Commit some cleanups to the format type code.

 - Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits.
 - Add a native slin16 type, so that 16kHz codecs can translate without losing resolution.
   (This doesn't affect anything immediately, until another codec has wb support.)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89071 f38db490-d61c-443f-a65b-d21fe96a405b

7 files changed, 15 insertions, 20 deletions

diff --git a/channels/chan_gtalk.c b/channels/chan_gtalk.c
index 5666af95c..6f106bde9 100644
--- a/channels/chan_gtalk.c
+++ b/channels/chan_gtalk.c
@@ -198,7 +198,7 @@ static int gtalk_get_codec(struct ast_channel *chan);
 static const struct ast_channel_tech gtalk_tech = {
 	.type = "Gtalk",
 	.description = "Gtalk Channel Driver",
-	.capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
+	.capabilities = AST_FORMAT_AUDIO_MASK,
 	.requester = gtalk_request,
 	.send_digit_begin = gtalk_digit_begin,
 	.send_digit_end = gtalk_digit_end,
diff --git a/channels/chan_h323.c b/channels/chan_h323.c
index da324cf6d..99b2086ae 100644
--- a/channels/chan_h323.c
+++ b/channels/chan_h323.c
@@ -249,7 +249,7 @@ static int oh323_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
 static const struct ast_channel_tech oh323_tech = {
 	.type = "H323",
 	.description = tdesc,
-	.capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
+	.capabilities = AST_FORMAT_AUDIO_MASK,
 	.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
 	.requester = oh323_request,
 	.send_digit_begin = oh323_digit_begin,
@@ -1739,7 +1739,7 @@ static struct ast_channel *oh323_request(const char *type, int format, void *dat
 		return NULL;
 	}
 	oldformat = format;
-	format &= ((AST_FORMAT_MAX_AUDIO << 1) - 1);
+	format &= AST_FORMAT_AUDIO_MASK;
 	if (!format) {
 		ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format '%d'\n", format);
 		oh323_destroy(pvt);
diff --git a/channels/chan_jingle.c b/channels/chan_jingle.c
index e332159db..16d906c91 100644
--- a/channels/chan_jingle.c
+++ b/channels/chan_jingle.c
@@ -198,7 +198,7 @@ static int jingle_get_codec(struct ast_channel *chan);
 static const struct ast_channel_tech jingle_tech = {
 	.type = "Jingle",
 	.description = "Jingle Channel Driver",
-	.capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
+	.capabilities = AST_FORMAT_AUDIO_MASK,
 	.requester = jingle_request,
 	.send_digit_begin = jingle_digit_begin,
 	.send_digit_end = jingle_digit_end,
@@ -340,10 +340,7 @@ static int jingle_accept_call(struct jingle *client, struct jingle_pvt *p)
 				continue;
 			if (alreadysent & pref_codec)
 				continue;
-			if (pref_codec <= AST_FORMAT_MAX_AUDIO)
-				add_codec_to_answer(p, pref_codec, dcodecs);
-			else
-				add_codec_to_answer(p, pref_codec, dcodecs);
+			add_codec_to_answer(p, pref_codec, dcodecs);
 			alreadysent |= pref_codec;
 		}
 		payload_red = iks_new("payload-type");
diff --git a/channels/chan_mgcp.c b/channels/chan_mgcp.c
index 9eb0fa112..b0b138a00 100644
--- a/channels/chan_mgcp.c
+++ b/channels/chan_mgcp.c
@@ -2076,7 +2076,7 @@ static int add_sdp(struct mgcp_request *resp, struct mgcp_subchannel *sub, struc
 	snprintf(c, sizeof(c), "c=IN IP4 %s\r\n", ast_inet_ntoa(dest.sin_addr));
 	ast_copy_string(t, "t=0 0\r\n", sizeof(t));
 	snprintf(m, sizeof(m), "m=audio %d RTP/AVP", ntohs(dest.sin_port));
-	for (x = 1; x <= AST_FORMAT_MAX_AUDIO; x <<= 1) {
+	for (x = 1; x <= AST_FORMAT_AUDIO_MASK; x <<= 1) {
 		if (p->capability & x) {
 			if (mgcpdebug) {
 				ast_verbose("Answering with capability %d\n", x);
@@ -2142,7 +2142,7 @@ static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp
 		return 0;
 	}
 	ast_copy_string(local, "p:20", sizeof(local));
-	for (x=1;x<= AST_FORMAT_MAX_AUDIO; x <<= 1) {
+	for (x = 1; x <= AST_FORMAT_AUDIO_MASK; x <<= 1) {
 		if (p->capability & x) {
 			snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype(1, x, 0));
 			strncat(local, tmp, sizeof(local) - strlen(local) - 1);
@@ -2172,7 +2172,7 @@ static int transmit_connect_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp
 	struct mgcp_endpoint *p = sub->parent;
 
