mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-23 16:50:47 +00:00
docs: more helper libraries docs fixes
Quieten gtk-doc a bit more.
This commit is contained in:
parent
4b06fad321
commit
e836151009
9 changed files with 34 additions and 8 deletions
|
@ -1362,6 +1362,7 @@ gst_rtp_buffer_list_set_timestamp
|
|||
<FILE>gstrtspdefs</FILE>
|
||||
<INCLUDE>gst/rtsp/gstrtspdefs.h</INCLUDE>
|
||||
GST_RTSP_CHECK
|
||||
GstRTSPEvent
|
||||
GstRTSPResult
|
||||
GstRTSPFamily
|
||||
GstRTSPState
|
||||
|
@ -1386,6 +1387,7 @@ gst_rtsp_find_method
|
|||
<INCLUDE>gst/rtsp/gstrtsptransport.h</INCLUDE>
|
||||
GstRTSPTransMode
|
||||
GstRTSPProfile
|
||||
GstRTSPRange
|
||||
GstRTSPLowerTrans
|
||||
GstRTSPTransport
|
||||
gst_rtsp_transport_new
|
||||
|
|
|
@ -152,6 +152,12 @@ gboolean gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf);
|
|||
/* functions useful for _getcaps functions */
|
||||
/**
|
||||
* GstAudioFieldFlag:
|
||||
* @GST_AUDIO_FIELD_RATE: add rate field to caps
|
||||
* @GST_AUDIO_FIELD_CHANNELS: add channels field to caps
|
||||
* @GST_AUDIO_FIELD_ENDIANNESS: add endianness field to caps
|
||||
* @GST_AUDIO_FIELD_WIDTH: add width field to caps
|
||||
* @GST_AUDIO_FIELD_DEPTH: add depth field to caps
|
||||
* @GST_AUDIO_FIELD_SIGNED: add signed field to caps
|
||||
*
|
||||
* Do not use anymore.
|
||||
*
|
||||
|
|
|
@ -44,7 +44,7 @@
|
|||
* </para></listitem>
|
||||
* <listitem><para>Either all or none of the channel positions are %GST_AUDIO_CHANNEL_POSITION_NONE.
|
||||
* </para></listitem>
|
||||
* <listitem><para>%GST_AUDIO_CHANNEL_POSITION_FRONT_MONO and %GST_AUDIO_CHANNEL_POSITION_LEFT or %GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT don't appear together in the given positions.
|
||||
* <listitem><para>%GST_AUDIO_CHANNEL_POSITION_FRONT_MONO and %GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT or %GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT don't appear together in the given positions.
|
||||
* </para></listitem>
|
||||
* </itemizedlist>
|
||||
*
|
||||
|
|
|
@ -25,6 +25,24 @@
|
|||
|
||||
G_BEGIN_DECLS
|
||||
|
||||
/**
|
||||
* GstAudioChannelPosition:
|
||||
* @GST_AUDIO_CHANNEL_POSITION_FRONT_MONO: front mono
|
||||
* @GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT: front left
|
||||
* @GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT: front right
|
||||
* @GST_AUDIO_CHANNEL_POSITION_REAR_CENTER: rear center
|
||||
* @GST_AUDIO_CHANNEL_POSITION_REAR_LEFT: rear left
|
||||
* @GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT: rear right
|
||||
* @GST_AUDIO_CHANNEL_POSITION_LFE: subwoofer
|
||||
* @GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER: front center
|
||||
* @GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER: front left of center
|
||||
* @GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER: front right of center
|
||||
* @GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT: side left
|
||||
* @GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT: side right
|
||||
* @GST_AUDIO_CHANNEL_POSITION_NONE: used for position-less channels, e.g.
|
||||
* from a sound card that records 1024 channels; mutually exclusive with
|
||||
* any other channel position
|
||||
*/
|
||||
typedef enum {
|
||||
GST_AUDIO_CHANNEL_POSITION_INVALID = -1,
|
||||
|
||||
|
|
|
@ -165,7 +165,7 @@ gst_property_probe_get_property (GstPropertyProbe * probe, const gchar * name)
|
|||
* @probe: the #GstPropertyProbe to check.
|
||||
* @pspec: #GParamSpec of the property.
|
||||
*
|
||||
* Runs a probe on the property specified by %pspec
|
||||
* Runs a probe on the property specified by @pspec
|
||||
*/
|
||||
void
|
||||
gst_property_probe_probe_property (GstPropertyProbe * probe,
|
||||
|
@ -188,7 +188,7 @@ gst_property_probe_probe_property (GstPropertyProbe * probe,
|
|||
* @probe: the #GstPropertyProbe to check.
|
||||
* @name: name of the property.
|
||||
*
|
||||
* Runs a probe on the property specified by %name.
|
||||
* Runs a probe on the property specified by @name.
|
||||
*/
|
||||
void
|
||||
gst_property_probe_probe_property_name (GstPropertyProbe * probe,
|
||||
|
|
|
@ -338,7 +338,7 @@ gst_tuner_get_norm (GstTuner * tuner)
|
|||
|
||||
/**
|
||||
* gst_tuner_set_frequency:
|
||||
* @tuner: The #Gsttuner (a #GstElement) that owns the given channel.
|
||||
* @tuner: The #GstTuner (a #GstElement) that owns the given channel.
|
||||
* @channel: The #GstTunerChannel to set the frequency on.
|
||||
* @frequency: The frequency to tune in to.
|
||||
*
|
||||
|
|
|
@ -88,7 +88,7 @@
|
|||
* </para>
|
||||
* <para>
|
||||
* The application will then call gst_install_plugins_async(), passing a
|
||||
* #NULL-terminated array of installer detail strings, and a function that
|
||||
* NULL-terminated array of installer detail strings, and a function that
|
||||
* should be called when the installation of the plugins has finished
|
||||
* (successfully or not). Optionally, a #GstInstallPluginsContext created
|
||||
* with gst_install_plugins_context_new() may be passed as well. This way
|
||||
|
|
|
@ -178,8 +178,8 @@ gst_rtp_buffer_new_copy_data (gpointer data, guint len)
|
|||
* @pad_len: the amount of padding
|
||||
* @csrc_count: the number of CSRC entries
|
||||
*
|
||||
* Allocate a new #Gstbuffer with enough data to hold an RTP packet with @csrc_count CSRCs,
|
||||
* a payload length of @payload_len and padding of @pad_len.
|
||||
* Allocate a new #GstBuffer with enough data to hold an RTP packet with
|
||||
* @csrc_count CSRCs, a payload length of @payload_len and padding of @pad_len.
|
||||
* All other RTP header fields will be set to 0/FALSE.
|
||||
*
|
||||
* Returns: A newly allocated buffer that can hold an RTP packet with given
|
||||
|
|
|
@ -94,7 +94,7 @@ typedef enum {
|
|||
} GstRTSPLowerTrans;
|
||||
|
||||
/**
|
||||
* RTSPRange:
|
||||
* GstRTSPRange:
|
||||
* @min: minimum value of the range
|
||||
* @max: maximum value of the range
|
||||
*
|
||||
|
|
Loading…
Reference in a new issue