diff --git a/gir-files/GES-1.0.gir b/gir-files/GES-1.0.gir index 0ff9aabe1..24fdf4634 100644 --- a/gir-files/GES-1.0.gir +++ b/gir-files/GES-1.0.gir @@ -7249,7 +7249,9 @@ could not be loaded. - + Moves @layer at @new_layer_priority meaning that @layer we land at that position in the stack of layers inside the timeline. If @new_layer_priority is superior than the number @@ -7726,7 +7728,9 @@ responsible for controlling its timing properties. - + The priority of the first layer the element is in (note that only groups can span over several layers). %GES_TIMELINE_ELEMENT_NO_LAYER_PRIORITY @@ -8240,7 +8244,8 @@ name/return location pairs, followed by NULL + c:identifier="ges_timeline_element_get_layer_priority" + version="1.16"> The priority of the first layer the element is in (note that only groups can span over several layers). %GES_TIMELINE_ELEMENT_NO_LAYER_PRIORITY @@ -8442,8 +8447,9 @@ usage. Paste @self inside the timeline. @self must have been created using ges_timeline_element_copy with recurse=TRUE set, otherwise it will fail. - - Paste @self copying the element + + New element resulting of pasting @self +or %NULL @@ -11791,6 +11797,7 @@ set as Metadata of the Asset. Finalize the request of an async #GESUriClipAsset @@ -12121,7 +12128,7 @@ by #GESUriClipAsset-s. - + @@ -13259,7 +13266,9 @@ using the ges_init_get_option_group() result. - + Use this function to check if GES has been initialized with ges_init() or ges_init_check(). diff --git a/gir-files/Gst-1.0.gir b/gir-files/Gst-1.0.gir index 963d5f03e..4cfdeea3f 100644 --- a/gir-files/Gst-1.0.gir +++ b/gir-files/Gst-1.0.gir @@ -10886,7 +10886,7 @@ device in the PLAYING state. version="1.4"> Creates the element with all of the required parameters set to use this device. - + a new #GstElement configured to use this device @@ -11088,7 +11088,7 @@ device in the PLAYING state. - + a new #GstElement configured to use this device @@ -23403,7 +23403,7 @@ was updated. response is received with a non-HTTP URL inside. (Since: 1.10) Message indicating a #GstDevice was changed @@ -45276,7 +45276,7 @@ determine a order for the two provided values. The major version of GStreamer at compile time: - + The micro version of GStreamer at compile time: diff --git a/gir-files/GstGL-1.0.gir b/gir-files/GstGL-1.0.gir index 630dc4754..97d414dc3 100644 --- a/gir-files/GstGL-1.0.gir +++ b/gir-files/GstGL-1.0.gir @@ -2776,7 +2776,7 @@ for details on what is a valid name. - + diff --git a/gir-files/GstPbutils-1.0.gir b/gir-files/GstPbutils-1.0.gir index 652ee9c00..bd72900d6 100644 --- a/gir-files/GstPbutils-1.0.gir +++ b/gir-files/GstPbutils-1.0.gir @@ -2752,7 +2752,7 @@ in debugging. The micro version of GStreamer's gst-plugins-base libraries at compile time. diff --git a/gir-files/GstRtp-1.0.gir b/gir-files/GstRtp-1.0.gir index a16270a05..bbfc42c36 100644 --- a/gir-files/GstRtp-1.0.gir +++ b/gir-files/GstRtp-1.0.gir @@ -16,9 +16,7 @@ and/or use gtk-doc annotations. --> c:identifier-prefixes="Gst" c:symbol-prefixes="gst"> - Note: The API in this module is not yet declared stable. + Note: The API in this module is not yet declared stable. The GstRTPCBuffer helper functions makes it easy to parse and create regular #GstBuffer objects that contain compound RTCP packets. These buffers are typically @@ -28,7 +26,6 @@ An RTCP buffer consists of 1 or more #GstRTCPPacket structures that you can retrieve with gst_rtcp_buffer_get_first_packet(). #GstRTCPPacket acts as a pointer into the RTCP buffer; you can move to the next packet with gst_rtcp_packet_move_to_next(). - @@ -36,271 +33,189 @@ gst_rtcp_packet_move_to_next(). - Add a new packet of @type to @rtcp. @packet will point to the newly created + Add a new packet of @type to @rtcp. @packet will point to the newly created packet. - - %TRUE if the packet could be created. This function returns %FALSE + %TRUE if the packet could be created. This function returns %FALSE if the max mtu is exceeded for the buffer. - a valid RTCP buffer + a valid RTCP buffer - the #GstRTCPType of the new packet + the #GstRTCPType of the new packet - pointer to new packet + pointer to new packet - Initialize a new #GstRTCPPacket pointer that points to the first packet in + Initialize a new #GstRTCPPacket pointer that points to the first packet in @rtcp. - - TRUE if the packet existed in @rtcp. + TRUE if the packet existed in @rtcp. - a valid RTCP buffer + a valid RTCP buffer - a #GstRTCPPacket + a #GstRTCPPacket - Get the number of RTCP packets in @rtcp. - + Get the number of RTCP packets in @rtcp. - the number of RTCP packets in @rtcp. + the number of RTCP packets in @rtcp. - a valid RTCP buffer + a valid RTCP buffer - Finish @rtcp after being constructed. This function is usually called + Finish @rtcp after being constructed. This function is usually called after gst_rtcp_buffer_map() and after adding the RTCP items to the new buffer. The function adjusts the size of @rtcp with the total length of all the added packets. - - a buffer with an RTCP packet + a buffer with an RTCP packet - Open @buffer for reading or writing, depending on @flags. The resulting RTCP + Open @buffer for reading or writing, depending on @flags. The resulting RTCP buffer state is stored in @rtcp. - - a buffer with an RTCP packet + a buffer with an RTCP packet - flags for the mapping + flags for the mapping - resulting #GstRTCPBuffer + resulting #GstRTCPBuffer - Create a new buffer for constructing RTCP packets. The packet will have a + Create a new buffer for constructing RTCP packets. The packet will have a maximum size of @mtu. - - A newly allocated buffer. + A newly allocated buffer. - the maximum mtu size. + the maximum mtu size. - Create a new buffer and set the data to a copy of @len + Create a new buffer and set the data to a copy of @len bytes of @data and the size to @len. The data will be freed when the buffer is freed. - - A newly allocated buffer with a copy of @data and of size @len. + A newly allocated buffer with a copy of @data and of size @len. - data for the new buffer + data for the new buffer - the length of data + the length of data - Create a new buffer and set the data and size of the buffer to @data and @len + Create a new buffer and set the data and size of the buffer to @data and @len respectively. @data will be freed when the buffer is unreffed, so this function transfers ownership of @data to the new buffer. - - A newly allocated buffer with @data and of size @len. + A newly allocated buffer with @data and of size @len. - data for the new buffer + data for the new buffer - the length of data + the length of data - Check if the data pointed to by @buffer is a valid RTCP packet using + Check if the data pointed to by @buffer is a valid RTCP packet using gst_rtcp_buffer_validate_data(). - - TRUE if @buffer is a valid RTCP packet. + TRUE if @buffer is a valid RTCP packet. - the buffer to validate + the buffer to validate - Check if the @data and @size point to the data of a valid compound, + Check if the @data and @size point to the data of a valid compound, non-reduced size RTCP packet. Use this function to validate a packet before using the other functions in this module. - - TRUE if the data points to a valid RTCP packet. + TRUE if the data points to a valid RTCP packet. - the data to validate + the data to validate - the length of @data to validate + the length of @data to validate @@ -308,35 +223,26 @@ this module. - Check if the @data and @size point to the data of a valid RTCP packet. + Check if the @data and @size point to the data of a valid RTCP packet. Use this function to validate a packet before using the other functions in this module. This function is updated to support reduced size rtcp packets according to RFC 5506 and will validate full compound RTCP packets as well as reduced size RTCP packets. - - TRUE if the data points to a valid RTCP packet. + TRUE if the data points to a valid RTCP packet. - the data to validate + the data to validate - the length of @data to validate + the length of @data to validate @@ -344,22 +250,15 @@ size RTCP packets. - Check if the data pointed to by @buffer is a valid RTCP packet using + Check if the data pointed to by @buffer is a valid RTCP packet using gst_rtcp_buffer_validate_reduced(). - - TRUE if @buffer is a valid RTCP packet. + TRUE if @buffer is a valid RTCP packet. - the buffer to validate + the buffer to validate @@ -369,132 +268,97 @@ gst_rtcp_buffer_validate_reduced(). glib:type-name="GstRTCPFBType" glib:get-type="gst_rtcpfb_type_get_type" c:type="GstRTCPFBType"> - Different types of feedback messages. + Different types of feedback messages. - Invalid type + Invalid type - Generic NACK + Generic NACK - Temporary Maximum Media Stream Bit Rate Request + Temporary Maximum Media Stream Bit Rate Request - Temporary Maximum Media Stream Bit Rate + Temporary Maximum Media Stream Bit Rate Notification - Request an SR packet for early + Request an SR packet for early synchronization - Picture Loss Indication + Picture Loss Indication - Slice Loss Indication + Slice Loss Indication - Reference Picture Selection Indication + Reference Picture Selection Indication - Application layer Feedback + Application layer Feedback - Full Intra Request Command + Full Intra Request Command - Temporal-Spatial Trade-off Request + Temporal-Spatial Trade-off Request - Temporal-Spatial Trade-off Notification + Temporal-Spatial Trade-off Notification - Video Back Channel Message + Video Back Channel Message - Data structure that points to a packet at @offset in @buffer. + Data structure that points to a packet at @offset in @buffer. The size of the structure is made public to allow stack allocations. - - pointer to RTCP buffer + pointer to RTCP buffer - offset of packet in buffer data + offset of packet in buffer data @@ -520,101 +384,69 @@ The size of the structure is made public to allow stack allocations. - Add profile-specific extension @data to @packet. If @packet already + Add profile-specific extension @data to @packet. If @packet already contains profile-specific extension @data will be appended to the existing extension. - - %TRUE if the profile specific extension data was added. + %TRUE if the profile specific extension data was added. - a valid SR or RR #GstRTCPPacket + a valid SR or RR #GstRTCPPacket - profile-specific data + profile-specific data - length of the profile-specific data in bytes + length of the profile-specific data in bytes - Add a new report block to @packet with the given values. - + Add a new report block to @packet with the given values. - %TRUE if the packet was created. This function can return %FALSE if + %TRUE if the packet was created. This function can return %FALSE if the max MTU is exceeded or the number of report blocks is greater than #GST_RTCP_MAX_RB_COUNT. - a valid SR or RR #GstRTCPPacket + a valid SR or RR #GstRTCPPacket - data source being reported + data source being reported - fraction lost since last SR/RR + fraction lost since last SR/RR - the cumululative number of packets lost + the cumululative number of packets lost - the extended last sequence number received + the extended last sequence number received - the interarrival jitter + the interarrival jitter - the last SR packet from this source + the last SR packet from this source - the delay since last SR packet + the delay since last SR packet @@ -622,21 +454,14 @@ the max MTU is exceeded or the number of report blocks is greater than - Get the application-dependent data attached to a RTPFB or PSFB @packet. - + Get the application-dependent data attached to a RTPFB or PSFB @packet. - A pointer to the data + A pointer to the data - a valid APP #GstRTCPPacket + a valid APP #GstRTCPPacket @@ -644,22 +469,15 @@ the max MTU is exceeded or the number of report blocks is greater than - Get the length of the application-dependent data attached to an APP + Get the length of the application-dependent data attached to an APP @packet. - - The length of data in 32-bit words. + The length of data in 32-bit words. - a valid APP #GstRTCPPacket + a valid APP #GstRTCPPacket @@ -667,21 +485,14 @@ the max MTU is exceeded or the number of report blocks is greater than - Get the name field of the APP @packet. - + Get the name field of the APP @packet. - The 4-byte name field, not zero-terminated. + The 4-byte name field, not zero-terminated. - a valid APP #GstRTCPPacket + a valid APP #GstRTCPPacket @@ -689,21 +500,14 @@ the max MTU is exceeded or the number of report blocks is greater than - Get the SSRC/CSRC field of the APP @packet. - + Get the SSRC/CSRC field of the APP @packet. - The SSRC/CSRC. + The SSRC/CSRC. - a valid APP #GstRTCPPacket + a valid APP #GstRTCPPacket @@ -711,21 +515,14 @@ the max MTU is exceeded or the number of report blocks is greater than - Get the subtype field of the APP @packet. - + Get the subtype field of the APP @packet. - The subtype. + The subtype. - a valid APP #GstRTCPPacket + a valid APP #GstRTCPPacket @@ -733,29 +530,20 @@ the max MTU is exceeded or the number of report blocks is greater than - Set the length of the application-dependent data attached to an APP + Set the length