From ccf511a5d495d542445087b4e0a12a115b6f389f Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Fri, 11 Nov 2011 12:24:08 +0100 Subject: [PATCH] rename BaseRTP -> RTPBase --- docs/libs/gst-plugins-base-libs-sections.txt | 105 ++++---- docs/libs/gst-plugins-base-libs.types | 6 +- gst-libs/gst/rtp/gstbasertpaudiopayload.c | 264 +++++++++---------- gst-libs/gst/rtp/gstbasertpaudiopayload.h | 66 ++--- gst-libs/gst/rtp/gstbasertpdepayload.c | 168 ++++++------ gst-libs/gst/rtp/gstbasertpdepayload.h | 62 +++-- gst-libs/gst/rtp/gstbasertppayload.c | 198 +++++++------- gst-libs/gst/rtp/gstbasertppayload.h | 94 +++---- gst-libs/gst/rtp/gstrtcpbuffer.c | 2 +- gst-libs/gst/rtp/gstrtpbuffer.c | 2 +- 10 files changed, 482 insertions(+), 485 deletions(-) diff --git a/docs/libs/gst-plugins-base-libs-sections.txt b/docs/libs/gst-plugins-base-libs-sections.txt index c321864ffb..ab4282e60e 100644 --- a/docs/libs/gst-plugins-base-libs-sections.txt +++ b/docs/libs/gst-plugins-base-libs-sections.txt @@ -1125,52 +1125,51 @@ gst_riff_strh
gstbasertpaudiopayload gst/rtp/gstbasertpaudiopayload.h -GstBaseRTPAudioPayload -GstBaseRTPAudioPayloadClass +GstRTPBaseAudioPayload +GstRTPBaseAudioPayloadClass -gst_base_rtp_audio_payload_set_frame_based -gst_base_rtp_audio_payload_set_frame_options -gst_base_rtp_audio_payload_set_sample_based -gst_base_rtp_audio_payload_set_sample_options -gst_base_rtp_audio_payload_get_adapter -gst_base_rtp_audio_payload_push -gst_base_rtp_audio_payload_flush -gst_base_rtp_audio_payload_set_samplebits_options +gst_rtp_base_audio_payload_set_frame_based +gst_rtp_base_audio_payload_set_frame_options +gst_rtp_base_audio_payload_set_sample_based +gst_rtp_base_audio_payload_set_sample_options +gst_rtp_base_audio_payload_get_adapter +gst_rtp_base_audio_payload_push +gst_rtp_base_audio_payload_flush +gst_rtp_base_audio_payload_set_samplebits_options -GST_TYPE_BASE_RTP_AUDIO_PAYLOAD -GST_BASE_RTP_AUDIO_PAYLOAD -GST_BASE_RTP_AUDIO_PAYLOAD_CLASS -GST_IS_BASE_RTP_AUDIO_PAYLOAD -GST_IS_BASE_RTP_AUDIO_PAYLOAD_CLASS -GST_BASE_RTP_AUDIO_PAYLOAD_CAST -gst_base_rtp_audio_payload_get_type -GstBaseRTPAudioPayloadPrivate +GST_TYPE_RTP_BASE_AUDIO_PAYLOAD +GST_RTP_BASE_AUDIO_PAYLOAD +GST_RTP_BASE_AUDIO_PAYLOAD_CLASS +GST_IS_RTP_BASE_AUDIO_PAYLOAD +GST_IS_RTP_BASE_AUDIO_PAYLOAD_CLASS +GST_RTP_BASE_AUDIO_PAYLOAD_CAST +gst_rtp_base_audio_payload_get_type +GstRTPBaseAudioPayloadPrivate
gstbasertpdepayload gst/rtp/gstbasertpdepayload.h -GstBaseRTPDepayload -GstBaseRTPDepayloadClass +GstRTPBaseDepayload +GstRTPBaseDepayloadClass -GST_BASE_RTP_DEPAYLOAD_SINKPAD -GST_BASE_RTP_DEPAYLOAD_SRCPAD +GST_RTP_BASE_DEPAYLOAD_SINKPAD +GST_RTP_BASE_DEPAYLOAD_SRCPAD -gst_base_rtp_depayload_push -gst_base_rtp_depayload_push_ts -gst_base_rtp_depayload_push_list +gst_rtp_base_depayload_push +gst_rtp_base_depayload_push_list -GstBaseRTPDepayloadPrivate -GST_TYPE_BASE_RTP_DEPAYLOAD -GST_BASE_RTP_DEPAYLOAD -GST_BASE_RTP_DEPAYLOAD_CLASS -GST_BASE_RTP_DEPAYLOAD_GET_CLASS -GST_IS_BASE_RTP_DEPAYLOAD -GST_IS_BASE_RTP_DEPAYLOAD_CLASS -GST_BASE_RTP_PAYLOAD_CAST -gst_base_rtp_depayload_get_type -GstBaseRTPPayloadPrivate +GstRTPBaseDepayloadPrivate +GST_TYPE_RTP_BASE_DEPAYLOAD +GST_RTP_BASE_DEPAYLOAD +GST_RTP_BASE_DEPAYLOAD_CLASS +GST_RTP_BASE_DEPAYLOAD_GET_CLASS +GST_IS_RTP_BASE_DEPAYLOAD +GST_IS_RTP_BASE_DEPAYLOAD_CLASS +GST_RTP_BASE_PAYLOAD_CAST +gst_rtp_base_depayload_get_type +GstRTPBasePayloadPrivate QUEUE_LOCK_INIT @@ -1182,27 +1181,27 @@ QUEUE_UNLOCK
gstbasertppayload gst/rtp/gstbasertppayload.h -GstBaseRTPPayload -GstBaseRTPPayloadClass +GstRTPBasePayload +GstRTPBasePayloadClass -GST_BASE_RTP_PAYLOAD_MTU -GST_BASE_RTP_PAYLOAD_PT -GST_BASE_RTP_PAYLOAD_SINKPAD -GST_BASE_RTP_PAYLOAD_SRCPAD +GST_RTP_BASE_PAYLOAD_MTU +GST_RTP_BASE_PAYLOAD_PT +GST_RTP_BASE_PAYLOAD_SINKPAD +GST_RTP_BASE_PAYLOAD_SRCPAD -gst_basertppayload_is_filled -gst_basertppayload_push -gst_basertppayload_push_list -gst_basertppayload_set_options -gst_basertppayload_set_outcaps +gst_rtp_base_payload_is_filled +gst_rtp_base_payload_push +gst_rtp_base_payload_push_list +gst_rtp_base_payload_set_options +gst_rtp_base_payload_set_outcaps -GST_TYPE_BASE_RTP_PAYLOAD -GST_BASE_RTP_PAYLOAD -GST_BASE_RTP_PAYLOAD_CLASS -GST_BASE_RTP_PAYLOAD_GET_CLASS -GST_IS_BASE_RTP_PAYLOAD -GST_IS_BASE_RTP_PAYLOAD_CLASS -gst_basertppayload_get_type +GST_TYPE_RTP_BASE_PAYLOAD +GST_RTP_BASE_PAYLOAD +GST_RTP_BASE_PAYLOAD_CLASS +GST_RTP_BASE_PAYLOAD_GET_CLASS +GST_IS_RTP_BASE_PAYLOAD +GST_IS_RTP_BASE_PAYLOAD_CLASS +gst_rtp_base_payload_get_type
diff --git a/docs/libs/gst-plugins-base-libs.types b/docs/libs/gst-plugins-base-libs.types index 37e13234ec..43444a7201 100644 --- a/docs/libs/gst-plugins-base-libs.types +++ b/docs/libs/gst-plugins-base-libs.types @@ -49,11 +49,11 @@ gst_video_overlay_get_type #include -gst_base_rtp_depayload_get_type +gst_rtp_base_depayload_get_type #include -gst_base_rtp_payload_get_type +gst_rtp_base_payload_get_type #include -gst_base_rtp_audio_payload_get_type +gst_rtp_base_audio_payload_get_type #include diff --git a/gst-libs/gst/rtp/gstbasertpaudiopayload.c b/gst-libs/gst/rtp/gstbasertpaudiopayload.c index 5425d2c6de..e5f29fe61a 100644 --- a/gst-libs/gst/rtp/gstbasertpaudiopayload.c +++ b/gst-libs/gst/rtp/gstbasertpaudiopayload.c @@ -24,7 +24,7 @@ * Provides a base class for audio RTP payloaders for frame or sample based * audio codecs (constant bitrate) * - * This class derives from GstBaseRTPPayload. It can be used for payloading + * This class derives from GstRTPBasePayload. It can be used for payloading * audio codecs. It will only work with constant bitrate codecs. It supports * both frame based and sample based codecs. It takes care of packing up the * audio data into RTP packets and filling up the headers accordingly. The @@ -40,16 +40,16 @@ * Usage * * To use this base class, your child element needs to call either - * gst_base_rtp_audio_payload_set_frame_based() or - * gst_base_rtp_audio_payload_set_sample_based(). This is usually done in the + * gst_rtp_base_audio_payload_set_frame_based() or + * gst_rtp_base_audio_payload_set_sample_based(). This is usually done in the * element's _init() function. Then, the child element must call either - * gst_base_rtp_audio_payload_set_frame_options(), - * gst_base_rtp_audio_payload_set_sample_options() or - * gst_base_rtp_audio_payload_set_samplebits_options. Since - * GstBaseRTPAudioPayload derives from GstBaseRTPPayload, the child element + * gst_rtp_base_audio_payload_set_frame_options(), + * gst_rtp_base_audio_payload_set_sample_options() or + * gst_rtp_base_audio_payload_set_samplebits_options. Since + * GstRTPBaseAudioPayload derives from GstRTPBasePayload, the child element * must set any variables or call/override any functions required by that base * class. The child element does not need to override any other functions - * specific to GstBaseRTPAudioPayload. + * specific to GstRTPBaseAudioPayload. * * */ @@ -78,16 +78,16 @@ enum }; /* function to convert bytes to a time */ -typedef GstClockTime (*GetBytesToTimeFunc) (GstBaseRTPAudioPayload * payload, +typedef GstClockTime (*GetBytesToTimeFunc) (GstRTPBaseAudioPayload * payload, guint64 bytes); /* function to convert bytes to a RTP time */ -typedef guint32 (*GetBytesToRTPTimeFunc) (GstBaseRTPAudioPayload * payload, +typedef guint32 (*GetBytesToRTPTimeFunc) (GstRTPBaseAudioPayload * payload, guint64 bytes); /* function to convert time to bytes */ -typedef guint64 (*GetTimeToBytesFunc) (GstBaseRTPAudioPayload * payload, +typedef guint64 (*GetTimeToBytesFunc) (GstRTPBaseAudioPayload * payload, GstClockTime time); -struct _GstBaseRTPAudioPayloadPrivate +struct _GstRTPBaseAudioPayloadPrivate { GetBytesToTimeFunc bytes_to_time; GetBytesToRTPTimeFunc bytes_to_rtptime; @@ -115,68 +115,68 @@ struct _GstBaseRTPAudioPayloadPrivate }; -#define GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE(o) \ - (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_BASE_RTP_AUDIO_PAYLOAD, \ - GstBaseRTPAudioPayloadPrivate)) +#define GST_RTP_BASE_AUDIO_PAYLOAD_GET_PRIVATE(o) \ + (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_BASE_AUDIO_PAYLOAD, \ + GstRTPBaseAudioPayloadPrivate)) -static void gst_base_rtp_audio_payload_finalize (GObject * object); +static void gst_rtp_base_audio_payload_finalize (GObject * object); -static void gst_base_rtp_audio_payload_set_property (GObject * object, +static void gst_rtp_base_audio_payload_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); -static void gst_base_rtp_audio_payload_get_property (GObject * object, +static void gst_rtp_base_audio_payload_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); /* bytes to time functions */ static GstClockTime -gst_base_rtp_audio_payload_frame_bytes_to_time (GstBaseRTPAudioPayload * +gst_rtp_base_audio_payload_frame_bytes_to_time (GstRTPBaseAudioPayload * payload, guint64 bytes); static GstClockTime -gst_base_rtp_audio_payload_sample_bytes_to_time (GstBaseRTPAudioPayload * +gst_rtp_base_audio_payload_sample_bytes_to_time (GstRTPBaseAudioPayload * payload, guint64 bytes); /* bytes to RTP time functions */ static guint32 -gst_base_rtp_audio_payload_frame_bytes_to_rtptime (GstBaseRTPAudioPayload * +gst_rtp_base_audio_payload_frame_bytes_to_rtptime (GstRTPBaseAudioPayload * payload, guint64 bytes); static guint32 -gst_base_rtp_audio_payload_sample_bytes_to_rtptime (GstBaseRTPAudioPayload * +gst_rtp_base_audio_payload_sample_bytes_to_rtptime (GstRTPBaseAudioPayload * payload, guint64 bytes); /* time to bytes functions */ static guint64 -gst_base_rtp_audio_payload_frame_time_to_bytes (GstBaseRTPAudioPayload * +gst_rtp_base_audio_payload_frame_time_to_bytes (GstRTPBaseAudioPayload * payload, GstClockTime time); static guint64 -gst_base_rtp_audio_payload_sample_time_to_bytes (GstBaseRTPAudioPayload * +gst_rtp_base_audio_payload_sample_time_to_bytes (GstRTPBaseAudioPayload * payload, GstClockTime time); -static GstFlowReturn gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload +static GstFlowReturn gst_rtp_base_audio_payload_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer); -static GstStateChangeReturn gst_base_rtp_payload_audio_change_state (GstElement +static GstStateChangeReturn gst_rtp_base_payload_audio_change_state (GstElement * element, GstStateChange transition); -static gboolean gst_base_rtp_payload_audio_handle_event (GstBaseRTPPayload +static gboolean gst_rtp_base_payload_audio_handle_event (GstRTPBasePayload * payload, GstEvent * event); -#define gst_base_rtp_audio_payload_parent_class parent_class -G_DEFINE_TYPE (GstBaseRTPAudioPayload, gst_base_rtp_audio_payload, - GST_TYPE_BASE_RTP_PAYLOAD); +#define gst_rtp_base_audio_payload_parent_class parent_class +G_DEFINE_TYPE (GstRTPBaseAudioPayload, gst_rtp_base_audio_payload, + GST_TYPE_RTP_BASE_PAYLOAD); static void -gst_base_rtp_audio_payload_class_init (GstBaseRTPAudioPayloadClass * klass) +gst_rtp_base_audio_payload_class_init (GstRTPBaseAudioPayloadClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; - GstBaseRTPPayloadClass *gstbasertppayload_class; + GstRTPBasePayloadClass *gstbasertppayload_class; - g_type_class_add_private (klass, sizeof (GstBaseRTPAudioPayloadPrivate)); + g_type_class_add_private (klass, sizeof (GstRTPBaseAudioPayloadPrivate)); gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; - gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; + gstbasertppayload_class = (GstRTPBasePayloadClass *) klass; - gobject_class->finalize = gst_base_rtp_audio_payload_finalize; - gobject_class->set_property = gst_base_rtp_audio_payload_set_property; - gobject_class->get_property = gst_base_rtp_audio_payload_get_property; + gobject_class->finalize = gst_rtp_base_audio_payload_finalize; + gobject_class->set_property = gst_rtp_base_audio_payload_set_property; + gobject_class->get_property = gst_rtp_base_audio_payload_get_property; g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BUFFER_LIST, g_param_spec_boolean ("buffer-list", "Buffer List", @@ -184,21 +184,21 @@ gst_base_rtp_audio_payload_class_init (GstBaseRTPAudioPayloadClass * klass) DEFAULT_BUFFER_LIST, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gstelement_class->change_state = - GST_DEBUG_FUNCPTR (gst_base_rtp_payload_audio_change_state); + GST_DEBUG_FUNCPTR (gst_rtp_base_payload_audio_change_state); gstbasertppayload_class->handle_buffer = - GST_DEBUG_FUNCPTR (gst_base_rtp_audio_payload_handle_buffer); + GST_DEBUG_FUNCPTR (gst_rtp_base_audio_payload_handle_buffer); gstbasertppayload_class->handle_event = - GST_DEBUG_FUNCPTR (gst_base_rtp_payload_audio_handle_event); + GST_DEBUG_FUNCPTR (gst_rtp_base_payload_audio_handle_event); GST_DEBUG_CATEGORY_INIT (basertpaudiopayload_debug, "basertpaudiopayload", 0, "base audio RTP payloader"); } static void -gst_base_rtp_audio_payload_init (GstBaseRTPAudioPayload * payload) +gst_rtp_base_audio_payload_init (GstRTPBaseAudioPayload * payload) { - payload->priv = GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE (payload); + payload->priv = GST_RTP_BASE_AUDIO_PAYLOAD_GET_PRIVATE (payload); /* these need to be set by child object if frame based */ payload->frame_size = 0; @@ -213,11 +213,11 @@ gst_base_rtp_audio_payload_init (GstBaseRTPAudioPayload * payload) } static void -gst_base_rtp_audio_payload_finalize (GObject * object) +gst_rtp_base_audio_payload_finalize (GObject * object) { - GstBaseRTPAudioPayload *payload; + GstRTPBaseAudioPayload *payload; - payload = GST_BASE_RTP_AUDIO_PAYLOAD (object); + payload = GST_RTP_BASE_AUDIO_PAYLOAD (object); g_object_unref (payload->priv->adapter); @@ -225,12 +225,12 @@ gst_base_rtp_audio_payload_finalize (GObject * object) } static void -gst_base_rtp_audio_payload_set_property (GObject * object, +gst_rtp_base_audio_payload_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { - GstBaseRTPAudioPayload *payload; + GstRTPBaseAudioPayload *payload; - payload = GST_BASE_RTP_AUDIO_PAYLOAD (object); + payload = GST_RTP_BASE_AUDIO_PAYLOAD (object); switch (prop_id) { case PROP_BUFFER_LIST: @@ -243,12 +243,12 @@ gst_base_rtp_audio_payload_set_property (GObject * object, } static void -gst_base_rtp_audio_payload_get_property (GObject * object, +gst_rtp_base_audio_payload_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { - GstBaseRTPAudioPayload *payload; + GstRTPBaseAudioPayload *payload; - payload = GST_BASE_RTP_AUDIO_PAYLOAD (object); + payload = GST_RTP_BASE_AUDIO_PAYLOAD (object); switch (prop_id) { case PROP_BUFFER_LIST: @@ -261,14 +261,14 @@ gst_base_rtp_audio_payload_get_property (GObject * object, } /** - * gst_base_rtp_audio_payload_set_frame_based: + * gst_rtp_base_audio_payload_set_frame_based: * @basertpaudiopayload: a pointer to the element. * - * Tells #GstBaseRTPAudioPayload that the child element is for a frame based + * Tells #GstRTPBaseAudioPayload that the child element is for a frame based * audio codec */ void -gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload * +gst_rtp_base_audio_payload_set_frame_based (GstRTPBaseAudioPayload * basertpaudiopayload) { g_return_if_fail (basertpaudiopayload != NULL); @@ -277,22 +277,22 @@ gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload * g_return_if_fail (basertpaudiopayload->priv->bytes_to_rtptime == NULL); basertpaudiopayload->priv->bytes_to_time = - gst_base_rtp_audio_payload_frame_bytes_to_time; + gst_rtp_base_audio_payload_frame_bytes_to_time; basertpaudiopayload->priv->bytes_to_rtptime = - gst_base_rtp_audio_payload_frame_bytes_to_rtptime; + gst_rtp_base_audio_payload_frame_bytes_to_rtptime; basertpaudiopayload->priv->time_to_bytes = - gst_base_rtp_audio_payload_frame_time_to_bytes; + gst_rtp_base_audio_payload_frame_time_to_bytes; } /** - * gst_base_rtp_audio_payload_set_sample_based: + * gst_rtp_base_audio_payload_set_sample_based: * @basertpaudiopayload: a pointer to the element. * - * Tells #GstBaseRTPAudioPayload that the child element is for a sample based + * Tells #GstRTPBaseAudioPayload that the child element is for a sample based * audio codec */ void -gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload * +gst_rtp_base_audio_payload_set_sample_based (GstRTPBaseAudioPayload * basertpaudiopayload) { g_return_if_fail (basertpaudiopayload != NULL); @@ -301,15 +301,15 @@ gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload * g_return_if_fail (basertpaudiopayload->priv->bytes_to_rtptime == NULL); basertpaudiopayload->priv->bytes_to_time = - gst_base_rtp_audio_payload_sample_bytes_to_time; + gst_rtp_base_audio_payload_sample_bytes_to_time; basertpaudiopayload->priv->bytes_to_rtptime = - gst_base_rtp_audio_payload_sample_bytes_to_rtptime; + gst_rtp_base_audio_payload_sample_bytes_to_rtptime; basertpaudiopayload->priv->time_to_bytes = - gst_base_rtp_audio_payload_sample_time_to_bytes; + gst_rtp_base_audio_payload_sample_time_to_bytes; } /** - * gst_base_rtp_audio_payload_set_frame_options: + * gst_rtp_base_audio_payload_set_frame_options: * @basertpaudiopayload: 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. @@ -318,10 +318,10 @@ gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload * * */ void -gst_base_rtp_audio_payload_set_frame_options (GstBaseRTPAudioPayload +gst_rtp_base_audio_payload_set_frame_options (GstRTPBaseAudioPayload * basertpaudiopayload, gint frame_duration, gint frame_size) { - GstBaseRTPAudioPayloadPrivate *priv; + GstRTPBaseAudioPayloadPrivate *priv; g_return_if_fail (basertpaudiopayload != NULL); @@ -339,25 +339,25 @@ gst_base_rtp_audio_payload_set_frame_options (GstBaseRTPAudioPayload } /** - * gst_base_rtp_audio_payload_set_sample_options: + * gst_rtp_base_audio_payload_set_sample_options: * @basertpaudiopayload: a pointer to the element. * @sample_size: Size per sample in bytes. * * Sets the options for sample based audio codecs. */ void -gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload +gst_rtp_base_audio_payload_set_sample_options (GstRTPBaseAudioPayload * basertpaudiopayload, gint sample_size) { g_return_if_fail (basertpaudiopayload != NULL); /* sample_size is in bits internally */ - gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload, + gst_rtp_base_audio_payload_set_samplebits_options (basertpaudiopayload, sample_size * 8); } /** - * gst_base_rtp_audio_payload_set_samplebits_options: + * gst_rtp_base_audio_payload_set_samplebits_options: * @basertpaudiopayload: a pointer to the element. * @sample_size: Size per sample in bits. * @@ -366,11 +366,11 @@ gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload * Since: 0.10.18 */ void -gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload +gst_rtp_base_audio_payload_set_samplebits_options (GstRTPBaseAudioPayload * basertpaudiopayload, gint sample_size) { guint fragment_size; - GstBaseRTPAudioPayloadPrivate *priv; + GstRTPBaseAudioPayloadPrivate *priv; g_return_if_fail (basertpaudiopayload != NULL); @@ -392,14 +392,14 @@ gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload } static void -gst_base_rtp_audio_payload_set_meta (GstBaseRTPAudioPayload * payload, +gst_rtp_base_audio_payload_set_meta (GstRTPBaseAudioPayload * payload, GstBuffer * buffer, guint payload_len, GstClockTime timestamp) { - GstBaseRTPPayload *basepayload; - GstBaseRTPAudioPayloadPrivate *priv; + GstRTPBasePayload *basepayload; + GstRTPBaseAudioPayloadPrivate *priv; GstRTPBuffer rtp; - basepayload = GST_BASE_RTP_PAYLOAD_CAST (payload); + basepayload = GST_RTP_BASE_PAYLOAD_CAST (payload); priv = payload->priv; /* set payload type */ @@ -431,8 +431,8 @@ gst_base_rtp_audio_payload_set_meta (GstBaseRTPAudioPayload * payload, } /** - * gst_base_rtp_audio_payload_push: - * @baseaudiopayload: a #GstBaseRTPPayload + * gst_rtp_base_audio_payload_push: + * @baseaudiopayload: a #GstRTPBasePayload * @data: data to set as payload * @payload_len: length of payload * @timestamp: a #GstClockTime @@ -446,16 +446,16 @@ gst_base_rtp_audio_payload_set_meta (GstBaseRTPAudioPayload * payload, * Since: 0.10.13 */ GstFlowReturn -gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload, +gst_rtp_base_audio_payload_push (GstRTPBaseAudioPayload * baseaudiopayload, const guint8 * data, guint payload_len, GstClockTime timestamp) { - GstBaseRTPPayload *basepayload; + GstRTPBasePayload *basepayload; GstBuffer *outbuf; guint8 *payload; GstFlowReturn ret; GstRTPBuffer rtp; - basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload); + basepayload = GST_RTP_BASE_PAYLOAD (baseaudiopayload); GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT, payload_len, GST_TIME_ARGS (timestamp)); @@ -470,27 +470,27 @@ gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload, gst_rtp_buffer_unmap (&rtp); /* set metadata */ - gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len, + gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len, timestamp); - ret = gst_base_rtp_payload_push (basepayload, outbuf); + ret = gst_rtp_base_payload_push (basepayload, outbuf); return ret; } static GstFlowReturn -gst_base_rtp_audio_payload_push_buffer (GstBaseRTPAudioPayload * +gst_rtp_base_audio_payload_push_buffer (GstRTPBaseAudioPayload * baseaudiopayload, GstBuffer * buffer, GstClockTime