From c2e620769806f48a73dc86ae67adb2cc4d501c44 Mon Sep 17 00:00:00 2001 From: Aaro Altonen Date: Tue, 23 Feb 2021 14:43:09 +0200 Subject: [PATCH] Document set_ts_info() --- docs/html/classuvgrtp_1_1media__stream.html | 11 +- docs/html/classuvgrtp_1_1rtcp-members.html | 2 +- docs/html/classuvgrtp_1_1rtcp.html | 51 +++- docs/html/functions.html | 3 + docs/html/functions_func.html | 3 + docs/html/media__stream_8hh_source.html | 190 ++++++------- docs/html/rtcp_8hh_source.html | 293 ++++++++++---------- docs/html/search/all_8.js | 3 +- docs/html/search/all_9.js | 4 +- docs/html/search/all_a.js | 2 +- docs/html/search/classes_0.js | 2 +- docs/html/search/classes_1.js | 2 +- docs/html/search/classes_2.js | 2 +- docs/html/search/classes_3.js | 2 +- docs/html/search/enums_0.js | 10 +- docs/html/search/enumvalues_0.js | 82 +++--- docs/html/search/files_0.js | 2 +- docs/html/search/functions_0.js | 2 +- docs/html/search/functions_1.js | 8 +- docs/html/search/functions_2.js | 4 +- docs/html/search/functions_3.js | 2 +- docs/html/search/functions_4.js | 2 +- docs/html/search/functions_5.js | 4 +- docs/html/search/functions_6.js | 2 +- docs/html/search/functions_7.html | 37 +++ docs/html/search/functions_7.js | 4 + docs/html/search/pages_0.js | 2 +- docs/html/search/searchdata.js | 2 +- include/rtcp.hh | 16 +- 29 files changed, 431 insertions(+), 318 deletions(-) create mode 100644 docs/html/search/functions_7.html create mode 100644 docs/html/search/functions_7.js diff --git a/docs/html/classuvgrtp_1_1media__stream.html b/docs/html/classuvgrtp_1_1media__stream.html index 68c90ee..598ac1c 100644 --- a/docs/html/classuvgrtp_1_1media__stream.html +++ b/docs/html/classuvgrtp_1_1media__stream.html @@ -138,6 +138,13 @@ Public Member Functions

Currently uvgRTP only supports key length of 16 bytes (128 bits) and salt length of 14 bytes (112 bits).

If the key or salt is longer, it is implicitly truncated to correct length and if the key or salt is shorter, a memory violation may occur

Notice that if user-managed SRTP has been enabled during media stream creation, this function must be called before anything else. All calls to other functions will fail with RTP_NOT_INITIALIZED until the SRTP context has been specified

+
Parameters
+ + + +
key128-bit long key
salt112-bit long salt
+
+
Returns
RTP error code
Return values
@@ -420,7 +427,7 @@ Public Member Functions

Send data to remote participant with a custom timestamp.

If so specified either by the selected media format and/or given RTP_CTX_ENABLE_FLAGS, uvgRTP fragments the input data into RTP packets of 1500 bytes, or to any other size defined by the application using RCC_MTU_SIZE

The frame is automatically reconstructed by the receiver if all fragments have been received successfully.

-

If application so wishes, it may override uvgRTP's own timestamp calculations and provide timestamping information for the stream itself. This requires that the application provides a sensible value for the ts parameter. If RTCP has been enabled, uvgrtp::rtcp::set_ts_info() should have been called.

+

If application so wishes, it may override uvgRTP's own timestamp calculations and provide timestamping information for the stream itself. This requires that the application provides a sensible value for the ts parameter. If RTCP has been enabled, uvgrtp::rtcp::set_ts_info() should have been called.

Parameters
@@ -543,7 +550,7 @@ Public Member Functions

Send data to remote participant with a custom timestamp.

If so specified either by the selected media format and/or given RTP_CTX_ENABLE_FLAGS, uvgRTP fragments the input data into RTP packets of 1500 bytes, or to any other size defined by the application using RCC_MTU_SIZE

The frame is automatically reconstructed by the receiver if all fragments have been received successfully.

-

If application so wishes, it may override uvgRTP's own timestamp calculations and provide timestamping information for the stream itself. This requires that the application provides a sensible value for the ts parameter. If RTCP has been enabled, uvgrtp::rtcp::set_ts_info() should have been called.

+

If application so wishes, it may override uvgRTP's own timestamp calculations and provide timestamping information for the stream itself. This requires that the application provides a sensible value for the ts parameter. If RTCP has been enabled, uvgrtp::rtcp::set_ts_info() should have been called.