of the application-dependent data attached to an APP @packet. - - %TRUE if there was enough space in the packet to add this much + %TRUE if there was enough space in the packet to add this much data. - a valid APP #GstRTCPPacket + a valid APP #GstRTCPPacket - Length of the data in 32-bit words + Length of the data in 32-bit words @@ -763,24 +551,17 @@ data. - Set the name field of the APP @packet. - + Set the name field of the APP @packet. - a valid APP #GstRTCPPacket + a valid APP #GstRTCPPacket - 4-byte ASCII name + 4-byte ASCII name @@ -788,24 +569,17 @@ data. - Set the SSRC/CSRC field of the APP @packet. - + Set the SSRC/CSRC field of the APP @packet. - a valid APP #GstRTCPPacket + a valid APP #GstRTCPPacket - SSRC/CSRC of the packet + SSRC/CSRC of the packet @@ -813,239 +587,164 @@ data. - Set the subtype field of the APP @packet. - + Set the subtype field of the APP @packet. - a valid APP #GstRTCPPacket + a valid APP #GstRTCPPacket - subtype of the packet + subtype of the packet - Add @ssrc to the BYE @packet. - + Add @ssrc to the BYE @packet. - %TRUE if the ssrc was added. This function can return %FALSE if + %TRUE if the ssrc was added. This function can return %FALSE if the max MTU is exceeded or the number of sources blocks is greater than #GST_RTCP_MAX_BYE_SSRC_COUNT. - a valid BYE #GstRTCPPacket + a valid BYE #GstRTCPPacket - an SSRC to add + an SSRC to add - Adds @len SSRCs in @ssrc to BYE @packet. - + Adds @len SSRCs in @ssrc to BYE @packet. - %TRUE if the all the SSRCs were added. This function can return %FALSE if + %TRUE if the all the SSRCs were added. This function can return %FALSE if the max MTU is exceeded or the number of sources blocks is greater than #GST_RTCP_MAX_BYE_SSRC_COUNT. - a valid BYE #GstRTCPPacket + a valid BYE #GstRTCPPacket - an array of SSRCs to add + an array of SSRCs to add - number of elements in @ssrc + number of elements in @ssrc - Get the @nth SSRC of the BYE @packet. - + Get the @nth SSRC of the BYE @packet. - The @nth SSRC of @packet. + The @nth SSRC of @packet. - a valid BYE #GstRTCPPacket + a valid BYE #GstRTCPPacket - the nth SSRC to get + the nth SSRC to get - Get the reason in @packet. - + Get the reason in @packet. - The reason for the BYE @packet or NULL if the packet did not contain + The reason for the BYE @packet or NULL if the packet did not contain a reason string. The string must be freed with g_free() after usage. - a valid BYE #GstRTCPPacket + a valid BYE #GstRTCPPacket - Get the length of the reason string. - + Get the length of the reason string. - The length of the reason string or 0 when there is no reason string + The length of the reason string or 0 when there is no reason string present. - a valid BYE #GstRTCPPacket + a valid BYE #GstRTCPPacket - Get the number of SSRC fields in @packet. - + Get the number of SSRC fields in @packet. - The number of SSRC fields in @packet. + The number of SSRC fields in @packet. - a valid BYE #GstRTCPPacket + a valid BYE #GstRTCPPacket - Set the reason string to @reason in @packet. - + Set the reason string to @reason in @packet. - TRUE if the string could be set. + TRUE if the string could be set. - a valid BYE #GstRTCPPacket + a valid BYE #GstRTCPPacket - a reason string + a reason string - The profile-specific extension data is copied into a new allocated + The profile-specific extension data is copied into a new allocated memory area @data. This must be freed with g_free() after usage. - - %TRUE if there was valid data. + %TRUE if there was valid data. - a valid SR or RR #GstRTCPPacket + a valid SR or RR #GstRTCPPacket - result profile-specific data + result profile-specific data @@ -1054,301 +753,206 @@ memory area @data. This must be freed with g_free() after usage. direction="out" caller-allocates="0" transfer-ownership="full"> - length of the profile-specific extension data + length of the profile-specific extension data - Get the Feedback Control Information attached to a RTPFB or PSFB @packet. - + Get the Feedback Control Information attached to a RTPFB or PSFB @packet. - a pointer to the FCI + a pointer to the FCI - a valid RTPFB or PSFB #GstRTCPPacket + a valid RTPFB or PSFB #GstRTCPPacket - Get the length of the Feedback Control Information attached to a + Get the length of the Feedback Control Information attached to a RTPFB or PSFB @packet. - - The length of the FCI in 32-bit words. + The length of the FCI in 32-bit words. - a valid RTPFB or PSFB #GstRTCPPacket + a valid RTPFB or PSFB #GstRTCPPacket - Get the media SSRC field of the RTPFB or PSFB @packet. - + Get the media SSRC field of the RTPFB or PSFB @packet. - the media SSRC. + the media SSRC. - a valid RTPFB or PSFB #GstRTCPPacket + a valid RTPFB or PSFB #GstRTCPPacket - Get the sender SSRC field of the RTPFB or PSFB @packet. - + Get the sender SSRC field of the RTPFB or PSFB @packet. - the sender SSRC. + the sender SSRC. - a valid RTPFB or PSFB #GstRTCPPacket + a valid RTPFB or PSFB #GstRTCPPacket - Get the feedback message type of the FB @packet. - + Get the feedback message type of the FB @packet. - The feedback message type. + The feedback message type. - a valid RTPFB or PSFB #GstRTCPPacket + a valid RTPFB or PSFB #GstRTCPPacket - Set the length of the Feedback Control Information attached to a + Set the length of the Feedback Control Information attached to a RTPFB or PSFB @packet. - - %TRUE if there was enough space in the packet to add this much FCI + %TRUE if there was enough space in the packet to add this much FCI - a valid RTPFB or PSFB #GstRTCPPacket + a valid RTPFB or PSFB #GstRTCPPacket - Length of the FCI in 32-bit words + Length of the FCI in 32-bit words - Set the media SSRC field of the RTPFB or PSFB @packet. - + Set the media SSRC field of the RTPFB or PSFB @packet. - a valid RTPFB or PSFB #GstRTCPPacket + a valid RTPFB or PSFB #GstRTCPPacket - a media SSRC + a media SSRC - Set the sender SSRC field of the RTPFB or PSFB @packet. - + Set the sender SSRC field of the RTPFB or PSFB @packet. - a valid RTPFB or PSFB #GstRTCPPacket + a valid RTPFB or PSFB #GstRTCPPacket - a sender SSRC + a sender SSRC - Set the feedback message type of the FB @packet. - + Set the feedback message type of the FB @packet. - a valid RTPFB or PSFB #GstRTCPPacket + a valid RTPFB or PSFB #GstRTCPPacket - the #GstRTCPFBType to set + the #GstRTCPFBType to set - Get the count field in @packet. - + Get the count field in @packet. - The count field in @packet or -1 if @packet does not point to a + The count field in @packet or -1 if @packet does not point to a valid packet. - a valid #GstRTCPPacket + a valid #GstRTCPPacket - Get the length field of @packet. This is the length of the packet in + Get the length field of @packet. This is the length of the packet in 32-bit words minus one. - - The length field of @packet. + The length field of @packet. - a valid #GstRTCPPacket + a valid #GstRTCPPacket - Get the packet padding of the packet pointed to by @packet. - + Get the packet padding of the packet pointed to by @packet. - If the packet has the padding bit set. + If the packet has the padding bit set. - a valid #GstRTCPPacket + a valid #GstRTCPPacket - - %TRUE if there was valid data. + %TRUE if there was valid data. - a valid SR or RR #GstRTCPPacket + a valid SR or RR #GstRTCPPacket - result profile-specific data + result profile-specific data @@ -1357,282 +961,199 @@ valid packet. direction="out" caller-allocates="0" transfer-ownership="full"> - result length of the profile-specific data + result length of the profile-specific data - - The number of 32-bit words containing profile-specific extension + The number of 32-bit words containing profile-specific extension data from @packet. - a valid SR or RR #GstRTCPPacket + a valid SR or RR #GstRTCPPacket - Parse the values of the @nth report block in @packet and store the result in + Parse the values of the @nth report block in @packet and store the result in the values. - - a valid SR or RR #GstRTCPPacket + a valid SR or RR #GstRTCPPacket - the nth report block in @packet + the nth report block in @packet - result for data source being reported + result for data source being reported - result for fraction lost since last SR/RR + result for fraction lost since last SR/RR - result for the cumululative number of packets lost + result for the cumululative number of packets lost - result for the extended last sequence number received + result for the extended last sequence number received - result for the interarrival jitter + result for the interarrival jitter - result for the last SR packet from this source + result for the last SR packet from this source - result for the delay since last SR packet + result for the delay since last SR packet - Get the number of report blocks in @packet. - + Get the number of report blocks in @packet. - The number of report blocks in @packet. + The number of report blocks in @packet. - a valid SR or RR #GstRTCPPacket + a valid SR or RR #GstRTCPPacket - Get the packet type of the packet pointed to by @packet. - + Get the packet type of the packet pointed to by @packet. - The packet type or GST_RTCP_TYPE_INVALID when @packet is not + The packet type or GST_RTCP_TYPE_INVALID when @packet is not pointing to a valid packet. - a valid #GstRTCPPacket + a valid #GstRTCPPacket - Move the packet pointer @packet to the next packet in the payload. + Move the packet pointer @packet to the next packet in the payload. Use gst_rtcp_buffer_get_first_packet() to initialize @packet. - - TRUE if @packet is pointing to a valid packet after calling this + TRUE if @packet is pointing to a valid packet after calling this function. - a #GstRTCPPacket + a #GstRTCPPacket - Removes the packet pointed to by @packet and moves pointer to the next one - + Removes the packet pointed to by @packet and moves pointer to the next one - TRUE if @packet is pointing to a valid packet after calling this + TRUE if @packet is pointing to a valid packet after calling this function. - a #GstRTCPPacket + a #GstRTCPPacket - Get the ssrc field of the RR @packet. - + Get the ssrc field of the RR @packet. - the ssrc. + the ssrc. - a valid RR #GstRTCPPacket + a valid RR #GstRTCPPacket - Set the ssrc field of the RR @packet. - + Set the ssrc field of the RR @packet. - a valid RR #GstRTCPPacket + a valid RR #GstRTCPPacket - the SSRC to set + the SSRC to set - Add a new SDES entry to the current item in @packet. - + Add a new SDES entry to the current item in @packet. - %TRUE if the item could be added, %FALSE if the MTU has been + %TRUE if the item could be added, %FALSE if the MTU has been reached. - a valid SDES #GstRTCPPacket + a valid SDES #GstRTCPPacket - the #GstRTCPSDESType of the SDES entry + the #GstRTCPSDESType of the SDES entry - the data length + the data length - the data + the data @@ -1641,74 +1162,52 @@ reached. - Add a new SDES item for @ssrc to @packet. - + Add a new SDES item for @ssrc to @packet. - %TRUE if the item could be added, %FALSE if the maximum amount of + %TRUE if the item could be added, %FALSE if the maximum amount of items has been exceeded for the SDES packet or the MTU has been reached. - a valid SDES #GstRTCPPacket + a valid SDES #GstRTCPPacket - the SSRC of the new item to add + the SSRC of the new item to add - This function is like gst_rtcp_packet_sdes_get_entry() but it returns a + This function is like gst_rtcp_packet_sdes_get_entry() but it returns a null-terminated copy of the data instead. use g_free() after usage. - - %TRUE if there was valid data. + %TRUE if there was valid data. - a valid SDES #GstRTCPPacket + a valid SDES #GstRTCPPacket - result of the entry type + result of the entry type - result length of the entry data + result length of the entry data - result entry data + result entry data @@ -1717,93 +1216,66 @@ null-terminated copy of the data instead. use g_free() after usage. - Move to the first SDES entry in the current item. - + Move to the first SDES entry in the current item. - %TRUE if there was a first entry. + %TRUE if there was a first entry. - a valid SDES #GstRTCPPacket + a valid SDES #GstRTCPPacket - Move to the first SDES item in @packet. - + Move to the first SDES item in @packet. - TRUE if there was a first item. + TRUE if there was a first item. - a valid SDES #GstRTCPPacket + a valid SDES #GstRTCPPacket - Get the data of the current SDES item entry. @type (when not NULL) will + Get the data of the current SDES item entry. @type (when not NULL) will contain the type of the entry. @data (when not NULL) will point to @len bytes. When @type refers to a text item, @data will point to a UTF8 string. Note that this UTF8 string is NOT null-terminated. Use gst_rtcp_packet_sdes_copy_entry() to get a null-terminated copy of the entry. - - %TRUE if there was valid data. + %TRUE if there was valid data. - a valid SDES #GstRTCPPacket + a valid SDES #GstRTCPPacket - result of the entry type + result of the entry type - result length of the entry data + result length of the entry data - result entry data + result entry data @@ -1812,262 +1284,183 @@ gst_rtcp_packet_sdes_copy_entry() to get a null-terminated copy of the entry. - Get the