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 of 'application/x-rtcp' #GstCaps. 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(). 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 if the max mtu is exceeded for the buffer. a valid RTCP buffer the #GstRTCPType of the new packet pointer to new packet Initialize a new #GstRTCPPacket pointer that points to the first packet in @rtcp. TRUE if the packet existed in @rtcp. a valid RTCP buffer a #GstRTCPPacket Get the number of RTCP packets in @rtcp. the number of RTCP packets in @rtcp. a valid RTCP buffer 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 Open @buffer for reading or writing, depending on @flags. The resulting RTCP buffer state is stored in @rtcp. a buffer with an RTCP packet flags for the mapping resulting #GstRTCPBuffer Create a new buffer for constructing RTCP packets. The packet will have a maximum size of @mtu. A newly allocated buffer. the maximum mtu size. 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. data for the new buffer the length of data 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. data for the new buffer the length of data 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. the buffer to validate 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. the data to validate the length of @data to validate 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. the data to validate the length of @data to validate 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. the buffer to validate Different types of feedback messages. Invalid type Generic NACK Temporary Maximum Media Stream Bit Rate Request Temporary Maximum Media Stream Bit Rate Notification Request an SR packet for early synchronization Picture Loss Indication Slice Loss Indication Reference Picture Selection Indication Application layer Feedback Full Intra Request Command Temporal-Spatial Trade-off Request Temporal-Spatial Trade-off Notification Video Back Channel Message 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 offset of packet in buffer data 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. a valid SR or RR #GstRTCPPacket profile-specific data length of the profile-specific data in bytes Add a new report block to @packet with the given values. %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 data source being reported fraction lost since last SR/RR the cumululative number of packets lost the extended last sequence number received the interarrival jitter the last SR packet from this source the delay since last SR packet Get the application-dependent data attached to a RTPFB or PSFB @packet. A pointer to the data a valid APP #GstRTCPPacket Get the length of the application-dependent data attached to an APP @packet. The length of data in 32-bit words. a valid APP #GstRTCPPacket Get the name field of the APP @packet. The 4-byte name field, not zero-terminated. a valid APP #GstRTCPPacket Get the SSRC/CSRC field of the APP @packet. The SSRC/CSRC. a valid APP #GstRTCPPacket Get the subtype field of the APP @packet. The subtype. a valid APP #GstRTCPPacket 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 data. a valid APP #GstRTCPPacket Length of the data in 32-bit words Set the name field of the APP @packet. a valid APP #GstRTCPPacket 4-byte ASCII name Set the SSRC/CSRC field of the APP @packet. a valid APP #GstRTCPPacket SSRC/CSRC of the packet Set the subtype field of the APP @packet. a valid APP #GstRTCPPacket subtype of the packet Add @ssrc to the BYE @packet. %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 an SSRC to add Adds @len SSRCs in @ssrc to BYE @packet. %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 an array of SSRCs to add number of elements in @ssrc Get the @nth SSRC of the BYE @packet. The @nth SSRC of @packet. a valid BYE #GstRTCPPacket the nth SSRC to get Get the reason in @packet. 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 Get the length of the reason string. The length of the reason string or 0 when there is no reason string present. a valid BYE #GstRTCPPacket Get the number of SSRC fields in @packet. The number of SSRC fields in @packet. a valid BYE #GstRTCPPacket Set the reason string to @reason in @packet. TRUE if the string could be set. a valid BYE #GstRTCPPacket a reason string 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. a valid SR or RR #GstRTCPPacket result profile-specific data length of the profile-specific extension data Get the Feedback Control Information attached to a RTPFB or PSFB @packet. a pointer to the FCI a valid RTPFB or PSFB #GstRTCPPacket Get the length of the Feedback Control Information attached to a RTPFB or PSFB @packet. The length of the FCI in 32-bit words. a valid RTPFB or PSFB #GstRTCPPacket Get the media SSRC field of the RTPFB or PSFB @packet. the media SSRC. a valid RTPFB or PSFB #GstRTCPPacket Get the sender SSRC field of the RTPFB or PSFB @packet. the sender SSRC. a valid RTPFB or PSFB #GstRTCPPacket Get the feedback message type of the FB @packet. The feedback message type. a valid RTPFB or PSFB #GstRTCPPacket 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 a valid RTPFB or PSFB #GstRTCPPacket Length of the FCI in 32-bit words Set the media SSRC field of the RTPFB or PSFB @packet. a valid RTPFB or PSFB #GstRTCPPacket a media SSRC Set the sender SSRC field of the RTPFB or PSFB @packet. a valid RTPFB or PSFB #GstRTCPPacket a sender SSRC Set the feedback message type of the FB @packet. a valid RTPFB or PSFB #GstRTCPPacket the #GstRTCPFBType to set Get the count field in @packet. The count field in @packet or -1 if @packet does not point to a valid packet. a valid #GstRTCPPacket Get the length field of @packet. This is the length of the packet in 32-bit words minus one. The length field of @packet. a valid #GstRTCPPacket Get the packet padding of the packet pointed to by @packet. If the packet has the padding bit set. a valid #GstRTCPPacket %TRUE if there was valid data. a valid SR or RR #GstRTCPPacket result profile-specific data result length of the profile-specific data The number of 32-bit words containing profile-specific extension data from @packet. a valid SR or RR #GstRTCPPacket Parse the values of the @nth report block in @packet and store the result in the values. a valid SR