[asterisk-dev] [Code Review] 3687: media format improvements: Update packetization handling; improve rtp_engine's ast_rtp_codecs handling

[Matt Jordan]

-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3687/
-----------------------------------------------------------

(Updated July 1, 2014, 8:15 p.m.)


Status
------

This change has been marked as submitted.


Review request for Asterisk Developers.


Changes
-------

Committed in revision 417740


Repository: Asterisk


Description
-------

This patch started out as an attempt to fix the BUGBUGs left over packetization calls into rtp_engine; it got a little bit bigger. Things now compile and work (see Testing), so this is a good place to stop before the renaming effort.

Primarily, this patch does the following:
(1) Removes ast_rtp_codecs_packetization_set. This call was effectively a NoOp with res_rtp_asterisk/res_rtp_multicast. The various channel drivers now call ast_rtp_codecs_set_framing where appropriate.
(2) A major overhaul of ast_rtp_codec was done. This includes:
    (a) Storing the framing on the structure. This allows for the smoother in res_rtp_asterisk to easily get the framing specified without having to do major gyrations.
    (b) Payload types (which are ao2 ref counted objects) are no longer stored in an ao2_container. This container had two patterns of usage: lookups by an integer key value and iteration. Vectors work well for this type of access and - for relatively small numbers of items (which is generally the case for payload types), are much faster on both counts.
(3) The 'use_ptime' setting in res_pjsip_sdp_rtp now works. Packetization is also handled a little bit better, as both the RTP engine and format_cap API already do the job of managing the framing.

A variety of ref leaks were cleaned up as well along the way.


Diffs
-----

  /team/group/media_formats-reviewed-trunk/tests/test_format_cap.c 417724 
  /team/group/media_formats-reviewed-trunk/res/res_speech.c 417724 
  /team/group/media_formats-reviewed-trunk/res/res_rtp_asterisk.c 417724 
  /team/group/media_formats-reviewed-trunk/res/res_pjsip_sdp_rtp.c 417724 
  /team/group/media_formats-reviewed-trunk/res/res_fax.c 417724 
  /team/group/media_formats-reviewed-trunk/main/rtp_engine.c 417724 
  /team/group/media_formats-reviewed-trunk/main/format_cap.c 417724 
  /team/group/media_formats-reviewed-trunk/main/format.c 417724 
  /team/group/media_formats-reviewed-trunk/include/asterisk/vector.h 417724 
  /team/group/media_formats-reviewed-trunk/include/asterisk/rtp_engine.h 417724 
  /team/group/media_formats-reviewed-trunk/include/asterisk/frame.h 417724 
  /team/group/media_formats-reviewed-trunk/include/asterisk/format_cap.h 417724 
  /team/group/media_formats-reviewed-trunk/include/asterisk/format.h 417724 
  /team/group/media_formats-reviewed-trunk/formats/format_h264.c 417724 
  /team/group/media_formats-reviewed-trunk/formats/format_h263.c 417724 
  /team/group/media_formats-reviewed-trunk/channels/chan_skinny.c 417724 
  /team/group/media_formats-reviewed-trunk/channels/chan_sip.c 417724 
  /team/group/media_formats-reviewed-trunk/channels/chan_motif.c 417724 
  /team/group/media_formats-reviewed-trunk/channels/chan_jingle.c 417724 
  /team/group/media_formats-reviewed-trunk/channels/chan_iax2.c 417724 
  /team/group/media_formats-reviewed-trunk/channels/chan_h323.c 417724 
  /team/group/media_formats-reviewed-trunk/channels/chan_gtalk.c 417724 
  /team/group/media_formats-reviewed-trunk/bridges/bridge_softmix.c 417724 
  /team/group/media_formats-reviewed-trunk/bridges/bridge_native_rtp.c 417724 
  /team/group/media_formats-reviewed-trunk/addons/ooh323cDriver.c 417724 
  /team/group/media_formats-reviewed-trunk/addons/chan_ooh323.c 417724 

Diff: https://reviewboard.asterisk.org/r/3687/diff/


Testing
-------

Back in February, I wrote a number of single audio stream tests for the PJSIP channel driver. Eventually these will get posted up for review, but the tests cover:
 * Basic Offer/Answer of different sets of codecs (using a variety of patterns, including allow=all (ew))
 * Packetization, including use_ptime=yes|no.
 * AVPF
 * Preferred codec only (by only specifying a single supported codec), subsets of offers, etc.

These tests will eventually get put up on another review, but they gave some confidence that the mucking around in the rtp_engine that is done on this patch works.


Thanks,

Matt Jordan

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140702/d45170ce/attachment.html>