mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-01-11 01:45:33 +00:00
Minor documentation fixes
This commit is contained in:
Thibault Saunier
parent
dce17521eb
commit
7fe3f36ac8
29 changed files with 127 additions and 115 deletions
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@ -47,7 +47,7 @@
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#include "io-sim.h"
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/**
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/*
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* @addtogroup Rawenc Raw VBI encoder
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* @ingroup Raw
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* @brief Converting sliced VBI data to raw VBI images.
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@ -375,7 +375,7 @@ clear_image (uint8_t * p,
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}
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}
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/**
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/*
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* @param raw Noise will be added to this raw VBI image.
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* @param sp Describes the raw VBI data in the buffer. @a sp->sampling_format
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* must be @c VBI_PIXFMT_Y8 (@c VBI_PIXFMT_YUV420 in libzvbi 0.2.x).
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@ -1008,12 +1008,12 @@ _vbi_raw_video_image (uint8_t * raw,
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return TRUE;
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}
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/**
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/*
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* @example examples/rawout.c
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* Raw VBI output example.
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*/
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/**
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/*
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* @param raw A raw VBI image will be stored here.
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* @param raw_size Size of the @a raw buffer in bytes. The buffer
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* must be large enough for @a sp->count[0] + count[1] lines
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@ -1079,7 +1079,7 @@ vbi_raw_vbi_image (uint8_t * raw,
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swap_fields ? _VBI_RAW_SWAP_FIELDS : 0, sliced, n_sliced_lines);
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}
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/**
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/*
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* @param raw A raw VBI image will be stored here.
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* @param raw_size Size of the @a raw buffer in bytes. The buffer
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* must be large enough for @a sp->count[0] + count[1] lines
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@ -21,7 +21,6 @@
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* SECTION:element-curlsink
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* @title: curlsink
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* @short_description: sink that uploads data to a server using libcurl
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* @see_also:
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*
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* This is a network sink that uses libcurl as a client to upload data to
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* a server (e.g. a HTTP/FTP server).
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@ -21,7 +21,6 @@
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* SECTION:element-curlfilesink
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* @title: curlfilesink
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* @short_description: sink that uploads data to a server using libcurl
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* @see_also:
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*
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* This is a network sink that uses libcurl as a client to upload data to
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* a local or network drive.
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@ -21,7 +21,6 @@
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* SECTION:element-curlftpsink
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* @title: curlftpsink
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* @short_description: sink that uploads data to a server using libcurl
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* @see_also:
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*
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* This is a network sink that uses libcurl as a client to upload data to
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* an FTP server.
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@ -21,7 +21,6 @@
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* SECTION:element-curlhttpsink
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* @title: curlhttpsink
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* @short_description: sink that uploads data to a server using libcurl
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* @see_also:
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*
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* This is a network sink that uses libcurl as a client to upload data to
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* an HTTP server.
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@ -21,7 +21,6 @@
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* SECTION:element-curlsftpsink
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* @title: curlsftpsink
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* @short_description: sink that uploads data to a server using libcurl
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* @see_also:
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*
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* This is a network sink that uses libcurl as a client to upload data to
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* a SFTP (SSH File Transfer Protocol) server.
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@ -21,7 +21,6 @@
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* SECTION:element-curlsink
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* @title: curlsink
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* @short_description: sink that uploads data to a server using libcurl
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* @see_also:
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*
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* This is a network sink that uses libcurl as a client to upload data to
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* an SMTP server.
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@ -21,7 +21,6 @@
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* SECTION:element-curlsshsink
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* @title: curlsshsink
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* @short_description: sink that uploads data to a server using libcurl
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* @see_also:
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*
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* This is a network sink that uses libcurl.
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*
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@ -21,7 +21,6 @@
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* SECTION:element-curltlssink
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* @title: curltlssink
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* @short_description: sink that uploads data to a server using libcurl
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* @see_also:
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*
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* This is a network sink that uses libcurl.