or RR #GstRTCPPacket the nth report block in @packet result for data source being reported result for fraction lost since last SR/RR result for the cumululative number of packets lost result for the extended last sequence number received result for the interarrival jitter result for the last SR packet from this source result for the delay since last SR packet Get the number of report blocks in @packet. The number of report blocks in @packet. a valid SR or RR #GstRTCPPacket Get the packet type of the packet pointed to by @packet. The packet type or GST_RTCP_TYPE_INVALID when @packet is not pointing to a valid packet. a valid #GstRTCPPacket 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 function. a #GstRTCPPacket 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 function. a #GstRTCPPacket Get the ssrc field of the RR @packet. the ssrc. a valid RR #GstRTCPPacket Set the ssrc field of the RR @packet. a valid RR #GstRTCPPacket the SSRC to set Add a new SDES entry to the current item in @packet. %TRUE if the item could be added, %FALSE if the MTU has been reached. a valid SDES #GstRTCPPacket the #GstRTCPSDESType of the SDES entry the data length the data Add a new SDES item for @ssrc to @packet. %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 the SSRC of the new item to add 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. a valid SDES #GstRTCPPacket result of the entry type result length of the entry data result entry data Move to the first SDES entry in the current item. %TRUE if there was a first entry. a valid SDES #GstRTCPPacket Move to the first SDES item in @packet. TRUE if there was a first item. a valid SDES #GstRTCPPacket 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. a valid SDES #GstRTCPPacket result of the entry type result length of the entry data result entry data Get the number of items in the SDES packet @packet. The number of items in @packet. a valid SDES #GstRTCPPacket Get the SSRC of the current SDES item. the SSRC of the current item. a valid SDES #GstRTCPPacket Move to the next SDES entry in the current item. %TRUE if there was a next entry. a valid SDES #GstRTCPPacket Move to the next SDES item in @packet. TRUE if there was a next item. a valid SDES #GstRTCPPacket Set the @nth new report block in @packet with the given values. Note: Not implemented. a valid SR or RR #GstRTCPPacket the nth report block to set data source being reported fraction lost since last SR/RR the cumululative number of packets lost the extended last sequence number received the interarrival jitter the last SR packet from this source the delay since last SR packet Parse the SR sender info and store the values. a valid SR #GstRTCPPacket result SSRC result NTP time result RTP time result packet count result octet count Set the given values in the SR packet @packet. a valid SR #GstRTCPPacket the SSRC the NTP time the RTP time the packet count the octet count Move to the first extended report block in XR @packet. TRUE if there was a first extended report block. a valid XR #GstRTCPPacket The number of 32-bit words containing type-specific block data from @packet. a valid XR #GstRTCPPacket Get the extended report block type of the XR @packet. The extended report block type. a valid XR #GstRTCPPacket Parse the extended report block for DLRR report block type. %TRUE if the report block is correctly parsed. a valid XR #GstRTCPPacket which has DLRR Report Block. the index of sub-block to retrieve. the SSRC of the receiver. the last receiver reference timestamp of @ssrc. the delay since @last_rr. Retrieve the packet receipt time of @seq which ranges in [begin_seq, end_seq). %TRUE if the report block returns the receipt time correctly. a valid XR #GstRTCPPacket which has the Packet Recept Times Report Block. the sequence to retrieve the time. the packet receipt time of @seq. Parse the Packet Recept Times Report Block from a XR @packet %TRUE if the report block is correctly parsed. 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 amount of thinning performed on the sequence number space. the first sequence number that this block reports on. the last sequence number that this block reports on plus one. Parse the extended report block for Loss RLE and Duplicated LRE block type. %TRUE if the report block is correctly parsed. 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 amount of thinning performed on the sequence number space. the first sequence number that this block reports on. the last sequence number that this block reports on plus one. the number of chunks calculated by block length. Retrieve actual chunk data. %TRUE if the report block returns chunk correctly. a valid XR #GstRTCPPacket which is Loss RLE or Duplicate RLE report. the index of chunk to retrieve. the @nth chunk. %TRUE if the report block returns the reference time correctly. a valid XR #GstRTCPPacket which has the Receiver Reference Time. NTP timestamp Get the ssrc field of the XR @packet. the ssrc. a valid XR #GstRTCPPacket Extract a basic information from static summary report block of XR @packet. %TRUE if the report block is correctly parsed. a valid XR #GstRTCPPacket which has Statics Summary Report Block. the SSRC of the source. the first sequence number that this block reports on. the last sequence number that this block reports on plus one. 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. a valid XR #GstRTCPPacket which has Statics Summary Report Block. the minimum relative transit time between two sequences. the maximum 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. 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. a valid XR #GstRTCPPacket which has Statics Summary Report Block. the number of lost packets between begin_seq and end_seq. the number of duplicate packets between begin_seq and end_seq. Extract the value of ttl for ipv4, or hop limit for ipv6. %TRUE if the report block is correctly parsed. 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 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 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. %TRUE if the report block is correctly parsed. 