----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4609/#review15183 -----------------------------------------------------------
Reviewed: I didn't have any findings. - Matt Jordan On April 9, 2015, 3:50 p.m., rmudgett wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/4609/ > ----------------------------------------------------------- > > (Updated April 9, 2015, 3:50 p.m.) > > > Review request for Asterisk Developers. > > > Bugs: ASTERISK-24841 > https://issues.asterisk.org/jira/browse/ASTERISK-24841 > > > Repository: Asterisk > > > Description > ------- > > With this patch, chan_pjsip/res_pjsip now sets the native formats to the > codecs negotiated by a call. > > * The changes in chan_pjsip.c and res_pjsip_sdp_rtp.c set the native > formats to include all the negotiated audio codecs instead of only the > initial preferred audio codec and later the currently received audio > codec. > > * The audio frame handling in channel.c:ast_read() is more streamlined and > will automatically adjust to changes in received frame formats. The new > policy is to remove translation and pass the new frame format to the > receiver except if the translation was to a signed linear format. A more > long winded version is commented in ast_read() along with some caveats. > > * The audio frame handling in channel.c:ast_write() is more streamlined > and will automatically adjust any needed translation to changes in the > frame formats sent. Frame formats sent can change for many reasons such > as a recording is being played back or the bridged peer changed the format > it sends. Since it is a normal expectation that sent formats can change, > the codec mismatch warning message is demoted to a debug message. > > * Removed the short circuit check in > channel.c:ast_channel_make_compatible_helper(). Two party bridges need to > make channels compatible with each other. However, transfers and moving > channels among bridges can result in otherwise compatible channels having > sub-optimal translation paths if the make compatible check is short > circuited. A result of forcing the reevaluation of channel compatibility > is that the asterisk.conf:transcode_via_slin and codecs.conf:genericplc > options take effect consistently now. It is unfortunate that these two > options are enabled by default and negate some of the benefits to the > changes in channel.c:ast_read() by forcing translation through signed > linear on a two party bridge. > > * Improved the softmix bridge technology to better control the translation > of frames to the bridge. All of the incoming translation is now normally > handled by ast_read() instead of splitting any translation steps between > ast_read() and the slin factory. If any frame comes in with an unexpected > format then the translation path in ast_read() is updated for the next > frame and the slin factory handles the current frame translation. > > This is the final patch in a series of patches aimed at improving > translation path choices. The other patches are on the following reviews: > https://reviewboard.asterisk.org/r/4600/ > https://reviewboard.asterisk.org/r/4605/ > > > Diffs > ----- > > /branches/13/res/res_pjsip_sdp_rtp.c 434526 > /branches/13/main/channel.c 434526 > /branches/13/include/asterisk/channel.h 434526 > /branches/13/channels/chan_pjsip.c 434526 > /branches/13/bridges/bridge_softmix.c 434526 > > Diff: https://reviewboard.asterisk.org/r/4609/diff/ > > > Testing > ------- > > * The testsuite still passes as well as it ever has. > > * Manual SIP and DTMF attended transfers still function. With all patches > in the series applied, if a low speed party transfers a higher speed party > to another high speed party then when the transfer completes the resulting > call works at the higher speed. Without the patch the resulting call may > go through a sub-optimal translation path with reduced audio quality. > > * ConfBridge bridges are able to change mixing rates as different speed > participants enter and leave the bridge. Sound files played back to > individual participants may go out with a different codec than the > participant sends to the conference. If the conference bridge is mixing > at a lower rate than a participant then the conference media may go out > with a different codec than the participant sends to the conference. > > * Used app_originate to setup a call through a non-optimizing local > channel. The resulting call used the same codecs as before the patch even > between parties with different speeds. > > > Thanks, > > rmudgett > >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
