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Minor documentation fixes

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This commit is contained in:
Thibault Saunier 2018-08-10 20:32:30 -04:00
parent dce17521eb
commit 7fe3f36ac8
29 changed files with 127 additions and 115 deletions
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10
ext/closedcaption/io-sim.c
View file

@ -47,7 +47,7 @@
#include "io-sim.h" #include "io-sim.h"
/** /*
* @addtogroup Rawenc Raw VBI encoder * @addtogroup Rawenc Raw VBI encoder
* @ingroup Raw * @ingroup Raw
* @brief Converting sliced VBI data to raw VBI images. * @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 raw Noise will be added to this raw VBI image.
* @param sp Describes the raw VBI data in the buffer. @a sp->sampling_format * @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). * 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; return TRUE;
} }
/** /*
* @example examples/rawout.c * @example examples/rawout.c
* Raw VBI output example. * Raw VBI output example.
*/ */
/** /*
* @param raw A raw VBI image will be stored here. * @param raw A raw VBI image will be stored here.
* @param raw_size Size of the @a raw buffer in bytes. The buffer * @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 * 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); swap_fields ? _VBI_RAW_SWAP_FIELDS : 0, sliced, n_sliced_lines);
} }
/** /*
* @param raw A raw VBI image will be stored here. * @param raw A raw VBI image will be stored here.
* @param raw_size Size of the @a raw buffer in bytes. The buffer * @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 * must be large enough for @a sp->count[0] + count[1] lines

1
ext/curl/gstcurlbasesink.c
View file

@ -21,7 +21,6 @@
* SECTION:element-curlsink * SECTION:element-curlsink
* @title: curlsink * @title: curlsink
* @short_description: sink that uploads data to a server using libcurl * @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 * This is a network sink that uses libcurl as a client to upload data to
* a server (e.g. a HTTP/FTP server). * a server (e.g. a HTTP/FTP server).

1
ext/curl/gstcurlfilesink.c
View file

@ -21,7 +21,6 @@
* SECTION:element-curlfilesink * SECTION:element-curlfilesink
* @title: curlfilesink * @title: curlfilesink
* @short_description: sink that uploads data to a server using libcurl * @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 * This is a network sink that uses libcurl as a client to upload data to
* a local or network drive. * a local or network drive.

1
ext/curl/gstcurlftpsink.c
View file

@ -21,7 +21,6 @@
* SECTION:element-curlftpsink * SECTION:element-curlftpsink
* @title: curlftpsink * @title: curlftpsink
* @short_description: sink that uploads data to a server using libcurl * @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 * This is a network sink that uses libcurl as a client to upload data to
* an FTP server. * an FTP server.

1
ext/curl/gstcurlhttpsink.c
View file

@ -21,7 +21,6 @@
* SECTION:element-curlhttpsink * SECTION:element-curlhttpsink
* @title: curlhttpsink * @title: curlhttpsink
* @short_description: sink that uploads data to a server using libcurl * @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 * This is a network sink that uses libcurl as a client to upload data to
* an HTTP server. * an HTTP server.

1
ext/curl/gstcurlsftpsink.c
View file

@ -21,7 +21,6 @@
* SECTION:element-curlsftpsink * SECTION:element-curlsftpsink
* @title: curlsftpsink * @title: curlsftpsink
* @short_description: sink that uploads data to a server using libcurl * @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 * This is a network sink that uses libcurl as a client to upload data to
* a SFTP (SSH File Transfer Protocol) server. * a SFTP (SSH File Transfer Protocol) server.

1
ext/curl/gstcurlsmtpsink.c
View file

@ -21,7 +21,6 @@
* SECTION:element-curlsink * SECTION:element-curlsink
* @title: curlsink * @title: curlsink
* @short_description: sink that uploads data to a server using libcurl * @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 * This is a network sink that uses libcurl as a client to upload data to
* an SMTP server. * an SMTP server.

