More 32->64 bit codec conversions.

In the process of swapping ULAW to a place in the extended codec space, we
found several unhandled cases, where a 32-bit integer was still being used to
handle a codec field.  Most of these have been fixed with this commit, although
there is at least one case (codec_dahdi) which depends upon outside headers to
be altered before a conversion can be made.
(Fixes AST-278, SWP-459)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231850 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 files changed, 17 insertions, 16 deletions

diff --git a/channels/chan_dahdi.c b/channels/chan_dahdi.c
index 0da2ded8c..40e7de230 100644
--- a/channels/chan_dahdi.c
+++ b/channels/chan_dahdi.c
@@ -8527,7 +8527,7 @@ static int dahdi_indicate(struct ast_channel *chan, int condition, const void *d
 static struct ast_channel *dahdi_new(struct dahdi_pvt *i, int state, int startpbx, int idx, int law, int transfercapability, const char *linkedid)
 {
 	struct ast_channel *tmp;
-	int deflaw;
+	format_t deflaw;
 	int res;
 	int x,y;
 	int features;
@@ -9909,7 +9909,7 @@ struct mwi_thread_data {
 	size_t len;
 };
 
-static int calc_energy(const unsigned char *buf, int len, int law)
+static int calc_energy(const unsigned char *buf, int len, format_t law)
 {
 	int x;
 	int sum = 0;
diff --git a/channels/chan_h323.c b/channels/chan_h323.c
index 3ccfde2d3..32d9443d4 100644
--- a/channels/chan_h323.c
+++ b/channels/chan_h323.c
@@ -3154,7 +3154,7 @@ static enum ast_rtp_glue_result oh323_get_rtp_peer(struct ast_channel *chan, str
 	return res;
 }
 
-static char *convertcap(int cap)
+static char *convertcap(format_t cap)
 {
 	switch (cap) {
 	case AST_FORMAT_G723_1:
@@ -3176,7 +3176,7 @@ static char *convertcap(int cap)
 	case AST_FORMAT_ILBC:
 		return "ILBC";
 	default:
-		ast_log(LOG_NOTICE, "Don't know how to deal with mode %d\n", cap);
+		ast_log(LOG_NOTICE, "Don't know how to deal with mode %" PRId64 "\n", cap);
 		return NULL;
 	}
 }
diff --git a/channels/chan_phone.c b/channels/chan_phone.c
index b84c39c9f..389f30401 100644
--- a/channels/chan_phone.c
+++ b/channels/chan_phone.c
@@ -94,7 +94,7 @@ static int echocancel = AEC_OFF;
 
 static int silencesupression = 0;
 
-static int prefformat = AST_FORMAT_G729A | AST_FORMAT_G723_1 | AST_FORMAT_SLINEAR | AST_FORMAT_ULAW;
+static format_t prefformat = AST_FORMAT_G729A | AST_FORMAT_G723_1 | AST_FORMAT_SLINEAR | AST_FORMAT_ULAW;
 
 /* Protect the interface list (of phone_pvt's) */
 AST_MUTEX_DEFINE_STATIC(iflock);
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b47e33084..eab73f7b8 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -212,6 +212,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include <signal.h>
 #include <sys/signal.h>
 #include <regex.h>
+#include <inttypes.h>
 
 #include "asterisk/network.h"
 #include "asterisk/paths.h"	/* need ast_config_AST_SYSTEM_NAME */
@@ -10353,7 +10354,7 @@ static void add_codec_to_sdp(const struct sip_pvt *p, format_t codec,
 
 
 	if (debug)
-		ast_verbose("Adding codec 0x%Lx (%s) to SDP\n", (long long) codec, ast_getformatname(codec));
+		ast_verbose("Adding codec 0x%" PRIx64 " (%s) to SDP\n", codec, ast_getformatname(codec));
 	if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 1, codec)) == -1)
 		return;
 
@@ -10401,7 +10402,7 @@ static void add_codec_to_sdp(const struct sip_pvt *p, format_t codec,
 