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*
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@ -975,7 +975,7 @@ gst_mss_manifest_get_duration (GstMssManifest * manifest)
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}
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/**
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/*
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* Gets the duration in nanoseconds
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*/
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GstClockTime
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@ -1194,7 +1194,7 @@ gst_mss_stream_type_name (GstMssStreamType streamtype)
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}
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}
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/**
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/*
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* Seeks all streams to the fragment that contains the set time
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*
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* @forward: if this is forward playback
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@ -1215,7 +1215,7 @@ gst_mss_manifest_seek (GstMssManifest * manifest, gboolean forward,
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((forward && (flags & GST_SEEK_FLAG_SNAP_AFTER)) || \
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(!forward && (flags & GST_SEEK_FLAG_SNAP_BEFORE)))
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/**
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/*
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* Seeks this stream to the fragment that contains the sample at time
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*
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* @time: time in nanoseconds
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@ -5054,7 +5054,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
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/**
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* GstWebRTCBin::create-offer:
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* @object: the #GstWebRtcBin
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* @object: the #webrtcbin
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* @options: create-offer options
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* @promise: a #GstPromise which will contain the offer
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*/
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@ -5067,7 +5067,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
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/**
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* GstWebRTCBin::create-answer:
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* @object: the #GstWebRtcBin
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* @object: the #webrtcbin
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* @options: create-answer options
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* @promise: a #GstPromise which will contain the answer
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*/
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@ -5080,7 +5080,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
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/**
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* GstWebRTCBin::set-local-description:
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* @object: the #GstWebRtcBin
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* @object: the #GstWebRTCBin
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* @desc: a #GstWebRTCSessionDescription description
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* @promise: (nullable): a #GstPromise to be notified when it's set
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*/
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@ -5093,7 +5093,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
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/**
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* GstWebRTCBin::set-remote-description:
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* @object: the #GstWebRtcBin
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* @object: the #GstWebRTCBin
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* @desc: a #GstWebRTCSessionDescription description
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* @promise: (nullable): a #GstPromise to be notified when it's set
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*/
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@ -5106,7 +5106,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
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/**
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* GstWebRTCBin::add-ice-candidate:
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* @object: the #GstWebRtcBin
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* @object: the #webrtcbin
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* @mline_index: the index of the media description in the SDP
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* @ice-candidate: an ice candidate
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*/
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@ -5118,7 +5118,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
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/**
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* GstWebRTCBin::get-stats:
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* @object: the #GstWebRtcBin
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* @object: the #webrtcbin
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* @pad: (nullable): A #GstPad to get the stats for, or %NULL for all
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* @promise: a #GstPromise for the result
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*
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@ -5195,7 +5195,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
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/**
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* GstWebRTCBin::on-negotiation-needed:
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* @object: the #GstWebRtcBin
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* @object: the #webrtcbin
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*/
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gst_webrtc_bin_signals[ON_NEGOTIATION_NEEDED_SIGNAL] =
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g_signal_new ("on-negotiation-needed", G_TYPE_FROM_CLASS (klass),
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@ -5204,7 +5204,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
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/**
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* GstWebRTCBin::on-ice-candidate:
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* @object: the #GstWebRtcBin
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* @object: the #webrtcbin
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* @mline_index: the index of the media description in the SDP
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* @candidate: the ICE candidate
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*/
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@ -5215,7 +5215,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
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/**
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* GstWebRTCBin::on-new-transceiver:
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* @object: the #GstWebRtcBin
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* @object: the #webrtcbin
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* @candidate: the new #GstWebRTCRTPTransceiver
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*/
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gst_webrtc_bin_signals[ON_NEW_TRANSCEIVER_SIGNAL] =
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@ -5225,8 +5225,8 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
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/**
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* GstWebRTCBin::on-data-channel:
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* @object: the #GstWebRtcBin
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* @candidate: the new #GstWebRTCDataChannel
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* @object: the #GstWebRTCBin
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* @candidate: the new `GstWebRTCDataChannel`
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*/
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gst_webrtc_bin_signals[ON_DATA_CHANNEL_SIGNAL] =
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g_signal_new ("on-data-channel", G_TYPE_FROM_CLASS (klass),
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@ -5235,7 +5235,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
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/**
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* GstWebRTCBin::add-transceiver:
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* @object: the #GstWebRtcBin
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* @object: the #webrtcbin
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* @direction: the direction of the new transceiver
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* @caps: (allow none): the codec preferences for this transceiver
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*
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@ -5250,7 +5250,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
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/**
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* GstWebRTCBin::get-transceivers:
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* @object: the #GstWebRtcBin
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* @object: the #webrtcbin
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*
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* Returns: a #GArray of #GstWebRTCRTPTransceivers
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*/
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@ -5262,7 +5262,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
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/**
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* GstWebRTCBin::get-transceiver:
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* @object: the #GstWebRtcBin
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* @object: the #GstWebRTCBin
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* @idx: The index of the transceiver
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*
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* Returns: the #GstWebRTCRTPTransceiver, or %NULL
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@ -5277,7 +5277,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
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/**
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* GstWebRTCBin::add-turn-server:
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* @object: the #GstWebRtcBin
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* @object: the #GstWebRTCBin
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* @uri: The uri of the server of the form turn(s)://username:password@host:port
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*
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* Add a turn server to obtain ICE candidates from
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@ -5290,7 +5290,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
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/*
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* GstWebRTCBin::create-data-channel:
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* @object: the #GstWebRtcBin
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* @object: the #GstWebRTCBin
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* @label: the label for the data channel
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* @options: a #GstStructure of options for creating the data channel
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*
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@ -941,7 +941,7 @@ gst_webrtc_ice_class_init (GstWebRTCICEClass * klass)
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/**
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* GstWebRTCICE::on-ice-candidate:
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* @object: the #GstWebRtcBin
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* @object: the #GstWebRTCBin
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* @candidate: the ICE candidate
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*/
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gst_webrtc_ice_signals[ON_ICE_CANDIDATE_SIGNAL] =
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@ -22,7 +22,6 @@
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/**
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* SECTION:gstadaptivedemux
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* @short_description: Base class for adaptive demuxers
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* @see_also:
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*
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* What is an adaptive demuxer?
