mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-15 12:56:33 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/subprojects/gst-plugins-base/gst-libs/gst/rtp/gstrtcpbuffer.h
Thibault Saunier 3296c678b3 rtcpbuffer: Allow padding on first reduced size packets 
It is valid to have the padding set to 1 on the first packet and it
happens very often from TWCC packets coming from libwebrtc. This means
that we were totally ignoring many TWCC packets.

Fix test that checked that a first packet with padding was not valid and
instead test a single twcc packet with padding to check precisely what
this patch was about.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2422>
2022-05-18 14:34:44 +00:00

671 lines
22 KiB
C
Raw Blame History

/* 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,
/* draft-holmer-rmcat-transport-wide-cc-extensions-01 */
GST_RTCP_RTPFB_TYPE_TWCC = 15,
/* 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.
*/
/**
* GST_RTCP_SDES_H323_CADDR:
*
* H.323 callable address
*
* Since: 1.20:
*/
/**
* GST_RTCP_SDES_APSI:
*
* Application Specific Identifier (RFC6776)
*
* Since: 1.20:
*/
/**
* GST_RTCP_SDES_RGRP:
*
* Reporting Group Identifier (RFC8861)
*
* Since: 1.20:
*/
/**
* GST_RTCP_SDES_RTP_STREAM_ID:
*
* RtpStreamId SDES item (RFC8852).
*
* Since: 1.20:
*/
/**
* GST_RTCP_SDES_REPAIRED_RTP_STREAM_ID:
*
* RepairedRtpStreamId SDES item (RFC8852).
*
* Since: 1.20:
*/
/**
* GST_RTCP_SDES_CCID:
*
* CLUE CaptId (RFC8849)
*
* Since: 1.20:
*/
/**
* GST_RTCP_SDES_MID:
*
* MID SDES item (RFC8843).
*
* Since: 1.20:
*/
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,
GST_RTCP_SDES_H323_CADDR = 9,
GST_RTCP_SDES_APSI = 10,
GST_RTCP_SDES_RGRP = 11,
GST_RTCP_SDES_RTP_STREAM_ID = 12,
GST_RTCP_SDES_REPAIRED_RTP_STREAM_ID = 13,
GST_RTCP_SDES_CCID = 14,
GST_RTCP_SDES_MID = 15,
} 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 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 | 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 * thinning,
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 * thinning,
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__ */
Reference in a new issue View git blame Copy permalink