Commit 7fe3f36a authored by Thibault Saunier's avatar Thibault Saunier

Minor documentation fixes

parent dce17521
......@@ -47,7 +47,7 @@
#include "io-sim.h"
/**
/*
* @addtogroup Rawenc Raw VBI encoder
* @ingroup Raw
* @brief Converting sliced VBI data to raw VBI images.
......@@ -375,7 +375,7 @@ clear_image (uint8_t * p,
}
}
/**
/*
* @param raw Noise will be added to this raw VBI image.
* @param sp Describes the raw VBI data in the buffer. @a sp->sampling_format
* must be @c VBI_PIXFMT_Y8 (@c VBI_PIXFMT_YUV420 in libzvbi 0.2.x).
......@@ -1008,12 +1008,12 @@ _vbi_raw_video_image (uint8_t * raw,
return TRUE;
}
/**
/*
* @example examples/rawout.c
* Raw VBI output example.
*/
/**
/*
* @param raw A raw VBI image will be stored here.
* @param raw_size Size of the @a raw buffer in bytes. The buffer
* must be large enough for @a sp->count[0] + count[1] lines
......@@ -1079,7 +1079,7 @@ vbi_raw_vbi_image (uint8_t * raw,
swap_fields ? _VBI_RAW_SWAP_FIELDS : 0, sliced, n_sliced_lines);
}
/**
/*
* @param raw A raw VBI image will be stored here.
* @param raw_size Size of the @a raw buffer in bytes. The buffer
* must be large enough for @a sp->count[0] + count[1] lines
......
......@@ -21,7 +21,6 @@
* SECTION:element-curlsink
* @title: curlsink
* @short_description: sink that uploads data to a server using libcurl
* @see_also:
*
* This is a network sink that uses libcurl as a client to upload data to
* a server (e.g. a HTTP/FTP server).
......
......@@ -21,7 +21,6 @@
* SECTION:element-curlfilesink
* @title: curlfilesink
* @short_description: sink that uploads data to a server using libcurl
* @see_also:
*
* This is a network sink that uses libcurl as a client to upload data to
* a local or network drive.
......
......@@ -21,7 +21,6 @@
* SECTION:element-curlftpsink
* @title: curlftpsink
* @short_description: sink that uploads data to a server using libcurl
* @see_also:
*
* This is a network sink that uses libcurl as a client to upload data to
* an FTP server.
......
......@@ -21,7 +21,6 @@
* SECTION:element-curlhttpsink
* @title: curlhttpsink
* @short_description: sink that uploads data to a server using libcurl
* @see_also:
*
* This is a network sink that uses libcurl as a client to upload data to
* an HTTP server.
......
......@@ -21,7 +21,6 @@
* SECTION:element-curlsftpsink
* @title: curlsftpsink
* @short_description: sink that uploads data to a server using libcurl
* @see_also:
*
* This is a network sink that uses libcurl as a client to upload data to
* a SFTP (SSH File Transfer Protocol) server.
......
......@@ -21,7 +21,6 @@
* SECTION:element-curlsink
* @title: curlsink
* @short_description: sink that uploads data to a server using libcurl
* @see_also:
*
* This is a network sink that uses libcurl as a client to upload data to
* an SMTP server.
......
......@@ -21,7 +21,6 @@
* SECTION:element-curlsshsink
* @title: curlsshsink
* @short_description: sink that uploads data to a server using libcurl
* @see_also:
*
* This is a network sink that uses libcurl.
*
......
......@@ -21,7 +21,6 @@
* SECTION:element-curltlssink
* @title: curltlssink
* @short_description: sink that uploads data to a server using libcurl
* @see_also:
*
* This is a network sink that uses libcurl.
*
......
