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[Astlinux-commits] SF.net SVN: astlinux:[7771] branches/1.0
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From: <abe...@us...> - 2016-07-22 13:51:40
|
Revision: 7771
http://sourceforge.net/p/astlinux/code/7771
Author: abelbeck
Date: 2016-07-22 13:51:37 +0000 (Fri, 22 Jul 2016)
Log Message:
-----------
asterisk, version bump to 11.23.0, 13.10.0, add SILK CODEC support to Asterisk 13
Modified Paths:
--------------
branches/1.0/astlinux-ast13.config
branches/1.0/package/asterisk/asterisk-11-configure-cross-fix.patch
branches/1.0/package/asterisk/asterisk-11-extension-changed-verbosity-chan_sip.patch
branches/1.0/package/asterisk/asterisk-11-voicemail-multiple-recipients.patch
branches/1.0/package/asterisk/asterisk-13-configure-menuselect-cross-fix.patch
branches/1.0/package/asterisk/asterisk-13-extension-changed-verbosity-chan_sip.patch
branches/1.0/package/asterisk/asterisk.mk
branches/1.0/x86_64-configs/astlinux-ast13.config
Added Paths:
-----------
branches/1.0/package/asterisk/asterisk-13-SILK-CODEC-support.patch
Removed Paths:
-------------
branches/1.0/package/asterisk/asterisk-11-app_queue-segfault-fix.patch
Modified: branches/1.0/astlinux-ast13.config
===================================================================
--- branches/1.0/astlinux-ast13.config 2016-07-21 21:45:19 UTC (rev 7770)
+++ branches/1.0/astlinux-ast13.config 2016-07-22 13:51:37 UTC (rev 7771)
@@ -169,7 +169,7 @@
# app_voicemail=IMAP_STORAGE - disabled (requires UW_IMAP)
#
# BR2_PACKAGE_ASTERISK_MENUSELECT is not set
-# BR2_PACKAGE_ASTERISK_CODEC_SILK is not set
+BR2_PACKAGE_ASTERISK_CODEC_SILK=y
BR2_PACKAGE_ASTERISK_FOP2=y
# BR2_PACKAGE_ASTERISK_GUI is not set
BR2_PACKAGE_PHONEPROV_TOOLS=y
Deleted: branches/1.0/package/asterisk/asterisk-11-app_queue-segfault-fix.patch
===================================================================
--- branches/1.0/package/asterisk/asterisk-11-app_queue-segfault-fix.patch 2016-07-21 21:45:19 UTC (rev 7770)
+++ branches/1.0/package/asterisk/asterisk-11-app_queue-segfault-fix.patch 2016-07-22 13:51:37 UTC (rev 7771)
@@ -1,15 +0,0 @@
-X-Git-Url: http://git.asterisk.org/gitweb/?p=asterisk%2Fasterisk.git;a=blobdiff_plain;f=apps%2Fapp_queue.c;h=fa432cb76f5e41ad1168b73d3cd415daf3c851c3;hp=4a8029080d4b88a34a84f8d2b20491ba1dcbf6f0;hb=32b4320d620ffff5c55f111fc22f22fdb07f8c60;hpb=981ed6091e513bc81d4863d409762638495c2ed1
-
-diff --git a/apps/app_queue.c b/apps/app_queue.c
-index 4a80290..fa432cb 100644
---- a/apps/app_queue.c
-+++ b/apps/app_queue.c
-@@ -3641,7 +3641,7 @@ static int can_ring_entry(struct queue_ent *qe, struct callattempt *call)
- return 0;
- }
-
-- if (call->member->in_call && call->lastqueue->wrapuptime) {
-+ if (call->member->in_call && call->lastqueue && call->lastqueue->wrapuptime) {
- ast_debug(1, "%s is in call, so not available (wrapuptime %d)\n",
- call->interface, call->lastqueue->wrapuptime);
- return 0;
Modified: branches/1.0/package/asterisk/asterisk-11-configure-cross-fix.patch
===================================================================