number of items in the SDES packet @packet. - + Get the number of items in the SDES packet @packet. - The number of items in @packet. + The number of items in @packet. - a valid SDES #GstRTCPPacket + a valid SDES #GstRTCPPacket - Get the SSRC of the current SDES item. - + Get the SSRC of the current SDES item. - the SSRC of the current item. + the SSRC of the current item. - a valid SDES #GstRTCPPacket + a valid SDES #GstRTCPPacket - Move to the next SDES entry in the current item. - + Move to the next SDES entry in the current item. - %TRUE if there was a next entry. + %TRUE if there was a next entry. - a valid SDES #GstRTCPPacket + a valid SDES #GstRTCPPacket - Move to the next SDES item in @packet. - + Move to the next SDES item in @packet. - TRUE if there was a next item. + TRUE if there was a next item. - a valid SDES #GstRTCPPacket + a valid SDES #GstRTCPPacket - Set the @nth new report block in @packet with the given values. + Set the @nth new report block in @packet with the given values. Note: Not implemented. - - a valid SR or RR #GstRTCPPacket + a valid SR or RR #GstRTCPPacket - the nth report block to set + the nth report block to set - data source being reported + data source being reported - fraction lost since last SR/RR + fraction lost since last SR/RR - the cumululative number of packets lost + the cumululative number of packets lost - the extended last sequence number received + the extended last sequence number received - the interarrival jitter + the interarrival jitter - the last SR packet from this source + the last SR packet from this source - the delay since last SR packet + the delay since last SR packet - Parse the SR sender info and store the values. - + Parse the SR sender info and store the values. - a valid SR #GstRTCPPacket + a valid SR #GstRTCPPacket - result SSRC + result SSRC - result NTP time + result NTP time - result RTP time + result RTP time - result packet count + result packet count - result octet count + result octet count - Set the given values in the SR packet @packet. - + Set the given values in the SR packet @packet. - a valid SR #GstRTCPPacket + a valid SR #GstRTCPPacket - the SSRC + the SSRC - the NTP time + the NTP time - the RTP time + the RTP time - the packet count + the packet count - the octet count + the octet count @@ -2075,21 +1468,14 @@ Note: Not implemented. - Move to the first extended report block in XR @packet. - + Move to the first extended report block in XR @packet. - TRUE if there was a first extended report block. + TRUE if there was a first extended report block. - a valid XR #GstRTCPPacket + a valid XR #GstRTCPPacket @@ -2097,19 +1483,14 @@ Note: Not implemented. - - The number of 32-bit words containing type-specific block + The number of 32-bit words containing type-specific block data from @packet. - a valid XR #GstRTCPPacket + a valid XR #GstRTCPPacket @@ -2117,21 +1498,14 @@ Note: Not implemented. - Get the extended report block type of the XR @packet. - + Get the extended report block type of the XR @packet. - The extended report block type. + The extended report block type. - a valid XR #GstRTCPPacket + a valid XR #GstRTCPPacket @@ -2139,45 +1513,30 @@ Note: Not implemented. - Parse the extended report block for DLRR report block type. - + Parse the extended report block for DLRR report block type. - %TRUE if the report block is correctly parsed. + %TRUE if the report block is correctly parsed. - a valid XR #GstRTCPPacket which has DLRR Report Block. + a valid XR #GstRTCPPacket which has DLRR Report Block. - the index of sub-block to retrieve. + the index of sub-block to retrieve. - the SSRC of the receiver. + the SSRC of the receiver. - the last receiver reference timestamp of @ssrc. + the last receiver reference timestamp of @ssrc. - the delay since @last_rr. + the delay since @last_rr. @@ -2185,33 +1544,22 @@ Note: Not implemented. - Retrieve the packet receipt time of @seq which ranges in [begin_seq, end_seq). - + Retrieve the packet receipt time of @seq which ranges in [begin_seq, end_seq). - %TRUE if the report block returns the receipt time correctly. + %TRUE if the report block returns the receipt time correctly. - a valid XR #GstRTCPPacket which has the Packet Recept Times Report Block. + a valid XR #GstRTCPPacket which has the Packet Recept Times Report Block. - the sequence to retrieve the time. + the sequence to retrieve the time. - the packet receipt time of @seq. + the packet receipt time of @seq. @@ -2219,45 +1567,30 @@ Note: Not implemented. - Parse the Packet Recept Times Report Block from a XR @packet - + Parse the Packet Recept Times Report Block from a XR @packet - %TRUE if the report block is correctly parsed. + %TRUE if the report block is correctly parsed. - a valid XR #GstRTCPPacket which has a Packet Receipt Times Report Block + a valid XR #GstRTCPPacket which has a Packet Receipt Times Report Block - the SSRC of the RTP data packet source being reported upon by this report block. + the SSRC of the RTP data packet source being reported upon by this report block. - the amount of thinning performed on the sequence number space. + the amount of thinning performed on the sequence number space. - the first sequence number that this block reports on. + the first sequence number that this block reports on. - the last sequence number that this block reports on plus one. + the last sequence number that this block reports on plus one. @@ -2265,51 +1598,34 @@ Note: Not implemented. - Parse the extended report block for Loss RLE and Duplicated LRE block type. - + Parse the extended report block for Loss RLE and Duplicated LRE block type. - %TRUE if the report block is correctly parsed. + %TRUE if the report block is correctly parsed. - a valid XR #GstRTCPPacket which is Loss RLE or Duplicate RLE report. + a valid XR #GstRTCPPacket which is Loss RLE or Duplicate RLE report. - the SSRC of the RTP data packet source being reported upon by this report block. + the SSRC of the RTP data packet source being reported upon by this report block. - the amount of thinning performed on the sequence number space. + the amount of thinning performed on the sequence number space. - the first sequence number that this block reports on. + the first sequence number that this block reports on. - the last sequence number that this block reports on plus one. + the last sequence number that this block reports on plus one. - the number of chunks calculated by block length. + the number of chunks calculated by block length. @@ -2317,33 +1633,22 @@ Note: Not implemented. - Retrieve actual chunk data. - + Retrieve actual chunk data. - %TRUE if the report block returns chunk correctly. + %TRUE if the report block returns chunk correctly. - a valid XR #GstRTCPPacket which is Loss RLE or Duplicate RLE report. + a valid XR #GstRTCPPacket which is Loss RLE or Duplicate RLE report. - the index of chunk to retrieve. + the index of chunk to retrieve. - the @nth chunk. + the @nth chunk. @@ -2351,24 +1656,17 @@ Note: Not implemented. - - %TRUE if the report block returns the reference time correctly. + %TRUE if the report block returns the reference time correctly. - a valid XR #GstRTCPPacket which has the Receiver Reference Time. + a valid XR #GstRTCPPacket which has the Receiver Reference Time. - NTP timestamp + NTP timestamp @@ -2376,21 +1674,14 @@ Note: Not implemented. - Get the ssrc field of the XR @packet. - + Get the ssrc field of the XR @packet. - the ssrc. + the ssrc. - a valid XR #GstRTCPPacket + a valid XR #GstRTCPPacket @@ -2398,39 +1689,26 @@ Note: Not implemented. - Extract a basic information from static summary report block of XR @packet. - + Extract a basic information from static summary report block of XR @packet. - %TRUE if the report block is correctly parsed. + %TRUE if the report block is correctly parsed. - a valid XR #GstRTCPPacket which has Statics Summary Report Block. + a valid XR #GstRTCPPacket which has Statics Summary Report Block. - the SSRC of the source. + the SSRC of the source. - the first sequence number that this block reports on. + the first sequence number that this block reports on. - the last sequence number that this block reports on plus one. + the last sequence number that this block reports on plus one. @@ -2438,46 +1716,31 @@ Note: Not implemented. - Extract jitter information from the statistics summary. If the jitter flag in + Extract jitter information from the statistics summary. If the jitter flag in a block header is set as zero, all of jitters will be zero. - - %TRUE if the report block is correctly parsed. + %TRUE if the report block is correctly parsed. - a valid XR #GstRTCPPacket which has Statics Summary Report Block. + a valid XR #GstRTCPPacket which has Statics Summary Report Block. - the minimum relative transit time between two sequences. + the minimum relative transit time between two sequences. - the maximum relative transit time between two sequences. + the maximum relative transit time between two sequences. - the mean relative transit time between two sequences. + the mean relative transit time between two sequences. - the standard deviation of the relative transit time between two sequences. + the standard deviation of the relative transit time between two sequences. @@ -2485,34 +1748,23 @@ a block header is set as zero, all of jitters will be zero. - Get the number of lost or duplicate packets. If the flag in a block header + Get the number of lost or duplicate packets. If the flag in a block header is set as zero, @lost_packets or @dup_packets will be zero. - - %TRUE if the report block is correctly parsed. + %TRUE if the report block is correctly parsed. - a valid XR #GstRTCPPacket which has Statics Summary Report Block. + a valid XR #GstRTCPPacket which has Statics Summary Report Block. - the number of lost packets between begin_seq and end_seq. + the number of lost packets between begin_seq and end_seq. - the number of duplicate packets between begin_seq and end_seq. + the number of duplicate packets between begin_seq and end_seq. @@ -2520,51 +1772,34 @@ is set as zero, @lost_packets or @dup_packets will be zero. - Extract the value of ttl for ipv4, or hop limit for ipv6. - + Extract the value of ttl for ipv4, or hop limit for ipv6. - %TRUE if the report block is correctly parsed. + %TRUE if the report block is correctly parsed. - a valid XR #GstRTCPPacket which has Statics Summary Report Block. + a valid XR #GstRTCPPacket which has Statics Summary Report Block. - the flag to indicate that the return values are ipv4 ttl or ipv6 hop limits. + the flag to indicate that the return values are ipv4 ttl or ipv6 hop limits. - the minimum TTL or Hop Limit value of data packets between two sequences. + the minimum TTL or Hop Limit value of data packets between two sequences. - the maximum TTL or Hop Limit value of data packets between two sequences. + the maximum TTL or Hop Limit value of data packets between two sequences. - the mean TTL or Hop Limit value of data packets between two sequences. + the mean TTL or Hop Limit value of data packets between two sequences. - the standard deviation of the TTL or Hop Limit value of data packets between two sequences. + the standard deviation of the TTL or Hop Limit value of data packets between two sequences. @@ -2572,42 +1807,29 @@ is set as zero, @lost_packets or @dup_packets will be zero. - - %TRUE if the report block is correctly parsed. + %TRUE if the report block is correctly parsed. - a valid XR #GstRTCPPacket which has VoIP Metrics Report Block. + a valid XR #GstRTCPPacket which has VoIP Metrics Report Block. - the fraction of RTP data packets within burst periods. + the fraction of RTP data packets within burst periods. - the fraction of RTP data packets within inter-burst gaps. + the fraction of RTP data packets within inter-burst gaps. - the mean duration(ms) of the burst periods. + the mean duration(ms) of the burst periods. - the mean duration(ms) of the gap periods. + the mean duration(ms) of the gap periods. @@ -2615,30 +1837,21 @@ is set as zero, @lost_packets or @dup_packets will be zero. - - %TRUE if the report block is correctly parsed. + %TRUE if the report block is correctly parsed. - a valid XR #GstRTCPPacket which has VoIP Metrics Report Block. + a valid XR #GstRTCPPacket which has VoIP Metrics Report Block. - the gap threshold. + the gap threshold. - the receiver configuration byte. + the receiver configuration byte. @@ -2646,30 +1859,21 @@ is set as zero, @lost_packets or @dup_packets will be zero. - - %TRUE if the report block is correctly parsed. + %TRUE if the report block is correctly parsed. - a valid XR #GstRTCPPacket which has VoIP Metrics Report Block. + a valid XR #GstRTCPPacket which has VoIP Metrics Report Block. - the most recently calculated round trip time between RTP interfaces(ms) + the most recently calculated round trip time between RTP interfaces(ms) - the most recently estimated end system delay(ms) + the most recently estimated end system delay(ms) @@ -2677,36 +1881,25 @@ is set as zero, @lost_packets or @dup_packets will be zero. - - %TRUE if the report block is correctly