timestamp) { - GstBaseRTPPayload *basepayload; - GstBaseRTPAudioPayloadPrivate *priv; + GstRTPBasePayload *basepayload; + GstRTPBaseAudioPayloadPrivate *priv; GstBuffer *outbuf; guint8 *payload; guint payload_len; GstFlowReturn ret; priv = baseaudiopayload->priv; - basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload); + basepayload = GST_RTP_BASE_PAYLOAD (baseaudiopayload); payload_len = gst_buffer_get_size (buffer); @@ -506,7 +506,7 @@ gst_base_rtp_audio_payload_push_buffer (GstBaseRTPAudioPayload * } /* set metadata */ - gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len, + gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len, timestamp); if (priv->buffer_list) { @@ -524,7 +524,7 @@ gst_base_rtp_audio_payload_push_buffer (GstBaseRTPAudioPayload * } GST_DEBUG_OBJECT (baseaudiopayload, "Pushing list %p", list); - ret = gst_base_rtp_payload_push_list (basepayload, list); + ret = gst_rtp_base_payload_push_list (basepayload, list); } else { GstRTPBuffer rtp; @@ -537,15 +537,15 @@ gst_base_rtp_audio_payload_push_buffer (GstBaseRTPAudioPayload * gst_buffer_unref (buffer); GST_DEBUG_OBJECT (baseaudiopayload, "Pushing buffer %p", outbuf); - ret = gst_base_rtp_payload_push (basepayload, outbuf); + ret = gst_rtp_base_payload_push (basepayload, outbuf); } return ret; } /** - * gst_base_rtp_audio_payload_flush: - * @baseaudiopayload: a #GstBaseRTPPayload + * gst_rtp_base_audio_payload_flush: + * @baseaudiopayload: a #GstRTPBasePayload * @payload_len: length of payload * @timestamp: a #GstClockTime * @@ -561,11 +561,11 @@ gst_base_rtp_audio_payload_push_buffer (GstBaseRTPAudioPayload * * Since: 0.10.25 */ GstFlowReturn -gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload, +gst_rtp_base_audio_payload_flush (GstRTPBaseAudioPayload * baseaudiopayload, guint payload_len, GstClockTime timestamp) { - GstBaseRTPPayload *basepayload; - GstBaseRTPAudioPayloadPrivate *priv; + GstRTPBasePayload *basepayload; + GstRTPBaseAudioPayloadPrivate *priv; GstBuffer *outbuf; guint8 *payload; GstFlowReturn ret; @@ -575,7 +575,7 @@ gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload, priv = baseaudiopayload->priv; adapter = priv->adapter; - basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload); + basepayload = GST_RTP_BASE_PAYLOAD (baseaudiopayload); if (payload_len == -1) payload_len = gst_adapter_available (adapter); @@ -609,7 +609,7 @@ gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload, buffer = gst_adapter_take_buffer (adapter, payload_len); ret = - gst_base_rtp_audio_payload_push_buffer (baseaudiopayload, buffer, + gst_rtp_base_audio_payload_push_buffer (baseaudiopayload, buffer, timestamp); } else { GstRTPBuffer rtp; @@ -625,10 +625,10 @@ gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload, gst_rtp_buffer_unmap (&rtp); /* set metadata */ - gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len, + gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len, timestamp); - ret = gst_base_rtp_payload_push (basepayload, outbuf); + ret = gst_rtp_base_payload_push (basepayload, outbuf); } return ret; @@ -640,24 +640,24 @@ gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload, * mtu and min/max_ptime values. We cache those so that we don't have to redo * all the calculations */ static gboolean -gst_base_rtp_audio_payload_get_lengths (GstBaseRTPPayload * +gst_rtp_base_audio_payload_get_lengths (GstRTPBasePayload * basepayload, guint * min_payload_len, guint * max_payload_len, guint * align) { - GstBaseRTPAudioPayload *payload; - GstBaseRTPAudioPayloadPrivate *priv; + GstRTPBaseAudioPayload *payload; + GstRTPBaseAudioPayloadPrivate *priv; guint max_mtu, mtu; guint maxptime_octets; guint minptime_octets; guint ptime_mult_octets; - payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload); + payload = GST_RTP_BASE_AUDIO_PAYLOAD_CAST (basepayload); priv = payload->priv; if (priv->align == 0) return FALSE; - mtu = GST_BASE_RTP_PAYLOAD_MTU (payload); + mtu = GST_RTP_BASE_PAYLOAD_MTU (payload); /* check cached values */ if (G_LIKELY (priv->cached_mtu == mtu @@ -726,7 +726,7 @@ gst_base_rtp_audio_payload_get_lengths (GstBaseRTPPayload * /* frame conversions functions */ static GstClockTime -gst_base_rtp_audio_payload_frame_bytes_to_time (GstBaseRTPAudioPayload * +gst_rtp_base_audio_payload_frame_bytes_to_time (GstRTPBaseAudioPayload * payload, guint64 bytes) { guint64 framecount; @@ -739,7 +739,7 @@ gst_base_rtp_audio_payload_frame_bytes_to_time (GstBaseRTPAudioPayload * } static guint32 -gst_base_rtp_audio_payload_frame_bytes_to_rtptime (GstBaseRTPAudioPayload * +gst_rtp_base_audio_payload_frame_bytes_to_rtptime (GstRTPBaseAudioPayload * payload, guint64 bytes) { guint64 framecount; @@ -752,11 +752,11 @@ gst_base_rtp_audio_payload_frame_bytes_to_rtptime (GstBaseRTPAudioPayload * time = framecount * payload->priv->frame_duration_ns; return gst_util_uint64_scale_int (time, - GST_BASE_RTP_PAYLOAD (payload)->clock_rate, GST_SECOND); + GST_RTP_BASE_PAYLOAD (payload)->clock_rate, GST_SECOND); } static guint64 -gst_base_rtp_audio_payload_frame_time_to_bytes (GstBaseRTPAudioPayload * +gst_rtp_base_audio_payload_frame_time_to_bytes (GstRTPBaseAudioPayload * payload, GstClockTime time) { return gst_util_uint64_scale (time, payload->frame_size, @@ -765,7 +765,7 @@ gst_base_rtp_audio_payload_frame_time_to_bytes (GstBaseRTPAudioPayload * /* sample conversion functions */ static GstClockTime -gst_base_rtp_audio_payload_sample_bytes_to_time (GstBaseRTPAudioPayload * +gst_rtp_base_audio_payload_sample_bytes_to_time (GstRTPBaseAudioPayload * payload, guint64 bytes) { guint64 rtptime; @@ -777,11 +777,11 @@ gst_base_rtp_audio_payload_sample_bytes_to_time (GstBaseRTPAudioPayload * rtptime = bytes; return gst_util_uint64_scale_int (rtptime, GST_SECOND, - GST_BASE_RTP_PAYLOAD (payload)->clock_rate); + GST_RTP_BASE_PAYLOAD (payload)->clock_rate); } static guint32 -gst_base_rtp_audio_payload_sample_bytes_to_rtptime (GstBaseRTPAudioPayload * +gst_rtp_base_audio_payload_sample_bytes_to_rtptime (GstRTPBaseAudioPayload * payload, guint64 bytes) { /* avoid division when we can */ @@ -792,13 +792,13 @@ gst_base_rtp_audio_payload_sample_bytes_to_rtptime (GstBaseRTPAudioPayload * } static guint64 -gst_base_rtp_audio_payload_sample_time_to_bytes (GstBaseRTPAudioPayload * +gst_rtp_base_audio_payload_sample_time_to_bytes (GstRTPBaseAudioPayload * payload, guint64 time) { guint64 samples; samples = gst_util_uint64_scale_int (time, - GST_BASE_RTP_PAYLOAD (payload)->clock_rate, GST_SECOND); + GST_RTP_BASE_PAYLOAD (payload)->clock_rate, GST_SECOND); /* avoid multiplication when we can */ if (G_LIKELY (payload->sample_size != 8)) @@ -808,11 +808,11 @@ gst_base_rtp_audio_payload_sample_time_to_bytes (GstBaseRTPAudioPayload * } static GstFlowReturn -gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload * +gst_rtp_base_audio_payload_handle_buffer (GstRTPBasePayload * basepayload, GstBuffer * buffer) { - GstBaseRTPAudioPayload *payload; - GstBaseRTPAudioPayloadPrivate *priv; + GstRTPBaseAudioPayload *payload; + GstRTPBaseAudioPayloadPrivate *priv; guint payload_len; GstFlowReturn ret; guint available; @@ -825,7 +825,7 @@ gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload * ret = GST_FLOW_OK; - payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload); + payload = GST_RTP_BASE_AUDIO_PAYLOAD_CAST (basepayload); priv = payload->priv; timestamp = GST_BUFFER_TIMESTAMP (buffer); @@ -834,7 +834,7 @@ gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload * GST_DEBUG_OBJECT (payload, "Got DISCONT"); /* flush everything out of the adapter, mark DISCONT */ - ret = gst_base_rtp_audio_payload_flush (payload, -1, -1); + ret = gst_rtp_base_audio_payload_flush (payload, -1, -1); priv->discont = TRUE; /* get the distance between the timestamp gap and produce the same gap in @@ -862,7 +862,7 @@ gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload * } } - if (!gst_base_rtp_audio_payload_get_lengths (basepayload, &min_payload_len, + if (!gst_rtp_base_audio_payload_get_lengths (basepayload, &min_payload_len, &max_payload_len, &align)) goto config_error; @@ -884,7 +884,7 @@ gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload * /* If buffer fits on an RTP packet, let's just push it through * this will check against max_ptime and max_mtu */ GST_DEBUG_OBJECT (payload, "Fast packet push"); - ret = gst_base_rtp_audio_payload_push_buffer (payload, buffer, timestamp); + ret = gst_rtp_base_audio_payload_push_buffer (payload, buffer, timestamp); } else { /* push the buffer in the adapter */ gst_adapter_push (priv->adapter, buffer); @@ -900,7 +900,7 @@ gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload * /* and flush out the bytes from the adapter, automatically set the * timestamp. */ - ret = gst_base_rtp_audio_payload_flush (payload, payload_len, -1); + ret = gst_rtp_base_audio_payload_flush (payload, payload_len, -1); available -= payload_len; GST_DEBUG_OBJECT (payload, "available after push %u", available); @@ -919,13 +919,13 @@ config_error: } static GstStateChangeReturn -gst_base_rtp_payload_audio_change_state (GstElement * element, +gst_rtp_base_payload_audio_change_state (GstElement * element, GstStateChange transition) { - GstBaseRTPAudioPayload *basertppayload; + GstRTPBaseAudioPayload *basertppayload; GstStateChangeReturn ret; - basertppayload = GST_BASE_RTP_AUDIO_PAYLOAD (element); + basertppayload = GST_RTP_BASE_AUDIO_PAYLOAD (element); switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: @@ -951,18 +951,18 @@ gst_base_rtp_payload_audio_change_state (GstElement * element, } static gboolean -gst_base_rtp_payload_audio_handle_event (GstBaseRTPPayload * basep, +gst_rtp_base_payload_audio_handle_event (GstRTPBasePayload * basep, GstEvent * event) { - GstBaseRTPAudioPayload *payload; + GstRTPBaseAudioPayload *payload; gboolean res = FALSE; - payload = GST_BASE_RTP_AUDIO_PAYLOAD (basep); + payload = GST_RTP_BASE_AUDIO_PAYLOAD (basep); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_EOS: /* flush remaining bytes in the adapter */ - gst_base_rtp_audio_payload_flush (payload, -1, -1); + gst_rtp_base_audio_payload_flush (payload, -1, -1); break; case GST_EVENT_FLUSH_STOP: gst_adapter_clear (payload->priv->adapter); @@ -972,14 +972,14 @@ gst_base_rtp_payload_audio_handle_event (GstBaseRTPPayload * basep, } /* let parent handle the remainder of the event */ - res = GST_BASE_RTP_PAYLOAD_CLASS (parent_class)->handle_event (basep, event); + res = GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_event (basep, event); return res; } /** - * gst_base_rtp_audio_payload_get_adapter: - * @basertpaudiopayload: a #GstBaseRTPAudioPayload + * gst_rtp_base_audio_payload_get_adapter: + * @basertpaudiopayload: a #GstRTPBaseAudioPayload * * Gets the internal adapter used by the depayloader. * @@ -988,7 +988,7 @@ gst_base_rtp_payload_audio_handle_event (GstBaseRTPPayload * basep, * Since: 0.10.13 */ GstAdapter * -gst_base_rtp_audio_payload_get_adapter (GstBaseRTPAudioPayload +gst_rtp_base_audio_payload_get_adapter (GstRTPBaseAudioPayload * basertpaudiopayload) { GstAdapter *adapter; diff --git a/gst-libs/gst/rtp/gstbasertpaudiopayload.h b/gst-libs/gst/rtp/gstbasertpaudiopayload.h index 13b93667b9..34a160cbf7 100644 --- a/gst-libs/gst/rtp/gstbasertpaudiopayload.h +++ b/gst-libs/gst/rtp/gstbasertpaudiopayload.h @@ -17,8 +17,8 @@ * Boston, MA 02111-1307, USA. */ -#ifndef __GST_BASE_RTP_AUDIO_PAYLOAD_H__ -#define __GST_BASE_RTP_AUDIO_PAYLOAD_H__ +#ifndef __GST_RTP_BASE_AUDIO_PAYLOAD_H__ +#define __GST_RTP_BASE_AUDIO_PAYLOAD_H__ #include #include @@ -26,31 +26,31 @@ G_BEGIN_DECLS -typedef struct _GstBaseRTPAudioPayload GstBaseRTPAudioPayload; -typedef struct _GstBaseRTPAudioPayloadClass GstBaseRTPAudioPayloadClass; +typedef struct _GstRTPBaseAudioPayload GstRTPBaseAudioPayload; +typedef struct _GstRTPBaseAudioPayloadClass GstRTPBaseAudioPayloadClass; -typedef struct _GstBaseRTPAudioPayloadPrivate GstBaseRTPAudioPayloadPrivate; +typedef struct _GstRTPBaseAudioPayloadPrivate GstRTPBaseAudioPayloadPrivate; -#define GST_TYPE_BASE_RTP_AUDIO_PAYLOAD \ - (gst_base_rtp_audio_payload_get_type()) -#define GST_BASE_RTP_AUDIO_PAYLOAD(obj) \ +#define GST_TYPE_RTP_BASE_AUDIO_PAYLOAD \ + (gst_rtp_base_audio_payload_get_type()) +#define GST_RTP_BASE_AUDIO_PAYLOAD(obj) \ (G_TYPE_CHECK_INSTANCE_CAST((obj), \ - GST_TYPE_BASE_RTP_AUDIO_PAYLOAD,GstBaseRTPAudioPayload)) -#define GST_BASE_RTP_AUDIO_PAYLOAD_CLASS(klass) \ + GST_TYPE_RTP_BASE_AUDIO_PAYLOAD,GstRTPBaseAudioPayload)) +#define GST_RTP_BASE_AUDIO_PAYLOAD_CLASS(klass) \ (G_TYPE_CHECK_CLASS_CAST((klass), \ - GST_TYPE_BASE_RTP_AUDIO_PAYLOAD,GstBaseRTPAudioPayloadClass)) -#define GST_IS_BASE_RTP_AUDIO_PAYLOAD(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD)) -#define GST_IS_BASE_RTP_AUDIO_PAYLOAD_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD)) -#define GST_BASE_RTP_AUDIO_PAYLOAD_CAST(obj) \ - ((GstBaseRTPAudioPayload *) (obj)) + GST_TYPE_RTP_BASE_AUDIO_PAYLOAD,GstRTPBaseAudioPayloadClass)) +#define GST_IS_RTP_BASE_AUDIO_PAYLOAD(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_BASE_AUDIO_PAYLOAD)) +#define GST_IS_RTP_BASE_AUDIO_PAYLOAD_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_BASE_AUDIO_PAYLOAD)) +#define GST_RTP_BASE_AUDIO_PAYLOAD_CAST(obj) \ + ((GstRTPBaseAudioPayload *) (obj)) -struct _GstBaseRTPAudioPayload +struct _GstRTPBaseAudioPayload { - GstBaseRTPPayload payload; + GstRTPBasePayload payload; - GstBaseRTPAudioPayloadPrivate *priv; + GstRTPBaseAudioPayloadPrivate *priv; GstClockTime base_ts; gint frame_size; @@ -62,44 +62,44 @@ struct _GstBaseRTPAudioPayload }; /** - * GstBaseRTPAudioPayloadClass: + * GstRTPBaseAudioPayloadClass: * @parent_class: the parent class * * Base class for audio RTP payloader. */ -struct _GstBaseRTPAudioPayloadClass +struct _GstRTPBaseAudioPayloadClass { - GstBaseRTPPayloadClass parent_class; + GstRTPBasePayloadClass parent_class; /*< private >*/ gpointer _gst_reserved[GST_PADDING]; }; -GType gst_base_rtp_audio_payload_get_type (void); +GType gst_rtp_base_audio_payload_get_type (void); /* configure frame based */ -void gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload *basertpaudiopayload); +void gst_rtp_base_audio_payload_set_frame_based (GstRTPBaseAudioPayload *basertpaudiopayload); -void gst_base_rtp_audio_payload_set_frame_options (GstBaseRTPAudioPayload *basertpaudiopayload, +void gst_rtp_base_audio_payload_set_frame_options (GstRTPBaseAudioPayload *basertpaudiopayload, gint frame_duration, gint frame_size); /* configure sample based */ -void gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload *basertpaudiopayload); -void gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload *basertpaudiopayload, +void gst_rtp_base_audio_payload_set_sample_based (GstRTPBaseAudioPayload *basertpaudiopayload); +void gst_rtp_base_audio_payload_set_sample_options (GstRTPBaseAudioPayload *basertpaudiopayload, gint sample_size); -void gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload *basertpaudiopayload, +void gst_rtp_base_audio_payload_set_samplebits_options (GstRTPBaseAudioPayload *basertpaudiopayload, gint sample_size); /* get the internal adapter */ -GstAdapter* gst_base_rtp_audio_payload_get_adapter (GstBaseRTPAudioPayload *basertpaudiopayload); +GstAdapter* gst_rtp_base_audio_payload_get_adapter (GstRTPBaseAudioPayload *basertpaudiopayload); /* push and flushing data */ -GstFlowReturn gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload, +GstFlowReturn gst_rtp_base_audio_payload_push (GstRTPBaseAudioPayload * baseaudiopayload, const guint8 * data, guint payload_len, GstClockTime timestamp); -GstFlowReturn gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload, +GstFlowReturn gst_rtp_base_audio_payload_flush (GstRTPBaseAudioPayload * baseaudiopayload, guint payload_len, GstClockTime timestamp); G_END_DECLS -#endif /* __GST_BASE_RTP_AUDIO_PAYLOAD_H__ */ +#endif /* __GST_RTP_BASE_AUDIO_PAYLOAD_H__ */ diff --git a/gst-libs/gst/rtp/gstbasertpdepayload.c b/gst-libs/gst/rtp/gstbasertpdepayload.c index 5a6892d88d..ad3f5be48a 100644 --- a/gst-libs/gst/rtp/gstbasertpdepayload.c +++ b/gst-libs/gst/rtp/gstbasertpdepayload.c @@ -30,10 +30,10 @@ GST_DEBUG_CATEGORY_STATIC (basertpdepayload_debug); #define GST_CAT_DEFAULT (basertpdepayload_debug) -#define GST_BASE_RTP_DEPAYLOAD_GET_PRIVATE(obj) \ - (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_RTP_DEPAYLOAD, GstBaseRTPDepayloadPrivate)) +#define GST_RTP_BASE_DEPAYLOAD_GET_PRIVATE(obj) \ + (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BASE_DEPAYLOAD, GstRTPBaseDepayloadPrivate)) -struct _GstBaseRTPDepayloadPrivate +struct _GstRTPBaseDepayloadPrivate { GstClockTime npt_start; GstClockTime npt_stop; @@ -63,58 +63,58 @@ enum PROP_LAST }; -static void gst_base_rtp_depayload_finalize (GObject * object); -static void gst_base_rtp_depayload_set_property (GObject * object, +static void gst_rtp_base_depayload_finalize (GObject * object); +static void gst_rtp_base_depayload_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); -static void gst_base_rtp_depayload_get_property (GObject * object, +static void gst_rtp_base_depayload_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); -static GstFlowReturn gst_base_rtp_depayload_chain (GstPad * pad, +static GstFlowReturn gst_rtp_base_depayload_chain (GstPad * pad, GstBuffer * in); -static gboolean gst_base_rtp_depayload_handle_sink_event (GstPad * pad, +static gboolean gst_rtp_base_depayload_handle_sink_event (GstPad * pad, GstEvent * event); -static GstStateChangeReturn gst_base_rtp_depayload_change_state (GstElement * +static GstStateChangeReturn gst_rtp_base_depayload_change_state (GstElement * element, GstStateChange transition); -static gboolean gst_base_rtp_depayload_packet_lost (GstBaseRTPDepayload * +static gboolean gst_rtp_base_depayload_packet_lost (GstRTPBaseDepayload * filter, GstEvent * event); -static gboolean gst_base_rtp_depayload_handle_event (GstBaseRTPDepayload * +static gboolean gst_rtp_base_depayload_handle_event (GstRTPBaseDepayload * filter, GstEvent * event); static GstElementClass *parent_class = NULL; -static void gst_base_rtp_depayload_class_init (GstBaseRTPDepayloadClass * +static void gst_rtp_base_depayload_class_init (GstRTPBaseDepayloadClass * klass); -static void gst_base_rtp_depayload_init (GstBaseRTPDepayload * basertppayload, - GstBaseRTPDepayloadClass * klass); +static void gst_rtp_base_depayload_init (GstRTPBaseDepayload * basertppayload, + GstRTPBaseDepayloadClass * klass); GType -gst_base_rtp_depayload_get_type (void) +gst_rtp_base_depayload_get_type (void) { - static GType base_rtp_depayload_type = 0; + static GType rtp_base_depayload_type = 0; - if (g_once_init_enter ((gsize *) & base_rtp_depayload_type)) { - static const GTypeInfo base_rtp_depayload_info = { - sizeof (GstBaseRTPDepayloadClass), + if (g_once_init_enter ((gsize *) & rtp_base_depayload_type)) { + static const GTypeInfo rtp_base_depayload_info = { + sizeof (GstRTPBaseDepayloadClass), NULL, NULL, - (GClassInitFunc) gst_base_rtp_depayload_class_init, + (GClassInitFunc) gst_rtp_base_depayload_class_init, NULL, NULL, - sizeof (GstBaseRTPDepayload), + sizeof (GstRTPBaseDepayload), 0, - (GInstanceInitFunc) gst_base_rtp_depayload_init, + (GInstanceInitFunc) gst_rtp_base_depayload_init, }; - g_once_init_leave ((gsize *) & base_rtp_depayload_type, - g_type_register_static (GST_TYPE_ELEMENT, "GstBaseRTPDepayload", - &base_rtp_depayload_info, G_TYPE_FLAG_ABSTRACT)); + g_once_init_leave ((gsize *) & rtp_base_depayload_type, + g_type_register_static (GST_TYPE_ELEMENT, "GstRTPBaseDepayload", + &rtp_base_depayload_info, G_TYPE_FLAG_ABSTRACT)); } - return base_rtp_depayload_type; + return rtp_base_depayload_type; } static void -gst_base_rtp_depayload_class_init (GstBaseRTPDepayloadClass * klass) +gst_rtp_base_depayload_class_init (GstRTPBaseDepayloadClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; @@ -123,29 +123,29 @@ gst_base_rtp_depayload_class_init (GstBaseRTPDepayloadClass * klass) gstelement_class = (GstElementClass *) klass; parent_class = g_type_class_peek_parent (klass); - g_type_class_add_private (klass, sizeof (GstBaseRTPDepayloadPrivate)); + g_type_class_add_private (klass, sizeof (GstRTPBaseDepayloadPrivate)); - gobject_class->finalize = gst_base_rtp_depayload_finalize; - gobject_class->set_property = gst_base_rtp_depayload_set_property; - gobject_class->get_property = gst_base_rtp_depayload_get_property; + gobject_class->finalize = gst_rtp_base_depayload_finalize; + gobject_class->set_property = gst_rtp_base_depayload_set_property; + gobject_class->get_property = gst_rtp_base_depayload_get_property; - gstelement_class->change_state = gst_base_rtp_depayload_change_state; + gstelement_class->change_state = gst_rtp_base_depayload_change_state; - klass->packet_lost = gst_base_rtp_depayload_packet_lost; - klass->handle_event = gst_base_rtp_depayload_handle_event; + 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, "Base class for RTP Depayloaders"); } static void -gst_base_rtp_depayload_init (GstBaseRTPDepayload * filter, - GstBaseRTPDepayloadClass * klass) +gst_rtp_base_depayload_init (GstRTPBaseDepayload * filter, + GstRTPBaseDepayloadClass * klass) { GstPadTemplate *pad_template; - GstBaseRTPDepayloadPrivate *priv; + GstRTPBaseDepayloadPrivate *priv; - priv = GST_BASE_RTP_DEPAYLOAD_GET_PRIVATE (filter); + priv = GST_RTP_BASE_DEPAYLOAD_GET_PRIVATE (filter); filter->priv = priv; GST_DEBUG_OBJECT (filter, "init"); @@ -154,9 +154,9 @@ gst_base_rtp_depayload_init (GstBaseRTPDepayload * filter, gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "sink"); g_return_if_fail (pad_template != NULL); filter->sinkpad = gst_pad_new_from_template (pad_template, "sink"); - gst_pad_set_chain_function (filter->sinkpad, gst_base_rtp_depayload_chain); + gst_pad_set_chain_function (filter->sinkpad, gst_rtp_base_depayload_chain); gst_pad_set_event_function (filter->sinkpad, - gst_base_rtp_depayload_handle_sink_event); + gst_rtp_base_depayload_handle_sink_event); gst_element_add_pad (GST_ELEMENT (filter), filter->sinkpad); pad_template = @@ -170,23 +170,23 @@ gst_base_rtp_depayload_init (GstBaseRTPDepayload * filter, } static void -gst_base_rtp_depayload_finalize (GObject * object) +gst_rtp_base_depayload_finalize (GObject * object) { G_OBJECT_CLASS (parent_class)->finalize (object); } static gboolean -gst_base_rtp_depayload_setcaps (GstBaseRTPDepayload * filter, GstCaps * caps) +gst_rtp_base_depayload_setcaps (GstRTPBaseDepayload * filter, GstCaps * caps) { - GstBaseRTPDepayloadClass *bclass; - GstBaseRTPDepayloadPrivate *priv; + GstRTPBaseDepayloadClass *bclass; + GstRTPBaseDepayloadPrivate *priv; gboolean res; GstStructure *caps_struct; const GValue *value; priv = filter->priv; - bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter); + bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (filter); GST_DEBUG_OBJECT (filter, "Set caps"); @@ -236,11 +236,11 @@ gst_base_rtp_depayload_setcaps (GstBaseRTPDepayload * filter, GstCaps * caps) } static GstFlowReturn -gst_base_rtp_depayload_chain (GstPad * pad, GstBuffer * in) +gst_rtp_base_depayload_chain (GstPad * pad, GstBuffer * in) { - GstBaseRTPDepayload *filter; - GstBaseRTPDepayloadPrivate *priv; - GstBaseRTPDepayloadClass *bclass; + GstRTPBaseDepayload *filter; + GstRTPBaseDepayloadPrivate *priv; + GstRTPBaseDepayloadClass *bclass; GstFlowReturn ret = GST_FLOW_OK; GstBuffer *out_buf; GstClockTime pts, dts; @@ -250,7 +250,7 @@ gst_base_rtp_depayload_chain (GstPad * pad, GstBuffer * in) gint gap; GstRTPBuffer rtp; - filter = GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad)); + filter = GST_RTP_BASE_DEPAYLOAD (GST_OBJECT_PARENT (pad)); priv = filter->priv; /* we must have a setcaps first */ @@ -326,7 +326,7 @@ gst_base_rtp_depayload_chain (GstPad * pad, GstBuffer * in) GST_BUFFER_FLAG_SET (in, GST_BUFFER_FLAG_DISCONT); } - bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter); + bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (filter); if (G_UNLIKELY (bclass->process == NULL)) goto no_process; @@ -334,7 +334,7 @@ gst_base_rtp_depayload_chain (GstPad * pad, GstBuffer * in) /* let's send it out to processing */ out_buf = bclass->process (filter, in); if (out_buf) { - ret = gst_base_rtp_depayload_push (filter, out_buf); + ret = gst_rtp_base_depayload_push (filter, out_buf); } gst_buffer_unref (in); @@ -380,7 +380,7 @@ no_process: } static gboolean -gst_base_rtp_depayload_handle_event (GstBaseRTPDepayload * filter, +gst_rtp_base_depayload_handle_event (GstRTPBaseDepayload * filter, GstEvent * event) { gboolean res = TRUE; @@ -398,7 +398,7 @@ gst_base_rtp_depayload_handle_event (GstBaseRTPDepayload * filter, gst_event_parse_caps (event, &caps); - res = gst_base_rtp_depayload_setcaps (filter, caps); + res = gst_rtp_base_depayload_setcaps (filter, caps); forward = FALSE; break; } @@ -412,9 +412,9 @@ gst_base_rtp_depayload_handle_event (GstBaseRTPDepayload * filter, } case GST_EVENT_CUSTOM_DOWNSTREAM: { - GstBaseRTPDepayloadClass *bclass; + GstRTPBaseDepayloadClass *bclass; - bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter); + bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (filter); if (gst_event_has_name (event, "GstRTPPacketLost")) { /* we get this event from the jitterbuffer when it considers a packet as @@ -446,19 +446,19 @@ gst_base_rtp_depayload_handle_event (GstBaseRTPDepayload * filter, } static gboolean -gst_base_rtp_depayload_handle_sink_event (GstPad * pad, GstEvent * event) +gst_rtp_base_depayload_handle_sink_event (GstPad * pad, GstEvent * event) { gboolean res = FALSE; - GstBaseRTPDepayload *filter; - GstBaseRTPDepayloadClass *bclass; + GstRTPBaseDepayload *filter; + GstRTPBaseDepayloadClass *bclass; - filter = GST_BASE_RTP_DEPAYLOAD (gst_pad_get_parent (pad)); + filter = GST_RTP_BASE_DEPAYLOAD (gst_pad_get_parent (pad)); if (G_UNLIKELY (filter == NULL)) { gst_event_unref (event); return FALSE; } - bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter); + bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (filter); if (bclass->handle_event) res = bclass->handle_event (filter, event); else @@ -469,12 +469,12 @@ gst_base_rtp_depayload_handle_sink_event (GstPad * pad, GstEvent * event) } static GstEvent * -create_segment_event (GstBaseRTPDepayload * filter, gboolean update, +create_segment_event (GstRTPBaseDepayload * filter, gboolean update, GstClockTime position) { GstEvent *event; GstClockTime stop; - GstBaseRTPDepayloadPrivate *priv; + GstRTPBaseDepayloadPrivate *priv; GstSegment segment; priv = filter->priv; @@ -499,15 +499,15 @@ create_segment_event (GstBaseRTPDepayload * filter, gboolean update, typedef struct { - GstBaseRTPDepayload *depayload; - GstBaseRTPDepayloadClass *bclass; + GstRTPBaseDepayload *depayload; + GstRTPBaseDepayloadClass *bclass; } HeaderData; static gboolean set_headers (GstBuffer ** buffer, guint idx, HeaderData * data) { - GstBaseRTPDepayload *depayload = data->depayload; - GstBaseRTPDepayloadPrivate *priv = depayload->priv; + GstRTPBaseDepayload *depayload = data->depayload; + GstRTPBaseDepayloadPrivate *priv = depayload->priv; GstClockTime pts, dts, duration; *buffer = gst_buffer_make_writable (*buffer); @@ -540,13 +540,13 @@ set_headers (GstBuffer ** buffer, guint idx, HeaderData * data) } static GstFlowReturn -gst_base_rtp_depayload_prepare_push (GstBaseRTPDepayload * filter, +gst_rtp_base_depayload_prepare_push (GstRTPBaseDepayload * filter, gboolean is_list, gpointer obj) { HeaderData data; data.depayload = filter; - data.bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter); + data.bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (filter); if (is_list) { GstBufferList **blist = obj; @@ -572,8 +572,8 @@ gst_base_rtp_depayload_prepare_push (GstBaseRTPDepayload * filter, } /** - * gst_base_rtp_depayload_push: - * @filter: a #GstBaseRTPDepayload + * gst_rtp_base_depayload_push: + * @filter: a #GstRTPBaseDepayload * @out_buf: a #GstBuffer * * Push @out_buf to the peer of @filter. This function takes ownership of @@ -585,11 +585,11 @@ gst_base_rtp_depayload_prepare_push (GstBaseRTPDepayload * filter, * Returns: a #GstFlowReturn. */ GstFlowReturn -gst_base_rtp_depayload_push (GstBaseRTPDepayload * filter, GstBuffer * out_buf) +gst_rtp_base_depayload_push (GstRTPBaseDepayload * filter, GstBuffer * out_buf) { GstFlowReturn res; - res = gst_base_rtp_depayload_prepare_push (filter, FALSE, &out_buf); + res = gst_rtp_base_depayload_prepare_push (filter, FALSE, &out_buf); if (G_LIKELY (res == GST_FLOW_OK)) res = gst_pad_push (filter->srcpad, out_buf); @@ -600,8 +600,8 @@ gst_base_rtp_depayload_push (GstBaseRTPDepayload * filter, GstBuffer * out_buf) } /** - * gst_base_rtp_depayload_push_list: - * @filter: a #GstBaseRTPDepayload + * gst_rtp_base_depayload_push_list: + * @filter: a #GstRTPBaseDepayload * @out_list: a #GstBufferList * * Push @out_list to the peer of @filter. This function takes ownership of @@ -612,12 +612,12 @@ gst_base_rtp_depayload_push (GstBaseRTPDepayload * filter, GstBuffer * out_buf) * Since: 0.10.32 */ GstFlowReturn -gst_base_rtp_depayload_push_list (GstBaseRTPDepayload * filter, +gst_rtp_base_depayload_push_list (GstRTPBaseDepayload * filter, GstBufferList * out_list) { GstFlowReturn res; - res = gst_base_rtp_depayload_prepare_push (filter, TRUE, &out_list); + res = gst_rtp_base_depayload_prepare_push (filter, TRUE, &out_list); if (G_LIKELY (res == GST_FLOW_OK)) res = gst_pad_push_list (filter->srcpad, out_list); @@ -630,7 +630,7 @@ gst_base_rtp_depayload_push_list (GstBaseRTPDepayload * filter, /* convert the PacketLost event form a jitterbuffer to a segment update. * subclasses can override this. */ static gboolean -gst_base_rtp_depayload_packet_lost (GstBaseRTPDepayload * filter, +gst_rtp_base_depayload_packet_lost (GstRTPBaseDepayload * filter, GstEvent * event) { GstClockTime timestamp, duration, position; @@ -657,14 +657,14 @@ gst_base_rtp_depayload_packet_lost (GstBaseRTPDepayload * filter, } static GstStateChangeReturn -gst_base_rtp_depayload_change_state (GstElement * element, +gst_rtp_base_depayload_change_state (GstElement * element, GstStateChange transition) { - GstBaseRTPDepayload *filter; - GstBaseRTPDepayloadPrivate *priv; + GstRTPBaseDepayload *filter; + GstRTPBaseDepayloadPrivate *priv; GstStateChangeReturn ret; - filter = GST_BASE_RTP_DEPAYLOAD (element); + filter = GST_RTP_BASE_DEPAYLOAD (element); priv = filter->priv; switch (transition) { @@ -702,7 +702,7 @@ gst_base_rtp_depayload_change_state (GstElement * element, } static void -gst_base_rtp_depayload_set_property (GObject * object, guint prop_id, +gst_rtp_base_depayload_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { switch (prop_id) { @@ -713,7 +713,7 @@ gst_base_rtp_depayload_set_property (GObject * object, guint prop_id, } static void -gst_base_rtp_depayload_get_property (GObject * object, guint prop_id, +gst_rtp_base_depayload_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { switch (prop_id) { diff --git a/gst-libs/gst/rtp/gstbasertpdepayload.h b/gst-libs/gst/rtp/gstbasertpdepayload.h index a02143f4a2..3373d4a403 100644 --- a/gst-libs/gst/rtp/gstbasertpdepayload.h +++ b/gst-libs/gst/rtp/gstbasertpdepayload.h @@ -17,34 +17,34 @@ * Boston, MA 02111-1307, USA. */ -#ifndef __GST_BASE_RTP_DEPAYLOAD_H__ -#define __GST_BASE_RTP_DEPAYLOAD_H__ +#ifndef __GST_RTP_BASE_DEPAYLOAD_H__ +#define __GST_RTP_BASE_DEPAYLOAD_H__ #include #include G_BEGIN_DECLS -#define GST_TYPE_BASE_RTP_DEPAYLOAD (gst_base_rtp_depayload_get_type()) -#define GST_BASE_RTP_DEPAYLOAD(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_RTP_DEPAYLOAD,GstBaseRTPDepayload)) -#define