--- branches/1.0/package/asterisk/asterisk-11-configure-cross-fix.patch 2016-07-21 21:45:19 UTC (rev 7770)
+++ branches/1.0/package/asterisk/asterisk-11-configure-cross-fix.patch 2016-07-22 13:51:37 UTC (rev 7771)
@@ -1,6 +1,6 @@
--- asterisk-1.8.25.0/configure.ac.orig 2014-01-24 16:52:23.000000000 -0600
+++ asterisk-1.8.25.0/configure.ac 2014-01-24 16:55:14.000000000 -0600
-@@ -2474,11 +2474,13 @@
+@@ -2473,11 +2473,13 @@
AC_OUTPUT
${ac_cv_path_EGREP} 'CURSES|GTK2|OSARCH|NEWT' makeopts > makeopts.acbak2
Modified: branches/1.0/package/asterisk/asterisk-11-extension-changed-verbosity-chan_sip.patch
===================================================================
--- branches/1.0/package/asterisk/asterisk-11-extension-changed-verbosity-chan_sip.patch 2016-07-21 21:45:19 UTC (rev 7770)
+++ branches/1.0/package/asterisk/asterisk-11-extension-changed-verbosity-chan_sip.patch 2016-07-22 13:51:37 UTC (rev 7771)
@@ -1,6 +1,6 @@
--- asterisk-11.17.1/channels/chan_sip.c.orig 2015-04-23 10:22:04.000000000 -0500
+++ asterisk-11.17.1/channels/chan_sip.c 2015-04-23 10:22:40.000000000 -0500
-@@ -16781,7 +16781,7 @@
+@@ -16799,7 +16799,7 @@
}
if (!force) {
Modified: branches/1.0/package/asterisk/asterisk-11-voicemail-multiple-recipients.patch
===================================================================
--- branches/1.0/package/asterisk/asterisk-11-voicemail-multiple-recipients.patch 2016-07-21 21:45:19 UTC (rev 7770)
+++ branches/1.0/package/asterisk/asterisk-11-voicemail-multiple-recipients.patch 2016-07-22 13:51:37 UTC (rev 7771)
@@ -1,6 +1,6 @@
--- asterisk-11.16.0/apps/app_voicemail.c.orig 2015-01-23 08:51:03.000000000 -0600
+++ asterisk-11.16.0/apps/app_voicemail.c 2015-02-10 09:58:04.000000000 -0600
-@@ -751,7 +751,7 @@
+@@ -753,7 +753,7 @@
char mailbox[AST_MAX_EXTENSION]; /*!< Mailbox id, unique within vm context */
char password[80]; /*!< Secret pin code, numbers only */
char fullname[80]; /*!< Full name, for directory app */
@@ -9,7 +9,7 @@
char *emailsubject; /*!< E-mail subject */
char *emailbody; /*!< E-mail body */
char pager[80]; /*!< E-mail address to pager (no attachment) */
-@@ -1194,6 +1194,8 @@
+@@ -1196,6 +1196,8 @@
vmu->maxdeletedmsg = maxdeletedmsg;
}
vmu->volgain = volgain;
@@ -18,7 +18,7 @@
ast_free(vmu->emailsubject);
vmu->emailsubject = NULL;
ast_free(vmu->emailbody);
-@@ -1505,7 +1507,8 @@
+@@ -1507,7 +1509,8 @@
} else if (!strcasecmp(var->name, "pager")) {
ast_copy_string(retval->pager, var->value, sizeof(retval->pager));
} else if (!strcasecmp(var->name, "email")) {
@@ -28,25 +28,26 @@
} else if (!strcasecmp(var->name, "fullname")) {
ast_copy_string(retval->fullname, var->value, sizeof(retval->fullname));
} else if (!strcasecmp(var->name, "context")) {
-@@ -1642,6 +1645,7 @@
- if ((vmu = (ivm ? ivm : ast_malloc(sizeof(*vmu))))) {
+@@ -1644,7 +1647,9 @@
+ if ((vmu = (ivm ? ivm : ast_calloc(1, sizeof(*vmu))))) {
++ ast_free(vmu->email);
+ ast_free(vmu->emailbody);
+ ast_free(vmu->emailsubject);
*vmu = *cur;
- if (!ivm) {
-+ vmu->email = ast_strdup(cur->email);