parsed. + %TRUE if the report block is correctly parsed. - a valid XR #GstRTCPPacket which has VoIP Metrics Report Block. + a valid XR #GstRTCPPacket which has VoIP Metrics Report Block. - the current nominal jitter buffer delay(ms) + the current nominal jitter buffer delay(ms) - the current maximum jitter buffer delay(ms) + the current maximum jitter buffer delay(ms) - the absolute maximum delay(ms) + the absolute maximum delay(ms) @@ -2714,24 +1907,17 @@ is set as zero, @lost_packets or @dup_packets will be zero. - - %TRUE if the report block is correctly parsed. + %TRUE if the report block is correctly parsed. - a valid XR #GstRTCPPacket which has VoIP Metrics Report Block. + a valid XR #GstRTCPPacket which has VoIP Metrics Report Block. - the SSRC of source + the SSRC of source @@ -2739,30 +1925,21 @@ is set as zero, @lost_packets or @dup_packets will be zero. - - %TRUE if the report block is correctly parsed. + %TRUE if the report block is correctly parsed. - a valid XR #GstRTCPPacket which has VoIP Metrics Report Block. + a valid XR #GstRTCPPacket which has VoIP Metrics Report Block. - the fraction of RTP data packets from the source lost. + the fraction of RTP data packets from the source lost. - the fraction of RTP data packets from the source that have been discarded. + the fraction of RTP data packets from the source that have been discarded. @@ -2770,42 +1947,29 @@ is set as zero, @lost_packets or @dup_packets will be zero. - - %TRUE if the report block is correctly parsed. + %TRUE if the report block is correctly parsed. - a valid XR #GstRTCPPacket which has VoIP Metrics Report Block. + a valid XR #GstRTCPPacket which has VoIP Metrics Report Block. - the R factor is a voice quality metric describing the segment of the call. + the R factor is a voice quality metric describing the segment of the call. - the external R factor is a voice quality metric. + the external R factor is a voice quality metric. - the estimated mean opinion score for listening quality. + the estimated mean opinion score for listening quality. - the estimated mean opinion score for conversational quality. + the estimated mean opinion score for conversational quality. @@ -2813,42 +1977,29 @@ is set as zero, @lost_packets or @dup_packets will be zero. - - %TRUE if the report block is correctly parsed. + %TRUE if the report block is correctly parsed. - a valid XR #GstRTCPPacket which has VoIP Metrics Report Block. + a valid XR #GstRTCPPacket which has VoIP Metrics Report Block. - the ratio of the signal level to a 0 dBm reference. + the ratio of the signal level to a 0 dBm reference. - the ratio of the silent period background noise level to a 0 dBm reference. + the ratio of the silent period background noise level to a 0 dBm reference. - the residual echo return loss value. + the residual echo return loss value. - the gap threshold. + the gap threshold. @@ -2856,21 +2007,14 @@ is set as zero, @lost_packets or @dup_packets will be zero. - Move to the next extended report block in XR @packet. - + Move to the next extended report block in XR @packet. - TRUE if there was a next extended report block. + TRUE if there was a next extended report block. - a valid XR #GstRTCPPacket + a valid XR #GstRTCPPacket @@ -2880,168 +2024,126 @@ is set as zero, @lost_packets or @dup_packets will be zero. glib:type-name="GstRTCPSDESType" glib:get-type="gst_rtcpsdes_type_get_type" c:type="GstRTCPSDESType"> - Different types of SDES content. + Different types of SDES content. - Invalid SDES entry + Invalid SDES entry - End of SDES list + End of SDES list - Canonical name + Canonical name - User name + User name - User's electronic mail address + User's electronic mail address - User's phone number + User's phone number - Geographic user location + Geographic user location - Name of application or tool + Name of application or tool - Notice about the source + Notice about the source - Private extensions + Private extensions - Different RTCP packet types. + Different RTCP packet types. - Invalid type + Invalid type - Sender report + Sender report - Receiver report + Receiver report - Source description + Source description - Goodbye + Goodbye - Application defined + Application defined - Transport layer feedback. + Transport layer feedback. - Payload-specific feedback. + Payload-specific feedback. - Extended report. + Extended report. glib:type-name="GstRTCPXRType" glib:get-type="gst_rtcpxr_type_get_type" c:type="GstRTCPXRType"> - Types of RTCP Extended Reports, those are defined in RFC 3611 and other RFCs + Types of RTCP Extended Reports, those are defined in RFC 3611 and other RFCs according to the [IANA registry](https://www.iana.org/assignments/rtcp-xr-block-types/rtcp-xr-block-types.xhtml). - Invalid XR Report Block + Invalid XR Report Block - Loss RLE Report Block + Loss RLE Report Block - Duplicate RLE Report Block + Duplicate RLE Report Block - Packet Receipt Times Report Block + Packet Receipt Times Report Block - Receiver Reference Time Report Block + Receiver Reference Time Report Block - Delay since the last Receiver Report + Delay since the last Receiver Report - Statistics Summary Report Block + Statistics Summary Report Block - VoIP Metrics Report Block + VoIP Metrics Report Block - The maximum amount of SSRCs in a BYE packet. - + The maximum amount of SSRCs in a BYE packet. - The maximum amount of Receiver report blocks in RR and SR messages. - + The maximum amount of Receiver report blocks in RR and SR messages. - The maximum text length for an SDES item. - + The maximum text length for an SDES item. - The maximum amount of SDES items. - + The maximum amount of SDES items. - Mask for version, padding bit and packet type pair allowing reduced size + Mask for version, padding bit and packet type pair allowing reduced size packets, basically it accepts other types than RR and SR - - Mask for version, padding bit and packet type pair - + Mask for version, padding bit and packet type pair - Valid value for the first two bytes of an RTCP packet after applying + Valid value for the first two bytes of an RTCP packet after applying #GST_RTCP_VALID_MASK to them. - - The supported RTCP version 2. - + The supported RTCP version 2. glib:type-name="GstRTPBaseAudioPayload" glib:get-type="gst_rtp_base_audio_payload_get_type" glib:type-struct="RTPBaseAudioPayloadClass"> - Provides a base class for audio RTP payloaders for frame or sample based + Provides a base class for audio RTP payloaders for frame or sample based audio codecs (constant bitrate) This class derives from GstRTPBasePayload. It can be used for payloading @@ -3225,227 +2283,162 @@ GstRTPBaseAudioPayload derives from GstRTPBasePayload, the child element must set any variables or call/override any functions required by that base class. The child element does not need to override any other functions specific to GstRTPBaseAudioPayload. - - Create an RTP buffer and store @payload_len bytes of the adapter as the + Create an RTP buffer and store @payload_len bytes of the adapter as the payload. Set the timestamp on the new buffer to @timestamp before pushing the buffer downstream. If @payload_len is -1, all pending bytes will be flushed. If @timestamp is -1, the timestamp will be calculated automatically. - - a #GstFlowReturn + a #GstFlowReturn - a #GstRTPBasePayload + a #GstRTPBasePayload - length of payload + length of payload - a #GstClockTime + a #GstClockTime - Gets the internal adapter used by the depayloader. - + Gets the internal adapter used by the depayloader. - a #GstAdapter. + a #GstAdapter. - a #GstRTPBaseAudioPayload + a #GstRTPBaseAudioPayload - Create an RTP buffer and store @payload_len bytes of @data as the + Create an RTP buffer and store @payload_len bytes of @data as the payload. Set the timestamp on the new buffer to @timestamp before pushing the buffer downstream. - - a #GstFlowReturn + a #GstFlowReturn - a #GstRTPBasePayload + a #GstRTPBasePayload - data to set as payload + data to set as payload - length of payload + length of payload - a #GstClockTime + a #GstClockTime - Tells #GstRTPBaseAudioPayload that the child element is for a frame based + Tells #GstRTPBaseAudioPayload that the child element is for a frame based audio codec - - a pointer to the element. + a pointer to the element. - Sets the options for frame based audio codecs. - + Sets the options for frame based audio codecs. - a pointer to the element. + a pointer to the element. - The duraction of an audio frame in milliseconds. + The duraction of an audio frame in milliseconds. - The size of an audio frame in bytes. + The size of an audio frame in bytes. - Tells #GstRTPBaseAudioPayload that the child element is for a sample based + Tells #GstRTPBaseAudioPayload that the child element is for a sample based audio codec - - a pointer to the element. + a pointer to the element. - Sets the options for sample based audio codecs. - + Sets the options for sample based audio codecs. - a pointer to the element. + a pointer to the element. - Size per sample in bytes. + Size per sample in bytes. - Sets the options for sample based audio codecs. - + Sets the options for sample based audio codecs. - a pointer to the element. + a pointer to the element. - Size per sample in bits. + Size per sample in bits. @@ -3473,7 +2466,7 @@ audio codec - + @@ -3481,18 +2474,13 @@ audio codec - Base class for audio RTP payloader. - + Base class for audio RTP payloader. - the parent class + the parent class - + @@ -3500,7 +2488,6 @@ audio codec - glib:type-name="GstRTPBaseDepayload" glib:get-type="gst_rtp_base_depayload_get_type" glib:type-struct="RTPBaseDepayloadClass"> - Provides a base class for RTP depayloaders - + Provides a base class for RTP depayloaders - @@ -3529,7 +2512,6 @@ audio codec - @@ -3543,7 +2525,6 @@ audio codec - @@ -3557,7 +2538,6 @@ audio codec - @@ -3571,7 +2551,6 @@ audio codec - @@ -3587,78 +2566,53 @@ audio codec - Queries whether #GstRTPSourceMeta will be added to depayloaded buffers. - + Queries whether #GstRTPSourceMeta will be added to depayloaded buffers. - %TRUE if source-info is enabled. + %TRUE if source-info is enabled. - a #GstRTPBaseDepayload + a #GstRTPBaseDepayload - Push @out_buf to the peer of @filter. This function takes ownership of + Push @out_buf to the peer of @filter. This function takes ownership of @out_buf. This function will by default apply the last incomming timestamp on the outgoing buffer when it didn't have a timestamp already. - - a #GstFlowReturn. + a #GstFlowReturn. - a #GstRTPBaseDepayload + a #GstRTPBaseDepayload - a #GstBuffer + a #GstBuffer - Push @out_list to the peer of @filter. This function takes ownership of + Push @out_list to the peer of @filter. This function takes ownership of @out_list. - - a #GstFlowReturn. + a #GstFlowReturn. - a #GstRTPBaseDepayload + a #GstRTPBaseDepayload - a #GstBufferList + a #GstBufferList @@ -3666,24 +2620,17 @@ the outgoing buffer when it didn't have a timestamp already. - Enable or disable adding #GstRTPSourceMeta to depayloaded buffers. - + Enable or disable adding #GstRTPSourceMeta to depayloaded buffers. - a #GstRTPBaseDepayload + a #GstRTPBaseDepayload - whether to add meta about RTP sources to buffer + whether to add meta about RTP sources to buffer @@ -3692,15 +2639,11 @@ the outgoing buffer when it didn't have a timestamp already. version="1.16" writable="1" transfer-ownership="none"> - Add RTP source information found in RTP header as meta to output buffer. + Add RTP source information found in RTP header as meta to output buffer. - Various depayloader statistics retrieved atomically (and are therefore + Various depayloader statistics retrieved atomically (and are therefore synchroized with each other). This property return a GstStructure named application/x-rtp-depayload-stats containing the following fields relating to the last processed buffer and current state of the stream being depayloaded: @@ -3741,7 +2684,7 @@ the last processed buffer and current state of the stream being depayloaded: c:type="GstRTPBaseDepayloadPrivate*"/> - + @@ -3749,19 +2692,13 @@ the last processed buffer and current state of the stream being depayloaded: - Base class for RTP depayloaders. - + Base class for RTP depayloaders. - the parent class + the parent class - @@ -3777,7 +2714,6 @@ the last processed buffer and current state of the stream being depayloaded: - @@ -3793,7 +2729,6 @@ the last processed buffer and current state of the stream being depayloaded: - @@ -3809,7 +2744,6 @@ the last processed buffer and current state of the stream being depayloaded: - @@ -3825,7 +2759,6 @@ the last processed buffer and current state of the stream being depayloaded: - @@ -3840,7 +2773,7 @@ the last processed buffer and current state of the stream being depayloaded: - + @@ -3848,7 +2781,6 @@ the last processed buffer and current state of the stream being depayloaded: - - Provides a base class for RTP payloaders - + Provides a base class for RTP payloaders - @@ -3880,7 +2808,6 @@ the last processed buffer and current state of the stream being depayloaded: - @@ -3894,7 +2821,6 @@ the last processed buffer and current state of the