Parameters
dataSmart pointer to data the that should be sent
diff --git a/docs/html/classuvgrtp_1_1rtcp-members.html b/docs/html/classuvgrtp_1_1rtcp-members.html index 033b429..6a61393 100644 --- a/docs/html/classuvgrtp_1_1rtcp-members.html +++ b/docs/html/classuvgrtp_1_1rtcp-members.html @@ -79,7 +79,7 @@ $(function() { - +
dataPointer to data the that should be sent
send_app_packet(char *name, uint8_t subtype, size_t payload_len, uint8_t *payload) (defined in uvgrtp::rtcp)uvgrtp::rtcp
send_bye_packet(std::vector< uint32_t > ssrcs) (defined in uvgrtp::rtcp)uvgrtp::rtcp
send_sdes_packet(std::vector< uvgrtp::frame::rtcp_sdes_item > &items) (defined in uvgrtp::rtcp)uvgrtp::rtcp
set_ts_info(uint64_t clock_start, uint32_t clock_rate, uint32_t rtp_ts_start) (defined in uvgrtp::rtcp)uvgrtp::rtcp
set_ts_info(uint64_t clock_start, uint32_t clock_rate, uint32_t rtp_ts_start)uvgrtp::rtcp
diff --git a/docs/html/classuvgrtp_1_1rtcp.html b/docs/html/classuvgrtp_1_1rtcp.html index 4626bb0..ce3ab3c 100644 --- a/docs/html/classuvgrtp_1_1rtcp.html +++ b/docs/html/classuvgrtp_1_1rtcp.html @@ -90,8 +90,8 @@ rtp_error_t send_app_packe rtp_error_t send_bye_packet (std::vector< uint32_t > ssrcs)   - -void set_ts_info (uint64_t clock_start, uint32_t clock_rate, uint32_t rtp_ts_start) +void set_ts_info (uint64_t clock_start, uint32_t clock_rate, uint32_t rtp_ts_start) + Provide timestamping information for RTCP. More...
  rtp_error_t install_sender_hook (void(*hook)(uvgrtp::frame::rtcp_sender_report *)) @@ -106,6 +106,53 @@ rtp_error_t install_sdes_h rtp_error_t install_app_hook (void(*hook)(uvgrtp::frame::rtcp_app_packet *))   +

Member Function Documentation

+ +

◆ set_ts_info()

+ +
+
+ + + + + + + + + + + + + + + + + + + + + + + + +
void uvgrtp::rtcp::set_ts_info (uint64_t clock_start,
uint32_t clock_rate,
uint32_t rtp_ts_start 
)
+
+ +

Provide timestamping information for RTCP.

+

If the application wishes to timestamp the stream itself AND it has enabled RTCP by using RCE_RTCP, it must provide timestamping information for RTCP so sensible synchronization values can be calculated for Sender Reports

+

The application can call uvgrtp::clock::ntp::now() to get the current wall clock reading as an NTP timestamp value

