[Asterisk-code-review] feat: AudioSocket channel, application, and ARI support. (asterisk[master])

George Joseph asteriskteam at digium.com
Wed Dec 11 10:33:12 CST 2019


George Joseph has posted comments on this change. ( https://gerrit.asterisk.org/c/asterisk/+/11579 )

Change subject: feat: AudioSocket channel, application, and ARI support.
......................................................................


Patch Set 32: Code-Review-1

(4 comments)

https://gerrit.asterisk.org/c/asterisk/+/11579/32/apps/app_audiosocket.c 
File apps/app_audiosocket.c:

PS32: 
I'm still confused as to why we need the app when the dialplan could just Dial using the audiosocket channel.  But OK.


https://gerrit.asterisk.org/c/asterisk/+/11579/32/apps/app_audiosocket.c@111 
PS32, Line 111: 	if (ast_set_write_format(chan, ast_format_slin)) {
              : 		ast_log(LOG_ERROR, "Failed to set write format to SLINEAR for channel %s\n", chanName);
              : 		ao2_ref(writeFormat, -1);
              : 		ao2_ref(readFormat, -1);
              : 		return -1;
              : 	}
              : 	if (ast_set_read_format(chan, ast_format_slin)) {
              : 		ast_log(LOG_ERROR, "Failed to set read format to SLINEAR for channel %s\n", chanName);
              : 
              : 		/* Attempt to restore previous write format even though it is likely to
              : 		 * fail, since setting the read format did.
              : 		 */
              : 		if (ast_set_write_format(chan, writeFormat)) {
              : 			ast_log(LOG_ERROR, "Failed to restore write format for channel %s\n", chanName);
              : 		}
              : 		ao2_ref(writeFormat, -1);
              : 		ao2_ref(readFormat, -1);
              : 		return -1;
              : 	}
This is the "calling" channel yes?  If so is this going to cause a renegoation with the caller?


https://gerrit.asterisk.org/c/asterisk/+/11579/32/apps/app_audiosocket.c@177 
PS32, Line 177: 		struct ast_channel *targetChan;
              : 		int ms = 0;
              : 		int outfd = 0;
              : 		struct ast_frame *f;
Might be better to declare these outside the loop since this loop runs for every frame.
Just initialize ms to 0.  outfd should probably be initialized to -1 at the top of the loop since 0 is a valid value.


https://gerrit.asterisk.org/c/asterisk/+/11579/32/res/res_audiosocket.exports.in 
File res/res_audiosocket.exports.in:

https://gerrit.asterisk.org/c/asterisk/+/11579/32/res/res_audiosocket.exports.in@3 
PS32, Line 3: 		LINKER_SYMBOL_PREFIXast_audiosocket_connect;
            : 		LINKER_SYMBOL_PREFIXast_audiosocket_init;
            : 		LINKER_SYMBOL_PREFIXast_audiosocket_send_frame;
            : 		LINKER_SYMBOL_PREFIX*ast_audiosocket_receive_frame;
You can condense this down to just
    LINKER_SYMBOL_PREFIXast_audiosocket*
unless there are ast_audiosocket functions that you _don't_ want exported.



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Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Change-Id: Ie866e6c4fa13178ec76f2a6971ad3590a3a588b5
Gerrit-Change-Number: 11579
Gerrit-PatchSet: 32
Gerrit-Owner: Seán C. McCord <ulexus at gmail.com>
Gerrit-Reviewer: Benjamin Keith Ford <bford at digium.com>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Seán C. McCord <ulexus at gmail.com>
Gerrit-Comment-Date: Wed, 11 Dec 2019 16:33:12 +0000
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