From: Wim Taymans Date: Fri, 11 Nov 2011 11:32:23 +0000 (+0100) Subject: rename files to match object names X-Git-Tag: 1.19.3~511^2~7167 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=5f1312b5d8ac5fc08f13b0ef6946847dfdb77f23;p=platform%2Fupstream%2Fgstreamer.git rename files to match object names --- diff --git a/docs/libs/gst-plugins-base-libs-sections.txt b/docs/libs/gst-plugins-base-libs-sections.txt index ab4282e..29af9b3 100644 --- a/docs/libs/gst-plugins-base-libs-sections.txt +++ b/docs/libs/gst-plugins-base-libs-sections.txt @@ -1123,8 +1123,8 @@ gst_riff_strh # rtp
-gstbasertpaudiopayload -gst/rtp/gstbasertpaudiopayload.h +gstrtpbaseaudiopayload +gst/rtp/gstrtpbaseaudiopayload.h GstRTPBaseAudioPayload GstRTPBaseAudioPayloadClass @@ -1148,8 +1148,8 @@ GstRTPBaseAudioPayloadPrivate
-gstbasertpdepayload -gst/rtp/gstbasertpdepayload.h +gstrtpbasedepayload +gst/rtp/gstrtpbasedepayload.h GstRTPBaseDepayload GstRTPBaseDepayloadClass @@ -1179,8 +1179,8 @@ QUEUE_UNLOCK
-gstbasertppayload -gst/rtp/gstbasertppayload.h +gstrtpbasepayload +gst/rtp/gstrtpbasepayload.h GstRTPBasePayload GstRTPBasePayloadClass diff --git a/docs/libs/gst-plugins-base-libs.types b/docs/libs/gst-plugins-base-libs.types index 43444a7..4b8731a 100644 --- a/docs/libs/gst-plugins-base-libs.types +++ b/docs/libs/gst-plugins-base-libs.types @@ -48,11 +48,11 @@ gst_video_orientation_get_type gst_video_overlay_get_type -#include +#include gst_rtp_base_depayload_get_type -#include +#include gst_rtp_base_payload_get_type -#include +#include gst_rtp_base_audio_payload_get_type diff --git a/gst-libs/gst/rtp/Makefile.am b/gst-libs/gst/rtp/Makefile.am index 79d8126..cd0946b 100644 --- a/gst-libs/gst/rtp/Makefile.am +++ b/gst-libs/gst/rtp/Makefile.am @@ -3,18 +3,18 @@ libgstrtpincludedir = $(includedir)/gstreamer-@GST_MAJORMINOR@/gst/rtp libgstrtpinclude_HEADERS = gstrtpbuffer.h \ gstrtcpbuffer.h \ gstrtppayloads.h \ - gstbasertpaudiopayload.h \ - gstbasertppayload.h \ - gstbasertpdepayload.h + gstrtpbaseaudiopayload.h \ + gstrtpbasepayload.h \ + gstrtpbasedepayload.h lib_LTLIBRARIES = libgstrtp-@GST_MAJORMINOR@.la libgstrtp_@GST_MAJORMINOR@_la_SOURCES = gstrtpbuffer.c \ gstrtcpbuffer.c \ gstrtppayloads.c \ - gstbasertpaudiopayload.c \ - gstbasertppayload.c \ - gstbasertpdepayload.c + gstrtpbaseaudiopayload.c \ + gstrtpbasepayload.c \ + gstrtpbasedepayload.c libgstrtp_@GST_MAJORMINOR@_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) libgstrtp_@GST_MAJORMINOR@_la_LIBADD = $(GST_BASE_LIBS) $(GST_LIBS) diff --git a/gst-libs/gst/rtp/gstbasertpaudiopayload.c b/gst-libs/gst/rtp/gstrtpbaseaudiopayload.c similarity index 91% rename from gst-libs/gst/rtp/gstbasertpaudiopayload.c rename to gst-libs/gst/rtp/gstrtpbaseaudiopayload.c index e5f29fe..95bfc3b 100644 --- a/gst-libs/gst/rtp/gstbasertpaudiopayload.c +++ b/gst-libs/gst/rtp/gstrtpbaseaudiopayload.c @@ -18,7 +18,7 @@ */ /** - * SECTION:gstbasertpaudiopayload + * SECTION:gstrtpbaseaudiopayload * @short_description: Base class for audio RTP payloader * * Provides a base class for audio RTP payloaders for frame or sample based @@ -63,10 +63,10 @@ #include #include -#include "gstbasertpaudiopayload.h" +#include "gstrtpbaseaudiopayload.h" -GST_DEBUG_CATEGORY_STATIC (basertpaudiopayload_debug); -#define GST_CAT_DEFAULT (basertpaudiopayload_debug) +GST_DEBUG_CATEGORY_STATIC (rtpbaseaudiopayload_debug); +#define GST_CAT_DEFAULT (rtpbaseaudiopayload_debug) #define DEFAULT_BUFFER_LIST FALSE @@ -166,13 +166,13 @@ gst_rtp_base_audio_payload_class_init (GstRTPBaseAudioPayloadClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; - GstRTPBasePayloadClass *gstbasertppayload_class; + GstRTPBasePayloadClass *gstrtpbasepayload_class; g_type_class_add_private (klass, sizeof (GstRTPBaseAudioPayloadPrivate)); gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; - gstbasertppayload_class = (GstRTPBasePayloadClass *) klass; + gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass; gobject_class->finalize = gst_rtp_base_audio_payload_finalize; gobject_class->set_property = gst_rtp_base_audio_payload_set_property; @@ -186,12 +186,12 @@ gst_rtp_base_audio_payload_class_init (GstRTPBaseAudioPayloadClass * klass) gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_base_payload_audio_change_state); - gstbasertppayload_class->handle_buffer = + gstrtpbasepayload_class->handle_buffer = GST_DEBUG_FUNCPTR (gst_rtp_base_audio_payload_handle_buffer); - gstbasertppayload_class->handle_event = + gstrtpbasepayload_class->handle_event = GST_DEBUG_FUNCPTR (gst_rtp_base_payload_audio_handle_event); - GST_DEBUG_CATEGORY_INIT (basertpaudiopayload_debug, "basertpaudiopayload", 0, + GST_DEBUG_CATEGORY_INIT (rtpbaseaudiopayload_debug, "rtpbaseaudiopayload", 