+
Parameters
+ + + + +
clock_startNTP timestamp for t = 0
clock_rateClock rate of the stream
rtp_ts_startRTP timestamp for t = 0
+
+
+ +
+
The documentation for this class was generated from the following file: diff --git a/docs/html/functions.html b/docs/html/functions.html index 36de06f..cfa80e8 100644 --- a/docs/html/functions.html +++ b/docs/html/functions.html @@ -95,6 +95,9 @@ $(function() {
  • push_frame() : uvgrtp::media_stream
    • +
  • set_ts_info() +: uvgrtp::rtcp +
  • ~context() : uvgrtp::context
    • diff --git a/docs/html/functions_func.html b/docs/html/functions_func.html index 396dbfa..896058b 100644 --- a/docs/html/functions_func.html +++ b/docs/html/functions_func.html @@ -95,6 +95,9 @@ $(function() {
  • push_frame() : uvgrtp::media_stream
    • +
  • set_ts_info() +: uvgrtp::rtcp +
  • ~context() : uvgrtp::context
    • diff --git a/docs/html/media__stream_8hh_source.html b/docs/html/media__stream_8hh_source.html index ef02d9b..57e4de4 100644 --- a/docs/html/media__stream_8hh_source.html +++ b/docs/html/media__stream_8hh_source.html @@ -112,105 +112,105 @@ $(function() {
    42  * Other error return codes are defined in {conn,writer,reader,srtp}.hh */
    43  rtp_error_t init(uvgrtp::zrtp *zrtp);
    45 
    -
    68  rtp_error_t add_srtp_ctx(uint8_t *key, uint8_t *salt);
    -
    69 
    -
    93  rtp_error_t push_frame(uint8_t *data, size_t data_len, int flags);
    -
    94 
    -
    118  rtp_error_t push_frame(std::unique_ptr<uint8_t[]> data, size_t data_len, int flags);
    -
    119 
    -
    149  rtp_error_t push_frame(uint8_t *data, size_t data_len, uint32_t ts, int flags);
    -
    150 
    -
    180  rtp_error_t push_frame(std::unique_ptr<uint8_t[]> data, size_t data_len, uint32_t ts, int flags);
    -
    181 
    -
    190  uvgrtp::frame::rtp_frame *pull_frame();
    -
    191 
    -
    201  uvgrtp::frame::rtp_frame *pull_frame(size_t timeout);
    -
    202 
    -
    221  rtp_error_t install_receive_hook(void *arg, void (*hook)(void *, uvgrtp::frame::rtp_frame *));
    -
    222 
    -
    224  /* If system call dispatcher is enabled and calling application has special requirements
    -
    225  * for the deallocation of a frame, it may install a deallocation hook which is called
    -
    226  * when SCD has processed the frame
    -
    227  *
    -
    228  * Return RTP_OK on success
    -
    229  * Return RTP_INVALID_VALUE if "hook" is nullptr */
    -
    230  rtp_error_t install_deallocation_hook(void (*hook)(void *));
    -
    231 
    -
    232  /* If needed, a notification hook can be installed to uvgRTP that can be used as
    -
    233  * an information side channel to the internal state of the library.
    -
    234  *
    -
    235  * When uvgRTP encouters a situation it doesn't know how to react to,
    -
    236  * it calls the notify hook with certain notify reason number (src/util.hh).
    -
    237  * Upon receiving a notification, application may ignore it or act on it somehow
    -
    238  *
    -
    239  * Currently only one notification type is supported and only receiver uses notifications
    -
    240  *
    -
    241  * "arg" is optional argument that is passed to hook when it is called. It may be nullptr
    -
    242  *
    -
    243  * Return RTP_OK on success
    -
    244  * Return RTP_INVALID_VALUE if "hook" is nullptr */
    -
    245  rtp_error_t install_notify_hook(void *arg, void (*hook)(void *, int));
    -
    247 
    -
    258  rtp_error_t configure_ctx(int flag, ssize_t value);
    -
    259 
    -
    261  /* Setter and getter for media-specific config that can be used f.ex with Opus */
    -
    262  void set_media_config(void *config);
    -
    263  void *get_media_config();
    -
    264 
    -
    265  /* Get unique key of the media stream
    -
    266  * Used by session to index media streams */
    -
    267  uint32_t get_key();
    -
    269 
    -
    282  uvgrtp::rtcp *get_rtcp();
    -
    283 
    -
    284  private:
    -
    285  /* Initialize the connection by initializing the socket
    -
    286  * and binding ourselves to specified interface and creating
    -
    287  * an outgoing address */
    -
    288  rtp_error_t init_connection();
    -
    289 
    -
    290  uint32_t key_;
    -
    291 
    -
    292  uvgrtp::srtp *srtp_;
    -
    293  uvgrtp::srtcp *srtcp_;
    -
    294  uvgrtp::socket *socket_;
    -
    295  uvgrtp::rtp *rtp_;
    -
    296  uvgrtp::rtcp *rtcp_;
    -
    297 
    -
    298  sockaddr_in addr_out_;
    -
    299  std::string addr_;
    -
    300  std::string laddr_;
    -
    301  int src_port_;
    -
    302  int dst_port_;
    -
    303  rtp_format_t fmt_;
    -
    304  int flags_;
    -
    305 
    -
    306  /* Media context config (SCD etc.) */
    -
    307  rtp_ctx_conf_t ctx_config_;
    +
    71  rtp_error_t add_srtp_ctx(uint8_t *key, uint8_t *salt);
    +
    72 
    +
    96  rtp_error_t push_frame(uint8_t *data, size_t data_len, int flags);
    +
    97 
    +
    121  rtp_error_t push_frame(std::unique_ptr<uint8_t[]> data, size_t data_len, int flags);
    +
    122 
    +
    152  rtp_error_t push_frame(uint8_t *data, size_t data_len, uint32_t ts, int flags);