stream being depayloaded: - @@ -3911,7 +2837,6 @@ the last processed buffer and current state of the stream being depayloaded: - @@ -3925,7 +2850,6 @@ the last processed buffer and current state of the stream being depayloaded: - @@ -3939,7 +2863,6 @@ the last processed buffer and current state of the stream being depayloaded: - @@ -3955,43 +2878,30 @@ the last processed buffer and current state of the stream being depayloaded: - Allocate a new #GstBuffer with enough data to hold an RTP packet with + Allocate a new #GstBuffer with enough data to hold an RTP packet with minimum @csrc_count CSRCs, a payload length of @payload_len and padding of @pad_len. If @payload has #GstRTPBasePayload:source-info %TRUE additional CSRCs may be allocated and filled with RTP source information. - - A newly allocated buffer that can hold an RTP packet with given + A newly allocated buffer that can hold an RTP packet with given parameters. - a #GstRTPBasePayload + a #GstRTPBasePayload - the length of the payload + the length of the payload - the amount of padding + the amount of padding - the minimum number of CSRC entries + the minimum number of CSRC entries @@ -3999,63 +2909,43 @@ parameters. - Count the total number of RTP sources found in the meta of @buffer, which + Count the total number of RTP sources found in the meta of @buffer, which will be automically added by gst_rtp_base_payload_allocate_output_buffer(). If #GstRTPBasePayload:source-info is %FALSE the count will be 0. - - The number of sources. + The number of sources. - a #GstRTPBasePayload + a #GstRTPBasePayload - a #GstBuffer, typically the buffer to payload + a #GstBuffer, typically the buffer to payload - Check if the packet with @size and @duration would exceed the configured + Check if the packet with @size and @duration would exceed the configured maximum size. - - %TRUE if the packet of @size and @duration would exceed the + %TRUE if the packet of @size and @duration would exceed the configured MTU or max_ptime. - a #GstRTPBasePayload + a #GstRTPBasePayload - the size of the packet + the size of the packet - the duration of the packet + the duration of the packet @@ -4063,124 +2953,86 @@ configured MTU or max_ptime. - Queries whether the payloader will add contributing sources (CSRCs) to the + Queries whether the payloader will add contributing sources (CSRCs) to the RTP header from #GstRTPSourceMeta. - - %TRUE if source-info is enabled. + %TRUE if source-info is enabled. - a #GstRTPBasePayload + a #GstRTPBasePayload - Push @buffer to the peer element of the payloader. The SSRC, payload type, + Push @buffer to the peer element of the payloader. The SSRC, payload type, seqnum and timestamp of the RTP buffer will be updated first. This function takes ownership of @buffer. - - a #GstFlowReturn. + a #GstFlowReturn. - a #GstRTPBasePayload + a #GstRTPBasePayload - a #GstBuffer + a #GstBuffer - Push @list to the peer element of the payloader. The SSRC, payload type, + Push @list to the peer element of the payloader. The SSRC, payload type, seqnum and timestamp of the RTP buffer will be updated first. This function takes ownership of @list. - - a #GstFlowReturn. + a #GstFlowReturn. - a #GstRTPBasePayload + a #GstRTPBasePayload - a #GstBufferList + a #GstBufferList - Set the rtp options of the payloader. These options will be set in the caps + Set the rtp options of the payloader. These options will be set in the caps of the payloader. Subclasses must call this method before calling gst_rtp_base_payload_push() or gst_rtp_base_payload_set_outcaps(). - - a #GstRTPBasePayload + a #GstRTPBasePayload - the media type (typically "audio" or "video") + the media type (typically "audio" or "video") - if the payload type is dynamic + if the payload type is dynamic - the encoding name + the encoding name - the clock rate of the media + the clock rate of the media @@ -4188,36 +3040,25 @@ gst_rtp_base_payload_push() or gst_rtp_base_payload_set_outcaps(). - Configure the output caps with the optional parameters. + Configure the output caps with the optional parameters. Variable arguments should be in the form field name, field type (as a GType), value(s). The last variable argument should be NULL. - - %TRUE if the caps could be set. + %TRUE if the caps could be set. - a #GstRTPBasePayload + a #GstRTPBasePayload - the first field name or %NULL + the first field name or %NULL - field values + field values @@ -4225,25 +3066,18 @@ Variable arguments should be in the form field name, field type - Enable or disable adding contributing sources to RTP packets from + Enable or disable adding contributing sources to RTP packets from #GstRTPSourceMeta. - - a #GstRTPBasePayload + a #GstRTPBasePayload - whether to add contributing sources to RTP packets + whether to add contributing sources to RTP packets @@ -4252,9 +3086,7 @@ Variable arguments should be in the form field name, field type - Minimum duration of the packet data in ns (can't go above MTU) + Minimum duration of the packet data in ns (can't go above MTU) @@ -4264,16 +3096,12 @@ Variable arguments should be in the form field name, field type version="1.16" writable="1" transfer-ownership="none"> - Make the payloader timestamp packets according to the Rate-Control=no + Make the payloader timestamp packets according to the Rate-Control=no behaviour specified in the ONVIF replay spec. - Try to use the offset fields to generate perfect RTP timestamps. When this + Try to use the offset fields to generate perfect RTP timestamps. When this option is disabled, RTP timestamps are generated from GST_BUFFER_PTS of each payloaded buffer. The PTSes of buffers may not necessarily increment with the amount of data in each input buffer, consider e.g. the case where @@ -4294,9 +3122,7 @@ timestamps for audio streams. - Force buffers to be multiples of this duration in ns (0 disables) + Force buffers to be multiples of this duration in ns (0 disables) @@ -4309,9 +3135,7 @@ timestamps for audio streams. version="1.16" writable="1" transfer-ownership="none"> - Enable writing the CSRC field in allocated RTP header based on RTP source + Enable writing the CSRC field in allocated RTP header based on RTP source information found in the input buffer's #GstRTPSourceMeta. @@ -4319,9 +3143,7 @@ information found in the input buffer's #GstRTPSourceMeta. - Various payloader statistics retrieved atomically (and are therefore + Various payloader statistics retrieved atomically (and are therefore synchroized with each other), these can be used e.g. to generate an RTP-Info header. This property return a GstStructure named application/x-rtp-payload-stats containing the following fields relating to @@ -4413,7 +3235,7 @@ the last processed buffer and current state of the stream being payloaded: - + @@ -4421,19 +3243,13 @@ the last processed buffer and current state of the stream being payloaded: - Base class for audio RTP payloader. - + Base class for audio RTP payloader. - the parent class + the parent class - @@ -4452,7 +3268,6 @@ the last processed buffer and current state of the stream being payloaded: - @@ -4468,7 +3283,6 @@ the last processed buffer and current state of the stream being payloaded: - @@ -4484,7 +3298,6 @@ the last processed buffer and current state of the stream being payloaded: - @@ -4500,7 +3313,6 @@ the last processed buffer and current state of the stream being payloaded: - @@ -4516,7 +3328,6 @@ the last processed buffer and current state of the stream being payloaded: - @@ -4534,7 +3345,7 @@ the last processed buffer and current state of the stream being payloaded: - + @@ -4542,214 +3353,149 @@ the last processed buffer and current state of the stream being payloaded: - - The GstRTPBuffer helper functions makes it easy to parse and create regular + The GstRTPBuffer helper functions makes it easy to parse and create regular #GstBuffer objects that contain RTP payloads. These buffers are typically of 'application/x-rtp' #GstCaps. - - pointer to RTP buffer + pointer to RTP buffer - internal state + internal state - array of data - + array of data + - array of size - + array of size + - array of #GstMapInfo - + array of #GstMapInfo + - Adds a RFC 5285 header extension with a one byte header to the end of the + Adds a RFC 5285 header extension with a one byte header to the end of the RTP header. If there is already a RFC 5285 header extension with a one byte header, the new extension will be appended. It will not work if there is already a header extension that does not follow the mecanism described in RFC 5285 or if there is a header extension with a two bytes header as described in RFC 5285. In that case, use gst_rtp_buffer_add_extension_twobytes_header() - - %TRUE if header extension could be added + %TRUE if header extension could be added - the RTP packet + the RTP packet - The ID of the header extension (between 1 and 14). + The ID of the header extension (between 1 and 14). - location for data + location for data - the size of the data in bytes + the size of the data in bytes - Adds a RFC 5285 header extension with a two bytes header to the end of the + Adds a RFC 5285 header extension with a two bytes header to the end of the RTP header. If there is already a RFC 5285 header extension with a two bytes header, the new extension will be appended. It will not work if there is already a header extension that does not follow the mecanism described in RFC 5285 or if there is a header extension with a one byte header as described in RFC 5285. In that case, use gst_rtp_buffer_add_extension_onebyte_header() - - %TRUE if header extension could be added + %TRUE if header extension could be added - the RTP packet + the RTP packet - Application specific bits + Application specific bits - The ID of the header extension + The ID of the header extension - location for data + location for data - the size of the data in bytes + the size of the data in bytes - Get the CSRC at index @idx in @buffer. - + Get the CSRC at index @idx in @buffer. - the CSRC at index @idx in host order. + the CSRC at index @idx in host order. - the RTP packet + the RTP packet - the index of the CSRC to get + the index of the CSRC to get - Get the CSRC count of the RTP packet in @buffer. - + Get the CSRC count of the RTP packet in @buffer. - the CSRC count of @buffer. + the CSRC count of @buffer. - the RTP packet + the RTP packet - Check if the extension bit is set on the RTP packet in @buffer. - + Check if the extension bit is set on the RTP packet in @buffer. - TRUE if @buffer has the extension bit set. + TRUE if @buffer has the extension bit set. - the RTP packet + the RTP packet @@ -4758,9 +3504,7 @@ gst_rtp_buffer_add_extension_onebyte_header() c:identifier="gst_rtp_buffer_get_extension_bytes" shadows="get_extension_data" version="1.2"> - Similar to gst_rtp_buffer_get_extension_data, but more suitable for language + Similar to gst_rtp_buffer_get_extension_data, but more suitable for language bindings usage. @bits will contain the extension 16 bits of custom data and the extension data (not including the extension header) is placed in a new #GBytes structure. @@ -4768,28 +3512,21 @@ the extension data (not including the extension header) is placed in a new If @rtp did not contain an extension, this function will return %NULL, with @bits unchanged. If there is an extension header but no extension data then an empty #GBytes will be returned. - - A new #GBytes if an extension header was present + A new #GBytes if an extension header was present and %NULL otherwise. - the RTP packet + the RTP packet - location for header bits + location for header bits @@ -4798,44 +3535,33 @@ and %NULL otherwise. c:identifier="gst_rtp_buffer_get_extension_data" shadowed-by="get_extension_bytes" introspectable="0"> - Get the extension data. @bits will contain the extension 16 bits of custom + Get the extension data. @bits will contain the extension 16 bits of custom data. @data will point to the data in the extension and @wordlen will contain the length of @data in 32 bits words. If @buffer did not contain an extension, this function will return %FALSE with @bits, @data and @wordlen unchanged. - - TRUE if @buffer had the extension bit set. + TRUE if @buffer had the extension bit set. - the RTP packet + the RTP packet - location for result bits + location for result bits - location for data + location for data @@ -4844,52 +3570,37 @@ with @bits, @data and @wordlen unchanged. direction="out" caller-allocates="0" transfer-ownership="full"> - location for length of @data in 32 bits words + location for length of @data in 32 bits words - Parses RFC 5285 style header extensions with a one byte header. It will + Parses RFC 5285 style header extensions with a one byte header. It will return the nth extension with the requested id. - - TRUE if @buffer had the requested header extension + TRUE if @buffer had the requested header extension - the RTP packet + the RTP packet - The ID of the header extension to be read (between 1 and 14). + The ID of the header extension to be read (between 1 and 14). - Read the nth extension packet with the requested ID + Read the nth extension