0, "base audio RTP payloader"); } @@ -262,55 +262,55 @@ gst_rtp_base_audio_payload_get_property (GObject * object, /** * gst_rtp_base_audio_payload_set_frame_based: - * @basertpaudiopayload: a pointer to the element. + * @rtpbaseaudiopayload: a pointer to the element. * * Tells #GstRTPBaseAudioPayload that the child element is for a frame based * audio codec */ void gst_rtp_base_audio_payload_set_frame_based (GstRTPBaseAudioPayload * - basertpaudiopayload) + rtpbaseaudiopayload) { - g_return_if_fail (basertpaudiopayload != NULL); - g_return_if_fail (basertpaudiopayload->priv->time_to_bytes == NULL); - g_return_if_fail (basertpaudiopayload->priv->bytes_to_time == NULL); - g_return_if_fail (basertpaudiopayload->priv->bytes_to_rtptime == NULL); + g_return_if_fail (rtpbaseaudiopayload != NULL); + g_return_if_fail (rtpbaseaudiopayload->priv->time_to_bytes == NULL); + g_return_if_fail (rtpbaseaudiopayload->priv->bytes_to_time == NULL); + g_return_if_fail (rtpbaseaudiopayload->priv->bytes_to_rtptime == NULL); - basertpaudiopayload->priv->bytes_to_time = + rtpbaseaudiopayload->priv->bytes_to_time = gst_rtp_base_audio_payload_frame_bytes_to_time; - basertpaudiopayload->priv->bytes_to_rtptime = + rtpbaseaudiopayload->priv->bytes_to_rtptime = gst_rtp_base_audio_payload_frame_bytes_to_rtptime; - basertpaudiopayload->priv->time_to_bytes = + rtpbaseaudiopayload->priv->time_to_bytes = gst_rtp_base_audio_payload_frame_time_to_bytes; } /** * gst_rtp_base_audio_payload_set_sample_based: - * @basertpaudiopayload: a pointer to the element. + * @rtpbaseaudiopayload: a pointer to the element. * * Tells #GstRTPBaseAudioPayload that the child element is for a sample based * audio codec */ void gst_rtp_base_audio_payload_set_sample_based (GstRTPBaseAudioPayload * - basertpaudiopayload) + rtpbaseaudiopayload) { - g_return_if_fail (basertpaudiopayload != NULL); - g_return_if_fail (basertpaudiopayload->priv->time_to_bytes == NULL); - g_return_if_fail (basertpaudiopayload->priv->bytes_to_time == NULL); - g_return_if_fail (basertpaudiopayload->priv->bytes_to_rtptime == NULL); + g_return_if_fail (rtpbaseaudiopayload != NULL); + g_return_if_fail (rtpbaseaudiopayload->priv->time_to_bytes == NULL); + g_return_if_fail (rtpbaseaudiopayload->priv->bytes_to_time == NULL); + g_return_if_fail (rtpbaseaudiopayload->priv->bytes_to_rtptime == NULL); - basertpaudiopayload->priv->bytes_to_time = + rtpbaseaudiopayload->priv->bytes_to_time = gst_rtp_base_audio_payload_sample_bytes_to_time; - basertpaudiopayload->priv->bytes_to_rtptime = + rtpbaseaudiopayload->priv->bytes_to_rtptime = gst_rtp_base_audio_payload_sample_bytes_to_rtptime; - basertpaudiopayload->priv->time_to_bytes = + rtpbaseaudiopayload->priv->time_to_bytes = gst_rtp_base_audio_payload_sample_time_to_bytes; } /** * gst_rtp_base_audio_payload_set_frame_options: - * @basertpaudiopayload: a pointer to the element. + * @rtpbaseaudiopayload: a pointer to the element. * @frame_duration: The duraction of an audio frame in milliseconds. * @frame_size: The size of an audio frame in bytes. * @@ -319,46 +319,46 @@ gst_rtp_base_audio_payload_set_sample_based (GstRTPBaseAudioPayload * */ void gst_rtp_base_audio_payload_set_frame_options (GstRTPBaseAudioPayload - * basertpaudiopayload, gint frame_duration, gint frame_size) + * rtpbaseaudiopayload, gint frame_duration, gint frame_size) { GstRTPBaseAudioPayloadPrivate *priv; - g_return_if_fail (basertpaudiopayload != NULL); + g_return_if_fail (rtpbaseaudiopayload != NULL); - priv = basertpaudiopayload->priv; + priv = rtpbaseaudiopayload->priv; - basertpaudiopayload->frame_duration = frame_duration; + rtpbaseaudiopayload->frame_duration = frame_duration; priv->frame_duration_ns = frame_duration * GST_MSECOND; - basertpaudiopayload->frame_size = frame_size; + rtpbaseaudiopayload->frame_size = frame_size; priv->align = frame_size; gst_adapter_clear (priv->adapter); - GST_DEBUG_OBJECT (basertpaudiopayload, "frame set to %d ms and size %d", + GST_DEBUG_OBJECT (rtpbaseaudiopayload, "frame set to %d ms and size %d", frame_duration, frame_size); } /** * gst_rtp_base_audio_payload_set_sample_options: - * @basertpaudiopayload: a pointer to the element. + * @rtpbaseaudiopayload: a pointer to the element. * @sample_size: Size per sample in bytes. * * Sets the options for sample based audio codecs. */ void gst_rtp_base_audio_payload_set_sample_options (GstRTPBaseAudioPayload - * basertpaudiopayload, gint sample_size) + * rtpbaseaudiopayload, gint sample_size) { - g_return_if_fail (basertpaudiopayload != NULL); + g_return_if_fail (rtpbaseaudiopayload != NULL); /* sample_size is in bits internally */ - gst_rtp_base_audio_payload_set_samplebits_options (basertpaudiopayload, + gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload, sample_size * 8); } /** * gst_rtp_base_audio_payload_set_samplebits_options: - * @basertpaudiopayload: a pointer to the element. + * @rtpbaseaudiopayload: a pointer to the element. * @sample_size: Size per sample in bits. * * Sets the options for sample based audio codecs. @@ -367,16 +367,16 @@ gst_rtp_base_audio_payload_set_sample_options (GstRTPBaseAudioPayload */ void gst_rtp_base_audio_payload_set_samplebits_options (GstRTPBaseAudioPayload - * basertpaudiopayload, gint sample_size) + * rtpbaseaudiopayload, gint sample_size) { guint fragment_size; GstRTPBaseAudioPayloadPrivate *priv; - g_return_if_fail (basertpaudiopayload != NULL); + g_return_if_fail (rtpbaseaudiopayload != NULL); - priv = basertpaudiopayload->priv; + priv = rtpbaseaudiopayload->priv; - basertpaudiopayload->sample_size = sample_size; + rtpbaseaudiopayload->sample_size = sample_size; /* sample_size is in bits and is converted into multiple bytes */ fragment_size = sample_size; @@ -387,7 +387,7 @@ gst_rtp_base_audio_payload_set_samplebits_options (GstRTPBaseAudioPayload gst_adapter_clear (priv->adapter); - GST_DEBUG_OBJECT (basertpaudiopayload, + GST_DEBUG_OBJECT (rtpbaseaudiopayload, "Samplebits set to sample size %d bits", sample_size); } @@ -922,16 +922,16 @@ static GstStateChangeReturn gst_rtp_base_payload_audio_change_state (GstElement * element, GstStateChange transition) { - GstRTPBaseAudioPayload *basertppayload; + GstRTPBaseAudioPayload *rtpbasepayload; GstStateChangeReturn ret; - basertppayload = GST_RTP_BASE_AUDIO_PAYLOAD (element); + rtpbasepayload = GST_RTP_BASE_AUDIO_PAYLOAD (element); switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: - basertppayload->priv->cached_mtu = -1; - basertppayload->priv->last_rtptime = -1; - basertppayload->priv->last_timestamp = -1; + rtpbasepayload->priv->cached_mtu = -1; + rtpbasepayload->priv->last_rtptime = -1; + rtpbasepayload->priv->last_timestamp = -1; break; default: break; @@ -941,7 +941,7 @@ gst_rtp_base_payload_audio_change_state (GstElement * element, switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY: - gst_adapter_clear (basertppayload->priv->adapter); + gst_adapter_clear (rtpbasepayload->priv->adapter); break; default: break; @@ -979,7 +979,7 @@ gst_rtp_base_payload_audio_handle_event (GstRTPBasePayload * basep, /** * gst_rtp_base_audio_payload_get_adapter: - * @basertpaudiopayload: a #GstRTPBaseAudioPayload + * @rtpbaseaudiopayload: a #GstRTPBaseAudioPayload * * Gets the internal adapter used by the depayloader. * @@ -989,11 +989,11 @@ gst_rtp_base_payload_audio_handle_event (GstRTPBasePayload * basep, */ GstAdapter * gst_rtp_base_audio_payload_get_adapter (GstRTPBaseAudioPayload - * basertpaudiopayload) + * rtpbaseaudiopayload) { GstAdapter *adapter; - if ((adapter = basertpaudiopayload->priv->adapter)) + if ((adapter = rtpbaseaudiopayload->priv->adapter)) g_object_ref (adapter); return adapter; diff --git a/gst-libs/gst/rtp/gstbasertpaudiopayload.h b/gst-libs/gst/rtp/gstrtpbaseaudiopayload.h similarity index 92% rename from gst-libs/gst/rtp/gstbasertpaudiopayload.h rename to gst-libs/gst/rtp/gstrtpbaseaudiopayload.h index 34a160c..6a9ae12 100644 --- a/gst-libs/gst/rtp/gstbasertpaudiopayload.h +++ b/gst-libs/gst/rtp/gstrtpbaseaudiopayload.h @@ -21,7 +21,7 @@ #define __GST_RTP_BASE_AUDIO_PAYLOAD_H__ #include -#include +#include #include G_BEGIN_DECLS @@ -78,20 +78,20 @@ struct _GstRTPBaseAudioPayloadClass GType gst_rtp_base_audio_payload_get_type (void); /* configure frame based */ -void gst_rtp_base_audio_payload_set_frame_based (GstRTPBaseAudioPayload *basertpaudiopayload); +void gst_rtp_base_audio_payload_set_frame_based (GstRTPBaseAudioPayload *rtpbaseaudiopayload); -void gst_rtp_base_audio_payload_set_frame_options (GstRTPBaseAudioPayload *basertpaudiopayload, +void gst_rtp_base_audio_payload_set_frame_options (GstRTPBaseAudioPayload *rtpbaseaudiopayload, gint frame_duration, gint frame_size); /* configure sample based */ -void gst_rtp_base_audio_payload_set_sample_based (GstRTPBaseAudioPayload *basertpaudiopayload); -void gst_rtp_base_audio_payload_set_sample_options (GstRTPBaseAudioPayload *basertpaudiopayload, +void gst_rtp_base_audio_payload_set_sample_based (GstRTPBaseAudioPayload *rtpbaseaudiopayload); +void gst_rtp_base_audio_payload_set_sample_options (GstRTPBaseAudioPayload *rtpbaseaudiopayload, gint sample_size); -void gst_rtp_base_audio_payload_set_samplebits_options (GstRTPBaseAudioPayload *basertpaudiopayload, +void gst_rtp_base_audio_payload_set_samplebits_options (GstRTPBaseAudioPayload *rtpbaseaudiopayload, gint sample_size); /* get the internal adapter */ -GstAdapter* gst_rtp_base_audio_payload_get_adapter (GstRTPBaseAudioPayload *basertpaudiopayload); +GstAdapter* gst_rtp_base_audio_payload_get_adapter (GstRTPBaseAudioPayload *rtpbaseaudiopayload); /* push and flushing data */ GstFlowReturn gst_rtp_base_audio_payload_push (GstRTPBaseAudioPayload * baseaudiopayload, diff --git a/gst-libs/gst/rtp/gstbasertpdepayload.c b/gst-libs/gst/rtp/gstrtpbasedepayload.c similarity index 98% rename from gst-libs/gst/rtp/gstbasertpdepayload.c rename to gst-libs/gst/rtp/gstrtpbasedepayload.c index ad3f5be..f5d1f3e 100644 --- a/gst-libs/gst/rtp/gstbasertpdepayload.c +++ b/gst-libs/gst/rtp/gstrtpbasedepayload.c @@ -19,16 +19,16 @@ */ /** - * SECTION:gstbasertpdepayload + * SECTION:gstrtpbasedepayload * @short_description: Base class for RTP depayloader * * Provides a base class for RTP depayloaders */ -#include "gstbasertpdepayload.h" +#include "gstrtpbasedepayload.h" -GST_DEBUG_CATEGORY_STATIC (basertpdepayload_debug); -#define GST_CAT_DEFAULT (basertpdepayload_debug) +GST_DEBUG_CATEGORY_STATIC (rtpbasedepayload_debug); +#define GST_CAT_DEFAULT (rtpbasedepayload_debug) #define GST_RTP_BASE_DEPAYLOAD_GET_PRIVATE(obj) \ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BASE_DEPAYLOAD, GstRTPBaseDepayloadPrivate)) @@ -85,7 +85,7 @@ static gboolean gst_rtp_base_depayload_handle_event (GstRTPBaseDepayload * static GstElementClass *parent_class = NULL; static void gst_rtp_base_depayload_class_init (GstRTPBaseDepayloadClass * klass); -static void gst_rtp_base_depayload_init (GstRTPBaseDepayload * basertppayload, +static void gst_rtp_base_depayload_init (GstRTPBaseDepayload * rtpbasepayload, GstRTPBaseDepayloadClass * klass); GType @@ -134,7 +134,7 @@ gst_rtp_base_depayload_class_init (GstRTPBaseDepayloadClass * klass) klass->packet_lost = gst_rtp_base_depayload_packet_lost; klass->handle_event = gst_rtp_base_depayload_handle_event; - GST_DEBUG_CATEGORY_INIT (basertpdepayload_debug, "basertpdepayload", 0, + GST_DEBUG_CATEGORY_INIT (rtpbasedepayload_debug, "rtpbasedepayload", 0, "Base class for RTP Depayloaders"); } diff --git a/gst-libs/gst/rtp/gstbasertpdepayload.h b/gst-libs/gst/rtp/gstrtpbasedepayload.h similarity index 100% rename from gst-libs/gst/rtp/gstbasertpdepayload.h rename to gst-libs/gst/rtp/gstrtpbasedepayload.h diff --git a/gst-libs/gst/rtp/gstbasertppayload.c b/gst-libs/gst/rtp/gstrtpbasepayload.c similarity index 78% rename from gst-libs/gst/rtp/gstbasertppayload.c rename to gst-libs/gst/rtp/gstrtpbasepayload.c index 2b14a1c..29bcb10 100644 --- a/gst-libs/gst/rtp/gstbasertppayload.c +++ b/gst-libs/gst/rtp/gstrtpbasepayload.c @@ -13,7 +13,7 @@ */ /** - * SECTION:gstbasertppayload + * SECTION:gstrtpbasepayload * @short_description: Base class for RTP payloader * * Provides a base class for RTP payloaders @@ -27,10 +27,10 @@ #include -#include "gstbasertppayload.h" +#include "gstrtpbasepayload.h" -GST_DEBUG_CATEGORY_STATIC (basertppayload_debug); -#define GST_CAT_DEFAULT (basertppayload_debug) +GST_DEBUG_CATEGORY_STATIC (rtpbasepayload_debug); +#define GST_CAT_DEFAULT (rtpbasepayload_debug) #define GST_RTP_BASE_PAYLOAD_GET_PRIVATE(obj) \ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BASE_PAYLOAD, GstRTPBasePayloadPrivate)) @@ -92,19 +92,19 @@ enum }; static void gst_rtp_base_payload_class_init (GstRTPBasePayloadClass * klass); -static void gst_rtp_base_payload_init (GstRTPBasePayload * basertppayload, +static void gst_rtp_base_payload_init (GstRTPBasePayload * rtpbasepayload, gpointer g_class); static void gst_rtp_base_payload_finalize (GObject * object); static GstCaps *gst_rtp_base_payload_sink_getcaps (GstPad * pad, GstCaps * filter); static gboolean gst_rtp_base_payload_event_default (GstRTPBasePayload * - basertppayload, GstEvent * event); + rtpbasepayload, GstEvent * event); static gboolean gst_rtp_base_payload_event (GstPad * pad, GstEvent * event); static GstFlowReturn gst_rtp_base_payload_chain (GstPad * pad, GstBuffer * buffer); static GstCaps *gst_rtp_base_payload_getcaps_default (GstRTPBasePayload * - basertppayload, GstPad * pad, GstCaps * filter); + rtpbasepayload, GstPad * pad, GstCaps * filter); static void gst_rtp_base_payload_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); @@ -121,10 +121,10 @@ static GstElementClass *parent_class = NULL; GType gst_rtp_base_payload_get_type (void) { - static GType basertppayload_type = 0; + static GType rtpbasepayload_type = 0; - if (g_once_init_enter ((gsize *) & basertppayload_type)) { - static const GTypeInfo basertppayload_info = { + if (g_once_init_enter ((gsize *) & rtpbasepayload_type)) { + static const GTypeInfo rtpbasepayload_info = { sizeof (GstRTPBasePayloadClass), NULL, NULL, @@ -136,11 +136,11 @@ gst_rtp_base_payload_get_type (void) (GInstanceInitFunc) gst_rtp_base_payload_init, }; - g_once_init_leave ((gsize *) & basertppayload_type, + g_once_init_leave ((gsize *) & rtpbasepayload_type, g_type_register_static (GST_TYPE_ELEMENT, "GstRTPBasePayload", - &basertppayload_info, G_TYPE_FLAG_ABSTRACT)); + &rtpbasepayload_info, G_TYPE_FLAG_ABSTRACT)); } - return basertppayload_type; + return rtpbasepayload_type; } static void @@ -243,81 +243,81 @@ gst_rtp_base_payload_class_init (GstRTPBasePayloadClass * klass) klass->get_caps = gst_rtp_base_payload_getcaps_default; klass->handle_event = gst_rtp_base_payload_event_default; - GST_DEBUG_CATEGORY_INIT (basertppayload_debug, "basertppayload", 0, + GST_DEBUG_CATEGORY_INIT (rtpbasepayload_debug, "rtpbasepayload", 0, "Base class for RTP Payloaders"); } static void -gst_rtp_base_payload_init (GstRTPBasePayload * basertppayload, gpointer g_class) +gst_rtp_base_payload_init (GstRTPBasePayload * rtpbasepayload, gpointer g_class) { GstPadTemplate *templ; GstRTPBasePayloadPrivate *priv; - basertppayload->priv = priv = - GST_RTP_BASE_PAYLOAD_GET_PRIVATE (basertppayload); + rtpbasepayload->priv = priv = + GST_RTP_BASE_PAYLOAD_GET_PRIVATE (rtpbasepayload); templ = gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "src"); g_return_if_fail (templ != NULL); - basertppayload->srcpad = gst_pad_new_from_template (templ, "src"); - gst_element_add_pad (GST_ELEMENT (basertppayload), basertppayload->srcpad); + rtpbasepayload->srcpad = gst_pad_new_from_template (templ, "src"); + gst_element_add_pad (GST_ELEMENT (rtpbasepayload), rtpbasepayload->srcpad); templ = gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "sink"); g_return_if_fail (templ != NULL); - basertppayload->sinkpad = gst_pad_new_from_template (templ, "sink"); - gst_pad_set_getcaps_function (basertppayload->sinkpad, + rtpbasepayload->sinkpad = gst_pad_new_from_template (templ, "sink"); + gst_pad_set_getcaps_function (rtpbasepayload->sinkpad, gst_rtp_base_payload_sink_getcaps); - gst_pad_set_event_function (basertppayload->sinkpad, + gst_pad_set_event_function (rtpbasepayload->sinkpad, gst_rtp_base_payload_event); - gst_pad_set_chain_function (basertppayload->sinkpad, + gst_pad_set_chain_function (rtpbasepayload->sinkpad, gst_rtp_base_payload_chain); - gst_element_add_pad (GST_ELEMENT (basertppayload), basertppayload->sinkpad); - - basertppayload->mtu = DEFAULT_MTU; - basertppayload->pt = DEFAULT_PT; - basertppayload->seqnum_offset = DEFAULT_SEQNUM_OFFSET; - basertppayload->ssrc = DEFAULT_SSRC; - basertppayload->ts_offset = DEFAULT_TIMESTAMP_OFFSET; - priv->seqnum_offset_random = (basertppayload->seqnum_offset == -1); - priv->ts_offset_random = (basertppayload->ts_offset == -1); - priv->ssrc_random = (basertppayload->ssrc == -1); - - basertppayload->max_ptime = DEFAULT_MAX_PTIME; - basertppayload->min_ptime = DEFAULT_MIN_PTIME; - basertppayload->priv->perfect_rtptime = DEFAULT_PERFECT_RTPTIME; - basertppayload->abidata.ABI.ptime_multiple = DEFAULT_PTIME_MULTIPLE; - basertppayload->priv->base_offset = GST_BUFFER_OFFSET_NONE; - basertppayload->priv->base_rtime = GST_BUFFER_OFFSET_NONE; - - basertppayload->media = NULL; - basertppayload->encoding_name = NULL; - - basertppayload->clock_rate = 0; - - basertppayload->priv->caps_max_ptime = DEFAULT_MAX_PTIME; - basertppayload->priv->prop_max_ptime = DEFAULT_MAX_PTIME; + gst_element_add_pad (GST_ELEMENT (rtpbasepayload), rtpbasepayload->sinkpad); + + rtpbasepayload->mtu = DEFAULT_MTU; + rtpbasepayload->pt = DEFAULT_PT; + rtpbasepayload->seqnum_offset = DEFAULT_SEQNUM_OFFSET; + rtpbasepayload->ssrc = DEFAULT_SSRC; + rtpbasepayload->ts_offset = DEFAULT_TIMESTAMP_OFFSET; + priv->seqnum_offset_random = (rtpbasepayload->seqnum_offset == -1); + priv->ts_offset_random = (rtpbasepayload->ts_offset == -1); + priv->ssrc_random = (rtpbasepayload->ssrc == -1); + + rtpbasepayload->max_ptime = DEFAULT_MAX_PTIME; + rtpbasepayload->min_ptime = DEFAULT_MIN_PTIME; + rtpbasepayload->priv->perfect_rtptime = DEFAULT_PERFECT_RTPTIME; + rtpbasepayload->abidata.ABI.ptime_multiple = DEFAULT_PTIME_MULTIPLE; + rtpbasepayload->priv->base_offset = GST_BUFFER_OFFSET_NONE; + rtpbasepayload->priv->base_rtime = GST_BUFFER_OFFSET_NONE; + + rtpbasepayload->media = NULL; + rtpbasepayload->encoding_name = NULL; + + rtpbasepayload->clock_rate = 0; + + rtpbasepayload->priv->caps_max_ptime = DEFAULT_MAX_PTIME; + rtpbasepayload->priv->prop_max_ptime = DEFAULT_MAX_PTIME; } static void gst_rtp_base_payload_finalize (GObject * object) { - GstRTPBasePayload *basertppayload; + GstRTPBasePayload *rtpbasepayload; - basertppayload = GST_RTP_BASE_PAYLOAD (object); + rtpbasepayload = GST_RTP_BASE_PAYLOAD (object); - g_free (basertppayload->media); - basertppayload->media = NULL; - g_free (basertppayload->encoding_name); - basertppayload->encoding_name = NULL; + g_free (rtpbasepayload->media); + rtpbasepayload->media = NULL; + g_free (rtpbasepayload->encoding_name); + rtpbasepayload->encoding_name = NULL; G_OBJECT_CLASS (parent_class)->finalize (object); } static GstCaps * -gst_rtp_base_payload_getcaps_default (GstRTPBasePayload * basertppayload, +gst_rtp_base_payload_getcaps_default (GstRTPBasePayload * rtpbasepayload, GstPad * pad, GstCaps * filter) { GstCaps *caps; @@ -338,48 +338,48 @@ gst_rtp_base_payload_getcaps_default (GstRTPBasePayload * basertppayload, static GstCaps * gst_rtp_base_payload_sink_getcaps (GstPad * pad, GstCaps * filter) { - GstRTPBasePayload *basertppayload; - GstRTPBasePayloadClass *basertppayload_class; + GstRTPBasePayload *rtpbasepayload; + GstRTPBasePayloadClass *rtpbasepayload_class; GstCaps *caps = NULL; GST_DEBUG_OBJECT (pad, "getting caps"); - basertppayload = GST_RTP_BASE_PAYLOAD (gst_pad_get_parent (pad)); - basertppayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (basertppayload); + rtpbasepayload = GST_RTP_BASE_PAYLOAD (gst_pad_get_parent (pad)); + rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload); - if (basertppayload_class->get_caps) - caps = basertppayload_class->get_caps (basertppayload, pad, filter); + if (rtpbasepayload_class->get_caps) + caps = rtpbasepayload_class->get_caps (rtpbasepayload, pad, filter); - gst_object_unref (basertppayload); + gst_object_unref (rtpbasepayload); return caps; } static gboolean -gst_rtp_base_payload_event_default (GstRTPBasePayload * basertppayload, +gst_rtp_base_payload_event_default (GstRTPBasePayload * rtpbasepayload, GstEvent * event) { gboolean res = FALSE; switch (GST_EVENT_TYPE (event)) { case GST_EVENT_FLUSH_START: - res = gst_pad_event_default (basertppayload->sinkpad, event); + res = gst_pad_event_default (rtpbasepayload->sinkpad, event); break; case GST_EVENT_FLUSH_STOP: - res = gst_pad_event_default (basertppayload->sinkpad, event); - gst_segment_init (&basertppayload->segment, GST_FORMAT_UNDEFINED); + res = gst_pad_event_default (rtpbasepayload->sinkpad, event); + gst_segment_init (&rtpbasepayload->segment, GST_FORMAT_UNDEFINED); break; case GST_EVENT_CAPS: { - GstRTPBasePayloadClass *basertppayload_class; + GstRTPBasePayloadClass *rtpbasepayload_class; GstCaps *caps; gst_event_parse_caps (event, &caps); - GST_DEBUG_OBJECT (basertppayload, "setting caps %" GST_PTR_FORMAT, caps); + GST_DEBUG_OBJECT (rtpbasepayload, "setting caps %" GST_PTR_FORMAT, caps); - basertppayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (basertppayload); - if (basertppayload_class->set_caps) - res = basertppayload_class->set_caps (basertppayload, caps); + rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload); + if (rtpbasepayload_class->set_caps) + res = rtpbasepayload_class->set_caps (rtpbasepayload, caps); gst_event_unref (event); break; @@ -388,18 +388,18 @@ gst_rtp_base_payload_event_default (GstRTPBasePayload * basertppayload, { GstSegment *segment; - segment = &basertppayload->segment; + segment = &rtpbasepayload->segment; gst_event_copy_segment (event, segment); - basertppayload->priv->base_offset = GST_BUFFER_OFFSET_NONE; + rtpbasepayload->priv->base_offset = GST_BUFFER_OFFSET_NONE; - GST_DEBUG_OBJECT (basertppayload, + GST_DEBUG_OBJECT (rtpbasepayload, "configured SEGMENT %" GST_SEGMENT_FORMAT, segment); - res = gst_pad_event_default (basertppayload->sinkpad, event); + res = gst_pad_event_default (rtpbasepayload->sinkpad, event); break; } default: - res = gst_pad_event_default (basertppayload->sinkpad, event); + res = gst_pad_event_default (rtpbasepayload->sinkpad, event); break; } return res; @@ -408,24 +408,24 @@ gst_rtp_base_payload_event_default (GstRTPBasePayload * basertppayload, static gboolean gst_rtp_base_payload_event (GstPad * pad, GstEvent * event) { - GstRTPBasePayload *basertppayload; - GstRTPBasePayloadClass *basertppayload_class; + GstRTPBasePayload *rtpbasepayload; + GstRTPBasePayloadClass *rtpbasepayload_class; gboolean res = FALSE; - basertppayload = GST_RTP_BASE_PAYLOAD (gst_pad_get_parent (pad)); - if (G_UNLIKELY (basertppayload == NULL)) { + rtpbasepayload = GST_RTP_BASE_PAYLOAD (gst_pad_get_parent (pad)); + if (G_UNLIKELY (rtpbasepayload == NULL)) { gst_event_unref (event); return FALSE; } - basertppayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (basertppayload); + rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload); - if (basertppayload_class->handle_event) - res = basertppayload_class->handle_event (basertppayload, event); + if (rtpbasepayload_class->handle_event) + res = rtpbasepayload_class->handle_event (rtpbasepayload, event); else gst_event_unref (event); - gst_object_unref (basertppayload); + gst_object_unref (rtpbasepayload); return res; } @@ -434,28 +434,28 @@ gst_rtp_base_payload_event (GstPad * pad, GstEvent * event) static GstFlowReturn gst_rtp_base_payload_chain (GstPad * pad, GstBuffer * buffer) { - GstRTPBasePayload *basertppayload; - GstRTPBasePayloadClass *basertppayload_class; + GstRTPBasePayload *rtpbasepayload; + GstRTPBasePayloadClass *rtpbasepayload_class; GstFlowReturn ret; - basertppayload = GST_RTP_BASE_PAYLOAD (gst_pad_get_parent (pad)); - basertppayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (basertppayload); + rtpbasepayload = GST_RTP_BASE_PAYLOAD (gst_pad_get_parent (pad)); + rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload); - if (!basertppayload_class->handle_buffer) + if (!rtpbasepayload_class->handle_buffer) goto no_function; - ret = basertppayload_class->handle_buffer (basertppayload, buffer); + ret = rtpbasepayload_class->handle_buffer (rtpbasepayload, buffer); - gst_object_unref (basertppayload); + gst_object_unref (rtpbasepayload); return ret; /* ERRORS */ no_function: { - GST_ELEMENT_ERROR (basertppayload, STREAM, NOT_IMPLEMENTED, (NULL), + GST_ELEMENT_ERROR (rtpbasepayload, STREAM, NOT_IMPLEMENTED, (NULL), ("subclass did not implement handle_buffer function")); - gst_object_unref (basertppayload); + gst_object_unref (rtpbasepayload); gst_buffer_unref (buffer); return GST_FLOW_ERROR; } @@ -499,18 +499,18 @@ copy_fixed (GQuark field_id, const GValue * value, GstStructure * dest) } static void -update_max_ptime (GstRTPBasePayload * basertppayload) +update_max_ptime (GstRTPBasePayload * rtpbasepayload) { - if (basertppayload->priv->caps_max_ptime != -1 && - basertppayload->priv->prop_max_ptime != -1) - basertppayload->max_ptime = MIN (basertppayload->priv->caps_max_ptime, - basertppayload->priv->prop_max_ptime); - else if (basertppayload->priv->caps_max_ptime != -1) - basertppayload->max_ptime = basertppayload->priv->caps_max_ptime; - else if (basertppayload->priv->prop_max_ptime != -1) - basertppayload->max_ptime = basertppayload->priv->prop_max_ptime; + if (rtpbasepayload->priv->caps_max_ptime != -1 && + rtpbasepayload->priv->prop_max_ptime != -1) + rtpbasepayload->max_ptime = MIN (rtpbasepayload->priv->caps_max_ptime, + rtpbasepayload->priv->prop_max_ptime); + else if (rtpbasepayload->priv->caps_max_ptime != -1) + rtpbasepayload->max_ptime = rtpbasepayload->priv->caps_max_ptime; + else if (rtpbasepayload->priv->prop_max_ptime != -1) + rtpbasepayload->max_ptime = rtpbasepayload->priv->prop_max_ptime; else - basertppayload->max_ptime = DEFAULT_MAX_PTIME; + rtpbasepayload->max_ptime = DEFAULT_MAX_PTIME; } /** @@ -904,49 +904,49 @@ static void gst_rtp_base_payload_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { - GstRTPBasePayload *basertppayload; + GstRTPBasePayload *rtpbasepayload; GstRTPBasePayloadPrivate *priv; gint64 val; - basertppayload = GST_RTP_BASE_PAYLOAD (object); - priv = basertppayload->priv; + rtpbasepayload = GST_RTP_BASE_PAYLOAD (object); + priv = rtpbasepayload->priv; switch (prop_id) { case PROP_MTU: - basertppayload->mtu = g_value_get_uint (value); + rtpbasepayload->mtu = g_value_get_uint (value); break; case PROP_PT: - basertppayload->pt = g_value_get_uint (value); + rtpbasepayload->pt = g_value_get_uint (value); break; case PROP_SSRC: val = g_value_get_uint (value); - basertppayload->ssrc = val; + rtpbasepayload->ssrc = val; priv->ssrc_random = FALSE; break; case PROP_TIMESTAMP_OFFSET: val = g_value_get_uint (value); - basertppayload->ts_offset = val; + rtpbasepayload->ts_offset = val; priv->ts_offset_random = FALSE; break; case PROP_SEQNUM_OFFSET: val = g_value_get_int (value); - basertppayload->seqnum_offset = val; + rtpbasepayload->seqnum_offset = val; priv->seqnum_offset_random = (val == -1); - GST_DEBUG_OBJECT (basertppayload, "seqnum offset 0x%04x, random %d", - basertppayload->seqnum_offset, priv->seqnum_offset_random); + GST_DEBUG_OBJECT (rtpbasepayload, "seqnum offset 0x%04x, random %d", + rtpbasepayload->seqnum_offset, priv->seqnum_offset_random); break; case PROP_MAX_PTIME: - basertppayload->priv->prop_max_ptime = g_value_get_int64 (value); - update_max_ptime (basertppayload); + rtpbasepayload->priv->prop_max_ptime = g_value_get_int64 (value); + update_max_ptime (rtpbasepayload); break; case PROP_MIN_PTIME: - basertppayload->min_ptime = g_value_get_int64 (value); + rtpbasepayload->min_ptime = g_value_get_int64 (value); break; case PROP_PERFECT_RTPTIME: priv->perfect_rtptime = g_value_get_boolean (value); break; case PROP_PTIME_MULTIPLE: - basertppayload->abidata.ABI.ptime_multiple = g_value_get_int64 (value); + rtpbasepayload->abidata.ABI.ptime_multiple = g_value_get_int64 (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); @@ -958,54 +958,54 @@ static void gst_rtp_base_payload_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { - GstRTPBasePayload *basertppayload; + GstRTPBasePayload *rtpbasepayload; GstRTPBasePayloadPrivate *priv; - basertppayload = GST_RTP_BASE_PAYLOAD (object); - priv = basertppayload->priv; + rtpbasepayload = GST_RTP_BASE_PAYLOAD (object); + priv = rtpbasepayload->priv; switch (prop_id) { case PROP_MTU: - g_value_set_uint (value, basertppayload->mtu); + g_value_set_uint (value, rtpbasepayload->mtu); break; case PROP_PT: - g_value_set_uint (value, basertppayload->pt); + g_value_set_uint (value, rtpbasepayload->pt); break; case PROP_SSRC: if (priv->ssrc_random) g_value_set_uint (value, -1); else - g_value_set_uint (value, basertppayload->ssrc); + g_value_set_uint (value, rtpbasepayload->ssrc); break; case PROP_TIMESTAMP_OFFSET: if (priv->ts_offset_random) g_value_set_uint (value, -1); else - g_value_set_uint (value, (guint32) basertppayload->ts_offset); + g_value_set_uint (value, (guint32) rtpbasepayload->ts_offset); break; case PROP_SEQNUM_OFFSET: if (priv->seqnum_offset_random) g_value_set_int (value, -1); else - g_value_set_int (value, (guint16) basertppayload->seqnum_offset); + g_value_set_int (value, (guint16) rtpbasepayload->seqnum_offset); break; case PROP_MAX_PTIME: - g_value_set_int64 (value, basertppayload->max_ptime); + g_value_set_int64 (value, rtpbasepayload->max_ptime); break; case PROP_MIN_PTIME: - g_value_set_int64 (value, basertppayload->min_ptime); + g_value_set_int64 (value, rtpbasepayload->min_ptime); break; case PROP_TIMESTAMP: - g_value_set_uint (value, basertppayload->timestamp); + g_value_set_uint (value, rtpbasepayload->timestamp); break; case PROP_SEQNUM: - g_value_set_uint (value, basertppayload->seqnum); + g_value_set_uint (value, rtpbasepayload->seqnum); break; case PROP_PERFECT_RTPTIME: g_value_set_boolean (value, priv->perfect_rtptime); break; case PROP_PTIME_MULTIPLE: - g_value_set_int64 (value, basertppayload->abidata.ABI.ptime_multiple); + g_value_set_int64 (value, rtpbasepayload->abidata.ABI.ptime_multiple); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); @@ -1017,37 +1017,37 @@ static GstStateChangeReturn gst_rtp_base_payload_change_state (GstElement * element, GstStateChange transition) { - GstRTPBasePayload *basertppayload; + GstRTPBasePayload *rtpbasepayload; GstRTPBasePayloadPrivate *priv; GstStateChangeReturn ret; - basertppayload = GST_RTP_BASE_PAYLOAD (element); - priv = basertppayload->priv; + rtpbasepayload = GST_RTP_BASE_PAYLOAD (element); + priv = rtpbasepayload->priv; switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: break; case GST_STATE_CHANGE_READY_TO_PAUSED: - gst_segment_init (&basertppayload->segment, GST_FORMAT_UNDEFINED); + gst_segment_init (&rtpbasepayload->segment, GST_FORMAT_UNDEFINED); if (priv->seqnum_offset_random) - basertppayload->seqnum_base = g_random_int_range (0, G_MAXUINT16); + rtpbasepayload->seqnum_base = g_random_int_range (0, G_MAXUINT16); else - basertppayload->seqnum_base = basertppayload->seqnum_offset; - priv->next_seqnum = basertppayload->seqnum_base; - basertppayload->seqnum = basertppayload->seqnum_base; + rtpbasepayload->seqnum_base = rtpbasepayload->seqnum_offset; + priv->next_seqnum = rtpbasepayload->seqnum_base; + rtpbasepayload->seqnum = rtpbasepayload->seqnum_base; if (priv->ssrc_random) - basertppayload->current_ssrc = g_random_int (); + rtpbasepayload->current_ssrc = g_random_int (); else - basertppayload->current_ssrc = basertppayload->ssrc; + rtpbasepayload->current_ssrc = rtpbasepayload->ssrc; if (priv->ts_offset_random) - basertppayload->ts_base = g_random_int (); + rtpbasepayload->ts_base = g_random_int (); else - basertppayload->ts_base = basertppayload->ts_offset; - basertppayload->timestamp = basertppayload->ts_base; - g_atomic_int_set (&basertppayload->priv->notified_first_timestamp, 1); + rtpbasepayload->ts_base = rtpbasepayload->ts_offset; + rtpbasepayload->timestamp = rtpbasepayload->ts_base; + g_atomic_int_set (&rtpbasepayload->priv->notified_first_timestamp, 1); priv->base_offset = GST_BUFFER_OFFSET_NONE; break; default: @@ -1058,7 +1058,7 @@ gst_rtp_base_payload_change_state (GstElement * element, switch (transition) { case GST_STATE_CHANGE_PLAYING_TO_PAUSED: - g_atomic_int_set (&basertppayload->priv->notified_first_timestamp, 1); + g_atomic_int_set (&rtpbasepayload->priv->notified_first_timestamp, 1); break; case GST_STATE_CHANGE_READY_TO_NULL: break; diff --git a/gst-libs/gst/rtp/gstbasertppayload.h b/gst-libs/gst/rtp/gstrtpbasepayload.h similarity index 100% rename from gst-libs/gst/rtp/gstbasertppayload.h rename to gst-libs/gst/rtp/gstrtpbasepayload.h