......@@ -975,7 +975,7 @@ gst_mss_manifest_get_duration (GstMssManifest * manifest)
}
/**
/*
* Gets the duration in nanoseconds
*/
GstClockTime
......@@ -1194,7 +1194,7 @@ gst_mss_stream_type_name (GstMssStreamType streamtype)
}
}
/**
/*
* Seeks all streams to the fragment that contains the set time
*
* @forward: if this is forward playback
......@@ -1215,7 +1215,7 @@ gst_mss_manifest_seek (GstMssManifest * manifest, gboolean forward,
((forward && (flags & GST_SEEK_FLAG_SNAP_AFTER)) || \
(!forward && (flags & GST_SEEK_FLAG_SNAP_BEFORE)))
/**
/*
* Seeks this stream to the fragment that contains the sample at time
*
* @time: time in nanoseconds
......
......@@ -1188,7 +1188,7 @@ _check_if_negotiation_is_needed (GstWebRTCBin * webrtc)
/* FIXME: emit when input caps/format changes? */
/* If connection has created any RTCDataChannel's, and no m= section has
* been negotiated yet for data, return "true".
* been negotiated yet for data, return "true".
* FIXME */
if (!webrtc->current_local_description) {
......@@ -3674,7 +3674,7 @@ _set_description_task (GstWebRTCBin * webrtc, struct set_description *sd)
if (sd->sdp->type == GST_WEBRTC_SDP_TYPE_ROLLBACK) {
/* FIXME:
* If the mid value of an RTCRtpTransceiver was set to a non-null value
* If the mid value of an RTCRtpTransceiver was set to a non-null value
* by the RTCSessionDescription that is being rolled back, set the mid
* value of that transceiver to null, as described by [JSEP]
* (section 4.1.7.2.).
......@@ -5054,7 +5054,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/**
* GstWebRTCBin::create-offer:
* @object: the #GstWebRtcBin
* @object: the #webrtcbin
* @options: create-offer options
* @promise: a #GstPromise which will contain the offer
*/
......@@ -5067,7 +5067,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/**
* GstWebRTCBin::create-answer:
* @object: the #GstWebRtcBin
* @object: the #webrtcbin
* @options: create-answer options
* @promise: a #GstPromise which will contain the answer
*/
......@@ -5080,7 +5080,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/**
* GstWebRTCBin::set-local-description:
* @object: the #GstWebRtcBin
* @object: the #GstWebRTCBin
* @desc: a #GstWebRTCSessionDescription description
* @promise: (nullable): a #GstPromise to be notified when it's set
*/
......@@ -5093,7 +5093,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/**
* GstWebRTCBin::set-remote-description:
* @object: the #GstWebRtcBin
* @object: the #GstWebRTCBin
* @desc: a #GstWebRTCSessionDescription description
* @promise: (nullable): a #GstPromise to be notified when it's set
*/
......@@ -5106,7 +5106,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/**
* GstWebRTCBin::add-ice-candidate:
* @object: the #GstWebRtcBin
* @object: the #webrtcbin
* @mline_index: the index of the media description in the SDP
* @ice-candidate: an ice candidate
*/
......@@ -5118,7 +5118,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/**
* GstWebRTCBin::get-stats:
* @object: the #GstWebRtcBin
* @object: the #webrtcbin
* @pad: (nullable): A #GstPad to get the stats for, or %NULL for all
* @promise: a #GstPromise for the result
*
......@@ -5195,7 +5195,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/**
* GstWebRTCBin::on-negotiation-needed:
* @object: the #GstWebRtcBin
* @object: the #webrtcbin
*/
gst_webrtc_bin_signals[ON_NEGOTIATION_NEEDED_SIGNAL] =
g_signal_new ("on-negotiation-needed", G_TYPE_FROM_CLASS (klass),
......@@ -5204,7 +5204,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/**
* GstWebRTCBin::on-ice-candidate:
* @object: the #GstWebRtcBin
* @object: the #webrtcbin
* @mline_index: the index of the media description in the SDP
* @candidate: the ICE candidate
*/
......@@ -5215,7 +5215,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/**
