allow for packetization on rtp channel drivers, need to add

option for setting our own packetization as apposed to just doing 
what is asked.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43243 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2 files changed, 167 insertions, 110 deletions

diff --git a/main/frame.c b/main/frame.c
index 07714e07f..be18148a3 100644
--- a/main/frame.c
+++ b/main/frame.c
@@ -101,24 +101,19 @@ struct ast_smoother {
 };
 
 /*! \brief Definition of supported media formats (codecs) */
-static struct ast_format_list {
-	int visible;	/*!< Can we see this entry */
-	int bits;	/*!< bitmask value */
-	char *name;	/*!< short name */
-	char *desc;	/*!< Description */
-} AST_FORMAT_LIST[] = {					/*!< Bit number: comment  - Bit numbers are hard coded in show_codec() */
-	{ 1, AST_FORMAT_G723_1 , "g723" , "G.723.1"},	/*!<  1 */
-	{ 1, AST_FORMAT_GSM, "gsm" , "GSM"},		/*!<  2: codec_gsm.c */
-	{ 1, AST_FORMAT_ULAW, "ulaw", "G.711 u-law" },	/*!<  3: codec_ulaw.c */
-	{ 1, AST_FORMAT_ALAW, "alaw", "G.711 A-law" },	/*!<  4: codec_alaw.c */
-	{ 1, AST_FORMAT_G726, "g726", "G.726 RFC3551" },/*!<  5: codec_g726.c */
-	{ 1, AST_FORMAT_ADPCM, "adpcm" , "ADPCM"},	/*!<  6: codec_adpcm.c */
-	{ 1, AST_FORMAT_SLINEAR, "slin",  "16 bit Signed Linear PCM"},	/*!< 7 */
-	{ 1, AST_FORMAT_LPC10, "lpc10", "LPC10" },	/*!<  8: codec_lpc10.c */
-	{ 1, AST_FORMAT_G729A, "g729", "G.729A" },	/*!<  9: Binary commercial distribution */
-	{ 1, AST_FORMAT_SPEEX, "speex", "SpeeX" },	/*!< 10: codec_speex.c */
-	{ 1, AST_FORMAT_ILBC, "ilbc", "iLBC"},		/*!< 11: codec_ilbc.c */
-	{ 1, AST_FORMAT_G726_AAL2, "g726aal2", "G.726 AAL2" },	/*!<  12: codec_g726.c */
+static struct ast_format_list AST_FORMAT_LIST[] = {					/*!< Bit number: comment  - Bit numbers are hard coded in show_codec() */
+	{ 1, AST_FORMAT_G723_1 , "g723" , "G.723.1", 24, 30, 300, 30, 30 },	/*!<  1 */
+	{ 1, AST_FORMAT_GSM, "gsm" , "GSM", 33, 20, 60, 20, 20 },		/*!<  2: codec_gsm.c */
+	{ 1, AST_FORMAT_ULAW, "ulaw", "G.711 u-law", 80, 10, 30, 10, 20 },	/*!<  3: codec_ulaw.c */
+	{ 1, AST_FORMAT_ALAW, "alaw", "G.711 A-law", 80, 10, 30, 10, 20 },	/*!<  4: codec_alaw.c */
+	{ 1, AST_FORMAT_G726, "g726", "G.726 RFC3551", 40, 10, 50, 10, 20 },/*!<  5: codec_g726.c */
+	{ 1, AST_FORMAT_ADPCM, "adpcm" , "ADPCM", 40, 10, 30, 10, 20 },	/*!<  6: codec_adpcm.c */
+	{ 1, AST_FORMAT_SLINEAR, "slin", "16 bit Signed Linear PCM", 160, 10, 70, 10, 20, AST_SMOOTHER_FLAG_BE },	/*!< 7 */
+	{ 1, AST_FORMAT_LPC10, "lpc10", "LPC10", 7, 20, 20, 20, 20 },	/*!<  8: codec_lpc10.c */ 
+	{ 1, AST_FORMAT_G729A, "g729", "G.729A", 10, 10, 230, 10, 20, AST_SMOOTHER_FLAG_G729 },	/*!<  9: Binary commercial distribution */
+	{ 1, AST_FORMAT_SPEEX, "speex", "SpeeX", 10, 10, 60, 10, 20 },	/*!< 10: codec_speex.c */
+	{ 1, AST_FORMAT_ILBC, "ilbc", "iLBC", 50, 30, 30, 30, 30 },		/*!< 11: codec_ilbc.c */ /* inc=30ms - workaround */
+	{ 1, AST_FORMAT_G726_AAL2, "g726aal2", "G.726 AAL2", 40, 10, 50, 10, 20 },	/*!<  12: codec_g726.c */
 	{ 0, 0, "nothing", "undefined" },
 	{ 0, 0, "nothing", "undefined" },
 	{ 0, 0, "nothing", "undefined" },
@@ -164,6 +159,11 @@ void ast_smoother_set_flags(struct ast_smoother *s, int flags)
 	s->flags = flags;
 }
 
