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gstreamer/gst-libs/gst/rtp/gstrtcpbuffer.h
Justin Kim 5303e2c32b rtcpbuffer: add support XR packet parsing 
According to RFC3611, the extended report blocks in XR packet can
have variable length. To visit each block, the iterator should look
into block header. Once XR type is extracted, users can parse the
detailed information by given functions.

Loss/Duplicate RLE
The Loss RLE and the Duplicate RLE have same format so
they can share parsers. For unit test, randomly generated
pseudo packet is used.

Packet Receipt Times
The packet receipt times report block has a list of receipt
times which are in [begin_seq, end_seq).

Receiver Reference Time paser for XR packet
The receiver reference time has ntptime which is 64 bit type.

DLRR
The DLRR report block consists of sub-blocks which has ssrc, last RR,
and delay since last RR. The number of sub-blocks should be calculated
from block length.

Statistics Summary
The Statistics Summary report block provides fixed length
information.

VoIP Metrics
VoIP Metrics consists of several metrics even though they are in
a report block. Data retrieving functions are added per metrics.

https://bugzilla.gnome.org/show_bug.cgi?id=789822
2018-12-13 14:01:06 -05:00

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/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
*
* gstrtcpbuffer.h: various helper functions to manipulate buffers
* with RTCP payload.
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifndef __GST_RTCPBUFFER_H__
#define __GST_RTCPBUFFER_H__
#include <gst/gst.h>
#include <gst/rtp/rtp-prelude.h>
G_BEGIN_DECLS
/**
* GST_RTCP_VERSION:
*
* The supported RTCP version 2.
*/
#define GST_RTCP_VERSION 2
/**
* GstRTCPType:
* @GST_RTCP_TYPE_INVALID: Invalid type
* @GST_RTCP_TYPE_SR: Sender report
* @GST_RTCP_TYPE_RR: Receiver report
* @GST_RTCP_TYPE_SDES: Source description
* @GST_RTCP_TYPE_BYE: Goodbye
* @GST_RTCP_TYPE_APP: Application defined
* @GST_RTCP_TYPE_RTPFB: Transport layer feedback.
* @GST_RTCP_TYPE_PSFB: Payload-specific feedback.
* @GST_RTCP_TYPE_XR: Extended report.
*
* Different RTCP packet types.
*/
typedef enum
{
GST_RTCP_TYPE_INVALID = 0,
GST_RTCP_TYPE_SR = 200,
GST_RTCP_TYPE_RR = 201,
GST_RTCP_TYPE_SDES = 202,
GST_RTCP_TYPE_BYE = 203,
GST_RTCP_TYPE_APP = 204,
GST_RTCP_TYPE_RTPFB = 205,
GST_RTCP_TYPE_PSFB = 206,
GST_RTCP_TYPE_XR = 207
} GstRTCPType;
/* FIXME 2.0: backwards compatibility define for enum typo */
#define GST_RTCP_RTPFB_TYPE_RCTP_SR_REQ GST_RTCP_RTPFB_TYPE_RTCP_SR_REQ
/**
* GstRTCPFBType:
* @GST_RTCP_FB_TYPE_INVALID: Invalid type
* @GST_RTCP_RTPFB_TYPE_NACK: Generic NACK
* @GST_RTCP_RTPFB_TYPE_TMMBR: Temporary Maximum Media Stream Bit Rate Request
* @GST_RTCP_RTPFB_TYPE_TMMBN: Temporary Maximum Media Stream Bit Rate
* Notification
* @GST_RTCP_RTPFB_TYPE_RTCP_SR_REQ: Request an SR packet for early
* synchronization
* @GST_RTCP_PSFB_TYPE_PLI: Picture Loss Indication
* @GST_RTCP_PSFB_TYPE_SLI: Slice Loss Indication
* @GST_RTCP_PSFB_TYPE_RPSI: Reference Picture Selection Indication
* @GST_RTCP_PSFB_TYPE_AFB: Application layer Feedback
* @GST_RTCP_PSFB_TYPE_FIR: Full Intra Request Command
* @GST_RTCP_PSFB_TYPE_TSTR: Temporal-Spatial Trade-off Request
* @GST_RTCP_PSFB_TYPE_TSTN: Temporal-Spatial Trade-off Notification
* @GST_RTCP_PSFB_TYPE_VBCN: Video Back Channel Message
*
* Different types of feedback messages.