- vmu->emailbody = ast_strdup(cur->emailbody);
- vmu->emailsubject = ast_strdup(cur->emailsubject);
- }
-@@ -1924,6 +1928,9 @@
- {
- if (ast_test_flag(vmu, VM_ALLOCED)) {
++ vmu->email = ast_strdup(cur->email);
+ vmu->emailbody = ast_strdup(cur->emailbody);
+ vmu->emailsubject = ast_strdup(cur->emailsubject);
+ ast_set2_flag(vmu, !ivm, VM_ALLOCED);
+@@ -1928,6 +1932,8 @@
+ return;
+ }
-+ ast_free(vmu->email);
-+ vmu->email = NULL;
-+
- ast_free(vmu->emailbody);
- vmu->emailbody = NULL;
-
-@@ -2571,7 +2578,7 @@
++ ast_free(vmu->email);
++ vmu->email = NULL;
+ ast_free(vmu->emailbody);
+ vmu->emailbody = NULL;
+ ast_free(vmu->emailsubject);
+@@ -2583,7 +2589,7 @@
* of this function, we will revert back to an empty string if tempcopy
* is 1.
*/
@@ -55,7 +56,7 @@
tempcopy = 1;
}
-@@ -2583,8 +2590,10 @@
+@@ -2595,8 +2601,10 @@
command hangs. */
if (!(p = vm_mkftemp(tmp))) {
ast_log(AST_LOG_WARNING, "Unable to store '%s' (can't create temporary file)\n", fn);
@@ -68,7 +69,7 @@
return -1;
}
-@@ -4929,6 +4938,9 @@
+@@ -4947,6 +4955,9 @@
struct ast_str *str1 = ast_str_create(16), *str2 = ast_str_create(16);
char *greeting_attachment;
char filename[256];
@@ -78,7 +79,7 @@
if (!str1 || !str2) {
ast_free(str1);
-@@ -4970,7 +4982,7 @@
+@@ -4988,7 +4999,7 @@
ast_str_substitute_variables(&str1, 0, ast, fromstring);
if (check_mime(ast_str_buffer(str1))) {
@@ -87,7 +88,7 @@
ast_str_encode_mime(&str2, 0, ast_str_buffer(str1), strlen("From: "), strlen(who) + 3);
while ((ptr = strchr(ast_str_buffer(str2), ' '))) {
*ptr = '\0';
-@@ -4991,20 +5003,25 @@
+@@ -5009,20 +5020,25 @@
fprintf(p, "From: Asterisk PBX <%s>" ENDL, who);
}
@@ -126,7 +127,7 @@
}
if (!ast_strlen_zero(emailsubject) || !ast_strlen_zero(vmu->emailsubject)) {
-@@ -5014,7 +5031,7 @@
+@@ -5032,7 +5048,7 @@
prep_email_sub_vars(ast, vmu, msgnum + 1, context, mailbox, fromfolder, cidnum, cidname, dur, date, category, flag);
ast_str_substitute_variables(&str1, 0, ast, e_subj);
if (check_mime(ast_str_buffer(str1))) {
@@ -135,7 +136,7 @@
char *ptr;
ast_str_encode_mime(&str2, 0, ast_str_buffer(str1), strlen("Subject: "), 0);
while ((ptr = strchr(ast_str_buffer(str2), ' '))) {
-@@ -11766,7 +11783,7 @@
+@@ -11802,7 +11818,7 @@
ast_copy_string(vmu->fullname, s, sizeof(vmu->fullname));
}
if (stringp && (s = strsep(&stringp, ","))) {
@@ -144,7 +145,7 @@
}
if (stringp && (s = strsep(&stringp, ","))) {
ast_copy_string(vmu->pager, s, sizeof(vmu->pager));
-@@ -14035,7 +14052,7 @@
+@@ -14084,7 +14100,7 @@
}
populate_defaults(vmu);
Added: branches/1.0/package/asterisk/asterisk-13-SILK-CODEC-support.patch
===================================================================
--- branches/1.0/package/asterisk/asterisk-13-SILK-CODEC-support.patch (rev 0)
+++ branches/1.0/package/asterisk/asterisk-13-SILK-CODEC-support.patch 2016-07-22 13:51:37 UTC (rev 7771)
@@ -0,0 +1,346 @@
+From 28501051b47e6bb8968bb016abf0b3493c05fa21 Mon Sep 17 00:00:00 2001
+From: Mark Michelson <mmi...@di...>
+Date: Thu, 30 Jun 2016 15:58:53 -0500
+Subject: [PATCH] Update support for SILK format.
