[asterisk-commits] russell: trunk r110268 - in /trunk: main/channel.c res/res_musiconhold.c

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Thu Mar 20 12:41:23 CDT 2008


Author: russell
Date: Thu Mar 20 12:41:22 2008
New Revision: 110268

URL: http://svn.digium.com/view/asterisk?view=rev&rev=110268
Log:
Add some fixes that I made in regards to wideband codec handling to get
G.722 music on hold working for me.

(issue #12164, reported by milazzo and jsmith, patches by me)

res/res_musiconhold.c:
 - I moved a single line so that the sample queue update happened before
   ast_write().  The reason that this was a bug is that the G.722 frame
   originally says it has 320 samples in it (which is correct).  However,
   when the frame is written to a channel that uses RTP, main/rtp.c modifies
   the frame to cut the number of samples in half before it sends it on
   the wire.  This is to account for the stupid incorrect G.722 spec that
   makes it so we have to lie about the number of samples with RTP.  I should
   probably go and re-work the RTP code so it doesn't modify the frame so
   that a bug like this won't happen in the future.  However, this change to
   MOH is harmless.

main/channel.c:
 - I made two fixes in regards to generator timing.  Generators use samples
   for timing.  However, this code assumed 8 kHz samples.  In one case, it was
   a hard coded 160 samples, that is now written as the sample rate / 50.  The
   other place was dealing with timing a generator based on frames coming from
   the other direction.  However, that would have only worked if the sample
   rates for the formats in both directions were the same.  The code now takes
   into account that the sample rates may differ, and scales the generator
   samples accordingly.

Modified:
    trunk/main/channel.c
    trunk/res/res_musiconhold.c

Modified: trunk/main/channel.c
URL: http://svn.digium.com/view/asterisk/trunk/main/channel.c?view=diff&rev=110268&r1=110267&r2=110268
==============================================================================
--- trunk/main/channel.c (original)
+++ trunk/main/channel.c Thu Mar 20 12:41:22 2008
@@ -1724,7 +1724,7 @@
 	if (!tmp || !generate)
 		return 0;
 
-	res = generate(chan, tmp, 0, 160);
+	res = generate(chan, tmp, 0, ast_format_rate(chan->writeformat & AST_FORMAT_AUDIO_MASK) / 50);
 
 	chan->generatordata = tmp;
 
@@ -2256,6 +2256,7 @@
 	if (chan->generatordata &&  !ast_internal_timing_enabled(chan)) {
 		void *tmp = chan->generatordata;
 		int res;
+		int samples;
 
 		if (chan->timingfunc) {
 			if (option_debug > 1)
@@ -2264,7 +2265,15 @@
 		}
 
 		chan->generatordata = NULL;     /* reset, to let writes go through */
-		res = chan->generator->generate(chan, tmp, f->datalen, f->samples);
+
+		if (f->subclass != chan->writeformat) {
+			float factor;
+			factor = ((float) ast_format_rate(chan->writeformat)) / ((float) ast_format_rate(f->subclass));
+			samples = (int) ( ((float) f->samples) * factor );
+		} else {
+			samples = f->samples;
+		}
+		res = chan->generator->generate(chan, tmp, f->datalen, samples);
 		chan->generatordata = tmp;
 		if (res) {
 			if (option_debug > 1)

Modified: trunk/res/res_musiconhold.c
URL: http://svn.digium.com/view/asterisk/trunk/res/res_musiconhold.c?view=diff&rev=110268&r1=110267&r2=110268
==============================================================================
--- trunk/res/res_musiconhold.c (original)
+++ trunk/res/res_musiconhold.c Thu Mar 20 12:41:22 2008
@@ -312,8 +312,8 @@
 	while (state->sample_queue > 0) {
 		if ((f = moh_files_readframe(chan))) {
 			state->samples += f->samples;
+			state->sample_queue -= f->samples;
 			res = ast_write(chan, f);
-			state->sample_queue -= f->samples;
 			ast_frfree(f);
 			if (res < 0) {
 				ast_log(LOG_WARNING, "Failed to write frame to '%s': %s\n", chan->name, strerror(errno));




More information about the asterisk-commits mailing list