*/
typedef enum
{
/* generic */
GST_RTCP_FB_TYPE_INVALID = 0,
/* RTPFB types */
GST_RTCP_RTPFB_TYPE_NACK = 1,
/* RTPFB types assigned in RFC 5104 */
GST_RTCP_RTPFB_TYPE_TMMBR = 3,
GST_RTCP_RTPFB_TYPE_TMMBN = 4,
/* RTPFB types assigned in RFC 6051 */
GST_RTCP_RTPFB_TYPE_RTCP_SR_REQ = 5,
/* PSFB types */
GST_RTCP_PSFB_TYPE_PLI = 1,
GST_RTCP_PSFB_TYPE_SLI = 2,
GST_RTCP_PSFB_TYPE_RPSI = 3,
GST_RTCP_PSFB_TYPE_AFB = 15,
/* PSFB types assigned in RFC 5104 */
GST_RTCP_PSFB_TYPE_FIR = 4,
GST_RTCP_PSFB_TYPE_TSTR = 5,
GST_RTCP_PSFB_TYPE_TSTN = 6,
GST_RTCP_PSFB_TYPE_VBCN = 7,
} GstRTCPFBType;
/**
* GstRTCPSDESType:
* @GST_RTCP_SDES_INVALID: Invalid SDES entry
* @GST_RTCP_SDES_END: End of SDES list
* @GST_RTCP_SDES_CNAME: Canonical name
* @GST_RTCP_SDES_NAME: User name
* @GST_RTCP_SDES_EMAIL: User's electronic mail address
* @GST_RTCP_SDES_PHONE: User's phone number
* @GST_RTCP_SDES_LOC: Geographic user location
* @GST_RTCP_SDES_TOOL: Name of application or tool
* @GST_RTCP_SDES_NOTE: Notice about the source
* @GST_RTCP_SDES_PRIV: Private extensions
*
* Different types of SDES content.
*/
typedef enum
{
GST_RTCP_SDES_INVALID = -1,
GST_RTCP_SDES_END = 0,
GST_RTCP_SDES_CNAME = 1,
GST_RTCP_SDES_NAME = 2,
GST_RTCP_SDES_EMAIL = 3,
GST_RTCP_SDES_PHONE = 4,
GST_RTCP_SDES_LOC = 5,
GST_RTCP_SDES_TOOL = 6,
GST_RTCP_SDES_NOTE = 7,
GST_RTCP_SDES_PRIV = 8
} GstRTCPSDESType;
/**
* GstRTCPXRType:
* @GST_RTCP_XR_TYPE_INVALID: Invalid XR Report Block
* @GST_RTCP_XR_TYPE_LRLE: Loss RLE Report Block
* @GST_RTCP_XR_TYPE_DRLE: Duplicate RLE Report Block
* @GST_RTCP_XR_TYPE_PRT: Packet Receipt Times Report Block
* @GST_RTCP_XR_TYPE_RRT: Receiver Reference Time Report Block
* @GST_RTCP_XR_TYPE_DLRR: Delay since the last Receiver Report
* @GST_RTCP_XR_TYPE_SSUMM: Statistics Summary Report Block
* @GST_RTCP_XR_TYPE_VOIP_METRICS: VoIP Metrics Report Block
*
* 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).
*
* Since: 1.16
*/
typedef enum
{
GST_RTCP_XR_TYPE_INVALID = -1,
GST_RTCP_XR_TYPE_LRLE = 1,
GST_RTCP_XR_TYPE_DRLE = 2,
GST_RTCP_XR_TYPE_PRT = 3,
GST_RTCP_XR_TYPE_RRT = 4,
GST_RTCP_XR_TYPE_DLRR = 5,
GST_RTCP_XR_TYPE_SSUMM = 6,
GST_RTCP_XR_TYPE_VOIP_METRICS = 7
} GstRTCPXRType;
/**
* GST_RTCP_MAX_SDES:
*
* The maximum text length for an SDES item.