    +
    153 
    +
    183  rtp_error_t push_frame(std::unique_ptr<uint8_t[]> data, size_t data_len, uint32_t ts, int flags);
    +
    184 
    +
    193  uvgrtp::frame::rtp_frame *pull_frame();
    +
    194 
    +
    204  uvgrtp::frame::rtp_frame *pull_frame(size_t timeout);
    +
    205 
    +
    224  rtp_error_t install_receive_hook(void *arg, void (*hook)(void *, uvgrtp::frame::rtp_frame *));
    +
    225 
    +
    227  /* If system call dispatcher is enabled and calling application has special requirements
    +
    228  * for the deallocation of a frame, it may install a deallocation hook which is called
    +
    229  * when SCD has processed the frame
    +
    230  *
    +
    231  * Return RTP_OK on success
    +
    232  * Return RTP_INVALID_VALUE if "hook" is nullptr */
    +
    233  rtp_error_t install_deallocation_hook(void (*hook)(void *));
    +
    234 
    +
    235  /* If needed, a notification hook can be installed to uvgRTP that can be used as
    +
    236  * an information side channel to the internal state of the library.
    +
    237  *
    +
    238  * When uvgRTP encouters a situation it doesn't know how to react to,
    +
    239  * it calls the notify hook with certain notify reason number (src/util.hh).
    +
    240  * Upon receiving a notification, application may ignore it or act on it somehow
    +
    241  *
    +
    242  * Currently only one notification type is supported and only receiver uses notifications
    +
    243  *
    +
    244  * "arg" is optional argument that is passed to hook when it is called. It may be nullptr
    +
    245  *
    +
    246  * Return RTP_OK on success
    +
    247  * Return RTP_INVALID_VALUE if "hook" is nullptr */
    +
    248  rtp_error_t install_notify_hook(void *arg, void (*hook)(void *, int));
    +
    250 
    +
    261  rtp_error_t configure_ctx(int flag, ssize_t value);
    +
    262 
    +
    264  /* Setter and getter for media-specific config that can be used f.ex with Opus */
    +
    265  void set_media_config(void *config);
    +
    266  void *get_media_config();
    +
    267 
    +
    268  /* Get unique key of the media stream
    +
    269  * Used by session to index media streams */
    +
    270  uint32_t get_key();
    +
    272 
    +
    285  uvgrtp::rtcp *get_rtcp();
    +
    286 
    +
    287  private:
    +
    288  /* Initialize the connection by initializing the socket
    +
    289  * and binding ourselves to specified interface and creating
    +
    290  * an outgoing address */
    +
    291  rtp_error_t init_connection();
    +
    292 
    +
    293  uint32_t key_;
    +
    294 
    +
    295  uvgrtp::srtp *srtp_;
    +
    296  uvgrtp::srtcp *srtcp_;
    +
    297  uvgrtp::socket *socket_;
    +
    298  uvgrtp::rtp *rtp_;
    +
    299  uvgrtp::rtcp *rtcp_;
    +
    300 
    +
    301  sockaddr_in addr_out_;
    +
    302  std::string addr_;
    +
    303  std::string laddr_;
    +
    304  int src_port_;
    +
    305  int dst_port_;
    +
    306  rtp_format_t fmt_;
    +
    307  int flags_;
    308 
    -
    309  /* Media config f.ex. for Opus */
    -
    310  void *media_config_;
    +
    309  /* Media context config (SCD etc.) */
    +
    310  rtp_ctx_conf_t ctx_config_;
    311 
    -
    312  /* Has the media stream been initialized */
    -
    313  bool initialized_;
    +
    312  /* Media config f.ex. for Opus */
    +
    313  void *media_config_;
    314 
    -
    315  /* Primary handler keys for the RTP packet dispatcher */
    -
    316  uint32_t rtp_handler_key_;
    -
    317  uint32_t zrtp_handler_key_;
    -
    318 
    -
    319  /* RTP packet dispatcher for the receiver */
    -
    320  uvgrtp::pkt_dispatcher *pkt_dispatcher_;
    -
    321  std::thread *dispatcher_thread_;
    -
    322 
    -
    323  /* Media object associated with this media stream. */
    -
    324  uvgrtp::formats::media *media_;
    +
    315  /* Has the media stream been initialized */
    +
    316  bool initialized_;
    +
    317 
    +
    318  /* Primary handler keys for the RTP packet dispatcher */
    +
    319  uint32_t rtp_handler_key_;
    +
    320  uint32_t zrtp_handler_key_;
    +
    321 
    +
    322  /* RTP packet dispatcher for the receiver */
    +
    323  uvgrtp::pkt_dispatcher *pkt_dispatcher_;
    +
    324  std::thread *dispatcher_thread_;
    325 
    -
    326  /* Thread that keeps the holepunched connection open for unidirectional streams */
    -
    327  uvgrtp::holepuncher *holepuncher_;
    -
    328  };
    -
    329 };
    -
    330 
    -
    331 namespace uvg_rtp = uvgrtp;
    +
    326  /* Media object associated with this media stream. */
    +
    327  uvgrtp::formats::media *media_;
    +
    328 
    +
    329  /* Thread that keeps the holepunched connection open for unidirectional streams */
    +
    330  uvgrtp::holepuncher *holepuncher_;
    +
    331  };
    +
    332 };
    +
    333 
    +
    334 namespace uvg_rtp = uvgrtp;
    uvgrtp::media_stream
    Definition: media_stream.hh:18
    uvgrtp::media_stream::add_srtp_ctx
    rtp_error_t add_srtp_ctx(uint8_t *key, uint8_t *salt)