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* Adaptive demuxers are special demuxers in the sense that they don't
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@ -658,7 +658,7 @@ struct _GstH264SPSExtMVCLevelValue
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* level values signalled for the coded video sequence.
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* @level_value: array of #GstH264SPSExtMVCLevelValue
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*
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* Represents the parsed seq_parameter_set_mvc_extension().
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* Represents the parsed `seq_parameter_set_mvc_extension()`.
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*
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* Since: 1.6
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*/
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@ -39,7 +39,7 @@
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*
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* * From #GST_H265_NAL_SLICE_TRAIL_N to #GST_H265_NAL_SLICE_CRA_NUT: gst_h265_parser_parse_slice_hdr()
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*
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* * #GST_H265_NAL_SEI: gst_h265_parser_parse_sei()
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* * `GST_H265_NAL_*_SEI`: gst_h265_parser_parse_sei()
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*
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* * #GST_H265_NAL_VPS: gst_h265_parser_parse_vps()
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*
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@ -2119,7 +2119,7 @@ gst_h265_parser_parse_pps (GstH265Parser * parser,
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/**
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* gst_h265_parser_parse_slice_hdr:
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* @parser: a #GstH265Parser
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* @nalu: The #GST_H265_NAL_SLICE #GstH265NalUnit to parse
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* @nalu: The `GST_H265_NAL_SLICE` #GstH265NalUnit to parse
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* @slice: The #GstH265SliceHdr to fill.
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*
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* Parses @data, and fills the @slice structure.
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@ -2656,7 +2656,7 @@ gst_h265_sei_free (GstH265SEIMessage * sei)
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/**
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* gst_h265_parser_parse_sei:
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* @nalparser: a #GstH265Parser
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* @nalu: The #GST_H265_NAL_SEI #GstH265NalUnit to parse
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* @nalu: The `GST_H265_NAL_*_SEI` #GstH265NalUnit to parse
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* @messages: The GArray of #GstH265SEIMessage to fill. The caller must free it when done.
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*
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* Parses @data, create and fills the @messages array.
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@ -27,7 +27,7 @@
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* Boston, MA 02110-1301, USA.
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*/
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/**
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/*
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* Common code for NAL parsing from h264 and h265 parsers.
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*/
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@ -1988,7 +1988,7 @@ gst_mpegts_descriptor_parse_dvb_multilingual_component (const
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* @private_data_specifier: (out): the private data specifier id
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* registered by http://www.dvbservices.com/
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* @private_data: (out) (transfer full) (allow-none) (array length=length): additional data or NULL
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* @length: (out) (allow-none): length of %private_data
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* @length: (out) (allow-none): length of @private_data
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*
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* Parses out the private data specifier from the @descriptor.
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*
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@ -2024,7 +2024,7 @@ gst_mpegts_descriptor_parse_dvb_private_data_specifier (const
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* @descriptor: a %GST_MTS_DESC_DVB_FREQUENCY_LIST #GstMpegtsDescriptor
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* @offset: (out): %FALSE in Hz, %TRUE in kHz
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* @list: (out) (transfer full) (element-type guint32): a list of all frequencies in Hz/kHz
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* depending on %offset
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* depending on @offset
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*
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* Parses out a list of frequencies from the @descriptor.
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*
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@ -2195,7 +2195,7 @@ gst_mpegts_descriptor_parse_dvb_scrambling (const GstMpegtsDescriptor *
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* @descriptor: a %GST_MTS_DESC_DVB_DATA_BROADCAST_ID #GstMpegtsDescriptor
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* @data_broadcast_id: (out): the data broadcast id
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* @id_selector_bytes: (out) (transfer full) (array length=len): the selector bytes, if present
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* @len: (out): the length of #id_selector_bytes
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* @len: (out): the length of @id_selector_bytes
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*
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* Parses out the data broadcast id from the @descriptor.
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*
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@ -464,8 +464,8 @@ struct _GstMpegtsDVBLinkageExtendedEvent
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* @transport_stream_id: the transport id
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* @original_network_id: the original network id
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* @service_id: the service id
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* @linkage_type: the type which %linkage_data has
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* @private_data_length: the length for %private_data_bytes
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* @linkage_type: the type which @linkage_data has
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* @private_data_length: the length for @private_data_bytes
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* @private_data_bytes: additional data bytes
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*/
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struct _GstMpegtsDVBLinkageDescriptor
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@ -200,7 +200,7 @@ static void
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* @application_context: (allow-none): GMainContext to use or %NULL
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*
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* Creates a new GstPlayerSignalDispatcher that uses @application_context,
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* or the thread default one if %NULL is used. See gst_player_new_full().
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* or the thread default one if %NULL is used. See gst_player_new().
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*
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* Returns: (transfer full): the new GstPlayerSignalDispatcher
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*/
|
||||
|
|
|
|||
|
|
@ -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.
|
||||
|
|
|
|||
Loading…
Reference in a new issue