1
ext/curl/gstcurlsshsink.c
View file

@ -21,7 +21,6 @@
* SECTION:element-curlsshsink * SECTION:element-curlsshsink
* @title: curlsshsink * @title: curlsshsink
* @short_description: sink that uploads data to a server using libcurl * @short_description: sink that uploads data to a server using libcurl
* @see_also:
* *
* This is a network sink that uses libcurl. * This is a network sink that uses libcurl.
* *

1
ext/curl/gstcurltlssink.c
View file

@ -21,7 +21,6 @@
* SECTION:element-curltlssink * SECTION:element-curltlssink
* @title: curltlssink * @title: curltlssink
* @short_description: sink that uploads data to a server using libcurl * @short_description: sink that uploads data to a server using libcurl
* @see_also:
* *
* This is a network sink that uses libcurl. * This is a network sink that uses libcurl.
* *

6
ext/smoothstreaming/gstmssmanifest.c
View file

@ -975,7 +975,7 @@ gst_mss_manifest_get_duration (GstMssManifest * manifest)
} }
/** /*
* Gets the duration in nanoseconds * Gets the duration in nanoseconds
*/ */
GstClockTime GstClockTime
@ -1194,7 +1194,7 @@ gst_mss_stream_type_name (GstMssStreamType streamtype)
} }
} }
/** /*
* Seeks all streams to the fragment that contains the set time * Seeks all streams to the fragment that contains the set time
* *
* @forward: if this is forward playback * @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_AFTER)) || \
(!forward && (flags & GST_SEEK_FLAG_SNAP_BEFORE))) (!forward && (flags & GST_SEEK_FLAG_SNAP_BEFORE)))
/** /*
* Seeks this stream to the fragment that contains the sample at time * Seeks this stream to the fragment that contains the sample at time
* *
* @time: time in nanoseconds * @time: time in nanoseconds

32
ext/webrtc/gstwebrtcbin.c
View file

@ -5054,7 +5054,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/** /**
* GstWebRTCBin::create-offer: * GstWebRTCBin::create-offer:
* @object: the #GstWebRtcBin * @object: the #webrtcbin
* @options: create-offer options * @options: create-offer options
* @promise: a #GstPromise which will contain the offer * @promise: a #GstPromise which will contain the offer
*/ */
@ -5067,7 +5067,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/** /**
* GstWebRTCBin::create-answer: * GstWebRTCBin::create-answer:
* @object: the #GstWebRtcBin * @object: the #webrtcbin
* @options: create-answer options * @options: create-answer options
* @promise: a #GstPromise which will contain the answer * @promise: a #GstPromise which will contain the answer
*/ */
@ -5080,7 +5080,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/** /**
* GstWebRTCBin::set-local-description: * GstWebRTCBin::set-local-description:
* @object: the #GstWebRtcBin * @object: the #GstWebRTCBin
* @desc: a #GstWebRTCSessionDescription description * @desc: a #GstWebRTCSessionDescription description
* @promise: (nullable): a #GstPromise to be notified when it's set * @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: * GstWebRTCBin::set-remote-description:
* @object: the #GstWebRtcBin * @object: the #GstWebRTCBin
* @desc: a #GstWebRTCSessionDescription description * @desc: a #GstWebRTCSessionDescription description
* @promise: (nullable): a #GstPromise to be notified when it's set * @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: * GstWebRTCBin::add-ice-candidate:
* @object: the #GstWebRtcBin * @object: the #webrtcbin
* @mline_index: the index of the media description in the SDP * @mline_index: the index of the media description in the SDP
* @ice-candidate: an ice candidate * @ice-candidate: an ice candidate
*/ */
@ -5118,7 +5118,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/** /**
* GstWebRTCBin::get-stats: * GstWebRTCBin::get-stats:
* @object: the #GstWebRtcBin * @object: the #webrtcbin
* @pad: (nullable): A #GstPad to get the stats for, or %NULL for all * @pad: (nullable): A #GstPad to get the stats for, or %NULL for all
* @promise: a #GstPromise for the result * @promise: a #GstPromise for the result
* *
@ -5195,7 +5195,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/** /**
* GstWebRTCBin::on-negotiation-needed: * GstWebRTCBin::on-negotiation-needed:
* @object: the #GstWebRtcBin * @object: the #webrtcbin
*/ */
gst_webrtc_bin_signals[ON_NEGOTIATION_NEEDED_SIGNAL] = gst_webrtc_bin_signals[ON_NEGOTIATION_NEEDED_SIGNAL] =
g_signal_new ("on-negotiation-needed", G_TYPE_FROM_CLASS (klass), 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: * GstWebRTCBin::on-ice-candidate:
* @object: the #GstWebRtcBin * @object: the #webrtcbin
* @mline_index: the index of the media description in the SDP * @mline_index: the index of the media description in the SDP
* @candidate: the ICE candidate * @candidate: the ICE candidate
*/ */
@ -5215,7 +5215,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/** /**
* GstWebRTCBin::on-new-transceiver: * GstWebRTCBin::on-new-transceiver:
* @object: the #GstWebRtcBin * @object: the #webrtcbin
* @candidate: the new #GstWebRTCRTPTransceiver * @candidate: the new #GstWebRTCRTPTransceiver
*/ */
gst_webrtc_bin_signals[ON_NEW_TRANSCEIVER_SIGNAL] = gst_webrtc_bin_signals[ON_NEW_TRANSCEIVER_SIGNAL] =
@ -5225,8 +5225,8 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/** /**
* GstWebRTCBin::on-data-channel: * GstWebRTCBin::on-data-channel:
* @object: the #GstWebRtcBin * @object: the #GstWebRTCBin
* @candidate: the new #GstWebRTCDataChannel * @candidate: the new `GstWebRTCDataChannel`
*/ */
gst_webrtc_bin_signals[ON_DATA_CHANNEL_SIGNAL] = gst_webrtc_bin_signals[ON_DATA_CHANNEL_SIGNAL] =
g_signal_new ("on-data-channel", G_TYPE_FROM_CLASS (klass), 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: * GstWebRTCBin::add-transceiver:
* @object: the #GstWebRtcBin * @object: the #webrtcbin
* @direction: the direction of the new transceiver * @direction: the direction of the new transceiver
* @caps: (allow none): the codec preferences for this transceiver * @caps: (allow none): the codec preferences for this transceiver
* *
@ -5250,7 +5250,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/** /**
* GstWebRTCBin::get-transceivers: * GstWebRTCBin::get-transceivers:
* @object: the #GstWebRtcBin * @object: the #webrtcbin
* *
* Returns: a #GArray of #GstWebRTCRTPTransceivers * Returns: a #GArray of #GstWebRTCRTPTransceivers
*/ */
@ -5262,7 +5262,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/** /**
* GstWebRTCBin::get-transceiver: * GstWebRTCBin::get-transceiver:
* @object: the #GstWebRtcBin * @object: the #GstWebRTCBin
* @idx: The index of the transceiver * @idx: The index of the transceiver
* *
* Returns: the #GstWebRTCRTPTransceiver, or %NULL * Returns: the #GstWebRTCRTPTransceiver, or %NULL
@ -5277,7 +5277,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/** /**
* GstWebRTCBin::add-turn-server: * 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 * @uri: The uri of the server of the form turn(s)://username:password@host:port
* *
* Add a turn server to obtain ICE candidates from * Add a turn server to obtain ICE candidates from
@ -5290,7 +5290,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
/* /*
* GstWebRTCBin::create-data-channel: * GstWebRTCBin::create-data-channel:
* @object: the #GstWebRtcBin * @object: the #GstWebRTCBin
* @label: the label for the data channel * @label: the label for the data channel
* @options: a #GstStructure of options for creating the data channel * @options: a #GstStructure of options for creating the data channel
* *