    Add keying information for user-managed SRTP session.
    uvgrtp::media_stream::get_rtcp
    uvgrtp::rtcp * get_rtcp()
    Get pointer to the RTCP object of the media stream.
    diff --git a/docs/html/rtcp_8hh_source.html b/docs/html/rtcp_8hh_source.html index 734491a..2d89a01 100644 --- a/docs/html/rtcp_8hh_source.html +++ b/docs/html/rtcp_8hh_source.html @@ -246,163 +246,162 @@ $(function() {
    180  /* Return SSRCs of all participants */
    181  std::vector<uint32_t> get_participants();
    183 
    -
    184  /* Set wallclock reading for t = 0 and random RTP timestamp from where the counting is started
    -
    185  * + clock rate for calculating the correct increment */
    -
    186  void set_ts_info(uint64_t clock_start, uint32_t clock_rate, uint32_t rtp_ts_start);
    -
    187 
    -
    188  /* Alternate way to get RTCP packets is to install a hook for them. So instead of
    -
    189  * polling an RTCP packet, user can install a function that is called when
    -
    190  * a specific RTCP packet is received. */
    -
    191  rtp_error_t install_sender_hook(void (*hook)(uvgrtp::frame::rtcp_sender_report *));
    -
    192  rtp_error_t install_receiver_hook(void (*hook)(uvgrtp::frame::rtcp_receiver_report *));
    -
    193  rtp_error_t install_sdes_hook(void (*hook)(uvgrtp::frame::rtcp_sdes_packet *));
    -
    194  rtp_error_t install_app_hook(void (*hook)(uvgrtp::frame::rtcp_app_packet *));
    -
    195 
    -
    197  /* Update RTCP-related sender statistics */
    -
    198  rtp_error_t update_sender_stats(size_t pkt_size);
    +
    198  void set_ts_info(uint64_t clock_start, uint32_t clock_rate, uint32_t rtp_ts_start);
    199 
    -
    200  /* Update RTCP-related receiver statistics */
    -
    201  static rtp_error_t recv_packet_handler(void *arg, int flags, frame::rtp_frame **out);
    -
    202 
    -
    203  /* Update RTCP-related sender statistics */
    -
    204  static rtp_error_t send_packet_handler_vec(void *arg, uvgrtp::buf_vec& buffers);
    -
    206 
    -
    207  private:
    -
    208  static void rtcp_runner(rtcp *rtcp);
    -
    209 
    -
    210  /* when we start the RTCP instance, we don't know what the SSRC of the remote is
    -
    211  * when an RTP packet is received, we must check if we've already received a packet
    -
    212  * from this sender and if not, create new entry to receiver_stats_ map */
    -
    213  bool is_participant(uint32_t ssrc);
    +
    200  /* Alternate way to get RTCP packets is to install a hook for them. So instead of
    +
    201  * polling an RTCP packet, user can install a function that is called when
    +
    202  * a specific RTCP packet is received. */
    +
    203  rtp_error_t install_sender_hook(void (*hook)(uvgrtp::frame::rtcp_sender_report *));
    +
    204  rtp_error_t install_receiver_hook(void (*hook)(uvgrtp::frame::rtcp_receiver_report *));
    +
    205  rtp_error_t install_sdes_hook(void (*hook)(uvgrtp::frame::rtcp_sdes_packet *));
    +
    206  rtp_error_t install_app_hook(void (*hook)(uvgrtp::frame::rtcp_app_packet *));
    +
    207 
    +
    209  /* Update RTCP-related sender statistics */
    +
    210  rtp_error_t update_sender_stats(size_t pkt_size);
    +
    211 
    +
    212  /* Update RTCP-related receiver statistics */
    +
    213  static rtp_error_t recv_packet_handler(void *arg, int flags, frame::rtp_frame **out);
    214 
    -
    215  /* When we receive an RTP or RTCP packet, we need to check the source address and see if it's
    -
    216  * the same address where we've received packets before.
    -
    217  *
    -
    218  * If the address is new, it means we have detected an SSRC collision and the paket should
    -
    219  * be dropped We also need to check whether this SSRC matches with our own SSRC and if it does
    -
    220  * we need to send RTCP BYE and rejoin to the session */
    -
    221  bool collision_detected(uint32_t ssrc, sockaddr_in& src_addr);
    -
    222 
    -
    223  /* Move participant from initial_peers_ to participants_ */
    -
    224  rtp_error_t add_participant(uint32_t ssrc);
    -
    225 
    -
    226  /* We've got a message from new source (the SSRC of the frame is not known to us)
    -
    227  * Initialize statistics for the peer and move it to participants_ */
    -
    228  rtp_error_t init_new_participant(uvgrtp::frame::rtp_frame *frame);
    -
    229 
    -
    230  /* Initialize the RTP Sequence related stuff of peer
    -
    231  * This function assumes that the peer already exists in the participants_ map */
    -
    232  rtp_error_t init_participant_seq(uint32_t ssrc, uint16_t base_seq);
    -
    233 
    -
    234  /* Update the SSRC's sequence related data in participants_ map
    -
    235  *
    -
    236  * Return RTP_OK if the received packet was OK
    -
    237  * Return RTP_GENERIC_ERROR if it wasn't and
    -
    238  * packet-related statistics should not be updated */
    -
    239  rtp_error_t update_participant_seq(uint32_t ssrc, uint16_t seq);
    -
    240 
    -
    241  /* Update the RTCP bandwidth variables
    -
    242  *
    -
