Namespace GstRtp

Attaches RTP source information to buffer.

Find the #GstRTPSourceMeta on buffer.

Open buffer for reading or writing, depending on flags. The resulting RTCP buffer state is stored in rtcp.

Create a new buffer for constructing RTCP packets. The packet will have a maximum size of mtu.

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.

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.

Check if the data pointed to by buffer is a valid RTCP packet using gst_rtcp_buffer_validate_data().

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.

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.

Check if the data pointed to by buffer is a valid RTCP packet using gst_rtcp_buffer_validate_reduced().

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.

Convert name into a GstRTCPSDESType. name is typically a key in a #GstStructure containing SDES items.

Converts type to the string equivalent. The string is typically used as a key in a #GstStructure containing SDES items.

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.

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.

Calculate the header length of an RTP packet with csrc_count CSRC entries. 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, 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, a padding of pad_len and a csrc_count CSRC entries.

Compare two sequence numbers, taking care of wraparounds. This function returns the difference between seqnum1 and seqnum2.

Get the default clock-rate for the static payload type 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.

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.

Map the contents of buffer into rtp.

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.

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.

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.

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.

Retrieve all the factories of the currently registered RTP header extensions. Call gst_element_factory_create() with each factory to create the associated #GstRTPHeaderExtension.

Reads the NTP time from the size NTP-56 extension bytes in data and store the result in ntptime.

Reads the NTP time from the size NTP-64 extension bytes in data and store the result in ntptime.

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.

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.

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.

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.