 	ast_copy_string(local, "p:20", sizeof(local));
-	for (x=1;x<= AST_FORMAT_MAX_AUDIO; x <<= 1) {
+	for (x = 1; x <= AST_FORMAT_AUDIO_MASK; x <<= 1) {
 		if (p->capability & x) {
 			snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype(1, x, 0));
 			strncat(local, tmp, sizeof(local) - strlen(local) - 1);
diff --git a/channels/chan_phone.c b/channels/chan_phone.c
index 33f515d5e..17204fda5 100644
--- a/channels/chan_phone.c
+++ b/channels/chan_phone.c
@@ -594,7 +594,7 @@ static struct ast_frame  *phone_read(struct ast_channel *ast)
 	}
 	p->fr.samples = 240;
 	p->fr.datalen = res;
-	p->fr.frametype = p->lastinput <= AST_FORMAT_MAX_AUDIO ?
+	p->fr.frametype = p->lastinput <= AST_FORMAT_AUDIO_MASK ?
                           AST_FRAME_VOICE : 
 			  p->lastinput <= AST_FORMAT_PNG ? AST_FRAME_IMAGE 
 			  : AST_FRAME_VIDEO;
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 7b4764b6d..2e231d35d 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -155,8 +155,6 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
 #define XMIT_ERROR		-2
 
-#define VIDEO_CODEC_MASK        0x1fc0000 /*!< Video codecs from H.261 thru AST_FORMAT_MAX_VIDEO */
-
 /* #define VOCAL_DATA_HACK */
 
 #define DEFAULT_DEFAULT_EXPIRY  120
@@ -7228,20 +7226,20 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
 	}
 
 	/* Now send any other common audio and video codecs, and non-codec formats: */
-	for (x = 1; x <= (needtext ? AST_FORMAT_MAX_TEXT : (needvideo ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO)); x <<= 1) {
+	for (x = 1; x <= (needtext ? AST_FORMAT_TEXT_MASK : (needvideo ? AST_FORMAT_VIDEO_MASK : AST_FORMAT_AUDIO_MASK)); x <<= 1) {
 		if (!(capability & x))	/* Codec not requested */
 			continue;
 
 		if (alreadysent & x)	/* Already added to SDP */
 			continue;
 
-		if (x <= AST_FORMAT_MAX_AUDIO)
+		if (x & AST_FORMAT_AUDIO_MASK)
 			add_codec_to_sdp(p, x, SDP_SAMPLE_RATE(x),
 				 &m_audio, &a_audio, debug, &min_audio_packet_size);
-		else if (x <= AST_FORMAT_MAX_VIDEO) 
+		else if (x & AST_FORMAT_VIDEO_MASK) 
 			add_vcodec_to_sdp(p, x, 90000,
 				 &m_video, &a_video, debug, &min_video_packet_size);
-		else if (x <= AST_FORMAT_MAX_TEXT)
+		else if (x & AST_FORMAT_TEXT_MASK)
 			add_tcodec_to_sdp(p, x, 1000,
 				 &m_text, &a_text, debug, &min_text_packet_size);
 	}
diff --git a/channels/chan_skinny.c b/channels/chan_skinny.c
index 86d2a5b95..6808d32a4 100644
--- a/channels/chan_skinny.c
+++ b/channels/chan_skinny.c
@@ -1239,7 +1239,7 @@ static int handle_time_date_req_message(struct skinny_req *req, struct skinnyses
 static const struct ast_channel_tech skinny_tech = {
 	.type = "Skinny",
 	.description = tdesc,
-	.capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
+	.capabilities = AST_FORMAT_AUDIO_MASK,
 	.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
 	.requester = skinny_request,
 	.devicestate = skinny_devicestate,
@@ -5395,7 +5395,7 @@ static struct ast_channel *skinny_request(const char *type, int format, void *da
 
 	oldformat = format;
 	
-	if (!(format &= ((AST_FORMAT_MAX_AUDIO << 1) - 1))) {
+	if (!(format &= AST_FORMAT_AUDIO_MASK)) {
 		ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format '%d'\n", format);
 		return NULL;	
 	}