    243  * "pkt_size" tells how much rtcp_byte_count_
    -
    244  * should be increased before calculating the new average */
    -
    245  void update_rtcp_bandwidth(size_t pkt_size);
    -
    246 
    -
    247  /* Functions for generating different kinds of reports.
    -
    248  * These functions will both generate the report and send it
    -
    249  *
    -
    250  * Return RTP_OK on success and RTP_ERROR on error */
    -
    251  rtp_error_t generate_sender_report();
    -
    252  rtp_error_t generate_receiver_report();
    -
    253 
    -
    254  /* Because struct statistics contains uvgRTP clock object we cannot
    -
    255  * zero it out without compiler complaining about it so all the fields
    -
    256  * must be set to zero manually */
    -
    257  void zero_stats(uvgrtp::rtcp_statistics *stats);
    +
    215  /* Update RTCP-related sender statistics */
    +
    216  static rtp_error_t send_packet_handler_vec(void *arg, uvgrtp::buf_vec& buffers);
    +
    218 
    +
    219  private:
    +
    220  static void rtcp_runner(rtcp *rtcp);
    +
    221 
    +
    222  /* when we start the RTCP instance, we don't know what the SSRC of the remote is
    +
    223  * when an RTP packet is received, we must check if we've already received a packet
    +
    224  * from this sender and if not, create new entry to receiver_stats_ map */
    +
    225  bool is_participant(uint32_t ssrc);
    +
    226 
    +
    227  /* When we receive an RTP or RTCP packet, we need to check the source address and see if it's
    +
    228  * the same address where we've received packets before.
    +
    229  *
    +
    230  * If the address is new, it means we have detected an SSRC collision and the paket should
    +
    231  * be dropped We also need to check whether this SSRC matches with our own SSRC and if it does
    +
    232  * we need to send RTCP BYE and rejoin to the session */
    +
    233  bool collision_detected(uint32_t ssrc, sockaddr_in& src_addr);
    +
    234 
    +
    235  /* Move participant from initial_peers_ to participants_ */
    +
    236  rtp_error_t add_participant(uint32_t ssrc);
    +
    237 
    +
    238  /* We've got a message from new source (the SSRC of the frame is not known to us)
    +
    239  * Initialize statistics for the peer and move it to participants_ */
    +
    240  rtp_error_t init_new_participant(uvgrtp::frame::rtp_frame *frame);
    +
    241 
    +
    242  /* Initialize the RTP Sequence related stuff of peer
    +
    243  * This function assumes that the peer already exists in the participants_ map */
    +
    244  rtp_error_t init_participant_seq(uint32_t ssrc, uint16_t base_seq);
    +
    245 
    +
    246  /* Update the SSRC's sequence related data in participants_ map
    +
    247  *
    +
    248  * Return RTP_OK if the received packet was OK
    +
    249  * Return RTP_GENERIC_ERROR if it wasn't and
    +
    250  * packet-related statistics should not be updated */
    +
    251  rtp_error_t update_participant_seq(uint32_t ssrc, uint16_t seq);
    +
    252 
    +
    253  /* Update the RTCP bandwidth variables
    +
    254  *
    +
    255  * "pkt_size" tells how much rtcp_byte_count_
    +
    256  * should be increased before calculating the new average */
    +
    257  void update_rtcp_bandwidth(size_t pkt_size);
    258 
    -
    259  /* Pointer to RTP context from which clock rate etc. info is collected and which is
    -
    260  * used to change SSRC if a collision is detected */
    -
    261  uvgrtp::rtp *rtp_;
    -
    262 
    -
    263  /* Secure RTCP context */
    -
    264  uvgrtp::srtcp *srtcp_;
    +
    259  /* Functions for generating different kinds of reports.
    +
    260  * These functions will both generate the report and send it
    +
    261  *
    +
    262  * Return RTP_OK on success and RTP_ERROR on error */
    +
    263  rtp_error_t generate_sender_report();
    +
    264  rtp_error_t generate_receiver_report();
    265 
    -
    266  /* RTP context flags */
    -
    267  int flags_;
    -
    268 
    -
    269  /* are we a sender or a receiver */
    -
    270  int our_role_;
    -
    271 
    -
    272  /* TODO: time_t?? */
    -
    273  size_t tp_; /* the last time an RTCP packet was transmitted */
    -
    274  size_t tc_; /* the current time */
    -
    275  size_t tn_; /* the next scheduled transmission time of an RTCP packet */
    -
    276  size_t pmembers_; /* the estimated number of session members at the time tn was last recomputed */
    -
    277  size_t members_; /* the most current estimate for the number of session members */
    -
    278  size_t senders_; /* the most current estimate for the number of senders in the session */
    -
    279 
    -
    280  /* The target RTCP bandwidth, i.e., the total bandwidth
    -
    281  * that will be used for RTCP packets by all members of this session,
    -
    282  * in octets per second. This will be a specified fraction of the
    -
    283  * "session bandwidth" parameter supplied to the application at startup. */
    -
    284  size_t rtcp_bandwidth_;
    -
    285 
    -
    286  /* Flag that is true if the application has sent data since