2
ext/webrtc/gstwebrtcice.c
View file

@ -941,7 +941,7 @@ gst_webrtc_ice_class_init (GstWebRTCICEClass * klass)
/** /**
* GstWebRTCICE::on-ice-candidate: * GstWebRTCICE::on-ice-candidate:
* @object: the #GstWebRtcBin * @object: the #GstWebRTCBin
* @candidate: the ICE candidate * @candidate: the ICE candidate
*/ */
gst_webrtc_ice_signals[ON_ICE_CANDIDATE_SIGNAL] = gst_webrtc_ice_signals[ON_ICE_CANDIDATE_SIGNAL] =

1
gst-libs/gst/adaptivedemux/gstadaptivedemux.c
View file

@ -22,7 +22,6 @@
/** /**
* SECTION:gstadaptivedemux * SECTION:gstadaptivedemux
* @short_description: Base class for adaptive demuxers * @short_description: Base class for adaptive demuxers
* @see_also:
* *
* What is an adaptive demuxer? * What is an adaptive demuxer?
* Adaptive demuxers are special demuxers in the sense that they don't * Adaptive demuxers are special demuxers in the sense that they don't

2
gst-libs/gst/codecparsers/gsth264parser.h
View file

@ -658,7 +658,7 @@ struct _GstH264SPSExtMVCLevelValue
* level values signalled for the coded video sequence. * level values signalled for the coded video sequence.
* @level_value: array of #GstH264SPSExtMVCLevelValue * @level_value: array of #GstH264SPSExtMVCLevelValue
* *
* Represents the parsed seq_parameter_set_mvc_extension(). * Represents the parsed `seq_parameter_set_mvc_extension()`.
* *
* Since: 1.6 * Since: 1.6
*/ */

