/* GStreamer * Copyright (C) <2007> Wim Taymans * * 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 #include 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: (value 32968) * * 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__ */