    -
    287  * the 2nd previous RTCP report was transmitted. */
    -
    288  bool we_sent_;
    -
    289 
    -
    290  /* The average compound RTCP packet size, in octets,
    -
    291  * over all RTCP packets sent and received by this participant. The
    -
    292  * size includes lower-layer transport and network protocol headers
    -
    293  * (e.g., UDP and IP) as explained in Section 6.2 */
    -
    294  size_t avg_rtcp_pkt_pize_;
    -
    295 
    -
    296  /* Number of RTCP packets and bytes sent and received by this participant */
    -
    297  size_t rtcp_pkt_count_;
    -
    298  size_t rtcp_byte_count_;
    -
    299 
    -
    300  /* Number of RTCP packets sent */
    -
    301  size_t rtcp_pkt_sent_count_;
    -
    302 
    -
    303  /* Flag that is true if the application has not yet sent an RTCP packet. */
    -
    304  bool initial_;
    -
    305 
    -
    306  /* Copy of our own current SSRC */
    -
    307  uint32_t ssrc_;
    -
    308 
    -
    309  /* NTP timestamp associated with initial RTP timestamp (aka t = 0) */
    -
    310  uint64_t clock_start_;
    +
    266  /* Because struct statistics contains uvgRTP clock object we cannot
    +
    267  * zero it out without compiler complaining about it so all the fields
    +
    268  * must be set to zero manually */
    +
    269  void zero_stats(uvgrtp::rtcp_statistics *stats);
    +
    270 
    +
    271  /* Pointer to RTP context from which clock rate etc. info is collected and which is
    +
    272  * used to change SSRC if a collision is detected */
    +
    273  uvgrtp::rtp *rtp_;
    +
    274 
    +
    275  /* Secure RTCP context */
    +
    276  uvgrtp::srtcp *srtcp_;
    +
    277 
    +
    278  /* RTP context flags */
    +
    279  int flags_;
    +
    280 
    +
    281  /* are we a sender or a receiver */
    +
    282  int our_role_;
    +
    283 
    +
    284  /* TODO: time_t?? */
    +
    285  size_t tp_; /* the last time an RTCP packet was transmitted */
    +
    286  size_t tc_; /* the current time */
    +
    287  size_t tn_; /* the next scheduled transmission time of an RTCP packet */
    +
    288  size_t pmembers_; /* the estimated number of session members at the time tn was last recomputed */
    +
    289  size_t members_; /* the most current estimate for the number of session members */
    +
    290  size_t senders_; /* the most current estimate for the number of senders in the session */
    +
    291 
    +
    292  /* The target RTCP bandwidth, i.e., the total bandwidth
    +
    293  * that will be used for RTCP packets by all members of this session,
    +
    294  * in octets per second. This will be a specified fraction of the
    +
    295  * "session bandwidth" parameter supplied to the application at startup. */
    +
    296  size_t rtcp_bandwidth_;
    +
    297 
    +
    298  /* Flag that is true if the application has sent data since
    +
    299  * the 2nd previous RTCP report was transmitted. */
    +
    300  bool we_sent_;
    +
    301 
    +
    302  /* The average compound RTCP packet size, in octets,
    +
    303  * over all RTCP packets sent and received by this participant. The
    +
    304  * size includes lower-layer transport and network protocol headers
    +
    305  * (e.g., UDP and IP) as explained in Section 6.2 */
    +
    306  size_t avg_rtcp_pkt_pize_;
    +
    307 
    +
    308  /* Number of RTCP packets and bytes sent and received by this participant */
    +
    309  size_t rtcp_pkt_count_;
    +
    310  size_t rtcp_byte_count_;
    311 
    -
    312  /* Clock rate of the media ie. how fast does the time increase */
    -
    313  uint32_t clock_rate_;
    +
    312  /* Number of RTCP packets sent */
    +
    313  size_t rtcp_pkt_sent_count_;
    314 
    -
    315  /* The first value of RTP timestamp (aka t = 0) */
    -
    316  uint32_t rtp_ts_start_;
    +
    315  /* Flag that is true if the application has not yet sent an RTCP packet. */
    +
    316  bool initial_;
    317 
    -
    318  std::map<uint32_t, rtcp_participant *> participants_;
    -
    319  size_t num_receivers_;
    +
    318  /* Copy of our own current SSRC */
    +
    319  uint32_t ssrc_;
    320 
    -
    321  /* statistics for RTCP Sender and Receiver Reports */
    -
    322  struct rtcp_statistics our_stats;
    +
    321  /* NTP timestamp associated with initial RTP timestamp (aka t = 0) */
    +
    322  uint64_t clock_start_;
    323 
    -
    324  /* If we expect frames from remote but haven't received anything from remote yet,
    -
    325  * the participant resides in this vector until he's moved to participants_ */
    -
    326  std::vector<rtcp_participant *> initial_participants_;
    -
    327 
    -
    328  /* Vector of sockets the RTCP runner is listening to
    -
    329  *
    -
    330  * The socket are also stored here (in addition to participants_ map) so they're easier
    -
    331  * to pass to poll when RTCP runner is listening to incoming packets */
    -
    332  std::vector<uvgrtp::socket> sockets_;
    -
    333 
    -
    334  void (*sender_hook_)(uvgrtp::frame::rtcp_sender_report *);