*/
#define GST_RTCP_MAX_SDES 255
/**
* GST_RTCP_MAX_RB_COUNT:
*
* The maximum amount of Receiver report blocks in RR and SR messages.
*/
#define GST_RTCP_MAX_RB_COUNT 31
/**
* GST_RTCP_MAX_SDES_ITEM_COUNT:
*
* The maximum amount of SDES items.
*/
#define GST_RTCP_MAX_SDES_ITEM_COUNT 31
/**
* GST_RTCP_MAX_BYE_SSRC_COUNT:
*
* The maximum amount of SSRCs in a BYE packet.
*/
#define GST_RTCP_MAX_BYE_SSRC_COUNT 31
/**
* GST_RTCP_VALID_MASK:
*
* Mask for version, padding bit and packet type pair
*/
#define GST_RTCP_VALID_MASK (0xc000 | 0x2000 | 0xfe)
/**
* GST_RTCP_REDUCED_SIZE_VALID_MASK:
*
* Mask for version, padding bit and packet type pair allowing reduced size
* packets, basically it accepts other types than RR and SR
*/
#define GST_RTCP_REDUCED_SIZE_VALID_MASK (0xc000 | 0x2000 | 0xf8)
/**
* GST_RTCP_VALID_VALUE:
*
* Valid value for the first two bytes of an RTCP packet after applying
* #GST_RTCP_VALID_MASK to them.
*/
#define GST_RTCP_VALID_VALUE ((GST_RTCP_VERSION << 14) | GST_RTCP_TYPE_SR)
typedef struct _GstRTCPBuffer GstRTCPBuffer;
typedef struct _GstRTCPPacket GstRTCPPacket;
struct _GstRTCPBuffer
{
GstBuffer *buffer;
GstMapInfo map;
};
#define GST_RTCP_BUFFER_INIT { NULL, GST_MAP_INFO_INIT }
/**
* GstRTCPPacket:
* @rtcp: pointer to RTCP buffer
* @offset: offset of packet in buffer data
*
* Data structure that points to a packet at @offset in @buffer.
* The size of the structure is made public to allow stack allocations.
*/
struct _GstRTCPPacket
{
/*< public >*/
GstRTCPBuffer *rtcp;
guint offset;
/*< private >*/
gboolean padding; /* padding field of current packet */
guint8 count; /* count field of current packet */
GstRTCPType type; /* type of current packet */
guint16 length; /* length of current packet in 32-bits words minus one, this is validated when doing _get_first_packet() and _move_to_next() */
guint item_offset; /* current item offset for navigating SDES */
guint item_count; /* current item count */
guint entry_offset; /* current entry offset for navigating SDES items */
};
/* creating buffers */
GST_RTP_API
GstBuffer* gst_rtcp_buffer_new_take_data (gpointer data, guint len);
GST_RTP_API
GstBuffer* gst_rtcp_buffer_new_copy_data (gconstpointer data, guint len);
GST_RTP_API
gboolean gst_rtcp_buffer_validate_data (guint8 *data, guint len);
GST_RTP_API
gboolean gst_rtcp_buffer_validate (GstBuffer *buffer);
GST_RTP_API
gboolean gst_rtcp_buffer_validate_data_reduced (guint8 *data, guint len);
GST_RTP_API
gboolean gst_rtcp_buffer_validate_reduced (GstBuffer *buffer);
GST_RTP_API
GstBuffer* gst_rtcp_buffer_new (guint mtu);
GST_RTP_API
gboolean gst_rtcp_buffer_map (GstBuffer *buffer, GstMapFlags flags, GstRTCPBuffer *rtcp);
GST_RTP_API
gboolean gst_rtcp_buffer_unmap (GstRTCPBuffer *rtcp);
/* adding/retrieving packets */
GST_RTP_API
guint gst_rtcp_buffer_get_packet_count (GstRTCPBuffer *rtcp);
GST_RTP_API
gboolean gst_rtcp_buffer_get_first_packet (GstRTCPBuffer *rtcp, GstRTCPPacket *packet);
GST_RTP_API