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 inter-burst gaps. the mean duration(ms) of the burst periods. the mean duration(ms) of the gap periods. %TRUE if the report block is correctly parsed. a valid XR #GstRTCPPacket which has VoIP Metrics Report Block. the gap threshold. the receiver configuration byte. %TRUE if the report block is correctly parsed. a valid XR #GstRTCPPacket which has VoIP Metrics Report Block. the most recently calculated round trip time between RTP interfaces(ms) the most recently estimated end system delay(ms) %TRUE if the report block is correctly parsed. a valid XR #GstRTCPPacket which has VoIP Metrics Report Block. the current nominal jitter buffer delay(ms) the current maximum jitter buffer delay(ms) the absolute maximum delay(ms) %TRUE if the report block is correctly parsed. a valid XR #GstRTCPPacket which has VoIP Metrics Report Block. the SSRC of source %TRUE if the report block is correctly parsed. 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 that have been discarded. %TRUE if the report block is correctly parsed. 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 external R factor is a voice quality metric. the estimated mean opinion score for listening quality. the estimated mean opinion score for conversational quality. %TRUE if the report block is correctly parsed. 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 silent period background noise level to a 0 dBm reference. the residual echo return loss value. the gap threshold. Move to the next extended report block in XR @packet. TRUE if there was a next extended report block. a valid XR #GstRTCPPacket Different types of SDES content. Invalid SDES entry End of SDES list Canonical name User name User's electronic mail address User's phone number Geographic user location Name of application or tool Notice about the source Private extensions H.323 callable address Application Specific Identifier (RFC6776) Reporting Group Identifier (RFC8861) RtpStreamId SDES item (RFC8852). RepairedRtpStreamId SDES item (RFC8852). CLUE CaptId (RFC8849) MID SDES item (RFC8843). Different RTCP packet types. Invalid type Sender report Receiver report Source description Goodbye Application defined Transport layer feedback. Payload-specific feedback. Extended report. 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 Loss RLE Report Block Duplicate RLE Report Block Packet Receipt Times Report Block Receiver Reference Time Report Block Delay since the last Receiver Report Statistics Summary Report Block VoIP Metrics Report Block The maximum amount of SSRCs in a BYE packet. The maximum amount of Receiver report blocks in RR and SR messages. The maximum text length for an SDES item. The maximum amount of SDES items. Mask for version 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 Valid value for the first two bytes of an RTCP packet after applying #GST_RTCP_VALID_MASK to them. The supported RTCP version 2. 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 audio codecs. It will only work with constant bitrate codecs. It supports both frame based and sample based codecs. It takes care of packing up the audio data into RTP packets and filling up the headers accordingly. The payloading is done based on the maximum MTU (mtu) and the maximum time per packet (max-ptime). The general idea is to divide large data buffers into smaller RTP packets. The RTP packet size is the minimum of either the MTU, max-ptime (if set) or available data. The RTP packet size is always larger or equal to min-ptime (if set). If min-ptime is not set, any residual data is sent in a last RTP packet. In the case of frame based codecs, the resulting RTP packets always contain full frames. ## Usage To use this base class, your child element needs to call either gst_rtp_base_audio_payload_set_frame_based() or gst_rtp_base_audio_payload_set_sample_based(). This is usually done in the element's `_init()` function. Then, the child element must call either gst_rtp_base_audio_payload_set_frame_options(), gst_rtp_base_audio_payload_set_sample_options() or gst_rtp_base_audio_payload_set_samplebits_options. Since 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 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 #GstRTPBasePayload length of payload a #GstClockTime Gets the internal adapter used by the depayloader. a #GstAdapter. a #GstRTPBaseAudioPayload 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 #GstRTPBasePayload data to set as payload length of payload a #GstClockTime Tells #GstRTPBaseAudioPayload that the child element is for a frame based audio codec a pointer to the element. Sets the options for frame based audio codecs. a pointer to the element. The duraction of an audio frame in milliseconds. The size of an audio frame in bytes. Tells #GstRTPBaseAudioPayload that the child element is for a sample based audio codec a pointer to the element. Sets the options for sample based audio codecs. a pointer to the element. Size per sample in bytes. Sets the options for sample based audio codecs. a pointer to the element. Size per sample in bits. Base class for audio RTP payloader. the parent class Provides a base class for RTP depayloaders In order to handle RTP header extensions correctly if the depayloader aggregates multiple RTP packet payloads into one output buffer this class provides the function gst_rtp_base_depayload_set_aggregate_hdrext_enabled(). If the aggregation is enabled the virtual functions @GstRTPBaseDepayload.process or @GstRTPBaseDepayload.process_rtp_packet must tell the base class what happens to the current RTP packet. By default the base class assumes that the packet payload is used with the next output buffer. If the RTP packet will not be used with an output buffer gst_rtp_base_depayload_dropped() must be called. A typical situation would be if we are waiting for a keyframe. If the RTP packet will be used but not with the current output buffer but with the next one gst_rtp_base_depayload_delayed() must be called. This may happen if the current RTP packet signals the start of a new output buffer and the currently processed output buffer will be pushed first. The undelay happens implicitly once the current buffer has been pushed or gst_rtp_base_depayload_flush() has been called. If gst_rtp_base_depayload_flush() is called all RTP packets that have not been dropped since the last output buffer are dropped, e.g. if an output buffer is