6
gst-libs/gst/codecparsers/gsth265parser.c
View file

@ -39,7 +39,7 @@
* *
* * From #GST_H265_NAL_SLICE_TRAIL_N to #GST_H265_NAL_SLICE_CRA_NUT: gst_h265_parser_parse_slice_hdr() * * 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() * * #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: * gst_h265_parser_parse_slice_hdr:
* @parser: a #GstH265Parser * @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. * @slice: The #GstH265SliceHdr to fill.
* *
* Parses @data, and fills the @slice structure. * Parses @data, and fills the @slice structure.
@ -2656,7 +2656,7 @@ gst_h265_sei_free (GstH265SEIMessage * sei)
/** /**
* gst_h265_parser_parse_sei: * gst_h265_parser_parse_sei:
* @nalparser: a #GstH265Parser * @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. * @messages: The GArray of #GstH265SEIMessage to fill. The caller must free it when done.
* *
* Parses @data, create and fills the @messages array. * Parses @data, create and fills the @messages array.

2
gst-libs/gst/codecparsers/nalutils.c
View file

@ -27,7 +27,7 @@
* Boston, MA 02110-1301, USA. * Boston, MA 02110-1301, USA.
*/ */
/** /*
* Common code for NAL parsing from h264 and h265 parsers. * Common code for NAL parsing from h264 and h265 parsers.
*/ */

6
gst-libs/gst/mpegts/gst-dvb-descriptor.c
View file

@ -1988,7 +1988,7 @@ gst_mpegts_descriptor_parse_dvb_multilingual_component (const
* @private_data_specifier: (out): the private data specifier id * @private_data_specifier: (out): the private data specifier id
* registered by http://www.dvbservices.com/ * registered by http://www.dvbservices.com/
* @private_data: (out) (transfer full) (allow-none) (array length=length): additional data or NULL * @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. * 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 * @descriptor: a %GST_MTS_DESC_DVB_FREQUENCY_LIST #GstMpegtsDescriptor
* @offset: (out): %FALSE in Hz, %TRUE in kHz * @offset: (out): %FALSE in Hz, %TRUE in kHz
* @list: (out) (transfer full) (element-type guint32): a list of all frequencies in Hz/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. * 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 * @descriptor: a %GST_MTS_DESC_DVB_DATA_BROADCAST_ID #GstMpegtsDescriptor
* @data_broadcast_id: (out): the data broadcast id * @data_broadcast_id: (out): the data broadcast id
* @id_selector_bytes: (out) (transfer full) (array length=len): the selector bytes, if present * @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. * Parses out the data broadcast id from the @descriptor.
* *

4
gst-libs/gst/mpegts/gst-dvb-descriptor.h
View file

@ -464,8 +464,8 @@ struct _GstMpegtsDVBLinkageExtendedEvent
* @transport_stream_id: the transport id * @transport_stream_id: the transport id
* @original_network_id: the original network id * @original_network_id: the original network id
* @service_id: the service id * @service_id: the service id
* @linkage_type: the type which %linkage_data has * @linkage_type: the type which @linkage_data has
* @private_data_length: the length for %private_data_bytes * @private_data_length: the length for @private_data_bytes
* @private_data_bytes: additional data bytes * @private_data_bytes: additional data bytes
*/ */
struct _GstMpegtsDVBLinkageDescriptor struct _GstMpegtsDVBLinkageDescriptor

2
gst-libs/gst/player/gstplayer-g-main-context-signal-dispatcher.c
View file

@ -200,7 +200,7 @@ static void
* @application_context: (allow-none): GMainContext to use or %NULL * @application_context: (allow-none): GMainContext to use or %NULL
* *
* Creates a new GstPlayerSignalDispatcher that uses @application_context, * 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 * Returns: (transfer full): the new GstPlayerSignalDispatcher
*/ */

2
gst-libs/gst/player/gstplayer-media-info.c
View file

@ -784,7 +784,7 @@ gst_player_media_info_get_container_format (const GstPlayerMediaInfo * info)
* @info: a #GstPlayerMediaInfo * @info: a #GstPlayerMediaInfo
* *
* Function to get the image (or preview-image) stored in taglist. * 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. * Returns: (transfer none): GstSample or NULL.
*/ */

