[asterisk-dev] [Code Review] 3687: media format improvements: Update packetization handling; improve rtp_engine's ast_rtp_codecs handling
Corey Farrell
reviewboard at asterisk.org
Tue Jul 1 12:55:41 CDT 2014
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Ship it!
/team/group/media_formats-reviewed-trunk/main/rtp_engine.c
<https://reviewboard.asterisk.org/r/3687/#comment22657>
Would be nice to BUGBUG this procedure so we can audit callers, make them check for/react to error.
- Corey Farrell
On July 1, 2014, 1:43 p.m., Matt Jordan wrote:
>
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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, 1:43 p.m.)
>
>
> Review request for Asterisk Developers.
>
>
> 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
>
>
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