gboolean gst_rtcp_packet_move_to_next (GstRTCPPacket *packet);
GST_RTP_API
gboolean gst_rtcp_buffer_add_packet (GstRTCPBuffer *rtcp, GstRTCPType type,
GstRTCPPacket *packet);
GST_RTP_API
gboolean gst_rtcp_packet_remove (GstRTCPPacket *packet);
/* working with packets */
GST_RTP_API
gboolean gst_rtcp_packet_get_padding (GstRTCPPacket *packet);
GST_RTP_API
guint8 gst_rtcp_packet_get_count (GstRTCPPacket *packet);
GST_RTP_API
GstRTCPType gst_rtcp_packet_get_type (GstRTCPPacket *packet);
GST_RTP_API
guint16 gst_rtcp_packet_get_length (GstRTCPPacket *packet);
/* sender reports */
GST_RTP_API
void gst_rtcp_packet_sr_get_sender_info (GstRTCPPacket *packet, guint32 *ssrc,
guint64 *ntptime, guint32 *rtptime,
guint32 *packet_count, guint32 *octet_count);
GST_RTP_API
void gst_rtcp_packet_sr_set_sender_info (GstRTCPPacket *packet, guint32 ssrc,
guint64 ntptime, guint32 rtptime,
guint32 packet_count, guint32 octet_count);
/* receiver reports */
GST_RTP_API
guint32 gst_rtcp_packet_rr_get_ssrc (GstRTCPPacket *packet);
GST_RTP_API
void gst_rtcp_packet_rr_set_ssrc (GstRTCPPacket *packet, guint32 ssrc);
/* report blocks for SR and RR */
GST_RTP_API
guint gst_rtcp_packet_get_rb_count (GstRTCPPacket *packet);
GST_RTP_API
void gst_rtcp_packet_get_rb (GstRTCPPacket *packet, guint nth, guint32 *ssrc,
guint8 *fractionlost, gint32 *packetslost,
guint32 *exthighestseq, guint32 *jitter,
guint32 *lsr, guint32 *dlsr);
GST_RTP_API
gboolean gst_rtcp_packet_add_rb (GstRTCPPacket *packet, guint32 ssrc,
guint8 fractionlost, gint32 packetslost,
guint32 exthighestseq, guint32 jitter,
guint32 lsr, guint32 dlsr);
GST_RTP_API
void gst_rtcp_packet_set_rb (GstRTCPPacket *packet, guint nth, guint32 ssrc,
guint8 fractionlost, gint32 packetslost,
guint32 exthighestseq, guint32 jitter,
guint32 lsr, guint32 dlsr);
/* profile-specific extensions for SR and RR */
GST_RTP_API
gboolean gst_rtcp_packet_add_profile_specific_ext (GstRTCPPacket * packet,
const guint8 * data, guint len);
GST_RTP_API
guint16 gst_rtcp_packet_get_profile_specific_ext_length (GstRTCPPacket * packet);
GST_RTP_API
gboolean gst_rtcp_packet_get_profile_specific_ext (GstRTCPPacket * packet,
guint8 ** data, guint * len);
GST_RTP_API
gboolean gst_rtcp_packet_copy_profile_specific_ext (GstRTCPPacket * packet,
guint8 ** data, guint * len);
/* source description packet */
GST_RTP_API
guint gst_rtcp_packet_sdes_get_item_count (GstRTCPPacket *packet);
GST_RTP_API
gboolean gst_rtcp_packet_sdes_first_item (GstRTCPPacket *packet);
GST_RTP_API
gboolean gst_rtcp_packet_sdes_next_item (GstRTCPPacket *packet);
GST_RTP_API
guint32 gst_rtcp_packet_sdes_get_ssrc (GstRTCPPacket *packet);
GST_RTP_API
gboolean gst_rtcp_packet_sdes_first_entry (GstRTCPPacket *packet);
GST_RTP_API
gboolean gst_rtcp_packet_sdes_next_entry (GstRTCPPacket *packet);
GST_RTP_API
gboolean gst_rtcp_packet_sdes_get_entry (GstRTCPPacket *packet,
GstRTCPSDESType *type, guint8 *len,
guint8 **data);
GST_RTP_API