3
gst-libs/gst/player/gstplayer.c
View file

@ -69,6 +69,9 @@ GST_DEBUG_CATEGORY_STATIC (gst_player_debug);
#define DEFAULT_AUDIO_VIDEO_OFFSET 0 #define DEFAULT_AUDIO_VIDEO_OFFSET 0
#define DEFAULT_SUBTITLE_VIDEO_OFFSET 0 #define DEFAULT_SUBTITLE_VIDEO_OFFSET 0
/**
* gst_player_error_quark:
*/
GQuark GQuark
gst_player_error_quark (void) gst_player_error_quark (void)
{ {

3
gst-libs/gst/webrtc/dtlstransport.h
View file

@ -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_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)) #define GST_WEBRTC_DTLS_TRANSPORT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_DTLS_TRANSPORT,GstWebRTCDTLSTransportClass))
/**
* GstWebRTCDTLSTransport:
*/
struct _GstWebRTCDTLSTransport struct _GstWebRTCDTLSTransport
{ {
GstObject parent; GstObject parent;

3
gst-libs/gst/webrtc/icetransport.h
View file

@ -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_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)) #define GST_WEBRTC_ICE_TRANSPORT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_ICE_TRANSPORT,GstWebRTCICETransportClass))
/**
* GstWebRTCICETransport:
*/
struct _GstWebRTCICETransport struct _GstWebRTCICETransport
{ {
GstObject parent; GstObject parent;

3
gst-libs/gst/webrtc/rtpreceiver.h
View file

@ -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_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)) #define GST_WEBRTC_RTP_RECEIVER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_RTP_RECEIVER,GstWebRTCRTPReceiverClass))
/**
* GstWebRTCRTPReceiver:
*/
struct _GstWebRTCRTPReceiver struct _GstWebRTCRTPReceiver
{ {
GstObject parent; GstObject parent;

3
gst-libs/gst/webrtc/rtpsender.h
View file

@ -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_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)) #define GST_WEBRTC_RTP_SENDER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_RTP_SENDER,GstWebRTCRTPSenderClass))
/**
* GstWebRTCRTPSender:
*/
struct _GstWebRTCRTPSender struct _GstWebRTCRTPSender
{ {
GstObject parent; GstObject parent;

3
gst-libs/gst/webrtc/rtptransceiver.h
View file

@ -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_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)) #define GST_WEBRTC_RTP_TRANSCEIVER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_RTP_TRANSCEIVER,GstWebRTCRTPTransceiverClass))
/**
* GstWebRTCRTPTransceiver:
*/
struct _GstWebRTCRTPTransceiver struct _GstWebRTCRTPTransceiver
{ {
GstObject parent; GstObject parent;