GST_BASE_RTP_DEPAYLOAD_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_RTP_DEPAYLOAD,GstBaseRTPDepayloadClass)) -#define GST_BASE_RTP_DEPAYLOAD_GET_CLASS(obj) \ - (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_BASE_RTP_DEPAYLOAD,GstBaseRTPDepayloadClass)) -#define GST_IS_BASE_RTP_DEPAYLOAD(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_RTP_DEPAYLOAD)) -#define GST_IS_BASE_RTP_DEPAYLOAD_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_RTP_DEPAYLOAD)) +#define GST_TYPE_RTP_BASE_DEPAYLOAD (gst_rtp_base_depayload_get_type()) +#define GST_RTP_BASE_DEPAYLOAD(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_BASE_DEPAYLOAD,GstRTPBaseDepayload)) +#define GST_RTP_BASE_DEPAYLOAD_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_BASE_DEPAYLOAD,GstRTPBaseDepayloadClass)) +#define GST_RTP_BASE_DEPAYLOAD_GET_CLASS(obj) \ + (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_RTP_BASE_DEPAYLOAD,GstRTPBaseDepayloadClass)) +#define GST_IS_RTP_BASE_DEPAYLOAD(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_BASE_DEPAYLOAD)) +#define GST_IS_RTP_BASE_DEPAYLOAD_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_BASE_DEPAYLOAD)) -#define GST_BASE_RTP_DEPAYLOAD_SINKPAD(depayload) (GST_BASE_RTP_DEPAYLOAD (depayload)->sinkpad) -#define GST_BASE_RTP_DEPAYLOAD_SRCPAD(depayload) (GST_BASE_RTP_DEPAYLOAD (depayload)->srcpad) +#define GST_RTP_BASE_DEPAYLOAD_SINKPAD(depayload) (GST_RTP_BASE_DEPAYLOAD (depayload)->sinkpad) +#define GST_RTP_BASE_DEPAYLOAD_SRCPAD(depayload) (GST_RTP_BASE_DEPAYLOAD (depayload)->srcpad) -typedef struct _GstBaseRTPDepayload GstBaseRTPDepayload; -typedef struct _GstBaseRTPDepayloadClass GstBaseRTPDepayloadClass; -typedef struct _GstBaseRTPDepayloadPrivate GstBaseRTPDepayloadPrivate; +typedef struct _GstRTPBaseDepayload GstRTPBaseDepayload; +typedef struct _GstRTPBaseDepayloadClass GstRTPBaseDepayloadClass; +typedef struct _GstRTPBaseDepayloadPrivate GstRTPBaseDepayloadPrivate; -struct _GstBaseRTPDepayload +struct _GstRTPBaseDepayload { GstElement parent; @@ -57,57 +57,55 @@ struct _GstBaseRTPDepayload gboolean need_newsegment; /*< private >*/ - GstBaseRTPDepayloadPrivate *priv; + GstRTPBaseDepayloadPrivate *priv; gpointer _gst_reserved[GST_PADDING-1]; }; /** - * GstBaseRTPDepayloadClass: + * GstRTPBaseDepayloadClass: * @parent_class: the parent class * @set_caps: configure the depayloader - * @add_to_queue: (deprecated) * @process: process incoming rtp packets - * @set_gst_timestamp: convert from RTP timestamp to GST timestamp * @packet_lost: signal the depayloader about packet loss * @handle_event: custom event handling * * Base class for audio RTP payloader. */ -struct _GstBaseRTPDepayloadClass +struct _GstRTPBaseDepayloadClass { GstElementClass parent_class; /* virtuals, inform the subclass of the caps. */ - gboolean (*set_caps) (GstBaseRTPDepayload *filter, GstCaps *caps); + gboolean (*set_caps) (GstRTPBaseDepayload *filter, GstCaps *caps); /* pure virtual function, child must use this to process incoming * rtp packets. If the child returns a buffer without a valid timestamp, * the timestamp of @in will be applied to the result buffer and the * buffer will be pushed. If this function returns %NULL, nothing is * pushed. */ - GstBuffer * (*process) (GstBaseRTPDepayload *base, GstBuffer *in); + GstBuffer * (*process) (GstRTPBaseDepayload *base, GstBuffer *in); /* non-pure function used to to signal the depayloader about packet loss. the * timestamp and duration are the estimated values of the lost packet. * The default implementation of this message pushes a segment update. */ - gboolean (*packet_lost) (GstBaseRTPDepayload *filter, GstEvent *event); + gboolean (*packet_lost) (GstRTPBaseDepayload *filter, GstEvent *event); /* the default implementation does the default actions for events but * implementation can override. * Since: 0.10.32 */ - gboolean (*handle_event) (GstBaseRTPDepayload * filter, GstEvent * event); + gboolean (*handle_event) (GstRTPBaseDepayload * filter, GstEvent * event); /*< private >*/ gpointer _gst_reserved[GST_PADDING-2]; }; -GType gst_base_rtp_depayload_get_type (void); +GType gst_rtp_base_depayload_get_type (void); -GstFlowReturn gst_base_rtp_depayload_push (GstBaseRTPDepayload *filter, GstBuffer *out_buf); -GstFlowReturn gst_base_rtp_depayload_push_list (GstBaseRTPDepayload *filter, GstBufferList *out_list); +GstFlowReturn gst_rtp_base_depayload_push (GstRTPBaseDepayload *filter, GstBuffer *out_buf); +GstFlowReturn gst_rtp_base_depayload_push_list (GstRTPBaseDepayload *filter, GstBufferList *out_list); G_END_DECLS -#endif /* __GST_BASE_RTP_DEPAYLOAD_H__ */ +#endif /* __GST_RTP_BASE_DEPAYLOAD_H__ */ diff --git a/gst-libs/gst/rtp/gstbasertppayload.c b/gst-libs/gst/rtp/gstbasertppayload.c index 4a6a533738..2b14a1c4dc 100644 --- a/gst-libs/gst/rtp/gstbasertppayload.c +++ b/gst-libs/gst/rtp/gstbasertppayload.c @@ -32,10 +32,10 @@ GST_DEBUG_CATEGORY_STATIC (basertppayload_debug); #define GST_CAT_DEFAULT (basertppayload_debug) -#define GST_BASE_RTP_PAYLOAD_GET_PRIVATE(obj) \ - (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_RTP_PAYLOAD, GstBaseRTPPayloadPrivate)) +#define GST_RTP_BASE_PAYLOAD_GET_PRIVATE(obj) \ + (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BASE_PAYLOAD, GstRTPBasePayloadPrivate)) -struct _GstBaseRTPPayloadPrivate +struct _GstRTPBasePayloadPrivate { gboolean ts_offset_random; gboolean seqnum_offset_random; @@ -51,7 +51,7 @@ struct _GstBaseRTPPayloadPrivate gint64 caps_max_ptime; }; -/* BaseRTPPayload signals and args */ +/* RTPBasePayload signals and args */ enum { /* FILL ME */ @@ -91,27 +91,27 @@ enum PROP_LAST }; -static void gst_base_rtp_payload_class_init (GstBaseRTPPayloadClass * klass); -static void gst_base_rtp_payload_init (GstBaseRTPPayload * basertppayload, +static void gst_rtp_base_payload_class_init (GstRTPBasePayloadClass * klass); +static void gst_rtp_base_payload_init (GstRTPBasePayload * basertppayload, gpointer g_class); -static void gst_base_rtp_payload_finalize (GObject * object); +static void gst_rtp_base_payload_finalize (GObject * object); -static GstCaps *gst_base_rtp_payload_sink_getcaps (GstPad * pad, +static GstCaps *gst_rtp_base_payload_sink_getcaps (GstPad * pad, GstCaps * filter); -static gboolean gst_base_rtp_payload_event_default (GstBaseRTPPayload * +static gboolean gst_rtp_base_payload_event_default (GstRTPBasePayload * basertppayload, GstEvent * event); -static gboolean gst_base_rtp_payload_event (GstPad * pad, GstEvent * event); -static GstFlowReturn gst_base_rtp_payload_chain (GstPad * pad, +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_base_rtp_payload_getcaps_default (GstBaseRTPPayload * +static GstCaps *gst_rtp_base_payload_getcaps_default (GstRTPBasePayload * basertppayload, GstPad * pad, GstCaps * filter); -static void gst_base_rtp_payload_set_property (GObject * object, guint prop_id, +static void gst_rtp_base_payload_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); -static void gst_base_rtp_payload_get_property (GObject * object, guint prop_id, +static void gst_rtp_base_payload_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); -static GstStateChangeReturn gst_base_rtp_payload_change_state (GstElement * +static GstStateChangeReturn gst_rtp_base_payload_change_state (GstElement * element, GstStateChange transition); static GstElementClass *parent_class = NULL; @@ -119,32 +119,32 @@ static GstElementClass *parent_class = NULL; /* FIXME 0.11: API should be changed to gst_base_typ_payload_xyz */ GType -gst_base_rtp_payload_get_type (void) +gst_rtp_base_payload_get_type (void) { static GType basertppayload_type = 0; if (g_once_init_enter ((gsize *) & basertppayload_type)) { static const GTypeInfo basertppayload_info = { - sizeof (GstBaseRTPPayloadClass), + sizeof (GstRTPBasePayloadClass), NULL, NULL, - (GClassInitFunc) gst_base_rtp_payload_class_init, + (GClassInitFunc) gst_rtp_base_payload_class_init, NULL, NULL, - sizeof (GstBaseRTPPayload), + sizeof (GstRTPBasePayload), 0, - (GInstanceInitFunc) gst_base_rtp_payload_init, + (GInstanceInitFunc) gst_rtp_base_payload_init, }; g_once_init_leave ((gsize *) & basertppayload_type, - g_type_register_static (GST_TYPE_ELEMENT, "GstBaseRTPPayload", + g_type_register_static (GST_TYPE_ELEMENT, "GstRTPBasePayload", &basertppayload_info, G_TYPE_FLAG_ABSTRACT)); } return basertppayload_type; } static void -gst_base_rtp_payload_class_init (GstBaseRTPPayloadClass * klass) +gst_rtp_base_payload_class_init (GstRTPBasePayloadClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; @@ -152,14 +152,14 @@ gst_base_rtp_payload_class_init (GstBaseRTPPayloadClass * klass) gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; - g_type_class_add_private (klass, sizeof (GstBaseRTPPayloadPrivate)); + g_type_class_add_private (klass, sizeof (GstRTPBasePayloadPrivate)); parent_class = g_type_class_peek_parent (klass); - gobject_class->finalize = gst_base_rtp_payload_finalize; + gobject_class->finalize = gst_rtp_base_payload_finalize; - gobject_class->set_property = gst_base_rtp_payload_set_property; - gobject_class->get_property = gst_base_rtp_payload_get_property; + gobject_class->set_property = gst_rtp_base_payload_set_property; + gobject_class->get_property = gst_rtp_base_payload_get_property; g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MTU, g_param_spec_uint ("mtu", "MTU", @@ -190,7 +190,7 @@ gst_base_rtp_payload_class_init (GstBaseRTPPayloadClass * klass) -1, G_MAXINT64, DEFAULT_MAX_PTIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** - * GstBaseRTPAudioPayload:min-ptime: + * GstRTPBaseAudioPayload:min-ptime: * * Minimum duration of the packet data in ns (can't go above MTU) * @@ -212,7 +212,7 @@ gst_base_rtp_payload_class_init (GstBaseRTPPayloadClass * klass) 0, G_MAXUINT16, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); /** - * GstBaseRTPAudioPayload:perfect-rtptime: + * GstRTPBaseAudioPayload:perfect-rtptime: * * Try to use the offset fields to generate perfect RTP timestamps. when this * option is disabled, RTP timestamps are generated from the GStreamer @@ -226,7 +226,7 @@ gst_base_rtp_payload_class_init (GstBaseRTPPayloadClass * klass) "Generate perfect RTP timestamps when possible", DEFAULT_PERFECT_RTPTIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** - * GstBaseRTPAudioPayload:ptime-multiple: + * GstRTPBaseAudioPayload:ptime-multiple: * * Force buffers to be multiples of this duration in ns (0 disables) * @@ -238,23 +238,23 @@ gst_base_rtp_payload_class_init (GstBaseRTPPayloadClass * klass) 0, G_MAXINT64, DEFAULT_PTIME_MULTIPLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); - gstelement_class->change_state = gst_base_rtp_payload_change_state; + gstelement_class->change_state = gst_rtp_base_payload_change_state; - klass->get_caps = gst_base_rtp_payload_getcaps_default; - klass->handle_event = gst_base_rtp_payload_event_default; + 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, "Base class for RTP Payloaders"); } static void -gst_base_rtp_payload_init (GstBaseRTPPayload * basertppayload, gpointer g_class) +gst_rtp_base_payload_init (GstRTPBasePayload * basertppayload, gpointer g_class) { GstPadTemplate *templ; - GstBaseRTPPayloadPrivate *priv; + GstRTPBasePayloadPrivate *priv; basertppayload->priv = priv = - GST_BASE_RTP_PAYLOAD_GET_PRIVATE (basertppayload); + GST_RTP_BASE_PAYLOAD_GET_PRIVATE (basertppayload); templ = gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "src"); @@ -269,11 +269,11 @@ gst_base_rtp_payload_init (GstBaseRTPPayload * basertppayload, gpointer g_class) basertppayload->sinkpad = gst_pad_new_from_template (templ, "sink"); gst_pad_set_getcaps_function (basertppayload->sinkpad, - gst_base_rtp_payload_sink_getcaps); + gst_rtp_base_payload_sink_getcaps); gst_pad_set_event_function (basertppayload->sinkpad, - gst_base_rtp_payload_event); + gst_rtp_base_payload_event); gst_pad_set_chain_function (basertppayload->sinkpad, - gst_base_rtp_payload_chain); + gst_rtp_base_payload_chain); gst_element_add_pad (GST_ELEMENT (basertppayload), basertppayload->sinkpad); basertppayload->mtu = DEFAULT_MTU; @@ -302,11 +302,11 @@ gst_base_rtp_payload_init (GstBaseRTPPayload * basertppayload, gpointer g_class) } static void -gst_base_rtp_payload_finalize (GObject * object) +gst_rtp_base_payload_finalize (GObject * object) { - GstBaseRTPPayload *basertppayload; + GstRTPBasePayload *basertppayload; - basertppayload = GST_BASE_RTP_PAYLOAD (object); + basertppayload = GST_RTP_BASE_PAYLOAD (object); g_free (basertppayload->media); basertppayload->media = NULL; @@ -317,7 +317,7 @@ gst_base_rtp_payload_finalize (GObject * object) } static GstCaps * -gst_base_rtp_payload_getcaps_default (GstBaseRTPPayload * basertppayload, +gst_rtp_base_payload_getcaps_default (GstRTPBasePayload * basertppayload, GstPad * pad, GstCaps * filter) { GstCaps *caps; @@ -336,16 +336,16 @@ gst_base_rtp_payload_getcaps_default (GstBaseRTPPayload * basertppayload, } static GstCaps * -gst_base_rtp_payload_sink_getcaps (GstPad * pad, GstCaps * filter) +gst_rtp_base_payload_sink_getcaps (GstPad * pad, GstCaps * filter) { - GstBaseRTPPayload *basertppayload; - GstBaseRTPPayloadClass *basertppayload_class; + GstRTPBasePayload *basertppayload; + GstRTPBasePayloadClass *basertppayload_class; GstCaps *caps = NULL; GST_DEBUG_OBJECT (pad, "getting caps"); - basertppayload = GST_BASE_RTP_PAYLOAD (gst_pad_get_parent (pad)); - basertppayload_class = GST_BASE_RTP_PAYLOAD_GET_CLASS (basertppayload); + basertppayload = GST_RTP_BASE_PAYLOAD (gst_pad_get_parent (pad)); + basertppayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (basertppayload); if (basertppayload_class->get_caps) caps = basertppayload_class->get_caps (basertppayload, pad, filter); @@ -356,7 +356,7 @@ gst_base_rtp_payload_sink_getcaps (GstPad * pad, GstCaps * filter) } static gboolean -gst_base_rtp_payload_event_default (GstBaseRTPPayload * basertppayload, +gst_rtp_base_payload_event_default (GstRTPBasePayload * basertppayload, GstEvent * event) { gboolean res = FALSE; @@ -371,13 +371,13 @@ gst_base_rtp_payload_event_default (GstBaseRTPPayload * basertppayload, break; case GST_EVENT_CAPS: { - GstBaseRTPPayloadClass *basertppayload_class; + GstRTPBasePayloadClass *basertppayload_class; GstCaps *caps; gst_event_parse_caps (event, &caps); GST_DEBUG_OBJECT (basertppayload, "setting caps %" GST_PTR_FORMAT, caps); - basertppayload_class = GST_BASE_RTP_PAYLOAD_GET_CLASS (basertppayload); + basertppayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (basertppayload); if (basertppayload_class->set_caps) res = basertppayload_class->set_caps (basertppayload, caps); @@ -406,19 +406,19 @@ gst_base_rtp_payload_event_default (GstBaseRTPPayload * basertppayload, } static gboolean -gst_base_rtp_payload_event (GstPad * pad, GstEvent * event) +gst_rtp_base_payload_event (GstPad * pad, GstEvent * event) { - GstBaseRTPPayload *basertppayload; - GstBaseRTPPayloadClass *basertppayload_class; + GstRTPBasePayload *basertppayload; + GstRTPBasePayloadClass *basertppayload_class; gboolean res = FALSE; - basertppayload = GST_BASE_RTP_PAYLOAD (gst_pad_get_parent (pad)); + basertppayload = GST_RTP_BASE_PAYLOAD (gst_pad_get_parent (pad)); if (G_UNLIKELY (basertppayload == NULL)) { gst_event_unref (event); return FALSE; } - basertppayload_class = GST_BASE_RTP_PAYLOAD_GET_CLASS (basertppayload); + basertppayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (basertppayload); if (basertppayload_class->handle_event) res = basertppayload_class->handle_event (basertppayload, event); @@ -432,14 +432,14 @@ gst_base_rtp_payload_event (GstPad * pad, GstEvent * event) static GstFlowReturn -gst_base_rtp_payload_chain (GstPad * pad, GstBuffer * buffer) +gst_rtp_base_payload_chain (GstPad * pad, GstBuffer * buffer) { - GstBaseRTPPayload *basertppayload; - GstBaseRTPPayloadClass *basertppayload_class; + GstRTPBasePayload *basertppayload; + GstRTPBasePayloadClass *basertppayload_class; GstFlowReturn ret; - basertppayload = GST_BASE_RTP_PAYLOAD (gst_pad_get_parent (pad)); - basertppayload_class = GST_BASE_RTP_PAYLOAD_GET_CLASS (basertppayload); + basertppayload = GST_RTP_BASE_PAYLOAD (gst_pad_get_parent (pad)); + basertppayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (basertppayload); if (!basertppayload_class->handle_buffer) goto no_function; @@ -462,8 +462,8 @@ no_function: } /** - * gst_base_rtp_payload_set_options: - * @payload: a #GstBaseRTPPayload + * gst_rtp_base_payload_set_options: + * @payload: a #GstRTPBasePayload * @media: the media type (typically "audio" or "video") * @dynamic: if the payload type is dynamic * @encoding_name: the encoding name @@ -471,10 +471,10 @@ no_function: * * Set the rtp options of the payloader. These options will be set in the caps * of the payloader. Subclasses must call this method before calling - * gst_base_rtp_payload_push() or gst_base_rtp_payload_set_outcaps(). + * gst_rtp_base_payload_push() or gst_rtp_base_payload_set_outcaps(). */ void -gst_base_rtp_payload_set_options (GstBaseRTPPayload * payload, +gst_rtp_base_payload_set_options (GstRTPBasePayload * payload, const gchar * media, gboolean dynamic, const gchar * encoding_name, guint32 clock_rate) { @@ -499,7 +499,7 @@ copy_fixed (GQuark field_id, const GValue * value, GstStructure * dest) } static void -update_max_ptime (GstBaseRTPPayload * basertppayload) +update_max_ptime (GstRTPBasePayload * basertppayload) { if (basertppayload->priv->caps_max_ptime != -1 && basertppayload->priv->prop_max_ptime != -1) @@ -514,8 +514,8 @@ update_max_ptime (GstBaseRTPPayload * basertppayload) } /** - * gst_base_rtp_payload_set_outcaps: - * @payload: a #GstBaseRTPPayload + * gst_rtp_base_payload_set_outcaps: + * @payload: a #GstRTPBasePayload * @fieldname: the first field name or %NULL * @...: field values * @@ -527,7 +527,7 @@ update_max_ptime (GstBaseRTPPayload * basertppayload) * Returns: %TRUE if the caps could be set. */ gboolean -gst_base_rtp_payload_set_outcaps (GstBaseRTPPayload * payload, +gst_rtp_base_payload_set_outcaps (GstRTPBasePayload * payload, const gchar * fieldname, ...) { GstCaps *srccaps, *peercaps; @@ -560,7 +560,7 @@ gst_base_rtp_payload_set_outcaps (GstBaseRTPPayload * payload, if (peercaps == NULL) { /* no peer caps, just add the other properties */ gst_caps_set_simple (srccaps, - "payload", G_TYPE_INT, GST_BASE_RTP_PAYLOAD_PT (payload), + "payload", G_TYPE_INT, GST_RTP_BASE_PAYLOAD_PT (payload), "ssrc", G_TYPE_UINT, payload->current_ssrc, "timestamp-offset", G_TYPE_UINT, payload->ts_base, "seqnum-offset", G_TYPE_UINT, payload->seqnum_base, NULL); @@ -597,18 +597,18 @@ gst_base_rtp_payload_set_outcaps (GstBaseRTPPayload * payload, if (gst_structure_get_int (s, "payload", &pt)) { /* use peer pt */ - GST_BASE_RTP_PAYLOAD_PT (payload) = pt; + GST_RTP_BASE_PAYLOAD_PT (payload) = pt; GST_LOG_OBJECT (payload, "using peer pt %d", pt); } else { if (gst_structure_has_field (s, "payload")) { /* can only fixate if there is a field */ gst_structure_fixate_field_nearest_int (s, "payload", - GST_BASE_RTP_PAYLOAD_PT (payload)); + GST_RTP_BASE_PAYLOAD_PT (payload)); gst_structure_get_int (s, "payload", &pt); GST_LOG_OBJECT (payload, "using peer pt %d", pt); } else { /* no pt field, use the internal pt */ - pt = GST_BASE_RTP_PAYLOAD_PT (payload); + pt = GST_RTP_BASE_PAYLOAD_PT (payload); gst_structure_set (s, "payload", G_TYPE_INT, pt, NULL); GST_LOG_OBJECT (payload, "using internal pt %d", pt); } @@ -664,15 +664,15 @@ gst_base_rtp_payload_set_outcaps (GstBaseRTPPayload * payload, update_max_ptime (payload); - res = gst_pad_set_caps (GST_BASE_RTP_PAYLOAD_SRCPAD (payload), srccaps); + res = gst_pad_set_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload), srccaps); gst_caps_unref (srccaps); return res; } /** - * gst_base_rtp_payload_is_filled: - * @payload: a #GstBaseRTPPayload + * gst_rtp_base_payload_is_filled: + * @payload: a #GstRTPBasePayload * @size: the size of the packet * @duration: the duration of the packet * @@ -683,7 +683,7 @@ gst_base_rtp_payload_set_outcaps (GstBaseRTPPayload * payload, * configured MTU or max_ptime. */ gboolean -gst_base_rtp_payload_is_filled (GstBaseRTPPayload * payload, +gst_rtp_base_payload_is_filled (GstRTPBasePayload * payload, guint size, GstClockTime duration) { if (size > payload->mtu) @@ -697,7 +697,7 @@ gst_base_rtp_payload_is_filled (GstBaseRTPPayload * payload, typedef struct { - GstBaseRTPPayload *payload; + GstRTPBasePayload *payload; guint32 ssrc; guint16 seqnum; guint8 pt; @@ -741,10 +741,10 @@ set_headers (GstBuffer ** buffer, guint group, guint idx, HeaderData * data) /* Updates the SSRC, payload type, seqnum and timestamp of the RTP buffer * before the buffer is pushed. */ static GstFlowReturn -gst_base_rtp_payload_prepare_push (GstBaseRTPPayload * payload, +gst_rtp_base_payload_prepare_push (GstRTPBasePayload * payload, gpointer obj, gboolean is_list) { - GstBaseRTPPayloadPrivate *priv; + GstRTPBasePayloadPrivate *priv; HeaderData data; if (payload->clock_rate == 0) @@ -844,8 +844,8 @@ no_rate: } /** - * gst_base_rtp_payload_push_list: - * @payload: a #GstBaseRTPPayload + * gst_rtp_base_payload_push_list: + * @payload: a #GstRTPBasePayload * @list: a #GstBufferList * * Push @list to the peer element of the payloader. The SSRC, payload type, @@ -858,12 +858,12 @@ no_rate: * Since: 0.10.24 */ GstFlowReturn -gst_base_rtp_payload_push_list (GstBaseRTPPayload * payload, +gst_rtp_base_payload_push_list (GstRTPBasePayload * payload, GstBufferList * list) { GstFlowReturn res; - res = gst_base_rtp_payload_prepare_push (payload, list, TRUE); + res = gst_rtp_base_payload_prepare_push (payload, list, TRUE); if (G_LIKELY (res == GST_FLOW_OK)) res = gst_pad_push_list (payload->srcpad, list); @@ -874,8 +874,8 @@ gst_base_rtp_payload_push_list (GstBaseRTPPayload * payload, } /** - * gst_base_rtp_payload_push: - * @payload: a #GstBaseRTPPayload + * gst_rtp_base_payload_push: + * @payload: a #GstRTPBasePayload * @buffer: a #GstBuffer * * Push @buffer to the peer element of the payloader. The SSRC, payload type, @@ -886,11 +886,11 @@ gst_base_rtp_payload_push_list (GstBaseRTPPayload * payload, * Returns: a #GstFlowReturn. */ GstFlowReturn -gst_base_rtp_payload_push (GstBaseRTPPayload * payload, GstBuffer * buffer) +gst_rtp_base_payload_push (GstRTPBasePayload * payload, GstBuffer * buffer) { GstFlowReturn res; - res = gst_base_rtp_payload_prepare_push (payload, buffer, FALSE); + res = gst_rtp_base_payload_prepare_push (payload, buffer, FALSE); if (G_LIKELY (res == GST_FLOW_OK)) res = gst_pad_push (payload->srcpad, buffer); @@ -901,14 +901,14 @@ gst_base_rtp_payload_push (GstBaseRTPPayload * payload, GstBuffer * buffer) } static void -gst_base_rtp_payload_set_property (GObject * object, guint prop_id, +gst_rtp_base_payload_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { - GstBaseRTPPayload *basertppayload; - GstBaseRTPPayloadPrivate *priv; + GstRTPBasePayload *basertppayload; + GstRTPBasePayloadPrivate *priv; gint64 val; - basertppayload = GST_BASE_RTP_PAYLOAD (object); + basertppayload = GST_RTP_BASE_PAYLOAD (object); priv = basertppayload->priv; switch (prop_id) { @@ -955,13 +955,13 @@ gst_base_rtp_payload_set_property (GObject * object, guint prop_id, } static void -gst_base_rtp_payload_get_property (GObject * object, guint prop_id, +gst_rtp_base_payload_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { - GstBaseRTPPayload *basertppayload; - GstBaseRTPPayloadPrivate *priv; + GstRTPBasePayload *basertppayload; + GstRTPBasePayloadPrivate *priv; - basertppayload = GST_BASE_RTP_PAYLOAD (object); + basertppayload = GST_RTP_BASE_PAYLOAD (object); priv = basertppayload->priv; switch (prop_id) { @@ -1014,14 +1014,14 @@ gst_base_rtp_payload_get_property (GObject * object, guint prop_id, } static GstStateChangeReturn -gst_base_rtp_payload_change_state (GstElement * element, +gst_rtp_base_payload_change_state (GstElement * element, GstStateChange transition) { - GstBaseRTPPayload *basertppayload; - GstBaseRTPPayloadPrivate *priv; + GstRTPBasePayload *basertppayload; + GstRTPBasePayloadPrivate *priv; GstStateChangeReturn ret; - basertppayload = GST_BASE_RTP_PAYLOAD (element); + basertppayload = GST_RTP_BASE_PAYLOAD (element); priv = basertppayload->priv; switch (transition) { diff --git a/gst-libs/gst/rtp/gstbasertppayload.h b/gst-libs/gst/rtp/gstbasertppayload.h index e8dff7281f..aeae973ee1 100644 --- a/gst-libs/gst/rtp/gstbasertppayload.h +++ b/gst-libs/gst/rtp/gstbasertppayload.h @@ -17,63 +17,63 @@ * Boston, MA 02111-1307, USA. */ -#ifndef __GST_BASE_RTP_PAYLOAD_H__ -#define __GST_BASE_RTP_PAYLOAD_H__ +#ifndef __GST_RTP_BASE_PAYLOAD_H__ +#define __GST_RTP_BASE_PAYLOAD_H__ #include G_BEGIN_DECLS -#define GST_TYPE_BASE_RTP_PAYLOAD \ - (gst_base_rtp_payload_get_type()) -#define GST_BASE_RTP_PAYLOAD(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_RTP_PAYLOAD,GstBaseRTPPayload)) -#define GST_BASE_RTP_PAYLOAD_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_RTP_PAYLOAD,GstBaseRTPPayloadClass)) -#define GST_BASE_RTP_PAYLOAD_GET_CLASS(obj) \ - (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_BASE_RTP_PAYLOAD, GstBaseRTPPayloadClass)) -#define GST_IS_BASE_RTP_PAYLOAD(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_RTP_PAYLOAD)) -#define GST_IS_BASE_RTP_PAYLOAD_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_RTP_PAYLOAD)) -#define GST_BASE_RTP_PAYLOAD_CAST(obj) \ - ((GstBaseRTPPayload*)(obj)) +#define GST_TYPE_RTP_BASE_PAYLOAD \ + (gst_rtp_base_payload_get_type()) +#define GST_RTP_BASE_PAYLOAD(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_BASE_PAYLOAD,GstRTPBasePayload)) +#define GST_RTP_BASE_PAYLOAD_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_BASE_PAYLOAD,GstRTPBasePayloadClass)) +#define GST_RTP_BASE_PAYLOAD_GET_CLASS(obj) \ + (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTP_BASE_PAYLOAD, GstRTPBasePayloadClass)) +#define GST_IS_RTP_BASE_PAYLOAD(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_BASE_PAYLOAD)) +#define GST_IS_RTP_BASE_PAYLOAD_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_BASE_PAYLOAD)) +#define GST_RTP_BASE_PAYLOAD_CAST(obj) \ + ((GstRTPBasePayload*)(obj)) -typedef struct _GstBaseRTPPayload GstBaseRTPPayload; -typedef struct _GstBaseRTPPayloadPrivate GstBaseRTPPayloadPrivate; -typedef struct _GstBaseRTPPayloadClass GstBaseRTPPayloadClass; +typedef struct _GstRTPBasePayload GstRTPBasePayload; +typedef struct _GstRTPBasePayloadPrivate GstRTPBasePayloadPrivate; +typedef struct _GstRTPBasePayloadClass GstRTPBasePayloadClass; /** - * GST_BASE_RTP_PAYLOAD_SINKPAD: - * @payload: a #GstBaseRTPPayload + * GST_RTP_BASE_PAYLOAD_SINKPAD: + * @payload: a #GstRTPBasePayload * * Get access to the sinkpad of @payload. */ -#define GST_BASE_RTP_PAYLOAD_SINKPAD(payload) (GST_BASE_RTP_PAYLOAD (payload)->sinkpad) +#define GST_RTP_BASE_PAYLOAD_SINKPAD(payload) (GST_RTP_BASE_PAYLOAD (payload)->sinkpad) /** - * GST_BASE_RTP_PAYLOAD_SRCPAD: - * @payload: a #GstBaseRTPPayload + * GST_RTP_BASE_PAYLOAD_SRCPAD: + * @payload: a #GstRTPBasePayload * * Get access to the srcpad of @payload. */ -#define GST_BASE_RTP_PAYLOAD_SRCPAD(payload) (GST_BASE_RTP_PAYLOAD (payload)->srcpad) +#define GST_RTP_BASE_PAYLOAD_SRCPAD(payload) (GST_RTP_BASE_PAYLOAD (payload)->srcpad) /** - * GST_BASE_RTP_PAYLOAD_PT: - * @payload: a #GstBaseRTPPayload + * GST_RTP_BASE_PAYLOAD_PT: + * @payload: a #GstRTPBasePayload * * Get access to the configured payload type of @payload. */ -#define GST_BASE_RTP_PAYLOAD_PT(payload) (GST_BASE_RTP_PAYLOAD (payload)->pt) +#define GST_RTP_BASE_PAYLOAD_PT(payload) (GST_RTP_BASE_PAYLOAD (payload)->pt) /** - * GST_BASE_RTP_PAYLOAD_MTU: - * @payload: a #GstBaseRTPPayload + * GST_RTP_BASE_PAYLOAD_MTU: + * @payload: a #GstRTPBasePayload * * Get access to the configured MTU of @payload. */ -#define GST_BASE_RTP_PAYLOAD_MTU(payload) (GST_BASE_RTP_PAYLOAD (payload)->mtu) +#define GST_RTP_BASE_PAYLOAD_MTU(payload) (GST_RTP_BASE_PAYLOAD (payload)->mtu) -struct _GstBaseRTPPayload +struct _GstRTPBasePayload { GstElement element; @@ -104,7 +104,7 @@ struct _GstBaseRTPPayload guint64 min_ptime; /*< private >*/ - GstBaseRTPPayloadPrivate *priv; + GstRTPBasePayloadPrivate *priv; union { struct { @@ -116,7 +116,7 @@ struct _GstBaseRTPPayload }; /** - * GstBaseRTPPayloadClass: + * GstRTPBasePayloadClass: * @parent_class: the parent class * @set_caps: configure the payloader * @handle_buffer: process data @@ -125,45 +125,45 @@ struct _GstBaseRTPPayload * * Base class for audio RTP payloader. */ -struct _GstBaseRTPPayloadClass +struct _GstRTPBasePayloadClass { GstElementClass parent_class; /* query accepted caps */ - GstCaps * (*get_caps) (GstBaseRTPPayload *payload, GstPad * pad, GstCaps * filter); + GstCaps * (*get_caps) (GstRTPBasePayload *payload, GstPad * pad, GstCaps * filter); /* receive caps on the sink pad, configure the payloader. */ - gboolean (*set_caps) (GstBaseRTPPayload *payload, GstCaps *caps); + gboolean (*set_caps) (GstRTPBasePayload *payload, GstCaps *caps); - /* handle a buffer, perform 0 or more gst_base_rtp_payload_push() on + /* handle a buffer, perform 0 or more gst_rtp_base_payload_push() on * the RTP buffers. This function takes ownership of the buffer. */ - GstFlowReturn (*handle_buffer) (GstBaseRTPPayload *payload, + GstFlowReturn (*handle_buffer) (GstRTPBasePayload *payload, GstBuffer *buffer); - gboolean (*handle_event) (GstBaseRTPPayload *payload, GstEvent * event); + gboolean (*handle_event) (GstRTPBasePayload *payload, GstEvent * event); /*< private >*/ gpointer _gst_reserved[GST_PADDING]; }; -GType gst_base_rtp_payload_get_type (void); +GType gst_rtp_base_payload_get_type (void); -void gst_base_rtp_payload_set_options (GstBaseRTPPayload *payload, +void gst_rtp_base_payload_set_options (GstRTPBasePayload *payload, const gchar *media, gboolean dynamic, const gchar *encoding_name, guint32 clock_rate); -gboolean gst_base_rtp_payload_set_outcaps (GstBaseRTPPayload *payload, +gboolean gst_rtp_base_payload_set_outcaps (GstRTPBasePayload *payload, const gchar *fieldname, ...); -gboolean gst_base_rtp_payload_is_filled (GstBaseRTPPayload *payload, +gboolean gst_rtp_base_payload_is_filled (GstRTPBasePayload *payload, guint size, GstClockTime duration); -GstFlowReturn gst_base_rtp_payload_push (GstBaseRTPPayload *payload, +GstFlowReturn gst_rtp_base_payload_push (GstRTPBasePayload *payload, GstBuffer *buffer); -GstFlowReturn gst_base_rtp_payload_push_list (GstBaseRTPPayload *payload, +GstFlowReturn gst_rtp_base_payload_push_list (GstRTPBasePayload *payload, GstBufferList *list); G_END_DECLS -#endif /* __GST_BASE_RTP_PAYLOAD_H__ */ +#endif /* __GST_RTP_BASE_PAYLOAD_H__ */ diff --git a/gst-libs/gst/rtp/gstrtcpbuffer.c b/gst-libs/gst/rtp/gstrtcpbuffer.c index 323c5cfc16..6f19a22d20 100644 --- a/gst-libs/gst/rtp/gstrtcpbuffer.c +++ b/gst-libs/gst/rtp/gstrtcpbuffer.c @@ -23,7 +23,7 @@ /** * SECTION:gstrtcpbuffer * @short_description: Helper methods for dealing with RTCP buffers - * @see_also: #GstBaseRTPPayload, #GstBaseRTPDepayload, #gstrtpbuffer + * @see_also: #GstRTPBasePayload, #GstRTPBaseDepayload, #gstrtpbuffer * * Note: The API in this module is not yet declared stable. * diff --git a/gst-libs/gst/rtp/gstrtpbuffer.c b/gst-libs/gst/rtp/gstrtpbuffer.c index 66df84121c..172d2a9c82 100644 --- a/gst-libs/gst/rtp/gstrtpbuffer.c +++ b/gst-libs/gst/rtp/gstrtpbuffer.c @@ -21,7 +21,7 @@ /** * SECTION:gstrtpbuffer * @short_description: Helper methods for dealing with RTP buffers - * @see_also: #GstBaseRTPPayload, #GstBaseRTPDepayload, gstrtcpbuffer + * @see_also: #GstRTPBasePayload, #GstRTPBaseDepayload, gstrtcpbuffer * * *