discarded due to malformed data. This may or may not include the current RTP packet depending on the 2nd parameter @keep_current. Be aware that in case gst_rtp_base_depayload_push_list() is used each buffer will see the same list of RTP header extensions. Called from @GstRTPBaseDepayload.process or @GstRTPBaseDepayload.process_rtp_packet when the depayloader needs to keep the current input RTP header for use with the next output buffer. The delayed buffer will remain until the end of processing the current output buffer and then enqueued for processing with the next output buffer. A typical use-case is when the depayloader implementation will start a new output buffer for the current input RTP buffer but push the current output buffer first. Must be called with the stream lock held. a #GstRTPBaseDepayload Called from @GstRTPBaseDepayload.process or @GstRTPBaseDepayload.process_rtp_packet if the depayloader does not use the current buffer for the output buffer. This will either drop the delayed buffer or the last buffer from the header extension cache. A typical use-case is when the depayloader implementation is dropping an input RTP buffer while waiting for the first keyframe. Must be called with the stream lock held. a #GstRTPBaseDepayload If @GstRTPBaseDepayload.process or @GstRTPBaseDepayload.process_rtp_packet drop an output buffer this function tells the base class to flush header extension cache as well. This will not drop an input RTP header marked as delayed from gst_rtp_base_depayload_delayed(). If @keep_current is %TRUE the current input RTP header will be kept and enqueued after flushing the previous input RTP headers. A typical use-case for @keep_current is when the depayloader implementation invalidates the current output buffer and starts a new one with the current RTP input buffer. Must be called with the stream lock held. a #GstRTPBaseDepayload if the current RTP buffer shall be kept Queries whether header extensions will be aggregated per depayloaded buffers. %TRUE if aggregate-header-extension is enabled. a #GstRTPBaseDepayload Queries whether #GstRTPSourceMeta will be added to depayloaded buffers. %TRUE if source-info is enabled. a #GstRTPBaseDepayload Push @out_buf to the peer of @filter. This function takes ownership of @out_buf. This function will by default apply the last incoming timestamp on the outgoing buffer when it didn't have a timestamp already. a #GstFlowReturn. a #GstRTPBaseDepayload a #GstBuffer Push @out_list to the peer of @filter. This function takes ownership of @out_list. a #GstFlowReturn. a #GstRTPBaseDepayload a #GstBufferList Enable or disable aggregating header extensions. a #GstRTPBaseDepayload whether to aggregate header extensions per output buffer Enable or disable adding #GstRTPSourceMeta to depayloaded buffers. a #GstRTPBaseDepayload whether to add meta about RTP sources to buffer If enabled, the depayloader will automatically try to enable all the RTP header extensions provided in the sink caps, saving the application the need to handle these extensions manually using the GstRTPBaseDepayload::request-extension: signal. Max seqnum reorder before the sender is assumed to have restarted. When max-reorder is set to 0 all reordered/duplicate packets are considered coming from a restarted sender. Add RTP source information found in RTP header as meta to output buffer. 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: * `clock-rate`: #G_TYPE_UINT, clock-rate of the stream * `npt-start`: #G_TYPE_UINT64, time of playback start * `npt-stop`: #G_TYPE_UINT64, time of playback stop * `play-speed`: #G_TYPE_DOUBLE, the playback speed * `play-scale`: #G_TYPE_DOUBLE, the playback scale * `running-time-dts`: #G_TYPE_UINT64, the last running-time of the last DTS * `running-time-pts`: #G_TYPE_UINT64, the last running-time of the last PTS * `seqnum`: #G_TYPE_UINT, the last seen seqnum * `timestamp`: #G_TYPE_UINT, the last seen RTP timestamp Add @ext as an extension for reading part of an RTP header extension from incoming RTP packets. the #GstRTPHeaderExtension Clear all RTP header extensions used by this depayloader. The returned @ext must be configured with the correct @ext_id and with the necessary attributes as required by the extension implementation. the #GstRTPHeaderExtension for @ext_id, or %NULL the extension id being requested the extension URI being requested Base class for RTP depayloaders. the parent class Provides a base class for RTP payloaders 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 parameters. a #GstRTPBasePayload the length of the payload the amount of padding the minimum number of CSRC entries 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. a #GstRTPBasePayload a #GstBuffer, typically the buffer to payload 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 configured MTU or max_ptime. a #GstRTPBasePayload the size of the packet the duration of the packet Queries whether the payloader will add contributing sources (CSRCs) to the RTP header from #GstRTPSourceMeta. %TRUE if source-info is enabled. a #GstRTPBasePayload 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 #GstRTPBasePayload a #GstBuffer 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 #GstRTPBasePayload a #GstBufferList 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 the media type (typically "audio" or "video") if the payload type is dynamic the encoding name the clock rate of the media 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. a #GstRTPBasePayload the first field name or %NULL field values Configure the output caps with the optional fields. %TRUE if the caps could be set. a #GstRTPBasePayload a #GstStructure with the caps fields Enable or disable adding contributing sources to RTP packets from #GstRTPSourceMeta. a #GstRTPBasePayload whether to add contributing sources to RTP packets If enabled, the payloader will automatically try to enable all the RTP header extensions provided in the src caps, saving the application the need to handle these extensions manually using the GstRTPBasePayload::request-extension: signal. Minimum duration of the packet data in ns (can't go above MTU) 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 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 the buffer arrives from