gboolean gst_rtcp_packet_sdes_copy_entry (GstRTCPPacket *packet,
GstRTCPSDESType *type, guint8 *len,
guint8 **data);
GST_RTP_API
gboolean gst_rtcp_packet_sdes_add_item (GstRTCPPacket *packet, guint32 ssrc);
GST_RTP_API
gboolean gst_rtcp_packet_sdes_add_entry (GstRTCPPacket *packet, GstRTCPSDESType type,
guint8 len, const guint8 *data);
/* bye packet */
GST_RTP_API
guint gst_rtcp_packet_bye_get_ssrc_count (GstRTCPPacket *packet);
GST_RTP_API
guint32 gst_rtcp_packet_bye_get_nth_ssrc (GstRTCPPacket *packet, guint nth);
GST_RTP_API
gboolean gst_rtcp_packet_bye_add_ssrc (GstRTCPPacket *packet, guint32 ssrc);
GST_RTP_API
gboolean gst_rtcp_packet_bye_add_ssrcs (GstRTCPPacket *packet, guint32 *ssrc, guint len);
GST_RTP_API
guint8 gst_rtcp_packet_bye_get_reason_len (GstRTCPPacket *packet);
GST_RTP_API
gchar* gst_rtcp_packet_bye_get_reason (GstRTCPPacket *packet);
GST_RTP_API
gboolean gst_rtcp_packet_bye_set_reason (GstRTCPPacket *packet, const gchar *reason);
/* app packets */
GST_RTP_API
void gst_rtcp_packet_app_set_subtype (GstRTCPPacket * packet, guint8 subtype);
GST_RTP_API
guint8 gst_rtcp_packet_app_get_subtype (GstRTCPPacket * packet);
GST_RTP_API
void gst_rtcp_packet_app_set_ssrc (GstRTCPPacket * packet, guint32 ssrc);
GST_RTP_API
guint32 gst_rtcp_packet_app_get_ssrc (GstRTCPPacket * packet);
GST_RTP_API
void gst_rtcp_packet_app_set_name (GstRTCPPacket * packet, const gchar *name);
GST_RTP_API
const gchar* gst_rtcp_packet_app_get_name (GstRTCPPacket * packet);
GST_RTP_API
guint16 gst_rtcp_packet_app_get_data_length (GstRTCPPacket * packet);
GST_RTP_API
gboolean gst_rtcp_packet_app_set_data_length (GstRTCPPacket * packet, guint16 wordlen);
GST_RTP_API
guint8* gst_rtcp_packet_app_get_data (GstRTCPPacket * packet);
/* feedback packets */
GST_RTP_API
guint32 gst_rtcp_packet_fb_get_sender_ssrc (GstRTCPPacket *packet);
GST_RTP_API
void gst_rtcp_packet_fb_set_sender_ssrc (GstRTCPPacket *packet, guint32 ssrc);
GST_RTP_API
guint32 gst_rtcp_packet_fb_get_media_ssrc (GstRTCPPacket *packet);
GST_RTP_API
void gst_rtcp_packet_fb_set_media_ssrc (GstRTCPPacket *packet, guint32 ssrc);
GST_RTP_API
GstRTCPFBType gst_rtcp_packet_fb_get_type (GstRTCPPacket *packet);
GST_RTP_API
void gst_rtcp_packet_fb_set_type (GstRTCPPacket *packet, GstRTCPFBType type);
GST_RTP_API
guint16 gst_rtcp_packet_fb_get_fci_length (GstRTCPPacket *packet);
GST_RTP_API
gboolean gst_rtcp_packet_fb_set_fci_length (GstRTCPPacket *packet, guint16 wordlen);
GST_RTP_API
guint8 * gst_rtcp_packet_fb_get_fci (GstRTCPPacket *packet);
/* helper functions */
GST_RTP_API
guint64 gst_rtcp_ntp_to_unix (guint64 ntptime);
GST_RTP_API
guint64 gst_rtcp_unix_to_ntp (guint64 unixtime);
GST_RTP_API
const gchar * gst_rtcp_sdes_type_to_name (GstRTCPSDESType type);
GST_RTP_API
GstRTCPSDESType gst_rtcp_sdes_name_to_type (const gchar *name);
/* extended report */
GST_RTP_API
guint32 gst_rtcp_packet_xr_get_ssrc (GstRTCPPacket *packet);
GST_RTP_API
gboolean gst_rtcp_packet_xr_first_rb (GstRTCPPacket *packet);