* GstWebRTCBin::on-new-transceiver:
* @object: the #GstWebRtcBin
* @object: the #webrtcbin
* @candidate: the new #GstWebRTCRTPTransceiver
*/
gst_webrtc_bin_signals[ON_NEW_TRANSCEIVER_SIGNAL] =
......@@ -5225,8 +5225,8 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/**
* GstWebRTCBin::on-data-channel:
* @object: the #GstWebRtcBin
* @candidate: the new #GstWebRTCDataChannel
* @object: the #GstWebRTCBin
* @candidate: the new `GstWebRTCDataChannel`
*/
gst_webrtc_bin_signals[ON_DATA_CHANNEL_SIGNAL] =
g_signal_new ("on-data-channel", G_TYPE_FROM_CLASS (klass),
......@@ -5235,7 +5235,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/**
* GstWebRTCBin::add-transceiver:
* @object: the #GstWebRtcBin
* @object: the #webrtcbin
* @direction: the direction of the new transceiver
* @caps: (allow none): the codec preferences for this transceiver
*
......@@ -5250,7 +5250,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/**
* GstWebRTCBin::get-transceivers:
* @object: the #GstWebRtcBin
* @object: the #webrtcbin
*
* Returns: a #GArray of #GstWebRTCRTPTransceivers
*/
......@@ -5262,7 +5262,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/**
* GstWebRTCBin::get-transceiver:
* @object: the #GstWebRtcBin
* @object: the #GstWebRTCBin
* @idx: The index of the transceiver
*
* Returns: the #GstWebRTCRTPTransceiver, or %NULL
......@@ -5277,7 +5277,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/**
* GstWebRTCBin::add-turn-server:
* @object: the #GstWebRtcBin
* @object: the #GstWebRTCBin
* @uri: The uri of the server of the form turn(s)://username:password@host:port
*
* Add a turn server to obtain ICE candidates from
......@@ -5290,7 +5290,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/*
* GstWebRTCBin::create-data-channel:
* @object: the #GstWebRtcBin
* @object: the #GstWebRTCBin
* @label: the label for the data channel
* @options: a #GstStructure of options for creating the data channel
*
......
......@@ -941,7 +941,7 @@ gst_webrtc_ice_class_init (GstWebRTCICEClass * klass)
/**
* GstWebRTCICE::on-ice-candidate:
* @object: the #GstWebRtcBin
* @object: the #GstWebRTCBin
* @candidate: the ICE candidate
*/
gst_webrtc_ice_signals[ON_ICE_CANDIDATE_SIGNAL] =
......
......@@ -22,7 +22,6 @@
/**
* SECTION:gstadaptivedemux
* @short_description: Base class for adaptive demuxers
* @see_also:
*
* What is an adaptive demuxer?
* Adaptive demuxers are special demuxers in the sense that they don't
......
......@@ -658,7 +658,7 @@ struct _GstH264SPSExtMVCLevelValue
* level values signalled for the coded video sequence.
* @level_value: array of #GstH264SPSExtMVCLevelValue
*
* Represents the parsed seq_parameter_set_mvc_extension().
* Represents the parsed `seq_parameter_set_mvc_extension()`.
*
* Since: 1.6
*/
......
......@@ -39,7 +39,7 @@
*
* * From #GST_H265_NAL_SLICE_TRAIL_N to #GST_H265_NAL_SLICE_CRA_NUT: gst_h265_parser_parse_slice_hdr()
*
* * #GST_H265_NAL_SEI: gst_h265_parser_parse_sei()
* * `GST_H265_NAL_*_SEI`: gst_h265_parser_parse_sei()
*
* * #GST_H265_NAL_VPS: gst_h265_parser_parse_vps()
*
......@@ -2119,7 +2119,7 @@ gst_h265_parser_parse_pps (GstH265Parser * parser,
/**
* gst_h265_parser_parse_slice_hdr:
* @parser: a #GstH265Parser
* @nalu: The #GST_H265_NAL_SLICE #GstH265NalUnit to parse
* @nalu: The `GST_H265_NAL_SLICE` #GstH265NalUnit to parse
* @slice: The #GstH265SliceHdr to fill.
*
* Parses @data, and fills the @slice structure.
......@@ -2656,7 +2656,7 @@ gst_h265_sei_free (GstH265SEIMessage * sei)
/**
* gst_h265_parser_parse_sei:
* @nalparser: a #GstH265Parser
* @nalu: The #GST_H265_NAL_SEI #GstH265NalUnit to parse
* @nalu: The `GST_H265_NAL_*_SEI` #GstH265NalUnit to parse
* @messages: The GArray of #GstH265SEIMessage to fill. The caller must free it when done.