+
+This commit adds scaffolding in order to support the SILK audio format
+on calls. Roughly, this is what is added:
+
+* Cached silk formats. One for each possible sample rate.
+* ast_codec structures for each possible sample rate.
+* RTP payload mappings for "SILK".
+
+In addition, this change overhauls the res_format_attr_silk file in the
+following ways:
+
+* The "samplerate" attribute is scrapped. That's native to the format.
+* There are far more checks to ensure that attributes have been
+ allocated before attempting to reference them.
+* We do not SDP fmtp lines for attributes set to 0.
+
+These changes make way to be able to install a codec_silk module and
+have it actually work. It also should allow for passthrough silk calls
+in Asterisk.
+
+Change-Id: Ieeb39c95a9fecc9246bcfd3c45a6c9b51c59380e
+---
+ include/asterisk/format_cache.h | 8 +++++
+ main/codec_builtin.c | 63 ++++++++++++++++++++++++++++++++++++++
+ main/format_cache.c | 20 ++++++++++++
+ main/rtp_engine.c | 10 ++++++
+ res/res_format_attr_silk.c | 64 ++++++++++++++++++++-------------------
+ 5 files changed, 134 insertions(+), 31 deletions(-)
+
+diff --git a/include/asterisk/format_cache.h b/include/asterisk/format_cache.h
+index 9f4e06a..ff03bb4 100644
+--- a/include/asterisk/format_cache.h
++++ b/include/asterisk/format_cache.h
+@@ -224,6 +224,14 @@ extern struct ast_format *ast_format_t140_red;
+ extern struct ast_format *ast_format_none;
+
+ /*!
++ * \brief Built-in SILK format.
++ */
++extern struct ast_format *ast_format_silk8;
++extern struct ast_format *ast_format_silk12;
++extern struct ast_format *ast_format_silk16;
++extern struct ast_format *ast_format_silk24;
++
++/*!
+ * \brief Initialize format cache support within the core.
+ *
+ * \retval 0 success
+diff --git a/main/codec_builtin.c b/main/codec_builtin.c
+index d3f6517..1d329bc 100644
+--- a/main/codec_builtin.c
++++ b/main/codec_builtin.c
+@@ -772,6 +772,65 @@ static struct ast_codec t140 = {
+ .type = AST_MEDIA_TYPE_TEXT,
+ };
+
++static int silk_samples(struct ast_frame *frame)
++{
++ /* XXX This is likely not at all what's intended from this callback. However,
++ * since SILK is variable bit rate, I have no idea how to take a frame of data
++ * and determine the number of samples present. Instead, we base this on the
++ * sample rate of the codec and the expected number of samples to receive in 20ms.
++ * In testing, this has worked just fine.