a network which means that the PTS is unrelated to the amount of data. Because the RTP timestamps are generated from GST_BUFFER_PTS this can result in RTP timestamps that also don't increment with the amount of data in the payloaded packet. To circumvent this it is possible to set the perfect rtptime option enabled. When this option is enabled the payloader will increment the RTP timestamps based on GST_BUFFER_OFFSET which relates to the amount of data in each packet rather than the GST_BUFFER_PTS of each buffer and therefore the RTP timestamps will more closely correlate with the amount of data in each buffer. Currently GstRTPBasePayload is limited to handling perfect RTP timestamps for audio streams. Force buffers to be multiples of this duration in ns (0 disables) Make the RTP packets' timestamps be scaled with the segment's rate (corresponding to RTSP speed parameter). Disabling this property means the timestamps will not be affected by the set delivery speed (RTSP speed). Example: A server wants to allow streaming a recorded video in double speed but still have the timestamps correspond to the position in the video. This is achieved by the client setting RTSP Speed to 2 while the server has this property disabled. Enable writing the CSRC field in allocated RTP header based on RTP source information found in the input buffer's #GstRTPSourceMeta. 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 the last processed buffer and current state of the stream being payloaded: * `clock-rate` :#G_TYPE_UINT, clock-rate of the stream * `running-time` :#G_TYPE_UINT64, running time * `seqnum` :#G_TYPE_UINT, sequence number, same as #GstRTPBasePayload:seqnum * `timestamp` :#G_TYPE_UINT, RTP timestamp, same as #GstRTPBasePayload:timestamp * `ssrc` :#G_TYPE_UINT, The SSRC in use * `pt` :#G_TYPE_UINT, The Payload type in use, same as #GstRTPBasePayload:pt * `seqnum-offset` :#G_TYPE_UINT, The current offset added to the seqnum * `timestamp-offset` :#G_TYPE_UINT, The current offset added to the timestamp Add @ext as an extension for writing part of an RTP header extension onto outgoing RTP packets. the #GstRTPHeaderExtension Clear all RTP header extensions used by this payloader. The returned @ext must be configured with the correct @ext_id and with the necessary attributes as required by the extension implementation. the #GstRTPHeaderExtension for @ext_id, or %NULL the extension id being requested the extension URI being requested Base class for audio RTP payloader. the parent class 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 internal state array of data array of size array of #GstMapInfo 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 mechanism 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 the RTP packet The ID of the header extension (between 1 and 14). location for data the size of the data in bytes 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 mechanism 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 the RTP packet Application specific bits The ID of the header extension location for data the size of the data in bytes Get the CSRC at index @idx in @buffer. the CSRC at index @idx in host order. the RTP packet the index of the CSRC to get Get the CSRC count of the RTP packet in @buffer. the CSRC count of @buffer. the RTP packet Check if the extension bit is set on the RTP packet in @buffer. TRUE if @buffer has the extension bit set. the RTP packet 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. 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 and %NULL otherwise. the RTP packet location for header bits 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. the RTP packet location for result bits location for data location for length of @data in 32 bits words 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 the RTP packet The ID of the header extension to be read (between 1 and 14). Read the nth extension packet with the requested ID location for data the size of the data in bytes 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 the RTP packet Application specific bits The ID of the header extension to be read (between 1 and 14). Read the nth extension packet with the requested ID location for data the size of the data in bytes 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 RTP packet Check if the marker bit is set on the RTP packet in @buffer. TRUE if @buffer has the marker bit set. the RTP packet Return the total length of the packet in @buffer. The total length of the packet in @buffer. the RTP packet Check if the padding bit is set on the RTP packet in @buffer. TRUE if @buffer has the padding bit set. the RTP packet Get a pointer to the payload data in @buffer. This pointer is valid as long as a reference to @buffer is held. A pointer to the payload data in @buffer. the RTP packet 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. the RTP packet 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. the RTP packet Get the length of the payload of the RTP packet in @buffer. The length of the payload in @buffer. the RTP packet 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. the RTP packet the offset in the payload the length in the payload Get the payload type of the RTP packet in @buffer. The payload type. the RTP packet Get the sequence number of the RTP packet in @buffer. The sequence number in host order. the RTP packet Get the SSRC of the RTP packet in @buffer. the SSRC of @buffer in host order. the RTP packet Get the timestamp of the RTP packet in @buffer. The timestamp in host order. the RTP packet Get the version number of the RTP packet in @buffer. The version of @buffer. the RTP packet 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 new amount of padding Unsets the extension bit of the RTP buffer and removes the extension header and data. If the RTP buffer has no header extension data, the action has no effect. The RTP buffer must be mapped READWRITE only once and the underlying GstBuffer must be writable. the RTP packet Modify the CSRC at index @idx in @buffer to @csrc. the RTP packet the CSRC index to set the CSRC in host order to set at @idx Set the extension bit on the RTP packet in @buffer to @extension. the RTP packet the new extension 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. Will