+int ast_smoother_test_flag(struct ast_smoother *s, int flag)
+{
+	return (s->flags & flag);
+}
+
 int __ast_smoother_feed(struct ast_smoother *s, struct ast_frame *f, int swap)
 {
 	if (f->frametype != AST_FRAME_VOICE) {
@@ -1099,6 +1099,7 @@ void ast_codec_pref_remove(struct ast_codec_pref *pref, int format)
 	struct ast_codec_pref oldorder;
 	int x, y = 0;
 	int slot;
+	int size;
 
 	if(!pref->order[0])
 		return;
@@ -1108,10 +1109,13 @@ void ast_codec_pref_remove(struct ast_codec_pref *pref, int format)
 
 	for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
 		slot = oldorder.order[x];
+		size = oldorder.framing[x];
 		if(! slot)
 			break;
-		if(AST_FORMAT_LIST[slot-1].bits != format)
-			pref->order[y++] = slot;
+		if(AST_FORMAT_LIST[slot-1].bits != format) {
+			pref->order[y] = slot;
+			pref->framing[y++] = size;
+		}
 	}
 	
 }
@@ -1143,6 +1147,84 @@ int ast_codec_pref_append(struct ast_codec_pref *pref, int format)
 }
 
 
+/*! \brief Set packet size for codec */
+int ast_codec_pref_setsize(struct ast_codec_pref *pref, int format, int framems)
+{
+	int x, index = -1;
+
+	for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
+		if(AST_FORMAT_LIST[x].bits == format) {
+			index = x;
+			break;
+		}
+	}
+
+	if(index < 0)
+		return -1;
+
+	/* size validation */
+	if(!framems)
+		framems = AST_FORMAT_LIST[index].def_ms;
+
+	if(AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */
+		framems -= framems % AST_FORMAT_LIST[index].inc_ms;
+
+	if(framems < AST_FORMAT_LIST[index].min_ms)
+		framems = AST_FORMAT_LIST[index].min_ms;
+
+	if(framems > AST_FORMAT_LIST[index].max_ms)
+		framems = AST_FORMAT_LIST[index].max_ms;
+
+
+	for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
+		if(pref->order[x] == (index + 1)) {
+			pref->framing[x] = framems;
+			break;
+		}
+	}
+
+	return x;
+}
+
+/*! \brief Get packet size for codec */
+struct ast_format_list ast_codec_pref_getsize(struct ast_codec_pref *pref, int format)
+{
+	int x, index = -1, framems = 0;
+	struct ast_format_list fmt;
+
+	for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
+		if(AST_FORMAT_LIST[x].bits == format) {
+			fmt = AST_FORMAT_LIST[x];
+			index = x;
+			break;
+		}
+	}
+
+	for (x = 0; x < sizeof(AST_FORMAT_LIST) / sizeof(AST_FORMAT_LIST[0]); x++) {
+		if(pref->order[x] == (index + 1)) {
+			framems = pref->framing[x];
+			break;
+		}
+	}
+
+	/* size validation */
+	if(!framems)
+		framems = AST_FORMAT_LIST[index].def_ms;
+
+	if(AST_FORMAT_LIST[index].inc_ms && framems % AST_FORMAT_LIST[index].inc_ms) /* avoid division by zero */
+		framems -= framems % AST_FORMAT_LIST[index].inc_ms;
+
+	if(framems < AST_FORMAT_LIST[index].min_ms)
+		framems = AST_FORMAT_LIST[index].min_ms;
+
+	if(framems > AST_FORMAT_LIST[index].max_ms)
+		framems = AST_FORMAT_LIST[index].max_ms;
+
+	fmt.cur_ms = framems;
+
+	return fmt;
+}
+
 /*! \brief Pick a codec */
 int ast_codec_choose(struct ast_codec_pref *pref, int formats, int find_best)
 {
@@ -1172,9 +1254,22 @@ void ast_parse_allow_disallow(struct ast_codec_pref *pref, int *mask, const char
 	char *parse;
 	char *this;
 	int format;
+	char *psize;
+	int framems;
 