    -
    335  void (*receiver_hook_)(uvgrtp::frame::rtcp_receiver_report *);
    -
    336  void (*sdes_hook_)(uvgrtp::frame::rtcp_sdes_packet *);
    -
    337  void (*app_hook_)(uvgrtp::frame::rtcp_app_packet *);
    -
    338  };
    -
    339 };
    -
    340 
    -
    341 namespace uvg_rtp = uvgrtp;
    +
    324  /* Clock rate of the media ie. how fast does the time increase */
    +
    325  uint32_t clock_rate_;
    +
    326 
    +
    327  /* The first value of RTP timestamp (aka t = 0) */
    +
    328  uint32_t rtp_ts_start_;
    +
    329 
    +
    330  std::map<uint32_t, rtcp_participant *> participants_;
    +
    331  size_t num_receivers_;
    +
    332 
    +
    333  /* statistics for RTCP Sender and Receiver Reports */
    +
    334  struct rtcp_statistics our_stats;
    +
    335 
    +
    336  /* If we expect frames from remote but haven't received anything from remote yet,
    +
    337  * the participant resides in this vector until he's moved to participants_ */
    +
    338  std::vector<rtcp_participant *> initial_participants_;
    +
    339 
    +
    340  /* Vector of sockets the RTCP runner is listening to
    +
    341  *
    +
    342  * The socket are also stored here (in addition to participants_ map) so they're easier
    +
    343  * to pass to poll when RTCP runner is listening to incoming packets */
    +
    344  std::vector<uvgrtp::socket> sockets_;
    +
    345 
    +
    346  void (*sender_hook_)(uvgrtp::frame::rtcp_sender_report *);
    +
    347  void (*receiver_hook_)(uvgrtp::frame::rtcp_receiver_report *);
    +
    348  void (*sdes_hook_)(uvgrtp::frame::rtcp_sdes_packet *);
    +
    349  void (*app_hook_)(uvgrtp::frame::rtcp_app_packet *);
    +
    350  };
    +
    351 };
    +
    352 
    +
    353 namespace uvg_rtp = uvgrtp;
    uvgrtp::rtcp
    Definition: rtcp.hh:74
    +
    uvgrtp::rtcp::set_ts_info
    void set_ts_info(uint64_t clock_start, uint32_t clock_rate, uint32_t rtp_ts_start)
    Provide timestamping information for RTCP.
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]; diff --git a/docs/html/search/functions_6.js b/docs/html/search/functions_6.js index 3c0ed5c..ee3803d 100644 --- a/docs/html/search/functions_6.js +++ b/docs/html/search/functions_6.js @@ -1,4 +1,4 @@ var searchData= [ - ['_7econtext_79',['~context',['../classuvgrtp_1_1context.html#a652a201ae11685961267455ab823eb60',1,'uvgrtp::context']]] + ['set_5fts_5finfo_80',['set_ts_info',['../classuvgrtp_1_1rtcp.html#a5d0d06f9b4b7779d96d8d60938627316',1,'uvgrtp::rtcp']]] ]; diff --git a/docs/html/search/functions_7.html b/docs/html/search/functions_7.html new file mode 100644 index 0000000..46b5c0f --- /dev/null +++ b/docs/html/search/functions_7.html @@ -0,0 +1,37 @@ + + + + + + + + + + +
    +
    Loading...
    +
    + +
    Searching...
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    No Matches
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    + + diff --git a/docs/html/search/functions_7.js b/docs/html/search/functions_7.js new file mode 100644 index 0000000..018a772 --- /dev/null +++ b/docs/html/search/functions_7.js @@ -0,0 +1,4 @@ +var searchData= +[ + ['_7econtext_81',['~context',['../classuvgrtp_1_1context.html#a652a201ae11685961267455ab823eb60',1,'uvgrtp::context']]] +]; diff --git a/docs/html/search/pages_0.js b/docs/html/search/pages_0.js index 1ba14d5..d310118 100644 --- a/docs/html/search/pages_0.js +++ b/docs/html/search/pages_0.js @@ -1,4 +1,4 @@ var searchData= [ - ['uvgrtp_20public_20api_20documentation_126',['uvgRTP public API documentation',['../index.html',1,'']]] + ['uvgrtp_20public_20api_20documentation_128',['uvgRTP public API documentation',['../index.html',1,'']]] ]; diff --git a/docs/html/search/searchdata.js b/docs/html/search/searchdata.js index 4ef6b76..0d871df 100644 --- a/docs/html/search/searchdata.js +++ b/docs/html/search/searchdata.js @@ -3,7 +3,7 @@ var indexSectionsWithContent = 0: "acdgimprsu~", 1: "cmrs", 2: "u", - 3: "acdgip~", + 3: "acdgips~", 4: "r", 5: "r", 6: "u" diff --git a/include/rtcp.hh b/include/rtcp.hh index 78fc534..5b1f204 100644 --- a/include/rtcp.hh +++ b/include/rtcp.hh @@ -181,8 +181,20 @@ namespace uvgrtp { std::vector get_participants(); /// \endcond - /* Set wallclock reading for t = 0 and random RTP timestamp from where the counting is started - * + clock rate for calculating the correct increment */ + /** + * \brief Provide timestamping information for RTCP + * + * \details If the application wishes to timestamp the stream itself AND it has + * enabled RTCP by using ::RCE_RTCP, it must provide timestamping information for + * RTCP so sensible synchronization values can be calculated for Sender Reports + * + * The application can call uvgrtp::clock::ntp::now() to get the current wall clock + * reading as an NTP timestamp value + * + * \param clock_start NTP timestamp for t = 0 + * \param clock_rate Clock rate of the stream + * \param rtp_ts_start RTP timestamp for t = 0 + */ void set_ts_info(uint64_t clock_start, uint32_t clock_rate, uint32_t rtp_ts_start); /* Alternate way to get RTCP packets is to install a hook for them. So instead of