also shorten the extension data from 1.20. True if done. the RTP packet the bits specific for the extension 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. the RTP packet the new marker 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 new packet length Set the padding bit on the RTP packet in @buffer to @padding. the buffer the new padding Set the payload type of the RTP packet in @buffer to @payload_type. the RTP packet the new type Set the sequence number of the RTP packet in @buffer to @seq. the RTP packet the new sequence number Set the SSRC on the RTP packet in @buffer to @ssrc. the RTP packet the new SSRC Set the timestamp of the RTP packet in @buffer to @timestamp. the RTP packet the new timestamp Set the version of the RTP packet in @buffer to @version. the RTP packet the new version Unmap @rtp previously mapped with gst_rtp_buffer_map(). a #GstRTPBuffer 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 the length of the payload the amount of padding the number of 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 number of CSRC entries 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 length of the payload the amount of padding the number of CSRC entries 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 total RTP packet the amount of padding the number of CSRC entries 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 are equal or a positive value if @seqnum1 is smaller than @segnum2. a sequence number a sequence number 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 clock-rate is undefined. the static payload type 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. 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. a previous extended timestamp a new timestamp Similar to gst_rtp_buffer_get_extension_onebyte_header, but working on the #GBytes you get from gst_rtp_buffer_get_extension_bytes. Parses RFC 5285 style header extensions with a one byte header. It will return the nth extension with the requested id. TRUE if @bytes had the requested header extension #GBytes The bit-pattern. Anything but 0xBEDE is rejected. The ID of the header extension to be read (between 1 and 14). Read the nth extension packet with the requested ID location for data the size of the data in bytes Map the contents of @buffer into @rtp. %TRUE if @buffer could be mapped. a #GstBuffer #GstMapFlags a #GstRTPBuffer 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 parameters. the length of the payload the amount of padding the number of CSRC entries 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. the total length of the packet the amount of padding the number of CSRC entries 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. data for the new buffer the length of data 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. data for the new buffer the length of data 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). They can conflict with other extended buffer flags. The #GstBuffer was once wrapped in a retransmitted packet as specified by RFC 4588. 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 Offset to define more flags. Additional mapping flags for gst_rtp_buffer_map(). Skip mapping and validation of RTP padding and RTP pad count when present. Useful for buffers where the padding may be encrypted. Offset to define more flags Instance struct for a RTP Audio/Video header extension. the #GstRTPHeaderExtension for @uri or %NULL the rtp header extension URI to search for This is used to know how much data a certain header extension will need for both allocating the resulting data, and deciding how much payload data can be generated. Implementations should return as accurate a value as is possible using the information given in the input @buffer. the maximum size of the data written by this extension a #GstRTPHeaderExtension a #GstBuffer the flags supported by this instance of @ext a #GstRTPHeaderExtension Read the RTP header extension from @data. whether the extension could be read from @data a #GstRTPHeaderExtension #GstRTPHeaderExtensionFlags for how the extension should be written location to read the rtp header extension from size of @data a #GstBuffer to modify if necessary gst_rtp_header_extension_set_id() must have been called with a valid extension id that is contained in these caps. The only current known caps format is based on the SDP standard as produced by gst_sdp_media_attributes_to_caps(). whether the configured attributes on @ext can successfully be set on @caps a #GstRTPHeaderExtension writable #GstCaps to modify Passes RTP payloader's sink (i.e. not payloaded) @caps to the header extension. Whether @caps could be read successfully a #GstRTPHeaderExtension sink #GstCaps Updates depayloader src caps based on the information received in RTP header. @caps must be writable as this function may modify them. whether @caps were modified successfully a #GstRTPHeaderExtension src #GstCaps to modify Writes the RTP header extension to @data using information available from the @input_meta. @data will be sized to be at least the value returned from gst_rtp_header_extension_get_max_size(). the size of the data written, < 0 on failure a #GstRTPHeaderExtension the input #GstBuffer to read information from if necessary #GstRTPHeaderExtensionFlags for how the extension should be written output RTP #GstBuffer location to write the rtp header extension into size of @data Retrieve the direction The direction the #GstRTPHeaderExtension the RTP extension id configured on @ext a #GstRTPHeaderExtension This is used to know how much data a certain header extension will need for both allocating the resulting data, and deciding how much payload data can be generated. Implementations should return as accurate a value as is possible using the information given in the input @buffer. the maximum size of the data written by this extension a #GstRTPHeaderExtension a #GstBuffer the #GstStructure field name used in SDP-like #GstCaps for this @ext configuration the #GstRTPHeaderExtension the flags supported by this instance of @ext a #GstRTPHeaderExtension the RTP extension URI for this object a #GstRTPHeaderExtension Read the RTP header extension from @data. whether the extension could be read from @data a #GstRTPHeaderExtension #GstRTPHeaderExtensionFlags for how the extension should be written location to read the rtp header extension from size of @data a #GstBuffer to modify if necessary gst_rtp_header_extension_set_id() must have been called with a valid extension id that is contained in these caps. The only current known caps format is based on the SDP standard as produced by gst_sdp_media_attributes_to_caps(). whether the @caps could be successfully set on @ext. a #GstRTPHeaderExtension the #GstCaps to configure this extension with gst_rtp_header_extension_set_id() must have been called with a valid extension id that is contained in these caps. The only current known caps format is based on the SDP standard as produced by gst_sdp_media_attributes_to_caps(). whether the configured attributes on @ext can successfully be set on @caps a #GstRTPHeaderExtension writable #GstCaps to modify Helper implementation for GstRTPExtensionClass::set_caps_from_attributes that sets the @ext uri on caps with the specified extension id as required for sdp #GstCaps. Requires that the extension does not have any attributes or direction advertised in @caps. whether the @ext attributes could be set on @caps. the #GstRTPHeaderExtension #GstCaps to write fields into Set the direction that this header extension should be used in. If #GST_RTP_HEADER_EXTENSION_DIRECTION_INHERITED is included, the direction will not be included in the caps (as it shouldn't be in the extmap line in the SDP). the #GstRTPHeaderExtension The direction sets the RTP extension id on @ext a #GstRTPHeaderExtension The id of this extension Passes RTP payloader's sink (i.e. not payloaded) @caps to the header extension. Whether @caps could be read successfully a #GstRTPHeaderExtension sink #GstCaps Call this function in a subclass from #GstRTPHeaderExtensionClass::read to tell the depayloader whether the data just parsed from RTP packet require updating its src (non-RTP) caps. If @state is TRUE, #GstRTPBaseDepayload will eventually invoke gst_rtp_header_extension_update_non_rtp_src_caps() to have the caps update applied. Applying the update also flips the internal "wants update" flag back to FALSE. a #GstRTPHeaderExtension TRUE if caps update is needed Updates depayloader src caps based on the information received in RTP header. @caps must be writable as this function may modify them. whether @caps were modified successfully a #GstRTPHeaderExtension src #GstCaps to modify Call this function after gst_rtp_header_extension_read() to check if the depayloader's src caps need updating with data received in the last RTP packet. Whether @ext wants to update depayloader's src caps. a #GstRTPHeaderExtension Writes the RTP header extension to @data using information available from the @input_meta. @data will be sized to be at least the value returned from gst_rtp_header_extension_get_max_size(). the size of the data written, < 0 on failure a #GstRTPHeaderExtension the input #GstBuffer to read information from if necessary #GstRTPHeaderExtensionFlags for how the extension should be written output RTP #GstBuffer location to write the rtp header extension into size of @data the parent #GObject Base class for RTP Header extensions. the parent class the flags supported by this instance of @ext a #GstRTPHeaderExtension the maximum size of the data written by this extension a #GstRTPHeaderExtension a #GstBuffer the size of the data written, < 0 on failure a #GstRTPHeaderExtension the input #GstBuffer to read information from if necessary #GstRTPHeaderExtensionFlags for how the extension should be written output RTP #GstBuffer location to write the rtp header extension into size of @data whether the extension could be read from @data a #GstRTPHeaderExtension #GstRTPHeaderExtensionFlags for how the extension should be written location to read the rtp header extension from size of @data a #GstBuffer to modify if necessary Whether @caps could be read successfully a #GstRTPHeaderExtension sink #GstCaps whether @caps were modified successfully a #GstRTPHeaderExtension src #GstCaps to modify whether the configured attributes on @ext can successfully be set on @caps a #GstRTPHeaderExtension writable #GstCaps to modify Set the URI for this RTP header extension implementation. the #GstRTPHeaderExtensionClass the RTP Header extension uri for @klass Direction to which to apply the RTP Header Extension Neither send nor receive RTP Header Extensions Only send RTP Header Extensions @GST_RTP_HEADER_EXTENSION_DIRECTION_RECVONLY: Only receive RTP Header Extensions Send and receive RTP Header Extensions ext RTP header extension direction is inherited from the stream Flags that apply to a RTP Audio/Video header extension. The one byte rtp extension header. 1-16 data bytes per extension with a maximum of 14 extension ids in total. The two byte rtp extension header. 256 data bytes per extension with a maximum of 255 (or 256 including appbits) extensions in total. Standard predefined fixed payload types. The official list is at: http://www.iana.org/assignments/rtp-parameters Audio: reserved: 19 unassigned: 20-23, Video: unassigned: 24, 27, 29, 30, 35-71, 77-95 Reserved for RTCP conflict avoidance: 72-76 ITU-T G.711. mu-law audio (RFC 3551) RFC 3551 says reserved RFC 3551 says reserved GSM audio ITU G.723.1 audio IMA ADPCM wave type (RFC 3551) IMA ADPCM wave type (RFC 3551) experimental linear predictive encoding ITU-T G.711 A-law audio (RFC 3551) ITU-T G.722 (RFC 3551) stereo PCM mono PCM EIA & TIA standard IS-733 Comfort Noise (RFC 3389) Audio MPEG 1-3. ITU-T G.728 Speech coder (RFC 3551) IMA ADPCM wave type (RFC 3551) IMA ADPCM wave type (RFC 3551) ITU-T G.729 Speech coder (RFC 3551) See RFC 2029 ISO Standards 10918-1 and 10918-2 (RFC 2435) nv encoding by Ron Frederick ITU-T Recommendation H.261 (RFC 2032) Video MPEG 1 & 2 (RFC 2250) MPEG-2 transport stream (RFC 2250) Video H263 (RFC 2190) Structure holding default payload type information. payload type, -1 means dynamic the media type(s), usually "audio", "video", "application", "text", "message". the encoding name of @pt default clock rate, 0 = unknown/variable encoding parameters. For audio this is the number of channels. NULL = not applicable. the bitrate of the media. 0 = unknown/variable. 