 	parse = ast_strdupa(list);
 	while ((this = strsep(&parse, ","))) {
+		framems = 0;
+		if ((psize = strrchr(this, ':'))) {
+			*psize = '\0';
+			psize++;
+			if (option_debug)
+				ast_log(LOG_DEBUG,"Packetization for codec: %s is %s\n", this, psize);
+			framems = strtol(psize,NULL,10);
+			if (framems == LONG_MIN || framems == LONG_MAX) {
+				framems = 0;
+			}
+		}
 		if (!(format = ast_getformatbyname(this))) {
 			ast_log(LOG_WARNING, "Cannot %s unknown format '%s'\n", allowing ? "allow" : "disallow", this);
 			continue;
@@ -1192,8 +1287,10 @@ void ast_parse_allow_disallow(struct ast_codec_pref *pref, int *mask, const char
 		 */
 		if (pref && (format & AST_FORMAT_AUDIO_MASK)) {
 			if (strcasecmp(this, "all")) {
-				if (allowing)
+				if (allowing) {
 					ast_codec_pref_append(pref, format);
+					ast_codec_pref_setsize(pref, format, framems);
+				}
 				else
 					ast_codec_pref_remove(pref, format);
 			} else if (!allowing) {
diff --git a/main/rtp.c b/main/rtp.c
index 9813ff996..5d426fe52 100644
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -160,6 +160,7 @@ struct ast_rtp {
 	int rtp_lookup_code_cache_code;
 	int rtp_lookup_code_cache_result;
 	struct ast_rtcp *rtcp;
+	struct ast_codec_pref pref;
 	struct ast_rtp *bridged;        /*!< Who we are Packet bridged to */
 };
 
@@ -2480,6 +2481,35 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
 	return 0;
 }
 
+int ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs)
+{
+	int x;
+	for (x = 0; x < 32; x++) {  /* Ugly way */
+		rtp->pref.order[x] = prefs->order[x];
+		rtp->pref.framing[x] = prefs->framing[x];
+	}
+	if (rtp->smoother)
+		ast_smoother_free(rtp->smoother);
+	rtp->smoother = NULL;
+	return 0;
+}
+
+struct ast_codec_pref *ast_rtp_codec_getpref(struct ast_rtp *rtp)
+{
+	return &rtp->pref;
+}
+
+int ast_rtp_codec_getformat(int pt)
+{
+	if (pt < 0 || pt > MAX_RTP_PT)
+		return 0; /* bogus payload type */
+
+	if (static_RTP_PT[pt].isAstFormat)
+		return static_RTP_PT[pt].code;
+	else
+		return 0;
+}
+
 int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
 {
 	struct ast_frame *f;
@@ -2522,99 +2552,29 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
 		rtp->smoother = NULL;
 	}
 
