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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/4609/">https://reviewboard.asterisk.org/r/4609/</a>
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<p>Ship it!</p>
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<a href="https://reviewboard.asterisk.org/r/4609/diff/1/?file=73842#file73842line4146" style="color: black; font-weight: bold; text-decoration: underline;">/branches/13/main/channel.c</a>
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(Diff revision 1)
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static struct ast_frame *__ast_read(struct ast_channel *chan, int dropaudio)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span> * Beware of the transcode_via_slin and genericplc options as</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span> * they force any transcoding to go through slin on a bridge.</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span> * Unfortuantely transcode_via_slin is enabled by default and</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span> * genericplc is enabled in the codecs.conf.sample file.</pre></td>
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<pre style="margin-left: 2em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">I figure I should point out why translate_via_slin is on by default. The big reason is for audiohooks or any other core audio manipulation features. Let's say that you do not have translate_via_slin enabled. Alice speaks g.722 and calls Bob who speaks ulaw. Asterisk might set up a translation path like so:
Alice read path: g.722->slin->ulaw
Alice write path: ulaw->slin->g.722
Bob read path: ulaw
Bob write path: ulaw
Every frame that comes in from Alice or Bob ends up getting extra translations to slin if audiohooks are in use. Meanwhile, if you do have translate_via_slin enabled, you'd be guaranteed to have these translation paths:
Alice read path: g.722->slin
Alice write path: slin->g.722
Bob read path: ulaw->slin
Bob write path: slin->ulaw
By having slin as the read and write formats of both channels, frames that come into the core don't have to go through any extra translations in order to be used in audiohooks.</pre>
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<p>- Mark Michelson</p>
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<p>On April 9th, 2015, 8:50 p.m. UTC, rmudgett wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By rmudgett.</div>
<p style="color: grey;"><i>Updated April 9, 2015, 8:50 p.m.</i></p>
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<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24841">ASTERISK-24841</a>
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<b style="color: #575012; font-size: 10pt;">Repository: </b>
Asterisk
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">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/
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">* 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.
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/branches/13/res/res_pjsip_sdp_rtp.c <span style="color: grey">(434526)</span></li>
<li>/branches/13/main/channel.c <span style="color: grey">(434526)</span></li>
<li>/branches/13/include/asterisk/channel.h <span style="color: grey">(434526)</span></li>
<li>/branches/13/channels/chan_pjsip.c <span style="color: grey">(434526)</span></li>
<li>/branches/13/bridges/bridge_softmix.c <span style="color: grey">(434526)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/4609/diff/" style="margin-left: 3em;">View Diff</a></p>
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