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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3687/
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(Updated July 1, 2014, 12:43 p.m.)
Review request for Asterisk Developers.
Changes
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- Removed locking in destruction routine
- Updated API to return an int in ast_rtp_codecs_payloads_default. Made memory
allocation failure there return an error.
Repository: Asterisk
Description
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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 (updated)
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/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
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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
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