++ */
++ return ast_format_get_sample_rate(frame->subclass.format) / 50;
++}
++
++static struct ast_codec silk8 = {
++ .name = "silk",
++ .description = "SILK Codec (8 KHz)",
++ .type = AST_MEDIA_TYPE_AUDIO,
++ .sample_rate = 8000,
++ .minimum_ms = 20,
++ .maximum_ms = 100,
++ .default_ms = 20,
++ .minimum_bytes = 160,
++ .samples_count = silk_samples
++};
++
++static struct ast_codec silk12 = {
++ .name = "silk",
++ .description = "SILK Codec (12 KHz)",
++ .type = AST_MEDIA_TYPE_AUDIO,
++ .sample_rate = 12000,
++ .minimum_ms = 20,
++ .maximum_ms = 100,
++ .default_ms = 20,
++ .minimum_bytes = 240,
++ .samples_count = silk_samples
++};
++
++static struct ast_codec silk16 = {
++ .name = "silk",
++ .description = "SILK Codec (16 KHz)",
++ .type = AST_MEDIA_TYPE_AUDIO,
++ .sample_rate = 16000,
++ .minimum_ms = 20,
++ .maximum_ms = 100,
++ .default_ms = 20,
++ .minimum_bytes = 320,
++ .samples_count = silk_samples
++};
++
++static struct ast_codec silk24 = {
++ .name = "silk",
++ .description = "SILK Codec (24 KHz)",
++ .type = AST_MEDIA_TYPE_AUDIO,
++ .sample_rate = 24000,
++ .minimum_ms = 20,
++ .maximum_ms = 100,
++ .default_ms = 20,
++ .minimum_bytes = 480,
++ .samples_count = silk_samples
++};
++
+ #define CODEC_REGISTER_AND_CACHE(codec) \
+ ({ \
+ int __res_ ## __LINE__ = 0; \
+@@ -843,6 +902,10 @@ int ast_codec_builtin_init(void)
+ res |= CODEC_REGISTER_AND_CACHE(t140red);
+ res |= CODEC_REGISTER_AND_CACHE(t140);
+ res |= CODEC_REGISTER_AND_CACHE(none);
++ res |= CODEC_REGISTER_AND_CACHE_NAMED("silk8", silk8);
++ res |= CODEC_REGISTER_AND_CACHE_NAMED("silk12", silk12);
++ res |= CODEC_REGISTER_AND_CACHE_NAMED("silk16", silk16);
++ res |= CODEC_REGISTER_AND_CACHE_NAMED("silk24", silk24);
+
+ return res;
+ }
+diff --git a/main/format_cache.c b/main/format_cache.c
+index 6638a78..74ebfe8 100644
+--- a/main/format_cache.c
++++ b/main/format_cache.c
+@@ -232,6 +232,14 @@ struct ast_format *ast_format_t140_red;
+ */
+ struct ast_format *ast_format_none;
+
++/*!
++ * \brief Built-in "silk" format
++ */
++struct ast_format *ast_format_silk8;
++struct ast_format *ast_format_silk12;
++struct ast_format *ast_format_silk16;
++struct ast_format *ast_format_silk24;
++
+ /*! \brief Number of buckets to use for the media format cache (should be prime for performance reasons) */
+ #define CACHE_BUCKETS 53
+
+@@ -331,6 +339,10 @@ static void format_cache_shutdown(void)
+ ao2_replace(ast_format_t140_red, NULL);
+ ao2_replace(ast_format_t140, NULL);
+ ao2_replace(ast_format_none, NULL);
++ ao2_replace(ast_format_silk8, NULL);
++ ao2_replace(ast_format_silk12, NULL);
++ ao2_replace(ast_format_silk16, NULL);
++ ao2_replace(ast_format_silk24, NULL);
+ }
+
+ int ast_format_cache_init(void)
+@@ -426,6 +438,14 @@ static void set_cached_format(const char *name, struct ast_format *format)
+ ao2_replace(ast_format_t140, format);
+ } else if (!strcmp(name, "none")) {
+ ao2_replace(ast_format_none, format);
++ } else if (!strcmp(name, "silk8")) {
++ ao2_replace(ast_format_silk8, format);
++ } else if (!strcmp(name, "silk12")) {
++ ao2_replace(ast_format_silk12, format);
