_slave_method (GstBaseAudioSrc * src, - GstBaseAudioSrcSlaveMethod method) +gst_audio_base_src_set_slave_method (GstAudioBaseSrc * src, + GstAudioBaseSrcSlaveMethod method) { - g_return_if_fail (GST_IS_BASE_AUDIO_SRC (src)); + g_return_if_fail (GST_IS_AUDIO_BASE_SRC (src)); GST_OBJECT_LOCK (src); src->priv->slave_method = method; @@ -414,8 +414,8 @@ gst_base_audio_src_set_slave_method (GstBaseAudioSrc * src, } /** - * gst_base_audio_src_get_slave_method: - * @src: a #GstBaseAudioSrc + * gst_audio_base_src_get_slave_method: + * @src: a #GstAudioBaseSrc * * Get the current slave method used by @src. * @@ -423,12 +423,12 @@ gst_base_audio_src_set_slave_method (GstBaseAudioSrc * src, * * Since: 0.10.20 */ -GstBaseAudioSrcSlaveMethod -gst_base_audio_src_get_slave_method (GstBaseAudioSrc * src) +GstAudioBaseSrcSlaveMethod +gst_audio_base_src_get_slave_method (GstAudioBaseSrc * src) { - GstBaseAudioSrcSlaveMethod result; + GstAudioBaseSrcSlaveMethod result; - g_return_val_if_fail (GST_IS_BASE_AUDIO_SRC (src), -1); + g_return_val_if_fail (GST_IS_AUDIO_BASE_SRC (src), -1); GST_OBJECT_LOCK (src); result = src->priv->slave_method; @@ -438,12 +438,12 @@ gst_base_audio_src_get_slave_method (GstBaseAudioSrc * src) } static void -gst_base_audio_src_set_property (GObject * object, guint prop_id, +gst_audio_base_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { - GstBaseAudioSrc *src; + GstAudioBaseSrc *src; - src = GST_BASE_AUDIO_SRC (object); + src = GST_AUDIO_BASE_SRC (object); switch (prop_id) { case PROP_BUFFER_TIME: @@ -453,10 +453,10 @@ gst_base_audio_src_set_property (GObject * object, guint prop_id, src->latency_time = g_value_get_int64 (value); break; case PROP_PROVIDE_CLOCK: - gst_base_audio_src_set_provide_clock (src, g_value_get_boolean (value)); + gst_audio_base_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)); + gst_audio_base_src_set_slave_method (src, g_value_get_enum (value)); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); @@ -465,12 +465,12 @@ gst_base_audio_src_set_property (GObject * object, guint prop_id, } static void -gst_base_audio_src_get_property (GObject * object, guint prop_id, +gst_audio_base_src_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { - GstBaseAudioSrc *src; + GstAudioBaseSrc *src; - src = GST_BASE_AUDIO_SRC (object); + src = GST_AUDIO_BASE_SRC (object); switch (prop_id) { case PROP_BUFFER_TIME: @@ -496,10 +496,10 @@ gst_base_audio_src_get_property (GObject * object, guint prop_id, GST_OBJECT_UNLOCK (src); break; case PROP_PROVIDE_CLOCK: - g_value_set_boolean (value, gst_base_audio_src_get_provide_clock (src)); + g_value_set_boolean (value, gst_audio_base_src_get_provide_clock (src)); break; case PROP_SLAVE_METHOD: - g_value_set_enum (value, gst_base_audio_src_get_slave_method (src)); + g_value_set_enum (value, gst_audio_base_src_get_slave_method (src)); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); @@ -508,7 +508,7 @@ gst_base_audio_src_get_property (GObject * object, guint prop_id, } static void -gst_base_audio_src_fixate (GstBaseSrc * bsrc, GstCaps * caps) +gst_audio_base_src_fixate (GstBaseSrc * bsrc, GstCaps * caps) { GstStructure *s; @@ -524,9 +524,9 @@ gst_base_audio_src_fixate (GstBaseSrc * bsrc, GstCaps * caps) } static gboolean -gst_base_audio_src_setcaps (GstBaseSrc * bsrc, GstCaps * caps) +gst_audio_base_src_setcaps (GstBaseSrc * bsrc, GstCaps * caps) { - GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc); + GstAudioBaseSrc *src = GST_AUDIO_BASE_SRC (bsrc); GstAudioRingBufferSpec *spec; gint bpf, rate; @@ -587,7 +587,7 @@ acquire_error: } static void -gst_base_audio_src_get_times (GstBaseSrc * bsrc, GstBuffer * buffer, +gst_audio_base_src_get_times (GstBaseSrc * bsrc, GstBuffer * buffer, GstClockTime * start, GstClockTime * end) { /* no need