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 insensitive way. a #GstRTPPayloadInfo or NULL when no info could be found. the media to find the encoding name to find 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. the payload_type to find The transfer profile to use. invalid profile the Audio/Visual profile (RFC 3551) the secure Audio/Visual profile (RFC 3711) the Audio/Visual profile with feedback (RFC 4585) the secure Audio/Visual profile with feedback (RFC 5124) Meta describing the source(s) of the buffer. parent #GstMeta the SSRC whether @ssrc is set and valid pointer to the CSRCs number of elements in @csrc Appends @csrc to the list of contributing sources in @meta. %TRUE if all elements in @csrc was added, %FALSE otherwise. a #GstRTPSourceMeta the csrcs to append number of elements in @csrc Count the total number of RTP sources found in @meta, both SSRC and CSRC. The number of RTP sources a #GstRTPSourceMeta Sets @ssrc in @meta. If @ssrc is %NULL the ssrc of @meta will be unset. %TRUE on success, %FALSE otherwise. a #GstRTPSourceMeta pointer to the SSRC Get access to the configured MTU of @payload. a #GstRTPBasePayload Get access to the configured payload type of @payload. a #GstRTPBasePayload Get access to the sinkpad of @payload. a #GstRTPBasePayload Get access to the srcpad of @payload. a #GstRTPBasePayload Constant string used in element classification to signal that this element is a RTP header extension. Check if @pt is a dynamic payload type. a payload type The supported RTP version 2. Attaches RTP source information to @buffer. the #GstRTPSourceMeta on @buffer. a #GstBuffer pointer to the SSRC pointer to the CSRCs number of elements in @csrc Find the #GstRTPSourceMeta on @buffer. the #GstRTPSourceMeta or %NULL when there is no such metadata on @buffer. a #GstBuffer Provides common defines for the RTP library. The GstRTPPayloads helper functions makes it easy to deal with static and dynamic payloads. Its main purpose is to retrieve properties such as the default clock-rate and get session bandwidth information. Open @buffer for reading or writing, depending on @flags. The resulting RTCP buffer state is stored in @rtcp. a buffer with an RTCP packet flags for the mapping resulting #GstRTCPBuffer Create a new buffer for constructing RTCP packets. The packet will have a maximum size of @mtu. A newly allocated buffer. the maximum mtu size. 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. data for the new buffer the length of data 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. data for the new buffer the length of data 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. the buffer to validate 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. the data to validate the length of @data to validate 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. the data to validate the length of @data to validate 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. the buffer to validate 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. an NTP timestamp 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 is a private sdes item. a SDES name 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 a #GstRTCPSDESType 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. an UNIX timestamp in nanoseconds 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 the length of the payload the amount of padding the number of 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 number of CSRC entries 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 length of the payload the amount of padding the number of CSRC entries 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 total RTP packet the amount of padding the number of CSRC entries 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 are equal or a positive value if @seqnum1 is smaller than @segnum2. a sequence number a sequence number 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 clock-rate is undefined. the static payload type 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. 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. a previous extended timestamp a new timestamp Similar to gst_rtp_buffer_get_extension_onebyte_header, but working on the #GBytes you get from gst_rtp_buffer_get_extension_bytes. Parses RFC 5285 style header extensions with a one byte header. It will return the nth extension with the requested id. TRUE if @bytes had the requested header extension #GBytes The bit-pattern. Anything but 0xBEDE is rejected. The ID of the header extension to be read (between 1 and 14). Read the nth extension packet with the requested ID location for data the size of the data in bytes Map the contents of @buffer into @rtp. %TRUE if @buffer could be mapped. a #GstBuffer #GstMapFlags a #GstRTPBuffer 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 parameters. the length of the payload the amount of padding the number of CSRC entries 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. the total length of the packet the amount of padding the number of CSRC entries 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. data for the new buffer the length of data 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. data for the new buffer the length of data Retrieve all the factories of the currently registered RTP header extensions. Call gst_element_factory_create() with each factory to create the associated #GstRTPHeaderExtension. a #GList of #GstElementFactory's. Use gst_plugin_feature_list_free() after use Reads the NTP time from the @size NTP-56 extension bytes in @data and store the result in @ntptime. %TRUE on success. the data to read from the size of @data the result NTP time Reads the NTP time from the @size NTP-64 extension bytes in @data and store the result in @ntptime. %TRUE on success. the data to read from the size of @data the result NTP time 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. the data to write to the size of @data the NTP time 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. the data to write to the size of @data the NTP time 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 insensitive way. a #GstRTPPayloadInfo or NULL when no info could be found. the media to find the encoding name to find 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. the payload_type to find