Import line: | GstRtp = imports.gi.GstRtp; |
GIR File: | GstRtp-0.10.gir |
C documentation: | GstRtp |
Namespace : | GstRtp |
Show / Hide Inherited methods, properties and signals |
Properties | |
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Number GstRtp.RTCP_MAX_BYE_SSRC_COUNT : 31
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Number GstRtp.RTCP_MAX_RB_COUNT : 31
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Number GstRtp.RTCP_MAX_SDES : 255
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Number GstRtp.RTCP_MAX_SDES_ITEM_COUNT : 31
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Number GstRtp.RTCP_VALID_MASK : 57598
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Number GstRtp.RTCP_VALID_VALUE : 0
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Number GstRtp.RTCP_VERSION : 2
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String GstRtp.RTP_PAYLOAD_1016_STRING : 1
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String GstRtp.RTP_PAYLOAD_CELLB_STRING : 25
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String GstRtp.RTP_PAYLOAD_CN_STRING : 13
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String GstRtp.RTP_PAYLOAD_DVI4_11025_STRING : 16
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String GstRtp.RTP_PAYLOAD_DVI4_16000_STRING : 6
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String GstRtp.RTP_PAYLOAD_DVI4_22050_STRING : 17
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String GstRtp.RTP_PAYLOAD_DVI4_8000_STRING : 5
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String GstRtp.RTP_PAYLOAD_DYNAMIC_STRING : [96, 127]
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String GstRtp.RTP_PAYLOAD_G721_STRING : 2
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String GstRtp.RTP_PAYLOAD_G722_STRING : 9
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Number GstRtp.RTP_PAYLOAD_G723_53 : 17
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String GstRtp.RTP_PAYLOAD_G723_53_STRING : 17
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Number GstRtp.RTP_PAYLOAD_G723_63 : 16
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String GstRtp.RTP_PAYLOAD_G723_63_STRING : 16
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String GstRtp.RTP_PAYLOAD_G723_STRING : 4
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String GstRtp.RTP_PAYLOAD_G728_STRING : 15
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String GstRtp.RTP_PAYLOAD_G729_STRING : 18
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String GstRtp.RTP_PAYLOAD_GSM_STRING : 3
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String GstRtp.RTP_PAYLOAD_H261_STRING : 31
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String GstRtp.RTP_PAYLOAD_H263_STRING : 34
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String GstRtp.RTP_PAYLOAD_JPEG_STRING : 26
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String GstRtp.RTP_PAYLOAD_L16_MONO_STRING : 11
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String GstRtp.RTP_PAYLOAD_L16_STEREO_STRING : 10
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String GstRtp.RTP_PAYLOAD_LPC_STRING : 7
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String GstRtp.RTP_PAYLOAD_MP2T_STRING : 33
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String GstRtp.RTP_PAYLOAD_MPA_STRING : 14
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String GstRtp.RTP_PAYLOAD_MPV_STRING : 32
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String GstRtp.RTP_PAYLOAD_NV_STRING : 28
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String GstRtp.RTP_PAYLOAD_PCMA_STRING : 8
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String GstRtp.RTP_PAYLOAD_PCMU_STRING : 0
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String GstRtp.RTP_PAYLOAD_QCELP_STRING : 12
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Number GstRtp.RTP_PAYLOAD_TS41 : 19
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String GstRtp.RTP_PAYLOAD_TS41_STRING : 19
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Number GstRtp.RTP_PAYLOAD_TS48 : 18
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String GstRtp.RTP_PAYLOAD_TS48_STRING : 18
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Number GstRtp.RTP_VERSION : 2
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None |
Method / Constructor | Defined By | |
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GstRtp.buffer_allocate_data
(Buffer buffer, Number payload_len, Number pad_len, Number csrc_count)
:
none
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.
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. MALLOCDATA of buffer will be overwritten and will not be freed. All other RTP header fields will be set to 0/FALSE.
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GstRtp | |
GstRtp | ||
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.
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GstRtp | |
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.
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GstRtp | |
Compare two sequence numbers, taking care of wraparounds.
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GstRtp | |
GstRtp | ||
Update the exttimestamp field with timestamp.
Update the exttimestamp field with timestamp. For the first call of the method, exttimestamp should point to a location with a value of -1. This function makes sure that the returned value is a constantly increasing value even in the case where there is a timestamp wraparound.