127
gst-libs/gst/webrtc/webrtc_fwd.h
View file

@ -27,6 +27,11 @@
#include <gst/gst.h> #include <gst/gst.h>
/**
* SECTION:webrtc_fwd.h
* @title: GstWebRTC Enumerations
*/
#ifndef GST_WEBRTC_API #ifndef GST_WEBRTC_API
# ifdef BUILDING_GST_WEBRTC # ifdef BUILDING_GST_WEBRTC
# define GST_WEBRTC_API GST_API_EXPORT /* from config.h */ # define GST_WEBRTC_API GST_API_EXPORT /* from config.h */
@ -56,11 +61,11 @@ typedef struct _GstWebRTCRTPTransceiverClass GstWebRTCRTPTransceiverClass;
/** /**
* GstWebRTCDTLSTransportState: * GstWebRTCDTLSTransportState:
* GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new * @GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new
* GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: closed * @GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: closed
* GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED: failed * @GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED: failed
* GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: connecting * @GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: connecting
* GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected * @GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected
*/ */
typedef enum /*< underscore_name=gst_webrtc_dtls_transport_state >*/ typedef enum /*< underscore_name=gst_webrtc_dtls_transport_state >*/
{ {
@ -73,9 +78,9 @@ typedef enum /*< underscore_name=gst_webrtc_dtls_transport_state >*/
/** /**
* GstWebRTCICEGatheringState: * GstWebRTCICEGatheringState:
* GST_WEBRTC_ICE_GATHERING_STATE_NEW: new * @GST_WEBRTC_ICE_GATHERING_STATE_NEW: new
* GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering * @GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering
* GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete * @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> * 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: * GstWebRTCICEConnectionState:
* GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new * @GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new
* GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: checking * @GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: checking
* GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED: connected * @GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED: connected
* GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED: completed * @GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED: completed
* GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: failed * @GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: failed
* GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: disconnected * @GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: disconnected
* GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: closed * @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> * 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: * GstWebRTCSignalingState:
* GST_WEBRTC_SIGNALING_STATE_STABLE: stable * @GST_WEBRTC_SIGNALING_STATE_STABLE: stable
* GST_WEBRTC_SIGNALING_STATE_CLOSED: closed * @GST_WEBRTC_SIGNALING_STATE_CLOSED: closed
* GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER: have-local-offer * @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_REMOTE_OFFER: have-remote-offer
* GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: have-local-pranswer * @GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: have-local-pranswer
* GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER: have-remote-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> * 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: * GstWebRTCPeerConnectionState:
* GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new * @GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new
* GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: connecting * @GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: connecting
* GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED: connected * @GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED: connected
* GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED: disconnected * @GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED: disconnected
* GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: failed * @GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: failed
* GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: closed * @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> * 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: * GstWebRTCICERole:
* GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled * @GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled
* GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling * @GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling
*/ */
typedef enum /*< underscore_name=gst_webrtc_ice_role >*/ typedef enum /*< underscore_name=gst_webrtc_ice_role >*/
{ {
@ -164,8 +169,8 @@ typedef enum /*< underscore_name=gst_webrtc_ice_role >*/
/** /**
* GstWebRTCICEComponent: * GstWebRTCICEComponent:
* GST_WEBRTC_ICE_COMPONENT_RTP, * @GST_WEBRTC_ICE_COMPONENT_RTP: RTP component
* GST_WEBRTC_ICE_COMPONENT_RTCP, * @GST_WEBRTC_ICE_COMPONENT_RTCP: RTCP component
*/ */
typedef enum /*< underscore_name=gst_webrtc_ice_component >*/ typedef enum /*< underscore_name=gst_webrtc_ice_component >*/
{ {
@ -175,10 +180,10 @@ typedef enum /*< underscore_name=gst_webrtc_ice_component >*/
/** /**
* GstWebRTCSDPType: * GstWebRTCSDPType:
* GST_WEBRTC_SDP_TYPE_OFFER: offer * @GST_WEBRTC_SDP_TYPE_OFFER: offer
* GST_WEBRTC_SDP_TYPE_PRANSWER: pranswer * @GST_WEBRTC_SDP_TYPE_PRANSWER: pranswer
* GST_WEBRTC_SDP_TYPE_ANSWER: answer * @GST_WEBRTC_SDP_TYPE_ANSWER: answer
* GST_WEBRTC_SDP_TYPE_ROLLBACK: rollback * @GST_WEBRTC_SDP_TYPE_ROLLBACK: rollback
* *
* See <ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype">http://w3c.github.io/webrtc-pc/#rtcsdptype</ulink> * 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; } GstWebRTCSDPType;
/** /**
* GstWebRTCRtpTransceiverDirection: * GstWebRTCRTPTransceiverDirection:
* GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE: none * @GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE: none
* GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE: inactive * @GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE: inactive
* GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY: sendonly * @GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY: sendonly
* GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY: recvonly * @GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY: recvonly
* GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV: sendrecv * @GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV: sendrecv
*/ */
typedef enum /*< underscore_name=gst_webrtc_rtp_transceiver_direction >*/ typedef enum /*< underscore_name=gst_webrtc_rtp_transceiver_direction >*/
{ {
@ -209,10 +214,10 @@ typedef enum /*< underscore_name=gst_webrtc_rtp_transceiver_direction >*/
/** /**
* GstWebRTCDTLSSetup: * GstWebRTCDTLSSetup:
* GST_WEBRTC_DTLS_SETUP_NONE: none * @GST_WEBRTC_DTLS_SETUP_NONE: none
* GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass * @GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass
* GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly * @GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly
* GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly * @GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly
*/ */
typedef enum /*< underscore_name=gst_webrtc_dtls_setup >*/ typedef enum /*< underscore_name=gst_webrtc_dtls_setup >*/
{ {
@ -224,20 +229,20 @@ typedef enum /*< underscore_name=gst_webrtc_dtls_setup >*/
/** /**
* GstWebRTCStatsType: * GstWebRTCStatsType:
* GST_WEBRTC_STATS_CODEC: codec * @GST_WEBRTC_STATS_CODEC: codec
* GST_WEBRTC_STATS_INBOUND_RTP: inbound-rtp * @GST_WEBRTC_STATS_INBOUND_RTP: inbound-rtp
* GST_WEBRTC_STATS_OUTBOUND_RTP: outbound-rtp * @GST_WEBRTC_STATS_OUTBOUND_RTP: outbound-rtp
* GST_WEBRTC_STATS_REMOTE_INBOUND_RTP: remote-inbound-rtp * @GST_WEBRTC_STATS_REMOTE_INBOUND_RTP: remote-inbound-rtp
* GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP: remote-outbound-rtp * @GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP: remote-outbound-rtp
* GST_WEBRTC_STATS_CSRC: csrc * @GST_WEBRTC_STATS_CSRC: csrc
* GST_WEBRTC_STATS_PEER_CONNECTION: peer-connectiion * @GST_WEBRTC_STATS_PEER_CONNECTION: peer-connectiion
* GST_WEBRTC_STATS_DATA_CHANNEL: data-channel * @GST_WEBRTC_STATS_DATA_CHANNEL: data-channel
* GST_WEBRTC_STATS_STREAM: stream * @GST_WEBRTC_STATS_STREAM: stream
* GST_WEBRTC_STATS_TRANSPORT: transport * @GST_WEBRTC_STATS_TRANSPORT: transport
* GST_WEBRTC_STATS_CANDIDATE_PAIR: candidate-pair * @GST_WEBRTC_STATS_CANDIDATE_PAIR: candidate-pair
* GST_WEBRTC_STATS_LOCAL_CANDIDATE: local-candidate * @GST_WEBRTC_STATS_LOCAL_CANDIDATE: local-candidate
* GST_WEBRTC_STATS_REMOTE_CANDIDATE: remote-candidate * @GST_WEBRTC_STATS_REMOTE_CANDIDATE: remote-candidate
* GST_WEBRTC_STATS_CERTIFICATE: certificate * @GST_WEBRTC_STATS_CERTIFICATE: certificate
*/ */
typedef enum /*< underscore_name=gst_webrtc_stats_type >*/ typedef enum /*< underscore_name=gst_webrtc_stats_type >*/
{ {
@ -259,8 +264,8 @@ typedef enum /*< underscore_name=gst_webrtc_stats_type >*/
/** /**
* GstWebRTCFECType: * GstWebRTCFECType:
* GST_WEBRTC_FEC_TYPE_NONE: none * @GST_WEBRTC_FEC_TYPE_NONE: none
* GST_WEBRTC_FEC_TYPE_ULP_RED: ulpfec + red * @GST_WEBRTC_FEC_TYPE_ULP_RED: ulpfec + red
*/ */
typedef enum /*< underscore_name=gst_webrtc_fec_type >*/ typedef enum /*< underscore_name=gst_webrtc_fec_type >*/
{ {

