> On June 30, 2014, 8:15 p.m., Corey Farrell wrote: > > /team/group/media_formats-reviewed-trunk/main/astobj2.c, lines 502-506 > > <https://reviewboard.asterisk.org/r/3687/diff/3/?file=61629#file61629line502> > > > > If you feel this won't interfere with re-merging trunk we can go ahead. > > I'd prefer that we bring all trunk updates into this branch with a > > separate commit. I fear that cherry picking will cause us to miss > > something and we could reverse a bug fix when we merge to trunk. > > Matt Jordan wrote: > I'm pretty sure there are going to be conflicts no matter what we do. > This was pretty handy in making sure that I didn't occur any additional ref > leaks as a result of this patch. > > I can revert it out of this changeset however.
I would like to do a remerge of trunk to media_formats-reviewed-trunk, I was thinking this would be better done after your next review (API renames). Even if we do cherry pick something from trunk I think a separate commit instead of combining with media_formats specific changes. - Corey ----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3687/#review12399 ----------------------------------------------------------- On July 1, 2014, 12:19 p.m., Matt Jordan wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/3687/ > ----------------------------------------------------------- > > (Updated July 1, 2014, 12:19 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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