packet with the requested ID - + location for data @@ -4899,61 +3610,44 @@ return the nth extension with the requested id. direction="out" caller-allocates="0" transfer-ownership="full"> - the size of the data in bytes + the size of the data in bytes - Parses RFC 5285 style header extensions with a two bytes header. It will + Parses RFC 5285 style header extensions with a two bytes header. It will return the nth extension with the requested id. - - TRUE if @buffer had the requested header extension + TRUE if @buffer had the requested header extension - the RTP packet + the RTP packet - Application specific bits + Application specific bits - The ID of the header extension to be read (between 1 and 14). + The ID of the header extension to be read (between 1 and 14). - Read the nth extension packet with the requested ID + Read the nth extension packet with the requested ID - + location for data @@ -4963,92 +3657,62 @@ return the nth extension with the requested id. direction="out" caller-allocates="0" transfer-ownership="full"> - the size of the data in bytes + the size of the data in bytes - Return the total length of the header in @buffer. This include the length of + Return the total length of the header in @buffer. This include the length of the fixed header, the CSRC list and the extension header. - - The total length of the header in @buffer. + The total length of the header in @buffer. - the RTP packet + the RTP packet - Check if the marker bit is set on the RTP packet in @buffer. - + Check if the marker bit is set on the RTP packet in @buffer. - TRUE if @buffer has the marker bit set. + TRUE if @buffer has the marker bit set. - the RTP packet + the RTP packet - Return the total length of the packet in @buffer. - + Return the total length of the packet in @buffer. - The total length of the packet in @buffer. + The total length of the packet in @buffer. - the RTP packet + the RTP packet - Check if the padding bit is set on the RTP packet in @buffer. - + Check if the padding bit is set on the RTP packet in @buffer. - TRUE if @buffer has the padding bit set. + TRUE if @buffer has the padding bit set. - the RTP packet + the RTP packet @@ -5057,15 +3721,10 @@ the fixed header, the CSRC list and the extension header. c:identifier="gst_rtp_buffer_get_payload" shadowed-by="get_payload_bytes" introspectable="0"> - Get a pointer to the payload data in @buffer. This pointer is valid as long + Get a pointer to the payload data in @buffer. This pointer is valid as long as a reference to @buffer is held. - - A pointer + A pointer to the payload data in @buffer. @@ -5073,32 +3732,23 @@ to the payload data in @buffer. - the RTP packet + the RTP packet - Create a buffer of the payload of the RTP packet in @buffer. This function + Create a buffer of the payload of the RTP packet in @buffer. This function will internally create a subbuffer of @buffer so that a memcpy can be avoided. - - A new buffer with the data of the payload. + A new buffer with the data of the payload. - the RTP packet + the RTP packet @@ -5107,688 +3757,475 @@ avoided. c:identifier="gst_rtp_buffer_get_payload_bytes" shadows="get_payload" version="1.2"> - Similar to gst_rtp_buffer_get_payload, but more suitable for language + Similar to gst_rtp_buffer_get_payload, but more suitable for language bindings usage. The return value is a pointer to a #GBytes structure containing the payload data in @rtp. - - A new #GBytes containing the payload data in @rtp. + A new #GBytes containing the payload data in @rtp. - the RTP packet + the RTP packet - Get the length of the payload of the RTP packet in @buffer. - + Get the length of the payload of the RTP packet in @buffer. - The length of the payload in @buffer. + The length of the payload in @buffer. - the RTP packet + the RTP packet - Create a subbuffer of the payload of the RTP packet in @buffer. @offset bytes + Create a subbuffer of the payload of the RTP packet in @buffer. @offset bytes are skipped in the payload and the subbuffer will be of size @len. If @len is -1 the total payload starting from @offset is subbuffered. - - A new buffer with the specified data of the payload. + A new buffer with the specified data of the payload. - the RTP packet + the RTP packet - the offset in the payload + the offset in the payload - the length in the payload + the length in the payload - Get the payload type of the RTP packet in @buffer. - + Get the payload type of the RTP packet in @buffer. - The payload type. + The payload type. - the RTP packet + the RTP packet - Get the sequence number of the RTP packet in @buffer. - + Get the sequence number of the RTP packet in @buffer. - The sequence number in host order. + The sequence number in host order. - the RTP packet + the RTP packet - Get the SSRC of the RTP packet in @buffer. - + Get the SSRC of the RTP packet in @buffer. - the SSRC of @buffer in host order. + the SSRC of @buffer in host order. - the RTP packet + the RTP packet - Get the timestamp of the RTP packet in @buffer. - + Get the timestamp of the RTP packet in @buffer. - The timestamp in host order. + The timestamp in host order. - the RTP packet + the RTP packet - Get the version number of the RTP packet in @buffer. - + Get the version number of the RTP packet in @buffer. - The version of @buffer. + The version of @buffer. - the RTP packet + the RTP packet - Set the amount of padding in the RTP packet in @buffer to + Set the amount of padding in the RTP packet in @buffer to @len. If @len is 0, the padding is removed. NOTE: This function does not work correctly. - - the RTP packet + the RTP packet - the new amount of padding + the new amount of padding - Modify the CSRC at index @idx in @buffer to @csrc. - + Modify the CSRC at index @idx in @buffer to @csrc. - the RTP packet + the RTP packet - the CSRC index to set + the CSRC index to set - the CSRC in host order to set at @idx + the CSRC in host order to set at @idx - Set the extension bit on the RTP packet in @buffer to @extension. - + Set the extension bit on the RTP packet in @buffer to @extension. - the RTP packet + the RTP packet - the new extension + the new extension - Set the extension bit of the rtp buffer and fill in the @bits and @length of the + Set the extension bit of the rtp buffer and fill in the @bits and @length of the extension header. If the existing extension data is not large enough, it will be made larger. - - True if done. + True if done. - the RTP packet + the RTP packet - the bits specific for the extension + the bits specific for the extension - the length that counts the number of 32-bit words in + the length that counts the number of 32-bit words in the extension, excluding the extension header ( therefore zero is a valid length) - Set the marker bit on the RTP packet in @buffer to @marker. - + Set the marker bit on the RTP packet in @buffer to @marker. - the RTP packet + the RTP packet - the new marker + the new marker - Set the total @rtp size to @len. The data in the buffer will be made + Set the total @rtp size to @len. The data in the buffer will be made larger if needed. Any padding will be removed from the packet. - - the RTP packet + the RTP packet - the new packet length + the new packet length - Set the padding bit on the RTP packet in @buffer to @padding. - + Set the padding bit on the RTP packet in @buffer to @padding. - the buffer + the buffer - the new padding + the new padding - Set the payload type of the RTP packet in @buffer to @payload_type. - + Set the payload type of the RTP packet in @buffer to @payload_type. - the RTP packet + the RTP packet - the new type + the new type - Set the sequence number of the RTP packet in @buffer to @seq. - + Set the sequence number of the RTP packet in @buffer to @seq. - the RTP packet + the RTP packet - the new sequence number + the new sequence number - Set the SSRC on the RTP packet in @buffer to @ssrc. - + Set the SSRC on the RTP packet in @buffer to @ssrc. - the RTP packet + the RTP packet - the new SSRC + the new SSRC - Set the timestamp of the RTP packet in @buffer to @timestamp. - + Set the timestamp of the RTP packet in @buffer to @timestamp. - the RTP packet + the RTP packet - the new timestamp + the new timestamp - Set the version of the RTP packet in @buffer to @version. - + Set the version of the RTP packet in @buffer to @version. - the RTP packet + the RTP packet - the new version + the new version - Unmap @rtp previously mapped with gst_rtp_buffer_map(). - + Unmap @rtp previously mapped with gst_rtp_buffer_map(). - a #GstRTPBuffer + a #GstRTPBuffer - Allocate enough data in @buffer to hold an RTP packet with @csrc_count CSRCs, + Allocate enough data in @buffer to hold an RTP packet with @csrc_count CSRCs, a payload length of @payload_len and padding of @pad_len. @buffer must be writable and all previous memory in @buffer will be freed. If @pad_len is >0, the padding bit will be set. All other RTP header fields will be set to 0/FALSE. - - a #GstBuffer + a #GstBuffer - the length of the payload + the length of the payload - the amount of padding + the amount of padding - the number of CSRC entries + the number of CSRC entries - Calculate the header length of an RTP packet with @csrc_count CSRC entries. + Calculate the header length of an RTP packet with @csrc_count CSRC entries. An RTP packet can have at most 15 CSRC entries. - - The length of an RTP header with @csrc_count CSRC entries. + The length of an RTP header with @csrc_count CSRC entries. - the number of CSRC entries + the number of CSRC entries - Calculate the total length of an RTP packet with a payload size of @payload_len, + Calculate the total length of an RTP packet with a payload size of @payload_len, a padding of @pad_len and a @csrc_count CSRC entries. - - The total length of an RTP header with given parameters. + The total length of an RTP header with given parameters. - the length of the payload + the length of the payload - the amount of padding + the amount of padding - the number of CSRC entries + the number of CSRC entries - Calculate the length of the payload of an RTP packet with size @packet_len, + Calculate the length of the payload of an RTP packet with size @packet_len, a padding of @pad_len and a @csrc_count CSRC entries. - - The length of the payload of an RTP packet with given parameters. + The length of the payload of an RTP packet with given parameters. - the length of the total RTP packet + the length of the total RTP packet - the amount of padding + the amount of padding - the number of CSRC entries + the number of CSRC entries - Compare two sequence numbers, taking care of wraparounds. This function + Compare two sequence numbers, taking care of wraparounds. This function returns the difference between @seqnum1 and @seqnum2. - - a negative value if @seqnum1 is bigger than @seqnum2, 0 if they + a negative value if @seqnum1 is bigger than @seqnum2, 0 if they are equal or a positive value if @seqnum1 is smaller than @segnum2. - a sequence number + a sequence number - a sequence number + a sequence number - Get the default clock-rate for the static payload type @payload_type. - + Get the default clock-rate for the static payload type @payload_type. - the default clock rate or -1 if the payload type is not static or + the default clock rate or -1 if the payload type is not static or the clock-rate is undefined. - the static payload type + the static payload type - Update the @exttimestamp field with the extended timestamp of @timestamp + Update the @exttimestamp field with the extended timestamp of @timestamp For the first call of the method, @exttimestamp should point to a location with a value of -1. @@ -5796,11 +4233,8 @@ This function is able to handle both forward and backward timestamps taking into account: - timestamp wraparound making sure that the returned value is properly increased. - timestamp unwraparound making sure that the returned value is properly decreased. - - The extended timestamp of @timestamp or 0 if the result can't go anywhere backwards. + The extended timestamp of @timestamp or 0 if the result can't go anywhere backwards. @@ -5808,185 +4242,130 @@ into account: direction="inout" caller-allocates="0" transfer-ownership="full"> - a previous extended timestamp + a previous extended timestamp - a new timestamp + a new timestamp - Map the contents of @buffer into @rtp. - + Map the contents of @buffer into @rtp. - %TRUE if @buffer could be mapped. + %TRUE if @buffer could be mapped. - a #GstBuffer + a #GstBuffer - #GstMapFlags + #GstMapFlags - a #GstRTPBuffer + a #GstRTPBuffer - Allocate a new #GstBuffer with enough data to hold an RTP packet with + Allocate a new #GstBuffer with enough data to hold an RTP packet with @csrc_count CSRCs, a payload length of @payload_len and padding of @pad_len. All other RTP header fields will be set to 0/FALSE. - - A newly allocated buffer that can hold an RTP packet with given + A newly allocated buffer that can hold an RTP packet with given parameters. - the length of the payload + the length of the payload - the amount of padding + the amount of padding - the number of CSRC entries + the number of CSRC