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GstRtp | |
Get the CSRC at index idx in buffer.
Get the CSRC at index idx in buffer.
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GstRtp | |
Get the CSRC count of the RTP packet in buffer.
Get the CSRC count of the RTP packet in buffer.
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GstRtp | |
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.
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GstRtp | |
Get the extension data.
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.
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GstRtp | |
Return the total length of the header in buffer.
Return the total length of the header in buffer. This include the length of the fixed header, the CSRC list and the extension header.
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GstRtp | |
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.
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GstRtp | |
Return the total length of the packet in buffer.
Return the total length of the packet in buffer.
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GstRtp | |
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.
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GstRtp | |
Get a pointer to the payload data in buffer.
Get a pointer to the payload data in buffer. This pointer is valid as long as a reference to buffer is held.
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GstRtp | |
Create a buffer of the payload of the RTP packet in buffer.
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.
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GstRtp | |
Get the length of the payload of the RTP packet in buffer.
Get the length of the payload of the RTP packet in buffer.
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GstRtp | |
Create a subbuffer of the payload of the RTP packet in buffer.
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 if subbuffered.
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GstRtp | |
Get the payload type of the RTP packet in buffer.
Get the payload type of the RTP packet in buffer.
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GstRtp | |
Get the sequence number of the RTP packet in buffer.
Get the sequence number of the RTP packet in buffer.
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GstRtp | |
Get the SSRC of the RTP packet in buffer.
Get the SSRC of the RTP packet in buffer.
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GstRtp | |
Get the timestamp of the RTP packet in buffer.
Get the timestamp of the RTP packet in buffer.
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GstRtp | |
Get the version number of the RTP packet in buffer.
Get the version number of the RTP packet in buffer.
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GstRtp | |
Get the length of the payload of the RTP packet in list.
Get the length of the payload of the RTP packet in list.
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GstRtp | |
Get the payload type of the first RTP packet in list.
Get the payload type of the first RTP packet in list. All packets in list should have the same payload type.
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GstRtp | |
Get the sequence number of the first RTP packet in list.
Get the sequence number of the first RTP packet in list. All packets within list have the same sequence number.
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GstRtp | |
Get the SSRC of the first RTP packet in list.
Get the SSRC of the first RTP packet in list. All RTP packets within list have the same SSRC.
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GstRtp | |
Get the timestamp of the first RTP packet in list.
Get the timestamp of the first RTP packet in list. All packets within list have the same timestamp.
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GstRtp | |
Set the payload type of each RTP packet in list to payload_type.
Set the payload type of each RTP packet in list to payload_type.
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GstRtp | |
Set the sequence number of each RTP packet in list to seq.
Set the sequence number of each RTP packet in list to seq.
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GstRtp | |
Set the SSRC on each RTP packet in list to ssrc.
Set the SSRC on each RTP packet in list to ssrc.
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GstRtp | |
Set the timestamp of each RTP packet in list to timestamp.
Set the timestamp of each RTP packet in list to timestamp.
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GstRtp | |
Check if all RTP packets in the list are valid using validate_data().
Check if all RTP packets in the list are valid using validate_data(). Use this function to validate an list before using the other functions in this module.
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GstRtp | |
Allocate a new GstBuffer with enough data to hold an RTP packet with All other RTP header fields will be set to 0/FALSE.
Allocate a new GstBuffer with enough data to hold an RTP packet with All other RTP header fields will be set to 0/FALSE. parameters.
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GstRtp | |
Create a new GstBuffer that can hold an RTP packet that is exactly All RTP header fields will be set to 0/FALSE.
Create a new GstBuffer that can hold an RTP packet that is exactly All RTP header fields will be set to 0/FALSE.
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GstRtp | |
Create a new buffer and set the data to a copy of len bytes of data and the size to 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.
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GstRtp | |
Create a new buffer and set the data and size of the buffer to data and len respectively.
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.
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GstRtp | |
Set the amount of padding in the RTP packet in buffer to
Set the amount of padding in the RTP packet in buffer to
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GstRtp | |
Modify the CSRC at index idx in buffer to csrc.
Modify the CSRC at index idx in buffer to csrc.
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GstRtp | |
Set the extension bit on the RTP packet in buffer to extension.