4
gst/geometrictransform/geometricmath.c
View file

@ -179,7 +179,7 @@ gst_gm_mod_float (gdouble a, gdouble b)
return a; return a;
} }
/** /*
* Returns a repeating triangle shape in the range 0..1 with wavelength 1.0 * Returns a repeating triangle shape in the range 0..1 with wavelength 1.0
*/ */
gdouble gdouble
@ -190,7 +190,7 @@ gst_gm_triangle (gdouble x)
return 2.0 * (r < 0.5 ? r : 1 - r); return 2.0 * (r < 0.5 ? r : 1 - r);
} }
/** /*
* Hermite interpolation * Hermite interpolation
*/ */
gdouble gdouble

6
sys/shm/shmpipe.c
View file

@ -263,8 +263,7 @@ sp_writer_create (const char *path, size_t size, mode_t perms)
return NULL; \ return NULL; \
} while (0) } while (0)
/** /* sp_open_shm:
* sp_open_shm:
* @path: Path of the shm area for a reader, * @path: Path of the shm area for a reader,
* NULL if this is a writer (then it will allocate its own path) * 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 i;
int had_client = 0; 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 * if a client closes connection but already decremented the use count
* for this buffer, but other clients didn't have time to decrement * for this buffer, but other clients didn't have time to decrement
* buffer will not be freed too early in sp_writer_close_client. * buffer will not be freed too early in sp_writer_close_client.