[Asterisk-cvs] asterisk Makefile,1.42,1.43 channel.c,1.50,1.51 frame.c,1.9,1.10 rtp.c,1.30,1.31
markster at lists.digium.com
markster at lists.digium.com
Mon Nov 3 20:14:04 CST 2003
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Update of /usr/cvsroot/asterisk
In directory mongoose.digium.com:/tmp/cvs-serv12608
Modified Files:
Makefile channel.c frame.c rtp.c
Log Message:
Remove really broken MP3 stuff in favor of G.726 in the near future
Index: Makefile
===================================================================
RCS file: /usr/cvsroot/asterisk/Makefile,v
retrieving revision 1.42
retrieving revision 1.43
diff -u -d -r1.42 -r1.43
--- Makefile 27 Oct 2003 20:00:40 -0000 1.42
+++ Makefile 4 Nov 2003 02:40:09 -0000 1.43
@@ -311,6 +311,7 @@
rm -f $(DESTDIR)$(MODULES_DIR)/chan_ixj.so
rm -f $(DESTDIR)$(MODULES_DIR)/chan_tor.so
rm -f $(DESTDIR)$(MODULES_DIR)/cdr_mysql.so
+ rm -f $(DESTDIR)$(MODULES_DIR)/codec_mp3_d.so
mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds
mkdir -p $(DESTDIR)$(ASTLOGDIR)/cdr-csv
mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/keys
Index: channel.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channel.c,v
retrieving revision 1.50
retrieving revision 1.51
diff -u -d -r1.50 -r1.51
--- channel.c 9 Oct 2003 14:12:26 -0000 1.50
+++ channel.c 4 Nov 2003 02:40:09 -0000 1.51
@@ -233,6 +233,8 @@
AST_FORMAT_ALAW,
/* Okay, well, signed linear is easy to translate into other stuff */
AST_FORMAT_SLINEAR,
+ /* G.726 is standard ADPCM */
+ AST_FORMAT_G726,
/* ADPCM has great sound quality and is still pretty easy to translate */
AST_FORMAT_ADPCM,
/* Okay, we're down to vocoders now, so pick GSM because it's small and easier to
@@ -249,8 +251,6 @@
AST_FORMAT_G729A,
/* Down to G.723.1 which is proprietary but at least designed for voice */
AST_FORMAT_G723_1,
- /* Last and least, MP3 which was of course never designed for real-time voice */
- AST_FORMAT_MP3,
};
Index: frame.c
===================================================================
RCS file: /usr/cvsroot/asterisk/frame.c,v
retrieving revision 1.9
retrieving revision 1.10
diff -u -d -r1.9 -r1.10
--- frame.c 8 Oct 2003 21:57:43 -0000 1.9
+++ frame.c 4 Nov 2003 02:40:09 -0000 1.10
@@ -353,8 +353,8 @@
return "ULAW";
else if (format == AST_FORMAT_ALAW)
return "ALAW";
- else if (format == AST_FORMAT_MP3)
- return "MP3";
+ else if (format == AST_FORMAT_G726)
+ return "G726";
else if (format == AST_FORMAT_SLINEAR)
return "SLINR";
else if (format == AST_FORMAT_LPC10)
@@ -388,8 +388,8 @@
return AST_FORMAT_ULAW;
else if (!strcasecmp(name, "alaw"))
return AST_FORMAT_ALAW;
- else if (!strcasecmp(name, "mp3"))
- return AST_FORMAT_MP3;
+ else if (!strcasecmp(name, "g726"))
+ return AST_FORMAT_G726;
else if (!strcasecmp(name, "slinear"))
return AST_FORMAT_SLINEAR;
else if (!strcasecmp(name, "lpc10"))
Index: rtp.c
===================================================================
RCS file: /usr/cvsroot/asterisk/rtp.c,v
retrieving revision 1.30
retrieving revision 1.31
diff -u -d -r1.30 -r1.31
--- rtp.c 3 Nov 2003 07:14:43 -0000 1.30
+++ rtp.c 4 Nov 2003 02:40:09 -0000 1.31
@@ -508,7 +508,7 @@
{{1, AST_FORMAT_GSM}, "audio", "GSM"},
{{1, AST_FORMAT_ULAW}, "audio", "PCMU"},
{{1, AST_FORMAT_ALAW}, "audio", "PCMA"},
- {{1, AST_FORMAT_MP3}, "audio", "MPA"},
+ {{1, AST_FORMAT_G726}, "audio", "G726-32"},
{{1, AST_FORMAT_ADPCM}, "audio", "DVI4"},
{{1, AST_FORMAT_SLINEAR}, "audio", "L16"},
{{1, AST_FORMAT_LPC10}, "audio", "LPC"},
@@ -529,6 +529,7 @@
table for transmission */
static struct rtpPayloadType static_RTP_PT[MAX_RTP_PT] = {
[0] = {1, AST_FORMAT_ULAW},
+ [2] = {1, AST_FORMAT_G726}, // Technically this is G.721, but if Cisco can do it, so can we...
[3] = {1, AST_FORMAT_GSM},
[4] = {1, AST_FORMAT_G723_1},
[5] = {1, AST_FORMAT_ADPCM}, // 8 kHz
@@ -538,7 +539,6 @@
[10] = {1, AST_FORMAT_SLINEAR}, // 2 channels
[11] = {1, AST_FORMAT_SLINEAR}, // 1 channel
[13] = {0, AST_RTP_CN},
- [14] = {1, AST_FORMAT_MP3},
[16] = {1, AST_FORMAT_ADPCM}, // 11.025 kHz
[17] = {1, AST_FORMAT_ADPCM}, // 22.050 kHz
[18] = {1, AST_FORMAT_G729A},
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