++ } else if (!strcmp(name, "silk16")) {
++ ao2_replace(ast_format_silk16, format);
++ } else if (!strcmp(name, "silk24")) {
++ ao2_replace(ast_format_silk24, format);
+ }
+ }
+
+diff --git a/main/rtp_engine.c b/main/rtp_engine.c
+index 462d4c5..8d46bfd 100644
+--- a/main/rtp_engine.c
++++ b/main/rtp_engine.c
+@@ -2198,6 +2198,11 @@ int ast_rtp_engine_init(void)
+ /* Opus and VP8 */
+ set_next_mime_type(ast_format_opus, 0, "audio", "opus", 48000);
+ set_next_mime_type(ast_format_vp8, 0, "video", "VP8", 90000);
++ /* DA SILK */
++ set_next_mime_type(ast_format_silk8, 0, "audio", "silk", 8000);
++ set_next_mime_type(ast_format_silk12, 0, "audio", "silk", 12000);
++ set_next_mime_type(ast_format_silk16, 0, "audio", "silk", 16000);
++ set_next_mime_type(ast_format_silk24, 0, "audio", "silk", 24000);
+
+ /* Define the static rtp payload mappings */
+ add_static_payload(0, ast_format_ulaw, 0);
+@@ -2243,6 +2248,11 @@ int ast_rtp_engine_init(void)
+ add_static_payload(100, ast_format_vp8, 0);
+ add_static_payload(107, ast_format_opus, 0);
+
++ add_static_payload(108, ast_format_silk8, 0);
++ add_static_payload(109, ast_format_silk12, 0);
++ add_static_payload(113, ast_format_silk16, 0);
++ add_static_payload(114, ast_format_silk24, 0);
++
+ return 0;
+ }
+
+diff --git a/res/res_format_attr_silk.c b/res/res_format_attr_silk.c
+index dcbbe4c..d52ec74 100644
+--- a/res/res_format_attr_silk.c
++++ b/res/res_format_attr_silk.c
+@@ -40,7 +40,6 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+ * \note The only attribute that affects compatibility here is the sample rate.
+ */
+ struct silk_attr {
+- unsigned int samplerate;
+ unsigned int maxbitrate;
+ unsigned int dtx;
+ unsigned int fec;
+@@ -54,10 +53,15 @@ static void silk_destroy(struct ast_format *format)
+ ast_free(attr);
+ }
+
++static void attr_init(struct silk_attr *attr)
++{
++ memset(attr, 0, sizeof(*attr));
++}
++
+ static int silk_clone(const struct ast_format *src, struct ast_format *dst)
+ {
+ struct silk_attr *original = ast_format_get_attribute_data(src);
+- struct silk_attr *attr = ast_calloc(1, sizeof(*attr));
++ struct silk_attr *attr = ast_malloc(sizeof(*attr));
+
+ if (!attr) {
+ return -1;
+@@ -65,6 +69,8 @@ static int silk_clone(const struct ast_format *src, struct ast_format *dst)
+
+ if (original) {
+ *attr = *original;
++ } else {
++ attr_init(attr);
+ }
+
+ ast_format_set_attribute_data(dst, attr);
+@@ -109,17 +115,17 @@ static void silk_generate_sdp_fmtp(const struct ast_format *format, unsigned int
+ ast_str_append(str, 0, "a=fmtp:%u maxaveragebitrate=%u\r\n", payload, attr->maxbitrate);
+ }
+
+- ast_str_append(str, 0, "a=fmtp:%u usedtx=%u\r\n", payload, attr->dtx);
+- ast_str_append(str, 0, "a=fmtp:%u useinbandfec=%u\r\n", payload, attr->fec);
++ if (attr->dtx) {
++ ast_str_append(str, 0, "a=fmtp:%u usedtx=%u\r\n", payload, attr->dtx);
++ }
++ if (attr->fec) {
++ ast_str_append(str, 0, "a=fmtp:%u useinbandfec=%u\r\n", payload, attr->fec);
++ }
+ }
+
+ static enum ast_format_cmp_res silk_cmp(const struct ast_format *format1, const struct ast_format *format2)