entries - Create a new #GstBuffer that can hold an RTP packet that is exactly + Create a new #GstBuffer that can hold an RTP packet that is exactly @packet_len long. The length of the payload depends on @pad_len and @csrc_count and can be calculated with gst_rtp_buffer_calc_payload_len(). All RTP header fields will be set to 0/FALSE. - - A newly allocated buffer that can hold an RTP packet of @packet_len. + A newly allocated buffer that can hold an RTP packet of @packet_len. - the total length of the packet + the total length of the packet - the amount of padding + the amount of padding - the number of CSRC entries + the number of CSRC entries - Create a new buffer and set the data to a copy of @len + Create a new buffer and set the data to a copy of @len bytes of @data and the size to @len. The data will be freed when the buffer is freed. - - A newly allocated buffer with a copy of @data and of size @len. + A newly allocated buffer with a copy of @data and of size @len. - data for the new + data for the new buffer - the length of data + the length of data - Create a new buffer and set the data and size of the buffer to @data and @len + Create a new buffer and set the data and size of the buffer to @data and @len respectively. @data will be freed when the buffer is unreffed, so this function transfers ownership of @data to the new buffer. - - A newly allocated buffer with @data and of size @len. + A newly allocated buffer with @data and of size @len. - + data for the new buffer - the length of data + the length of data @@ -5997,9 +4376,7 @@ function transfers ownership of @data to the new buffer. glib:type-name="GstRTPBufferFlags" glib:get-type="gst_rtp_buffer_flags_get_type" c:type="GstRTPBufferFlags"> - Additional RTP buffer flags. These flags can potentially be used on any + Additional RTP buffer flags. These flags can potentially be used on any buffers carrying RTP packets. Note that these are only valid for #GstCaps of type: application/x-rtp (x-rtcp). @@ -6008,18 +4385,14 @@ They can conflict with other extended buffer flags. value="1048576" c:identifier="GST_RTP_BUFFER_FLAG_RETRANSMISSION" glib:nick="retransmission"> - The #GstBuffer was once wrapped + The #GstBuffer was once wrapped in a retransmitted packet as specified by RFC 4588. - The packet represents redundant RTP packet. + The packet represents redundant RTP packet. The flag is used in gstrtpstorage to be able to hold the packetback and use it only for recovery from packet loss. Since: 1.14 @@ -6028,9 +4401,7 @@ They can conflict with other extended buffer flags. value="268435456" c:identifier="GST_RTP_BUFFER_FLAG_LAST" glib:nick="last"> - Offset to define more flags. + Offset to define more flags. glib:type-name="GstRTPBufferMapFlags" glib:get-type="gst_rtp_buffer_map_flags_get_type" c:type="GstRTPBufferMapFlags"> - Additional mapping flags for gst_rtp_buffer_map(). + Additional mapping flags for gst_rtp_buffer_map(). - Skip mapping and validation of RTP + Skip mapping and validation of RTP padding and RTP pad count when present. Useful for buffers where the padding may be encrypted. @@ -6055,18 +4422,14 @@ They can conflict with other extended buffer flags. value="16777216" c:identifier="GST_RTP_BUFFER_MAP_FLAG_LAST" glib:nick="last"> - Offset to define more flags + Offset to define more flags - Standard predefined fixed payload types. + Standard predefined fixed payload types. The official list is at: http://www.iana.org/assignments/rtp-parameters @@ -6082,307 +4445,224 @@ Reserved for RTCP conflict avoidance: 72-76 value="0" c:identifier="GST_RTP_PAYLOAD_PCMU" glib:nick="pcmu"> - ITU-T G.711. mu-law audio (RFC 3551) + ITU-T G.711. mu-law audio (RFC 3551) - RFC 3551 says reserved + RFC 3551 says reserved - RFC 3551 says reserved + RFC 3551 says reserved - GSM audio + GSM audio - ITU G.723.1 audio + ITU G.723.1 audio - IMA ADPCM wave type (RFC 3551) + IMA ADPCM wave type (RFC 3551) - IMA ADPCM wave type (RFC 3551) + IMA ADPCM wave type (RFC 3551) - experimental linear predictive encoding + experimental linear predictive encoding - ITU-T G.711 A-law audio (RFC 3551) + ITU-T G.711 A-law audio (RFC 3551) - ITU-T G.722 (RFC 3551) + ITU-T G.722 (RFC 3551) - stereo PCM + stereo PCM - mono PCM + mono PCM - EIA & TIA standard IS-733 + EIA & TIA standard IS-733 - Comfort Noise (RFC 3389) + Comfort Noise (RFC 3389) - Audio MPEG 1-3. + Audio MPEG 1-3. - ITU-T G.728 Speech coder (RFC 3551) + ITU-T G.728 Speech coder (RFC 3551) - IMA ADPCM wave type (RFC 3551) + IMA ADPCM wave type (RFC 3551) - IMA ADPCM wave type (RFC 3551) + IMA ADPCM wave type (RFC 3551) - ITU-T G.729 Speech coder (RFC 3551) + ITU-T G.729 Speech coder (RFC 3551) - See RFC 2029 + See RFC 2029 - ISO Standards 10918-1 and 10918-2 (RFC 2435) + ISO Standards 10918-1 and 10918-2 (RFC 2435) - nv encoding by Ron Frederick + nv encoding by Ron Frederick - ITU-T Recommendation H.261 (RFC 2032) + ITU-T Recommendation H.261 (RFC 2032) - Video MPEG 1 & 2 (RFC 2250) + Video MPEG 1 & 2 (RFC 2250) - MPEG-2 transport stream (RFC 2250) + MPEG-2 transport stream (RFC 2250) - Video H263 (RFC 2190) + Video H263 (RFC 2190) - Structure holding default payload type information. - + Structure holding default payload type information. - payload type, -1 means dynamic + payload type, -1 means dynamic - the media type(s), usually "audio", "video", "application", "text", + the media type(s), usually "audio", "video", "application", "text", "message". - the encoding name of @pt + the encoding name of @pt - default clock rate, 0 = unknown/variable + default clock rate, 0 = unknown/variable - encoding parameters. For audio this is the number of + encoding parameters. For audio this is the number of channels. NULL = not applicable. - the bitrate of the media. 0 = unknown/variable. + the bitrate of the media. 0 = unknown/variable. - + - Get the #GstRTPPayloadInfo for @media and @encoding_name. This function is + Get the #GstRTPPayloadInfo for @media and @encoding_name. This function is mostly used to get the default clock-rate and bandwidth for dynamic payload types specified with @media and @encoding name. The search for @encoding_name will be performed in a case insensitve way. - - a #GstRTPPayloadInfo or NULL when no info could be found. + a #GstRTPPayloadInfo or NULL when no info could be found. - the media to find + the media to find - the encoding name to find + the encoding name to find - Get the #GstRTPPayloadInfo for @payload_type. This function is + Get the #GstRTPPayloadInfo for @payload_type. This function is mostly used to get the default clock-rate and bandwidth for static payload types specified with @payload_type. - - a #GstRTPPayloadInfo or NULL when no info could be found. + a #GstRTPPayloadInfo or NULL when no info could be found. - the payload_type to find + the payload_type to find @@ -6393,117 +4673,81 @@ types specified with @payload_type. glib:type-name="GstRTPProfile" glib:get-type="gst_rtp_profile_get_type" c:type="GstRTPProfile"> - The transfer profile to use. + The transfer profile to use. - invalid profile + invalid profile - the Audio/Visual profile (RFC 3551) + the Audio/Visual profile (RFC 3551) - the secure Audio/Visual profile (RFC 3711) + the secure Audio/Visual profile (RFC 3711) - the Audio/Visual profile with feedback (RFC 4585) + the Audio/Visual profile with feedback (RFC 4585) - the secure Audio/Visual profile with feedback (RFC 5124) + the secure Audio/Visual profile with feedback (RFC 5124) - Meta describing the source(s) of the buffer. - + Meta describing the source(s) of the buffer. - parent #GstMeta + parent #GstMeta - the SSRC + the SSRC - whether @ssrc is set and valid + whether @ssrc is set and valid - pointer to the CSRCs - + pointer to the CSRCs + - number of elements in @csrc + number of elements in @csrc - Appends @csrc to the list of contributing sources in @meta. - + Appends @csrc to the list of contributing sources in @meta. - %TRUE if all elements in @csrc was added, %FALSE otherwise. + %TRUE if all elements in @csrc was added, %FALSE otherwise. - a #GstRTPSourceMeta + a #GstRTPSourceMeta - the csrcs to append + the csrcs to append - number of elements in @csrc + number of elements in @csrc @@ -6511,21 +4755,14 @@ types specified with @payload_type. - Count the total number of RTP sources found in @meta, both SSRC and CSRC. - + Count the total number of RTP sources found in @meta, both SSRC and CSRC. - The number of RTP sources + The number of RTP sources - a #GstRTPSourceMeta + a #GstRTPSourceMeta @@ -6533,36 +4770,26 @@ types specified with @payload_type. - Sets @ssrc in @meta. If @ssrc is %NULL the ssrc of @meta will be unset. - + Sets @ssrc in @meta. If @ssrc is %NULL the ssrc of @meta will be unset. - %TRUE on success, %FALSE otherwise. + %TRUE on success, %FALSE otherwise. - a #GstRTPSourceMeta + a #GstRTPSourceMeta - pointer to the SSRC + pointer to the SSRC - @@ -6571,294 +4798,237 @@ types specified with @payload_type. - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - The supported RTP version 2. - + The supported RTP version 2. - Attaches RTP source information to @buffer. - + Attaches RTP source information to @buffer. - the #GstRTPSourceMeta on @buffer. + the #GstRTPSourceMeta on @buffer. - a #GstBuffer + a #GstBuffer - pointer to the SSRC + pointer to the SSRC - pointer to the CSRCs + pointer to the CSRCs - number of elements in @csrc + number of elements in @csrc @@ -6866,22 +5036,15 @@ types specified with @payload_type. - Find the #GstRTPSourceMeta on @buffer. - + Find the #GstRTPSourceMeta on @buffer. - the #GstRTPSourceMeta or %NULL when there + the #GstRTPSourceMeta or %NULL when there is no such metadata on @buffer. - a #GstBuffer + a #GstBuffer @@ -6889,31 +5052,22 @@ is no such metadata on @buffer. - Open @buffer for reading or writing, depending on @flags. The resulting RTCP + Open @buffer for reading or writing, depending on @flags. The resulting RTCP buffer state is stored in @rtcp. - - a buffer with an RTCP packet + a buffer with an RTCP packet - flags for the mapping + flags for the mapping - resulting #GstRTCPBuffer + resulting #GstRTCPBuffer @@ -6921,22 +5075,15 @@ buffer state is stored in @rtcp. - Create a new buffer for constructing RTCP packets. The packet will have a + Create a new buffer for constructing RTCP packets. The packet will have a maximum size of @mtu. - - A newly allocated buffer. + A newly allocated buffer. - the maximum mtu size. + the maximum mtu size. @@ -6944,31 +5091,22 @@ maximum size of @mtu. - Create a new buffer and set the data to a copy of @len + Create a new buffer and set the data to a copy of @len bytes of @data and the size to @len. The data will be freed when the buffer is freed. - - A newly allocated buffer with a copy of @data and of size @len. + A newly allocated buffer with a copy of @data and of size @len. - data for the new buffer + data for the new buffer - the length of data + the length of data @@ -6976,31 +5114,22 @@ is freed. - Create a new buffer and set the data and size of the buffer to @data and @len + Create a new buffer and set the data and size of the buffer to @data and @len respectively. @data will be freed when the buffer is unreffed, so this function transfers ownership of @data to the new buffer. - - A newly allocated buffer with @data and of size @len. + A newly allocated buffer with @data and of size @len. - data for the new buffer + data for the new buffer - the length of data + the length of data @@ -7008,22 +5137,15 @@ function transfers ownership of @data to the new buffer. - Check if the data pointed to by @buffer is a valid RTCP packet using + Check if the data pointed to by @buffer is a valid RTCP packet using gst_rtcp_buffer_validate_data(). - - TRUE if @buffer is a valid RTCP packet. + TRUE if @buffer is a valid RTCP packet. - the buffer to validate + the buffer to validate @@ -7031,32 +5153,23 @@ gst_rtcp_buffer_validate_data(). - Check if the @data and @size point to the data of a valid compound, + Check if the @data and @size point to the data of a valid compound, non-reduced size RTCP packet. Use this function to validate a packet before using the other functions in this module. - - TRUE if the data points to a valid RTCP packet. + TRUE if the data points to a valid RTCP packet. - the data to validate + the data to validate - the length of @data to validate + the length of @data to validate @@ -7065,35 +5178,26 @@ this module. c:identifier="gst_rtcp_buffer_validate_data_reduced" moved-to="RTCPBuffer.validate_data_reduced" version="1.6"> - Check if the @data and @size point to the data of a valid RTCP packet. + Check if the @data and @size point to the data of a valid RTCP packet. Use this function to validate a packet before using the other functions in this module. This function is updated to support reduced size rtcp packets according to RFC 5506 and will validate full compound RTCP packets as well as reduced size RTCP packets. - - TRUE if the data points to a valid RTCP packet. + TRUE if the data points to a valid RTCP packet. - the data to validate + the data to validate - the length of @data to validate + the length of @data to validate @@ -7102,114 +5206,79 @@ size RTCP packets. c:identifier="gst_rtcp_buffer_validate_reduced" moved-to="RTCPBuffer.validate_reduced" version="1.6"> - Check if the data pointed to by @buffer is a valid RTCP packet using + Check if the data pointed to by @buffer is a valid RTCP packet using gst_rtcp_buffer_validate_reduced(). - - TRUE if @buffer is a valid RTCP packet. + TRUE if @buffer is a valid RTCP packet. - the buffer to validate + the buffer to validate - Converts an NTP time to UNIX nanoseconds. @ntptime can typically be + Converts an NTP time to UNIX nanoseconds. @ntptime can typically be the NTP time of an SR RTCP message and contains, in the upper 32 bits, the number of seconds since 1900 and, in the lower 32 bits, the fractional seconds. The resulting value will be the number of nanoseconds since 1970. - - the UNIX time for @ntptime in nanoseconds. + the UNIX time for @ntptime in nanoseconds. - an NTP timestamp + an NTP timestamp - Convert @name into a @GstRTCPSDESType. @name is typically a key in a + Convert @name into a @GstRTCPSDESType. @name is typically a key in a #GstStructure containing SDES items. - - the #GstRTCPSDESType for @name or #GST_RTCP_SDES_PRIV when @name + the #GstRTCPSDESType for @name or #GST_RTCP_SDES_PRIV when @name is a private sdes item. - a SDES name + a SDES name - Converts @type to the string equivalent. The string is typically used as a + Converts @type to the string equivalent. The string is typically used as a key in a #GstStructure containing SDES items. - - the string equivalent of @type + the string equivalent of @type - a #GstRTCPSDESType + a #GstRTCPSDESType - Converts a UNIX timestamp in nanoseconds to an NTP time. The caller should + Converts a UNIX timestamp in nanoseconds to an NTP time. The caller should pass a value with nanoseconds since 1970. The NTP time will, in the upper 32 bits, contain the number of seconds since 1900 and, in the lower 32 bits, the fractional seconds. The resulting value can be used as an ntptime for constructing SR RTCP packets. - - the NTP time for @unixtime. + the NTP time for @unixtime. - an UNIX timestamp in nanoseconds + an UNIX timestamp in nanoseconds @@ -7217,40 +5286,29 @@ for constructing SR RTCP packets. - Allocate enough data in @buffer to hold an RTP packet with @csrc_count CSRCs, + Allocate enough data in @buffer to hold an RTP packet with @csrc_count CSRCs, a payload length of @payload_len and padding of @pad_len. @buffer must be writable and all previous memory in @buffer will be freed. If @pad_len is >0, the padding bit will be set. All other RTP header fields will be set to 0/FALSE. - - a #GstBuffer + a #GstBuffer - the length of the payload + the length of the payload - the amount of padding + the amount of padding - the number of CSRC entries + the number of CSRC entries @@ -7258,22 +5316,15 @@ will be set to 0/FALSE. - Calculate the header length of an RTP packet with @csrc_count CSRC entries. + Calculate the header length of an RTP packet with @csrc_count CSRC entries. An RTP packet can have at most 15 CSRC entries. - - The length of an RTP header with @csrc_count CSRC entries. + The length of an RTP header with @csrc_count CSRC entries. - the number of CSRC entries + the number of CSRC entries @@ -7281,34 +5332,23 @@ An RTP packet can have at most 15 CSRC entries. - Calculate the total length of an RTP packet with a payload size of @payload_len, + Calculate the total length of an RTP packet with a payload size of @payload_len, a padding of @pad_len and a @csrc_count CSRC entries. - - The total length of an RTP header with given parameters. + The total length of an RTP header with given parameters. - the length of the payload + the length of the payload - the amount of padding + the amount of padding - the number of CSRC entries + the number of CSRC entries @@ -7316,34 +5356,23 @@ a padding of @pad_len and a @csrc_count CSRC entries. - Calculate the length of the payload of an RTP packet with size @packet_len, + Calculate the length of the payload of an RTP packet with size @packet_len, a padding of @pad_len and a @csrc_count CSRC entries. - - The length of the payload of an RTP packet with given parameters. + The length of the payload of an RTP packet with given parameters. - the length of the total RTP packet + the length of the total RTP packet - the amount of padding + the amount of padding - the number of CSRC entries + the number of CSRC entries @@ -7351,29 +5380,20 @@ a padding of @pad_len and a @csrc_count CSRC entries. - Compare two sequence numbers, taking care of wraparounds. This function + Compare two sequence numbers, taking care of wraparounds. This function returns the difference between @seqnum1 and @seqnum2. - - a negative value if @seqnum1 is bigger than @seqnum2, 0 if they + a negative value if @seqnum1 is bigger than @seqnum2, 0 if they are equal or a positive value if @seqnum1 is smaller than @segnum2. - a sequence number + a sequence number - a sequence number + a sequence number @@ -7381,22 +5401,15 @@ are equal or a positive value if @seqnum1 is smaller than @segnum2. - Get the default clock-rate for the static payload type @payload_type. - + Get the default clock-rate for the static payload type @payload_type. - the default clock rate or -1 if the payload type is not static or + the default clock rate or -1 if the payload type is not static or the clock-rate is undefined. - the static payload type + the static payload type @@ -7404,9 +5417,7 @@ the clock-rate is undefined. - Update the @exttimestamp field with the extended timestamp of @timestamp + Update the @exttimestamp field with the extended timestamp of @timestamp For the first call of the method, @exttimestamp should point to a location with a value of -1. @@ -7414,11 +5425,8 @@ This function is able to handle both forward and backward timestamps taking into account: - timestamp wraparound making sure that the returned value is properly increased. - timestamp unwraparound making sure that the returned value is properly decreased. - - The extended timestamp of @timestamp or 0 if the result can't go anywhere backwards. + The extended timestamp of @timestamp or 0 if the result can't go anywhere backwards. @@ -7426,15 +5434,11 @@ into account: direction="inout" caller-allocates="0" transfer-ownership="full"> - a previous extended timestamp + a previous extended timestamp - a new timestamp + a new timestamp @@ -7442,36 +5446,25 @@ into account: - Map the contents of @buffer into @rtp. - + Map the contents of @buffer into @rtp. - %TRUE if @buffer could be mapped. + %TRUE if @buffer could be mapped. - a #GstBuffer + a #GstBuffer - #GstMapFlags + #GstMapFlags - a #GstRTPBuffer + a #GstRTPBuffer @@ -7479,36 +5472,25 @@ into account: - Allocate a new #GstBuffer with enough data to hold an RTP packet with + Allocate a new #GstBuffer with enough data to hold an RTP packet with @csrc_count CSRCs, a payload length of @payload_len and padding of @pad_len. All other RTP header fields will be set to 0/FALSE. - - A newly allocated buffer that can hold an RTP packet with given + A newly allocated buffer that can hold an RTP packet with given parameters. - the length of the payload + the length of the payload - the amount of padding + the amount of padding - the number of CSRC entries + the number of CSRC entries @@ -7516,36 +5498,25 @@ parameters. - Create a new #GstBuffer that can hold an RTP packet that is exactly + Create a new #GstBuffer that can hold an RTP packet that is exactly @packet_len long. The length of the payload depends on @pad_len and @csrc_count and can be calculated with gst_rtp_buffer_calc_payload_len(). All RTP header fields will be set to 0/FALSE. - - A newly allocated buffer that can hold an RTP packet of @packet_len. + A newly allocated buffer that can hold an RTP packet of @packet_len. - the total length of the packet + the total length of the packet - the amount of padding + the amount of padding - the number of CSRC entries + the number of CSRC entries @@ -7553,32 +5524,23 @@ All RTP header fields will be set to 0/FALSE. - Create a new buffer and set the data to a copy of @len + Create a new buffer and set the data to a copy of @len bytes of @data and the size to @len. The data will be freed when the buffer is freed. - - A newly allocated buffer with a copy of @data and of size @len. + A newly allocated buffer with a copy of @data and of size @len. - data for the new + data for the new buffer - the length of data + the length of data @@ -7586,125 +5548,89 @@ is freed. - Create a new buffer and set the data and size of the buffer to @data and @len + Create a new buffer and set the data and size of the buffer to @data and @len respectively. @data will be freed when the buffer is unreffed, so this function transfers ownership of @data to the new buffer. - - A newly allocated buffer with @data and of size @len. + A newly allocated buffer with @data and of size @len. - + data for the new buffer - the length of data + the length of data - Reads the NTP time from the @size NTP-56 extension bytes in @data and store the + Reads the NTP time from the @size NTP-56 extension bytes in @data and store the result in @ntptime. - - %TRUE on success. + %TRUE on success. - the data to read from + the data to read from - the size of @data + the size of @data - the result NTP time + the result NTP time - Reads the NTP time from the @size NTP-64 extension bytes in @data and store the + Reads the NTP time from the @size NTP-64 extension bytes in @data and store the result in @ntptime. - - %TRUE on success. + %TRUE on success. - the data to read from + the data to read from - the size of @data + the size of @data - the result NTP time + the result NTP time - Writes the NTP time in @ntptime to the format required for the NTP-56 header + Writes the NTP time in @ntptime to the format required for the NTP-56 header extension. @data must hold at least #GST_RTP_HDREXT_NTP_56_SIZE bytes. - - %TRUE on success. + %TRUE on success. @@ -7712,36 +5638,25 @@ extension. @data must hold at least #GST_RTP_HDREXT_NTP_56_SIZE bytes. transfer-ownership="none" nullable="1" allow-none="1"> - the data to write to + the data to write to - the size of @data + the size of @data - the NTP time + the NTP time - Writes the NTP time in @ntptime to the format required for the NTP-64 header + Writes the NTP time in @ntptime to the format required for the NTP-64 header extension. @data must hold at least #GST_RTP_HDREXT_NTP_64_SIZE bytes. - - %TRUE on success. + %TRUE on success. @@ -7749,21 +5664,15 @@ extension. @data must hold at least #GST_RTP_HDREXT_NTP_64_SIZE bytes. transfer-ownership="none" nullable="1" allow-none="1"> - the data to write to + the data to write to - the size of @data + the size of @data - the NTP time + the NTP time @@ -7771,31 +5680,22 @@ extension. @data must hold at least #GST_RTP_HDREXT_NTP_64_SIZE bytes. - Get the #GstRTPPayloadInfo for @media and @encoding_name. This function is + Get the #GstRTPPayloadInfo for @media and @encoding_name. This function is mostly used to get the default clock-rate and bandwidth for dynamic payload types specified with @media and @encoding name. The search for @encoding_name will be performed in a case insensitve way. - - a #GstRTPPayloadInfo or NULL when no info could be found. + a #GstRTPPayloadInfo or NULL when no info could be found. - the media to find + the media to find - the encoding name to find + the encoding name to find @@ -7803,30 +5703,22 @@ The search for @encoding_name will be performed in a case insensitve way. - Get the #GstRTPPayloadInfo for @payload_type. This function is + Get the #GstRTPPayloadInfo for @payload_type. This function is mostly used to get the default clock-rate and bandwidth for static payload types specified with @payload_type. - - a #GstRTPPayloadInfo or NULL when no info could be found. + a #GstRTPPayloadInfo or NULL when no info could be found. - the payload_type to find + the payload_type to find - @@ -7834,7 +5726,6 @@ types specified with @payload_type. - diff --git a/gir-files/GstVideo-1.0.gir b/gir-files/GstVideo-1.0.gir index dea5c0caa..69aa65ff5 100644 --- a/gir-files/GstVideo-1.0.gir +++ b/gir-files/GstVideo-1.0.gir @@ -1498,7 +1498,9 @@ Performs the multiplication, meta->matrix X matrix. a 4x4 transformation matrix to be applied - + + + @@ -2069,12 +2071,15 @@ the parity check bits). Some know types of Ancillary Data identifiers. - + CEA 708 Ancillary data according to SMPTE 334 - + CEA 608 Ancillary data according to SMPTE 334 + + AFD/Bar Ancillary data according to SMPTE 2016-3 (Since: 1.18) + Additional video buffer flags. These flags can potentially be used on any diff --git a/gir-files/GstWebRTC-1.0.gir b/gir-files/GstWebRTC-1.0.gir index f3874aea5..74f1119ca 100644 --- a/gir-files/GstWebRTC-1.0.gir +++ b/gir-files/GstWebRTC-1.0.gir @@ -257,17 +257,17 @@ See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate">h glib:type-name="GstWebRTCFECType" glib:get-type="gst_webrtc_fec_type_get_type" c:type="GstWebRTCFECType"> - GST_WEBRTC_FEC_TYPE_NONE: none -GST_WEBRTC_FEC_TYPE_ULP_RED: ulpfec + red + none + ulpfec + red