to sync to a clock here, we schedule the samples based @@ -597,9 +597,9 @@ gst_base_audio_src_get_times (GstBaseSrc * bsrc, GstBuffer * buffer, } static gboolean -gst_base_audio_src_query (GstBaseSrc * bsrc, GstQuery * query) +gst_audio_base_src_query (GstBaseSrc * bsrc, GstQuery * query) { - GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc); + GstAudioBaseSrc *src = GST_AUDIO_BASE_SRC (bsrc); gboolean res = FALSE; switch (GST_QUERY_TYPE (query)) { @@ -660,9 +660,9 @@ done: } static gboolean -gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event) +gst_audio_base_src_event (GstBaseSrc * bsrc, GstEvent * event) { - GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc); + GstAudioBaseSrc *src = GST_AUDIO_BASE_SRC (bsrc); gboolean res, forward; res = FALSE; @@ -698,7 +698,7 @@ gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event) * If the next sample is too far away, this function will position itself to the * next most recent sample, creating discontinuity */ static guint64 -gst_base_audio_src_get_offset (GstBaseAudioSrc * src) +gst_audio_base_src_get_offset (GstAudioBaseSrc * src) { guint64 sample; gint readseg, segdone, segtotal, sps; @@ -745,11 +745,11 @@ gst_base_audio_src_get_offset (GstBaseAudioSrc * src) } static GstFlowReturn -gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length, +gst_audio_base_src_create (GstBaseSrc * bsrc, guint64 offset, guint length, GstBuffer ** outbuf) { GstFlowReturn ret; - GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc); + GstAudioBaseSrc *src = GST_AUDIO_BASE_SRC (bsrc); GstBuffer *buf; guchar *data, *ptr; guint samples, total_samples; @@ -787,7 +787,7 @@ gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length, } else { /* calculate the sequentially next sample we need to read. This can jump and * create a DISCONT. */ - sample = gst_base_audio_src_get_offset (src); + sample = gst_audio_base_src_get_offset (src); } GST_DEBUG_OBJECT (src, "reading from sample %" G_GUINT64_FORMAT " length %u", @@ -850,11 +850,11 @@ gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length, if (clock != src->clock) { /* we are slaved, check how to handle this */ switch (src->priv->slave_method) { - case GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE: + case GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE: /* not implemented, use skew 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: + case GST_AUDIO_BASE_SRC_SLAVE_SKEW: { GstClockTime running_time; GstClockTime base_time; @@ -955,7 +955,7 @@ gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length, } break; } - case GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP: + case GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP: { GstClockTime base_time, latency; @@ -977,7 +977,7 @@ gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length, else timestamp = 0; } - case GST_BASE_AUDIO_SRC_SLAVE_NONE: + case GST_AUDIO_BASE_SRC_SLAVE_NONE: break; } } else { @@ -1042,8 +1042,8 @@ stopped: } /** - * gst_base_audio_src_create_ringbuffer: - * @src: a #GstBaseAudioSrc. + * gst_audio_base_src_create_ringbuffer: + * @src: a #GstAudioBaseSrc. * * Create and return the #GstAudioRingBuffer for @src. This function will call the * ::create_ringbuffer vmethod and will set @src as the parent of the returned @@ -1052,12 +1052,12 @@ stopped: * Returns: The new ringbuffer of @src. */ GstAudioRingBuffer * -gst_base_audio_src_create_ringbuffer (GstBaseAudioSrc * src) +gst_audio_base_src_create_ringbuffer (GstAudioBaseSrc * src) { - GstBaseAudioSrcClass *bclass; + GstAudioBaseSrcClass *bclass; GstAudioRingBuffer *buffer = NULL; - bclass = GST_BASE_AUDIO_SRC_GET_CLASS (src); + bclass = GST_AUDIO_BASE_SRC_GET_CLASS (src); if (bclass->create_ringbuffer) buffer = bclass->create_ringbuffer (src); @@ -1068,11 +1068,11 @@ gst_base_audio_src_create_ringbuffer (GstBaseAudioSrc * src) } static GstStateChangeReturn -gst_base_audio_src_change_state (GstElement * element, +gst_audio_base_src_change_state (GstElement * element, GstStateChange transition) { GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; - GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (element); + GstAudioBaseSrc *src = GST_AUDIO_BASE_SRC (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: @@ -1080,7 +1080,7 @@ gst_base_audio_src_change_state (GstElement * element, GST_OBJECT_LOCK (src); if (src->ringbuffer == NULL) { gst_audio_clock_reset (GST_AUDIO_CLOCK (src->clock), 0); - src->ringbuffer = gst_base_audio_src_create_ringbuffer (src); + src->ringbuffer = gst_audio_base_src_create_ringbuffer (src); } GST_OBJECT_UNLOCK (src); if (!gst_audio_ring_buffer_open_device (src->ringbuffer)) @@ -1096,7 +1096,7 @@ gst_base_audio_src_change_state (GstElement * element, * should post this messages whenever necessary */ if (src->clock && GST_IS_AUDIO_CLOCK (src->clock) && GST_AUDIO_CLOCK_CAST (src->clock)->func == - (GstAudioClockGetTimeFunc) gst_base_audio_src_get_time) + (GstAudioClockGetTimeFunc) gst_audio_base_src_get_time) gst_element_post_message (element, gst_message_new_clock_provide (GST_OBJECT_CAST (element), src->clock, TRUE)); @@ -1117,7 +1117,7 @@ gst_base_audio_src_change_state (GstElement * element, * should post this messages whenever necessary */ if (src->clock && GST_IS_AUDIO_CLOCK (src->clock) && GST_AUDIO_CLOCK_CAST (src->clock)->func == - (GstAudioClockGetTimeFunc) gst_base_audio_src_get_time) + (GstAudioClockGetTimeFunc) gst_audio_base_src_get_time) gst_element_post_message (element, gst_message_new_clock_lost (GST_OBJECT_CAST (element), src->clock)); gst_audio_ring_buffer_set_flushing (src->ringbuffer, TRUE); diff --git a/gst-libs/gst/audio/gstbaseaudiosrc.h b/gst-libs/gst/audio/gstbaseaudiosrc.h index 9db16b4654..96738ab871 100644 --- a/gst-libs/gst/audio/gstbaseaudiosrc.h +++ b/gst-libs/gst/audio/gstbaseaudiosrc.h @@ -23,8 +23,8 @@ /* a base class for audio sources. */ -#ifndef __GST_BASE_AUDIO_SRC_H__ -#define __GST_BASE_AUDIO_SRC_H__ +#ifndef __GST_AUDIO_BASE_SRC_H__ +#define __GST_AUDIO_BASE_SRC_H__ #include #include @@ -33,60 +33,60 @@ G_BEGIN_DECLS -#define GST_TYPE_BASE_AUDIO_SRC (gst_base_audio_src_get_type()) -#define GST_BASE_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_SRC,GstBaseAudioSrc)) -#define GST_BASE_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_SRC,GstBaseAudioSrcClass)) -#define GST_BASE_AUDIO_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_BASE_AUDIO_SRC, GstBaseAudioSrcClass)) -#define GST_IS_BASE_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_SRC)) -#define GST_IS_BASE_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_SRC)) +#define GST_TYPE_AUDIO_BASE_SRC (gst_audio_base_src_get_type()) +#define GST_AUDIO_BASE_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_BASE_SRC,GstAudioBaseSrc)) +#define GST_AUDIO_BASE_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_BASE_SRC,GstAudioBaseSrcClass)) +#define GST_AUDIO_BASE_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIO_BASE_SRC, GstAudioBaseSrcClass)) +#define GST_IS_AUDIO_BASE_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_BASE_SRC)) +#define GST_IS_AUDIO_BASE_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_BASE_SRC)) /** - * GST_BASE_AUDIO_SRC_CLOCK: - * @obj: a #GstBaseAudioSrc + * GST_AUDIO_BASE_SRC_CLOCK: + * @obj: a #GstAudioBaseSrc * * Get the #GstClock of @obj. */ -#define GST_BASE_AUDIO_SRC_CLOCK(obj) (GST_BASE_AUDIO_SRC (obj)->clock) +#define GST_AUDIO_BASE_SRC_CLOCK(obj) (GST_AUDIO_BASE_SRC (obj)->clock) /** - * GST_BASE_AUDIO_SRC_PAD: - * @obj: a #GstBaseAudioSrc + * GST_AUDIO_BASE_SRC_PAD: + * @obj: a #GstAudioBaseSrc * * Get the source #GstPad of @obj. */ -#define GST_BASE_AUDIO_SRC_PAD(obj) (GST_BASE_SRC (obj)->srcpad) +#define GST_AUDIO_BASE_SRC_PAD(obj) (GST_BASE_SRC (obj)->srcpad) -typedef struct _GstBaseAudioSrc GstBaseAudioSrc; -typedef struct _GstBaseAudioSrcClass GstBaseAudioSrcClass; -typedef struct _GstBaseAudioSrcPrivate GstBaseAudioSrcPrivate; +typedef struct _GstAudioBaseSrc GstAudioBaseSrc; +typedef struct _GstAudioBaseSrcClass GstAudioBaseSrcClass; +typedef struct _GstAudioBaseSrcPrivate GstAudioBaseSrcPrivate; /** - * GstBaseAudioSrcSlaveMethod: - * @GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE: Resample to match the master clock. - * @GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP: Retimestamp output buffers with master + * GstAudioBaseSrcSlaveMethod: + * @GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE: Resample to match the master clock. + * @GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP: Retimestamp output buffers with master * clock time. - * @GST_BASE_AUDIO_SRC_SLAVE_SKEW: Adjust capture pointer when master clock + * @GST_AUDIO_BASE_SRC_SLAVE_SKEW: Adjust capture pointer when master clock * drifts too much. - * @GST_BASE_AUDIO_SRC_SLAVE_NONE: No adjustment is done. + * @GST_AUDIO_BASE_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; + GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE, + GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP, + GST_AUDIO_BASE_SRC_SLAVE_SKEW, + GST_AUDIO_BASE_SRC_SLAVE_NONE +} GstAudioBaseSrcSlaveMethod; -#define GST_TYPE_BASE_AUDIO_SRC_SLAVE_METHOD (gst_base_audio_src_slave_method_get_type ()) +#define GST_TYPE_AUDIO_BASE_SRC_SLAVE_METHOD (gst_audio_base_src_slave_method_get_type ()) /** - * GstBaseAudioSrc: + * GstAudioBaseSrc: * - * Opaque #GstBaseAudioSrc. + * Opaque #GstAudioBaseSrc. */ -struct _GstBaseAudioSrc { +struct _GstAudioBaseSrc { GstPushSrc element; /*< protected >*/ /* with LOCK */ @@ -104,44 +104,44 @@ struct _GstBaseAudioSrc { GstClock *clock; /*< private >*/ - GstBaseAudioSrcPrivate *priv; + GstAudioBaseSrcPrivate *priv; gpointer _gst_reserved[GST_PADDING - 1]; }; /** - * GstBaseAudioSrcClass: + * GstAudioBaseSrcClass: * @parent_class: the parent class. * @create_ringbuffer: create and return a #GstAudioRingBuffer to read from. * - * #GstBaseAudioSrc class. Override the vmethod to implement + * #GstAudioBaseSrc class. Override the vmethod to implement * functionality. */ -struct _GstBaseAudioSrcClass { +struct _GstAudioBaseSrcClass { GstPushSrcClass parent_class; /* subclass ringbuffer allocation */ - GstAudioRingBuffer* (*create_ringbuffer) (GstBaseAudioSrc *src); + GstAudioRingBuffer* (*create_ringbuffer) (GstAudioBaseSrc *src); /*< private >*/ gpointer _gst_reserved[GST_PADDING]; }; -GType gst_base_audio_src_get_type(void); -GType gst_base_audio_src_slave_method_get_type (void); +GType gst_audio_base_src_get_type(void); +GType gst_audio_base_src_slave_method_get_type (void); GstAudioRingBuffer * - gst_base_audio_src_create_ringbuffer (GstBaseAudioSrc *src); + gst_audio_base_src_create_ringbuffer (GstAudioBaseSrc *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_audio_base_src_set_provide_clock (GstAudioBaseSrc *src, gboolean provide); +gboolean gst_audio_base_src_get_provide_clock (GstAudioBaseSrc *src); -void gst_base_audio_src_set_slave_method (GstBaseAudioSrc *src, - GstBaseAudioSrcSlaveMethod method); -GstBaseAudioSrcSlaveMethod - gst_base_audio_src_get_slave_method (GstBaseAudioSrc *src); +void gst_audio_base_src_set_slave_method (GstAudioBaseSrc *src, + GstAudioBaseSrcSlaveMethod method); +GstAudioBaseSrcSlaveMethod + gst_audio_base_src_get_slave_method (GstAudioBaseSrc *src); G_END_DECLS -#endif /* __GST_BASE_AUDIO_SRC_H__ */ +#endif /* __GST_AUDIO_BASE_SRC_H__ */