Set the extension bit on the RTP packet in buffer to extension.
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GstRtp | |
the extension, excluding the extension header ( therefore zero is a valid length) Set the extension bit of the rtp buffer and fill in the bits and length of the extension header.
the extension, excluding the extension header ( therefore zero is a valid length) Set the extension bit of the rtp buffer and fill in the bits and length of the extension header. It will refuse to set the extension data if the buffer is not large enough.
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GstRtp | |
Set the marker bit on the RTP packet in buffer to marker.
Set the marker bit on the RTP packet in buffer to marker.
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GstRtp | |
Set the total buffer size to len.
Set the total buffer size to len. The data in the buffer will be made larger if needed. Any padding will be removed from the packet.
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GstRtp | |
Set the padding bit on the RTP packet in buffer to padding.
Set the padding bit on the RTP packet in buffer to padding.
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GstRtp | |
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.
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GstRtp | |
Set the sequence number of the RTP packet in buffer to seq.
Set the sequence number of the RTP packet in buffer to seq.
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GstRtp | |
Set the SSRC on the RTP packet in buffer to ssrc.
Set the SSRC on the RTP packet in buffer to ssrc.
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GstRtp | |
Set the timestamp of the RTP packet in buffer to timestamp.
Set the timestamp of the RTP packet in buffer to timestamp.
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GstRtp | |
Set the version of the RTP packet in buffer to version.
Set the version of the RTP packet in buffer to version.
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GstRtp | |
Check if the data pointed to by buffer is a valid RTP packet using validate_data().
Check if the data pointed to by buffer is a valid RTP packet using validate_data(). Use this function to validate a packet before using the other functions in this module.
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GstRtp | |
Check if the data and size point to the data of a valid RTP packet.
Check if the data and size point to the data of a valid RTP packet. This function checks the length, version and padding of the packet data. Use this function to validate a packet before using the other functions in this module.
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GstRtp | |
Get the GstRTPPayloadInfo for media and encoding_name.
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.
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GstRtp | |
Get the GstRTPPayloadInfo for payload_type.
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.
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GstRtp | |
Add a new packet of type to buffer.
Add a new packet of type to buffer. packet will point to the newly created packet. if the max mtu is exceeded for the buffer.
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GstRtp | |
GstRtp.rtcp_buffer_end
(Buffer buffer)
:
none
Finish buffer after being constructured.
Finish buffer after being constructured. This function is usually called after gst_rtcp_buffer_new() and after adding the RTCP items to the new buffer. The function adjusts the size of buffer with the total length of all the added packets.
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GstRtp | |
Initialize a new GstRTCPPacket pointer that points to the first packet in
Initialize a new GstRTCPPacket pointer that points to the first packet in
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GstRtp | |
Get the number of RTCP packets in buffer.
Get the number of RTCP packets in buffer.
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GstRtp | |
Create a new buffer for constructing RTCP packets.
Create a new buffer for constructing RTCP packets. The packet will have a maximum size of mtu.
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GstRtp | |
Create a new buffer and set the data to a copy of len bytes of data and the size to 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.
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GstRtp | |
Create a new buffer and set the data and size of the buffer to data and len respectively.
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.
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GstRtp | |
Check if the data pointed to by buffer is a valid RTCP packet using gst_rtcp_buffer_validate_data().
Check if the data pointed to by buffer is a valid RTCP packet using gst_rtcp_buffer_validate_data().
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GstRtp | |
Check if the data and size point to the data of a valid RTCP (compound) packet.
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GstRtp | |
Converts an NTP time to UNIX nanoseconds.
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.
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GstRtp | |
Get the feedback message type of the FB packet.
Get the feedback message type of the FB packet.
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GstRtp | |
Get the packet type of the packet pointed to by packet.
Get the packet type of the packet pointed to by packet. pointing to a valid packet.
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GstRtp | |
Convert name into a GstRTCPSDESType.
Convert name into a GstRTCPSDESType. name is typically a key in a GstStructure containing SDES items. is a private sdes item.
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GstRtp | |
Converts type to the string equivalent.
Converts type to the string equivalent. The string is typically used as a key in a GstStructure containing SDES items.
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GstRtp | |
Converts a UNIX timestamp in nanoseconds to an NTP time.
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.
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GstRtp |
None |