 /*! \brief Add video codec offer to SDP offer/answer body in INVITE or 200 OK */
 /* This is different to the audio one now so we can add more caps later */
-static void add_vcodec_to_sdp(const struct sip_pvt *p, int codec,
+static void add_vcodec_to_sdp(const struct sip_pvt *p, format_t codec,
 			     struct ast_str **m_buf, struct ast_str **a_buf,
 			     int debug, int *min_packet_size)
 {
@@ -10411,7 +10412,7 @@ static void add_vcodec_to_sdp(const struct sip_pvt *p, int codec,
 		return;
 
 	if (debug)
-		ast_verbose("Adding video codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
+		ast_verbose("Adding video codec 0x%" PRIx64 " (%s) to SDP\n", codec, ast_getformatname(codec));
 
 	if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->vrtp), 1, codec)) == -1)
 		return;
@@ -10726,7 +10727,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
 		}
 
 		/* Now send any other common audio and video codecs, and non-codec formats: */
-		for (x = 1LL; x <= (needtext ? AST_FORMAT_TEXT_MASK : (needvideo ? AST_FORMAT_VIDEO_MASK : AST_FORMAT_AUDIO_MASK)); x <<= 1) {
+		for (x = 1ULL; x <= (needtext ? AST_FORMAT_TEXT_MASK : (needvideo ? AST_FORMAT_VIDEO_MASK : AST_FORMAT_AUDIO_MASK)); x <<= 1) {
 			if (!(capability & x))	/* Codec not requested */
 				continue;
 
diff --git a/channels/chan_skinny.c b/channels/chan_skinny.c
index 354414e23..79eb2fb2b 100644
--- a/channels/chan_skinny.c
+++ b/channels/chan_skinny.c
@@ -1715,7 +1715,7 @@ static struct skinny_speeddial *find_speeddial_by_instance(struct skinny_device
 	return sd;
 }
 
-static int codec_skinny2ast(enum skinny_codecs skinnycodec)
+static format_t codec_skinny2ast(enum skinny_codecs skinnycodec)
 {
 	switch (skinnycodec) {
 	case SKINNY_CODEC_ALAW:
@@ -1737,7 +1737,7 @@ static int codec_skinny2ast(enum skinny_codecs skinnycodec)
 	}
 }
 
-static int codec_ast2skinny(int astcodec)
+static int codec_ast2skinny(format_t astcodec)
 {
 	switch (astcodec) {
 	case AST_FORMAT_ALAW:
@@ -5307,7 +5307,7 @@ static int handle_capabilities_res_message(struct skinny_req *req, struct skinny
 	struct skinny_device *d = s->device;
 	struct skinny_line *l;
 	uint32_t count = 0;
-	int codecs = 0;
+	format_t codecs = 0;
 	int i;
 	char buf[256];
 
@@ -5318,12 +5318,12 @@ static int handle_capabilities_res_message(struct skinny_req *req, struct skinny
 	}
 
 	for (i = 0; i < count; i++) {
-		int acodec = 0;
+		format_t acodec = 0;
 		int scodec = 0;
 		scodec = letohl(req->data.caps.caps[i].codec);
 		acodec = codec_skinny2ast(scodec);
 		if (skinnydebug)
-			ast_verb(1, "Adding codec capability '%d (%d)'\n", acodec, scodec);
+			ast_verb(1, "Adding codec capability '%" PRId64 " (%d)'\n", acodec, scodec);
 		codecs |= acodec;
 	}
 
diff --git a/channels/chan_unistim.c b/channels/chan_unistim.c
index 53ac9f94e..79df46650 100644
--- a/channels/chan_unistim.c
+++ b/channels/chan_unistim.c
@@ -407,7 +407,7 @@ struct unistim_line {
 	/*! AMA flags (for billing) */
 	int amaflags;
 	/*! Codec supported */
-	int capability;
+	format_t capability;
 	/*! Parkinglot */
 	char parkinglot[AST_MAX_CONTEXT];
 	struct unistim_line *next;
@@ -4710,7 +4710,7 @@ static char *unistim_info(struct ast_cli_entry *e, int cmd, struct ast_cli_args
 		line = device->lines;
 		while (line) {
 			ast_cli(a->fd,
-					"->name=%s fullname=%s exten=%s callid=%s cap=%d device=%p line=%p\n",
+					"->name=%s fullname=%s exten=%s callid=%s cap=%" PRId64 " device=%p line=%p\n",
 					line->name, line->fullname, line->exten, line->cid_num,
 					line->capability, line->parent, line);
 			for (i = 0; i < MAX_SUBS; i++) {