+ {
+- struct silk_attr *attr1 = ast_format_get_attribute_data(format1);
+- struct silk_attr *attr2 = ast_format_get_attribute_data(format2);
+-
+- if (((!attr1 || !attr1->samplerate) && (!attr2 || !attr2->samplerate)) ||
+- (attr1->samplerate == attr2->samplerate)) {
++ if (ast_format_get_sample_rate(format1) == ast_format_get_sample_rate(format2)) {
+ return AST_FORMAT_CMP_EQUAL;
+ }
+
+@@ -130,13 +136,10 @@ static struct ast_format *silk_getjoint(const struct ast_format *format1, const
+ {
+ struct silk_attr *attr1 = ast_format_get_attribute_data(format1);
+ struct silk_attr *attr2 = ast_format_get_attribute_data(format2);
+- unsigned int samplerate;
+ struct ast_format *jointformat;
+ struct silk_attr *attr_res;
+
+- samplerate = attr1->samplerate & attr2->samplerate;
+- /* sample rate is the only attribute that has any bearing on if joint capabilities exist or not */
+- if (samplerate) {
++ if (ast_format_get_sample_rate(format1) != ast_format_get_sample_rate(format2)) {
+ return NULL;
+ }
+
+@@ -145,22 +148,25 @@ static struct ast_format *silk_getjoint(const struct ast_format *format1, const
+ return NULL;
+ }
+ attr_res = ast_format_get_attribute_data(jointformat);
+- attr_res->samplerate = samplerate;
+
+- /* Take the lowest max bitrate */
+- attr_res->maxbitrate = MIN(attr1->maxbitrate, attr2->maxbitrate);
++ if (!attr1 || !attr2) {
++ attr_init(attr_res);
++ } else {
++ /* Take the lowest max bitrate */
++ attr_res->maxbitrate = MIN(attr1->maxbitrate, attr2->maxbitrate);
+
+- /* Only do dtx if both sides want it. DTX is a trade off between
+- * computational complexity and bandwidth. */
+- attr_res->dtx = attr1->dtx && attr2->dtx ? 1 : 0;
++ /* Only do dtx if both sides want it. DTX is a trade off between
++ * computational complexity and bandwidth. */
++ attr_res->dtx = attr1->dtx && attr2->dtx ? 1 : 0;
+
+- /* Only do FEC if both sides want it. If a peer specifically requests not
+- * to receive with FEC, it may be a waste of bandwidth. */
+- attr_res->fec = attr1->fec && attr2->fec ? 1 : 0;
++ /* Only do FEC if both sides want it. If a peer specifically requests not
++ * to receive with FEC, it may be a waste of bandwidth. */
++ attr_res->fec = attr1->fec && attr2->fec ? 1 : 0;
+
+- /* Use the maximum packetloss percentage between the two attributes. This affects how
+- * much redundancy is used in the FEC. */
+- attr_res->packetloss_percentage = MAX(attr1->packetloss_percentage, attr2->packetloss_percentage);
++ /* Use the maximum packetloss percentage between the two attributes. This affects how
++ * much redundancy is used in the FEC. */
++ attr_res->packetloss_percentage = MAX(attr1->packetloss_percentage, attr2->packetloss_percentage);
++ }
+
+ return jointformat;
+ }
+@@ -183,9 +189,7 @@ static struct ast_format *silk_set(const struct ast_format *format, const char *
+ }
+ attr = ast_format_get_attribute_data(cloned);
+
+- if (!strcasecmp(name, "sample_rate")) {
+- attr->samplerate = val;
+- } else if (!strcasecmp(name, "max_bitrate")) {
++ if (!strcasecmp(name, "max_bitrate")) {