*
* Parses @data, create and fills the @messages array.
......
......@@ -27,7 +27,7 @@
* Boston, MA 02110-1301, USA.
*/
/**
/*
* Common code for NAL parsing from h264 and h265 parsers.
*/
......
......@@ -1988,7 +1988,7 @@ gst_mpegts_descriptor_parse_dvb_multilingual_component (const
* @private_data_specifier: (out): the private data specifier id
* registered by http://www.dvbservices.com/
* @private_data: (out) (transfer full) (allow-none) (array length=length): additional data or NULL
* @length: (out) (allow-none): length of %private_data
* @length: (out) (allow-none): length of @private_data
*
* Parses out the private data specifier from the @descriptor.
*
......@@ -2024,7 +2024,7 @@ gst_mpegts_descriptor_parse_dvb_private_data_specifier (const
* @descriptor: a %GST_MTS_DESC_DVB_FREQUENCY_LIST #GstMpegtsDescriptor
* @offset: (out): %FALSE in Hz, %TRUE in kHz
* @list: (out) (transfer full) (element-type guint32): a list of all frequencies in Hz/kHz
* depending on %offset
* depending on @offset
*
* Parses out a list of frequencies from the @descriptor.
*
......@@ -2195,7 +2195,7 @@ gst_mpegts_descriptor_parse_dvb_scrambling (const GstMpegtsDescriptor *
* @descriptor: a %GST_MTS_DESC_DVB_DATA_BROADCAST_ID #GstMpegtsDescriptor
* @data_broadcast_id: (out): the data broadcast id
* @id_selector_bytes: (out) (transfer full) (array length=len): the selector bytes, if present
* @len: (out): the length of #id_selector_bytes
* @len: (out): the length of @id_selector_bytes
*
* Parses out the data broadcast id from the @descriptor.
*
......
......@@ -464,8 +464,8 @@ struct _GstMpegtsDVBLinkageExtendedEvent
* @transport_stream_id: the transport id
* @original_network_id: the original network id
* @service_id: the service id
* @linkage_type: the type which %linkage_data has
* @private_data_length: the length for %private_data_bytes
* @linkage_type: the type which @linkage_data has
* @private_data_length: the length for @private_data_bytes
* @private_data_bytes: additional data bytes
*/
struct _GstMpegtsDVBLinkageDescriptor
......
......@@ -200,7 +200,7 @@ static void
* @application_context: (allow-none): GMainContext to use or %NULL
*
* Creates a new GstPlayerSignalDispatcher that uses @application_context,
* or the thread default one if %NULL is used. See gst_player_new_full().
* or the thread default one if %NULL is used. See gst_player_new().
*
* Returns: (transfer full): the new GstPlayerSignalDispatcher
*/
......
......@@ -784,7 +784,7 @@ gst_player_media_info_get_container_format (const GstPlayerMediaInfo * info)
* @info: a #GstPlayerMediaInfo
*
* Function to get the image (or preview-image) stored in taglist.
* Application can use gst_sample_*_() API's to get caps, buffer etc.
* Application can use `gst_sample_*_()` API's to get caps, buffer etc.
*
* Returns: (transfer none): GstSample or NULL.
*/
......
......@@ -69,6 +69,9 @@ GST_DEBUG_CATEGORY_STATIC (gst_player_debug);
#define DEFAULT_AUDIO_VIDEO_OFFSET 0
#define DEFAULT_SUBTITLE_VIDEO_OFFSET 0
/**
* gst_player_error_quark:
*/
GQuark
gst_player_error_quark (void)
{
......
......@@ -35,6 +35,9 @@ GType gst_webrtc_dtls_transport_get_type(void);
#define GST_IS_WEBRTC_DTLS_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_DTLS_TRANSPORT))
#define GST_WEBRTC_DTLS_TRANSPORT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_DTLS_TRANSPORT,GstWebRTCDTLSTransportClass))
/**
* GstWebRTCDTLSTransport:
*/
struct _GstWebRTCDTLSTransport
{
GstObject parent;
......