GST_RTP_API
gboolean gst_rtcp_packet_xr_next_rb (GstRTCPPacket * packet);
GST_RTP_API
GstRTCPXRType gst_rtcp_packet_xr_get_block_type (GstRTCPPacket * packet);
GST_RTP_API
guint16 gst_rtcp_packet_xr_get_block_length (GstRTCPPacket * packet);
GST_RTP_API
gboolean gst_rtcp_packet_xr_get_rle_info (GstRTCPPacket * packet,
guint32 * ssrc, guint8 * thining,
guint16 * begin_seq, guint16 * end_seq,
guint32 * chunk_count);
GST_RTP_API
gboolean gst_rtcp_packet_xr_get_rle_nth_chunk (GstRTCPPacket * packet, guint nth,
guint16 * chunk);
GST_RTP_API
gboolean gst_rtcp_packet_xr_get_prt_info (GstRTCPPacket * packet,
guint32 * ssrc, guint8 * thining,
guint16 * begin_seq, guint16 * end_seq);
GST_RTP_API
gboolean gst_rtcp_packet_xr_get_prt_by_seq (GstRTCPPacket * packet, guint16 seq,
guint32 * receipt_time);
GST_RTP_API
gboolean gst_rtcp_packet_xr_get_rrt (GstRTCPPacket * packet, guint64 * timestamp);
GST_RTP_API
gboolean gst_rtcp_packet_xr_get_dlrr_block (GstRTCPPacket * packet,
guint nth, guint32 * ssrc,
guint32 * last_rr, guint32 * delay);
GST_RTP_API
gboolean gst_rtcp_packet_xr_get_summary_info (GstRTCPPacket * packet, guint32 * ssrc,
guint16 * begin_seq, guint16 * end_seq);
GST_RTP_API
gboolean gst_rtcp_packet_xr_get_summary_pkt (GstRTCPPacket * packet,
guint32 * lost_packets, guint32 * dup_packets);
GST_RTP_API
gboolean gst_rtcp_packet_xr_get_summary_jitter (GstRTCPPacket * packet,
guint32 * min_jitter, guint32 * max_jitter,
guint32 * mean_jitter, guint32 * dev_jitter);
GST_RTP_API
gboolean gst_rtcp_packet_xr_get_summary_ttl (GstRTCPPacket * packet, gboolean * is_ipv4,
guint8 * min_ttl, guint8 * max_ttl,
guint8 * mean_ttl, guint8 * dev_ttl);
GST_RTP_API
gboolean gst_rtcp_packet_xr_get_voip_metrics_ssrc (GstRTCPPacket * packet, guint32 * ssrc);
GST_RTP_API
gboolean gst_rtcp_packet_xr_get_voip_packet_metrics (GstRTCPPacket * packet,
guint8 * loss_rate, guint8 * discard_rate);
GST_RTP_API
gboolean gst_rtcp_packet_xr_get_voip_burst_metrics (GstRTCPPacket * packet,
guint8 * burst_density, guint8 * gap_density,
guint16 * burst_duration, guint16 * gap_duration);
GST_RTP_API
gboolean gst_rtcp_packet_xr_get_voip_delay_metrics (GstRTCPPacket * packet,
guint16 * roundtrip_delay,
guint16 * end_system_delay);
GST_RTP_API
gboolean gst_rtcp_packet_xr_get_voip_signal_metrics (GstRTCPPacket * packet,
guint8 * signal_level, guint8 * noise_level,
guint8 * rerl, guint8 * gmin);
GST_RTP_API
gboolean gst_rtcp_packet_xr_get_voip_quality_metrics (GstRTCPPacket * packet,
guint8 * r_factor, guint8 * ext_r_factor,
guint8 * mos_lq, guint8 * mos_cq);
GST_RTP_API
gboolean gst_rtcp_packet_xr_get_voip_configuration_params (GstRTCPPacket * packet,
guint8 * gmin, guint8 * rx_config);
GST_RTP_API
gboolean gst_rtcp_packet_xr_get_voip_jitter_buffer_params (GstRTCPPacket * packet,
guint16 * jb_nominal,
guint16 * jb_maximum,
guint16 * jb_abs_max);
G_END_DECLS
#endif /* __GST_RTCPBUFFER_H__ */
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