-
-	switch(subclass) {
-	case AST_FORMAT_SLINEAR:
-		if (!rtp->smoother) {
-			rtp->smoother = ast_smoother_new(320);
-		}
-		if (!rtp->smoother) {
-			ast_log(LOG_WARNING, "Unable to create smoother :(\n");
-			return -1;
-		}
-		ast_smoother_feed_be(rtp->smoother, _f);
-		
-		while((f = ast_smoother_read(rtp->smoother)))
-			ast_rtp_raw_write(rtp, f, codec);
-		break;
-	case AST_FORMAT_ULAW:
-	case AST_FORMAT_ALAW:
-		if (!rtp->smoother) {
-			rtp->smoother = ast_smoother_new(160);
-		}
-		if (!rtp->smoother) {
-			ast_log(LOG_WARNING, "Unable to create smoother :(\n");
-			return -1;
-		}
-		ast_smoother_feed(rtp->smoother, _f);
-		
-		while((f = ast_smoother_read(rtp->smoother)))
-			ast_rtp_raw_write(rtp, f, codec);
-		break;
-	case AST_FORMAT_ADPCM:
-	case AST_FORMAT_G726:
-	case AST_FORMAT_G726_AAL2:
-		if (!rtp->smoother) {
-			rtp->smoother = ast_smoother_new(80);
-		}
-		if (!rtp->smoother) {
-			ast_log(LOG_WARNING, "Unable to create smoother :(\n");
-			return -1;
-		}
-		ast_smoother_feed(rtp->smoother, _f);
-		
-		while((f = ast_smoother_read(rtp->smoother)))
-			ast_rtp_raw_write(rtp, f, codec);
-		break;
-	case AST_FORMAT_G729A:
-		if (!rtp->smoother) {
-			rtp->smoother = ast_smoother_new(20);
-			if (rtp->smoother)
-				ast_smoother_set_flags(rtp->smoother, AST_SMOOTHER_FLAG_G729);
-		}
-		if (!rtp->smoother) {
-			ast_log(LOG_WARNING, "Unable to create g729 smoother :(\n");
-			return -1;
-		}
-		ast_smoother_feed(rtp->smoother, _f);
-		
-		while((f = ast_smoother_read(rtp->smoother)))
-			ast_rtp_raw_write(rtp, f, codec);
-		break;
-	case AST_FORMAT_GSM:
-		if (!rtp->smoother) {
-			rtp->smoother = ast_smoother_new(33);
-		}
-		if (!rtp->smoother) {
-			ast_log(LOG_WARNING, "Unable to create GSM smoother :(\n");
-			return -1;
-		}
-		ast_smoother_feed(rtp->smoother, _f);
-		while((f = ast_smoother_read(rtp->smoother)))
-			ast_rtp_raw_write(rtp, f, codec);
-		break;
-	case AST_FORMAT_ILBC:
-		if (!rtp->smoother) {
-			rtp->smoother = ast_smoother_new(50);
+	if (!rtp->smoother) {
+		struct ast_format_list fmt = ast_codec_pref_getsize(&rtp->pref, subclass);
+		if (fmt.inc_ms) { /* if codec parameters is set / avoid division by zero */
+			if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) {
+				ast_log(LOG_WARNING, "Unable to create smoother: format: %d ms: %d len: %d\n", subclass, fmt.cur_ms, ((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms));
+				return -1;
+			}
+			if (fmt.flags)
+				ast_smoother_set_flags(rtp->smoother, fmt.flags);
+			if (option_debug)
+				ast_log(LOG_DEBUG, "Created smoother: format: %d ms: %d len: %d\n", subclass, fmt.cur_ms, ((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms));
 		}
-		if (!rtp->smoother) {
-			ast_log(LOG_WARNING, "Unable to create ILBC smoother :(\n");
-			return -1;
+	}
+	if (rtp->smoother) {
+		if (ast_smoother_test_flag(rtp->smoother, AST_SMOOTHER_FLAG_BE)) {
+			ast_smoother_feed_be(rtp->smoother, _f);
+		} else {
+			ast_smoother_feed(rtp->smoother, _f);
 		}
-		ast_smoother_feed(rtp->smoother, _f);
+
 		while((f = ast_smoother_read(rtp->smoother)))
 			ast_rtp_raw_write(rtp, f, codec);
-		break;
-	default:	
-		ast_log(LOG_WARNING, "Not sure about sending format %s packets\n", ast_getformatname(subclass));
-		/* fall through to... */
-	case AST_FORMAT_H261:
-	case AST_FORMAT_H263:
-	case AST_FORMAT_H263_PLUS:
-	case AST_FORMAT_H264:
-	case AST_FORMAT_G723_1:
-	case AST_FORMAT_LPC10:
-	case AST_FORMAT_SPEEX:
+	} else {
 	        /* Don't buffer outgoing frames; send them one-per-packet: */
 		if (_f->offset < hdrlen) {
 			f = ast_frdup(_f);