......@@ -34,6 +34,9 @@ GType gst_webrtc_ice_transport_get_type(void);
#define GST_IS_WEBRTC_ICE_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_ICE_TRANSPORT))
#define GST_WEBRTC_ICE_TRANSPORT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_ICE_TRANSPORT,GstWebRTCICETransportClass))
/**
* GstWebRTCICETransport:
*/
struct _GstWebRTCICETransport
{
GstObject parent;
......
......@@ -35,6 +35,9 @@ GType gst_webrtc_rtp_receiver_get_type(void);
#define GST_IS_WEBRTC_RTP_RECEIVER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_RTP_RECEIVER))
#define GST_WEBRTC_RTP_RECEIVER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_RTP_RECEIVER,GstWebRTCRTPReceiverClass))
/**
* GstWebRTCRTPReceiver:
*/
struct _GstWebRTCRTPReceiver
{
GstObject parent;
......
......@@ -35,6 +35,9 @@ GType gst_webrtc_rtp_sender_get_type(void);
#define GST_IS_WEBRTC_RTP_SENDER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_RTP_SENDER))
#define GST_WEBRTC_RTP_SENDER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_RTP_SENDER,GstWebRTCRTPSenderClass))
/**
* GstWebRTCRTPSender:
*/
struct _GstWebRTCRTPSender
{
GstObject parent;
......
......@@ -36,6 +36,9 @@ GType gst_webrtc_rtp_transceiver_get_type(void);
#define GST_IS_WEBRTC_RTP_TRANSCEIVER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_RTP_TRANSCEIVER))
#define GST_WEBRTC_RTP_TRANSCEIVER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_RTP_TRANSCEIVER,GstWebRTCRTPTransceiverClass))
/**
* GstWebRTCRTPTransceiver:
*/
struct _GstWebRTCRTPTransceiver
{
GstObject parent;
......
......@@ -27,6 +27,11 @@
#include <gst/gst.h>
/**
* SECTION:webrtc_fwd.h
* @title: GstWebRTC Enumerations
*/
#ifndef GST_WEBRTC_API
# ifdef BUILDING_GST_WEBRTC
# define GST_WEBRTC_API GST_API_EXPORT /* from config.h */
......@@ -56,11 +61,11 @@ typedef struct _GstWebRTCRTPTransceiverClass GstWebRTCRTPTransceiverClass;
/**
* GstWebRTCDTLSTransportState:
* GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new
* GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: closed
* GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED: failed
* GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: connecting
* GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected
* @GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new
* @GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: closed
* @GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED: failed
* @GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: connecting
* @GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected
*/
typedef enum /*< underscore_name=gst_webrtc_dtls_transport_state >*/
{
......@@ -73,9 +78,9 @@ typedef enum /*< underscore_name=gst_webrtc_dtls_transport_state >*/
/**
* GstWebRTCICEGatheringState:
* GST_WEBRTC_ICE_GATHERING_STATE_NEW: new
* GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering
* GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete
* @GST_WEBRTC_ICE_GATHERING_STATE_NEW: new
* @GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering
* @GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete
*
* See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate">http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate</ulink>
*/
......@@ -88,13 +93,13 @@ typedef enum /*< underscore_name=gst_webrtc_ice_gathering_state >*/
/**
* GstWebRTCICEConnectionState:
* GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new
* GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: checking
* GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED: connected
* GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED: completed
* GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: failed
* GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: disconnected
* GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: closed
* @GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new
* @GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: checking
* @GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED: connected
* @GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED: completed