+ attr->maxbitrate = val;
+ } else if (!strcasecmp(name, "dtx")) {
+ attr->dtx = val;
+@@ -205,9 +209,7 @@ static const void *silk_get(const struct ast_format *format, const char *name)
+ struct silk_attr *attr = ast_format_get_attribute_data(format);
+ unsigned int *val;
+
+- if (!strcasecmp(name, "sample_rate")) {
+- val = &attr->samplerate;
+- } else if (!strcasecmp(name, "max_bitrate")) {
++ if (!strcasecmp(name, "max_bitrate")) {
+ val = &attr->maxbitrate;
+ } else if (!strcasecmp(name, "dtx")) {
+ val = &attr->dtx;
+--
+1.7.9.5
Modified: branches/1.0/package/asterisk/asterisk-13-configure-menuselect-cross-fix.patch
===================================================================
--- branches/1.0/package/asterisk/asterisk-13-configure-menuselect-cross-fix.patch 2016-07-21 21:45:19 UTC (rev 7770)
+++ branches/1.0/package/asterisk/asterisk-13-configure-menuselect-cross-fix.patch 2016-07-22 13:51:37 UTC (rev 7771)
@@ -1,6 +1,6 @@
--- asterisk-13.1.0/configure.ac.orig 2014-01-24 16:52:23.000000000 -0600
+++ asterisk-13.1.0/configure.ac 2014-01-24 16:55:14.000000000 -0600
-@@ -2630,11 +2630,13 @@
+@@ -2636,11 +2636,13 @@
AC_OUTPUT
${ac_cv_path_EGREP} 'CURSES|GTK2|OSARCH|NEWT' makeopts > makeopts.acbak2
Modified: branches/1.0/package/asterisk/asterisk-13-extension-changed-verbosity-chan_sip.patch
===================================================================
--- branches/1.0/package/asterisk/asterisk-13-extension-changed-verbosity-chan_sip.patch 2016-07-21 21:45:19 UTC (rev 7770)
+++ branches/1.0/package/asterisk/asterisk-13-extension-changed-verbosity-chan_sip.patch 2016-07-22 13:51:37 UTC (rev 7771)
@@ -1,6 +1,6 @@
--- asterisk-11.17.1/channels/chan_sip.c.orig 2015-04-23 10:22:04.000000000 -0500
+++ asterisk-11.17.1/channels/chan_sip.c 2015-04-23 10:22:40.000000000 -0500
-@@ -17333,7 +17333,7 @@
+@@ -17351,7 +17351,7 @@
}
if (!force) {
Modified: branches/1.0/package/asterisk/asterisk.mk
===================================================================
--- branches/1.0/package/asterisk/asterisk.mk 2016-07-21 21:45:19 UTC (rev 7770)
+++ branches/1.0/package/asterisk/asterisk.mk 2016-07-22 13:51:37 UTC (rev 7771)
@@ -7,9 +7,9 @@
ASTERISK_VERSION := 1.8.32.3
else
ifeq ($(BR2_PACKAGE_ASTERISK_v11),y)
-ASTERISK_VERSION := 11.22.0
+ASTERISK_VERSION := 11.23.0
else
-ASTERISK_VERSION := 13.9.1
+ASTERISK_VERSION := 13.10.0
endif
endif
ASTERISK_SOURCE := asterisk-$(ASTERISK_VERSION).tar.gz
Modified: branches/1.0/x86_64-configs/astlinux-ast13.config
===================================================================
--- branches/1.0/x86_64-configs/astlinux-ast13.config 2016-07-21 21:45:19 UTC (rev 7770)
+++ branches/1.0/x86_64-configs/astlinux-ast13.config 2016-07-22 13:51:37 UTC (rev 7771)
@@ -150,7 +150,7 @@
# app_voicemail=IMAP_STORAGE - disabled (requires UW_IMAP)
#
# BR2_PACKAGE_ASTERISK_MENUSELECT is not set
-# BR2_PACKAGE_ASTERISK_CODEC_SILK is not set
+BR2_PACKAGE_ASTERISK_CODEC_SILK=y
BR2_PACKAGE_ASTERISK_FOP2=y
# BR2_PACKAGE_ASTERISK_GUI is not set
BR2_PACKAGE_PHONEPROV_TOOLS=y
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