* @GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: failed
* @GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: disconnected
* @GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: closed
*
* See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate</ulink>
*/
......@@ -111,12 +116,12 @@ typedef enum /*< underscore_name=gst_webrtc_ice_connection_state >*/
/**
* GstWebRTCSignalingState:
* GST_WEBRTC_SIGNALING_STATE_STABLE: stable
* GST_WEBRTC_SIGNALING_STATE_CLOSED: closed
* GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER: have-local-offer
* GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER: have-remote-offer
* GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: have-local-pranswer
* GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER: have-remote-pranswer
* @GST_WEBRTC_SIGNALING_STATE_STABLE: stable
* @GST_WEBRTC_SIGNALING_STATE_CLOSED: closed
* @GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER: have-local-offer
* @GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER: have-remote-offer
* @GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: have-local-pranswer
* @GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER: have-remote-pranswer
*
* See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate">http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate</ulink>
*/
......@@ -132,12 +137,12 @@ typedef enum /*< underscore_name=gst_webrtc_signaling_state >*/
/**
* GstWebRTCPeerConnectionState:
* GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new
* GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: connecting
* GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED: connected
* GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED: disconnected
* GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: failed
* GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: closed
* @GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new
* @GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: connecting
* @GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED: connected
* @GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED: disconnected
* @GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: failed
* @GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: closed
*
* See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate</ulink>
*/
......@@ -153,8 +158,8 @@ typedef enum /*< underscore_name=gst_webrtc_peer_connection_state >*/
/**
* GstWebRTCICERole:
* GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled
* GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling
* @GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled
* @GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling
*/
typedef enum /*< underscore_name=gst_webrtc_ice_role >*/
{
......@@ -164,8 +169,8 @@ typedef enum /*< underscore_name=gst_webrtc_ice_role >*/
/**
* GstWebRTCICEComponent:
* GST_WEBRTC_ICE_COMPONENT_RTP,
* GST_WEBRTC_ICE_COMPONENT_RTCP,
* @GST_WEBRTC_ICE_COMPONENT_RTP: RTP component
* @GST_WEBRTC_ICE_COMPONENT_RTCP: RTCP component
*/
typedef enum /*< underscore_name=gst_webrtc_ice_component >*/
{
......@@ -175,10 +180,10 @@ typedef enum /*< underscore_name=gst_webrtc_ice_component >*/
/**
* GstWebRTCSDPType:
* GST_WEBRTC_SDP_TYPE_OFFER: offer
* GST_WEBRTC_SDP_TYPE_PRANSWER: pranswer
* GST_WEBRTC_SDP_TYPE_ANSWER: answer
* GST_WEBRTC_SDP_TYPE_ROLLBACK: rollback
* @GST_WEBRTC_SDP_TYPE_OFFER: offer
* @GST_WEBRTC_SDP_TYPE_PRANSWER: pranswer
* @GST_WEBRTC_SDP_TYPE_ANSWER: answer
* @GST_WEBRTC_SDP_TYPE_ROLLBACK: rollback
*
* See <ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype">http://w3c.github.io/webrtc-pc/#rtcsdptype</ulink>
*/
......@@ -191,12 +196,12 @@ typedef enum /*< underscore_name=gst_webrtc_sdp_type >*/
} GstWebRTCSDPType;
/**
* GstWebRTCRtpTransceiverDirection:
* GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE: none
* GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE: inactive
* GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY: sendonly
* GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY: recvonly
* GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV: sendrecv
* GstWebRTCRTPTransceiverDirection:
* @GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE: none
* @GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE: inactive
* @GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY: sendonly
* @GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY: recvonly
* @GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV: sendrecv
*/
typedef enum /*< underscore_name=gst_webrtc_rtp_transceiver_direction >*/
{
......@@ -209,10 +214,10 @@ typedef enum /*< underscore_name=gst_webrtc_rtp_transceiver_direction >*/
/**
* GstWebRTCDTLSSetup:
* GST_WEBRTC_DTLS_SETUP_NONE: none
* GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass
* GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly
* GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly
* @GST_WEBRTC_DTLS_SETUP_NONE: none
* @GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass
* @GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly
* @GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly
*/
typedef enum /*< underscore_name=gst_webrtc_dtls_setup >*/
{
......@@ -224,20 +229,20 @@ typedef enum /*< underscore_name=gst_webrtc_dtls_setup >*/
/**
* GstWebRTCStatsType:
* GST_WEBRTC_STATS_CODEC: codec
* GST_WEBRTC_STATS_INBOUND_RTP: inbound-rtp
* GST_WEBRTC_STATS_OUTBOUND_RTP: outbound-rtp
* GST_WEBRTC_STATS_REMOTE_INBOUND_RTP: remote-inbound-rtp
* GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP: remote-outbound-rtp
* GST_WEBRTC_STATS_CSRC: csrc
* GST_WEBRTC_STATS_PEER_CONNECTION: peer-connectiion
* GST_WEBRTC_STATS_DATA_CHANNEL: data-channel
* GST_WEBRTC_STATS_STREAM: stream
* GST_WEBRTC_STATS_TRANSPORT: transport
* GST_WEBRTC_STATS_CANDIDATE_PAIR: candidate-pair
* GST_WEBRTC_STATS_LOCAL_CANDIDATE: local-candidate
* GST_WEBRTC_STATS_REMOTE_CANDIDATE: remote-candidate
* GST_WEBRTC_STATS_CERTIFICATE: certificate
* @GST_WEBRTC_STATS_CODEC: codec
* @GST_WEBRTC_STATS_INBOUND_RTP: inbound-rtp
* @GST_WEBRTC_STATS_OUTBOUND_RTP: outbound-rtp
* @GST_WEBRTC_STATS_REMOTE_INBOUND_RTP: remote-inbound-rtp
* @GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP: remote-outbound-rtp
* @GST_WEBRTC_STATS_CSRC: csrc
* @GST_WEBRTC_STATS_PEER_CONNECTION: peer-connectiion
* @GST_WEBRTC_STATS_DATA_CHANNEL: data-channel
* @GST_WEBRTC_STATS_STREAM: stream
* @GST_WEBRTC_STATS_TRANSPORT: transport
* @GST_WEBRTC_STATS_CANDIDATE_PAIR: candidate-pair
* @GST_WEBRTC_STATS_LOCAL_CANDIDATE: local-candidate
* @GST_WEBRTC_STATS_REMOTE_CANDIDATE: remote-candidate
* @GST_WEBRTC_STATS_CERTIFICATE: certificate
*/
typedef enum /*< underscore_name=gst_webrtc_stats_type >*/
{
......@@ -259,8 +264,8 @@ typedef enum /*< underscore_name=gst_webrtc_stats_type >*/
/**
* GstWebRTCFECType:
* GST_WEBRTC_FEC_TYPE_NONE: none
* GST_WEBRTC_FEC_TYPE_ULP_RED: ulpfec + red
* @GST_WEBRTC_FEC_TYPE_NONE: none
* @GST_WEBRTC_FEC_TYPE_ULP_RED: ulpfec + red
*/
typedef enum /*< underscore_name=gst_webrtc_fec_type >*/
{
......
......@@ -179,7 +179,7 @@ gst_gm_mod_float (gdouble a, gdouble b)
return a;
}
/**
/*
* Returns a repeating triangle shape in the range 0..1 with wavelength 1.0
*/
gdouble
......@@ -190,7 +190,7 @@ gst_gm_triangle (gdouble x)
return 2.0 * (r < 0.5 ? r : 1 - r);
}
/**
/*
* Hermite interpolation
*/
gdouble
......
......@@ -263,8 +263,7 @@ sp_writer_create (const char *path, size_t size, mode_t perms)
return NULL; \
} while (0)
/**
* sp_open_shm:
/* sp_open_shm:
* @path: Path of the shm area for a reader,
* NULL if this is a writer (then it will allocate its own path)
*
......@@ -857,8 +856,7 @@ sp_shmbuf_dec (ShmPipe * self, ShmBuffer * buf, ShmBuffer * prev_buf,
int i;
int had_client = 0;
/**
* Remove client from the list of buffer users. Here we make sure that
/* Remove client from the list of buffer users. Here we make sure that
* if a client closes connection but already decremented the use count
* for this buffer, but other clients didn't have time to decrement
* buffer will not be freed too early in sp_writer_close_client.
......
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