move 'struct ast_rtp' back to rtp.c where it belongs

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@33133 f38db490-d61c-443f-a65b-d21fe96a405b

3 files changed, 90 insertions, 78 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 06e728d6a..d84768cba 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -4112,7 +4112,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 	int sendonly = 0;
 	int numberofports;
 	struct ast_channel *bridgepeer = NULL;
-	struct ast_rtp newaudiortp, newvideortp;	/* Buffers for codec handling */
+	struct ast_rtp *newaudiortp, *newvideortp;	/* Buffers for codec handling */
 	int newjointcapability;				/* Negotiated capability */
 	int newpeercapability;
 	int newnoncodeccapability;
@@ -4125,10 +4125,13 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 	}
 
 	/* Initialize the temporary RTP structures we use to evaluate the offer from the peer */
-	memset(&newaudiortp, 0, sizeof(newaudiortp));
-	memset(&newvideortp, 0, sizeof(newvideortp));
-	ast_rtp_pt_default(&newaudiortp);
-	ast_rtp_pt_default(&newvideortp);
+	newaudiortp = alloca(ast_rtp_alloc_size());
+	memset(newaudiortp, 0, ast_rtp_alloc_size());
+	ast_rtp_pt_default(newaudiortp);
+
+	newvideortp = alloca(ast_rtp_alloc_size());
+	memset(newvideortp, 0, ast_rtp_alloc_size());
+	ast_rtp_pt_default(newvideortp);
 
 	/* Update our last rtprx when we receive an SDP, too */
 	p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
@@ -4168,7 +4171,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 		numberofmediastreams++;
 
 		if (p->vrtp)
-			ast_rtp_pt_clear(&newvideortp);  /* Must be cleared in case no m=video line exists */
+			ast_rtp_pt_clear(newvideortp);  /* Must be cleared in case no m=video line exists */
 		numberofports = 1;
 		if ((sscanf(m, "audio %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) ||
 		    (sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1)) {
@@ -4176,7 +4179,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 			/* Found audio stream in this media definition */
 			portno = x;
 			/* Scan through the RTP payload types specified in a "m=" line: */
-			ast_rtp_pt_clear(&newaudiortp);
+			ast_rtp_pt_clear(newaudiortp);
 			for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
 				if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
 					ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
@@ -4184,7 +4187,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 				}
 				if (debug)
 					ast_verbose("Found RTP audio format %d\n", codec);
-				ast_rtp_set_m_type(&newaudiortp, codec);
+				ast_rtp_set_m_type(newaudiortp, codec);
 			}
 		} else if ((sscanf(m, "video %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) ||
 		    (sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1)) {
@@ -4199,7 +4202,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 				}
 				if (debug)
 					ast_verbose("Found RTP video format %d\n", codec);
-				ast_rtp_set_m_type(&newvideortp, codec);
+				ast_rtp_set_m_type(newvideortp, codec);
 			}
 		} else 
 			ast_log(LOG_WARNING, "Unsupported SDP media type in offer: %s\n", m);
@@ -4305,14 +4308,14 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 			ast_verbose("Found description format %s for ID %d\n", mimeSubtype, codec);
 
 		/* Note: should really look at the 'freq' and '#chans' params too */
-		ast_rtp_set_rtpmap_type(&newaudiortp, codec, "audio", mimeSubtype);
+		ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype);
 		if (p->vrtp)
-			ast_rtp_set_rtpmap_type(&newvideortp, codec, "video", mimeSubtype);
+			ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype);
 	}
 
 	/* Now gather all of the codecs that we are asked for: */
-	ast_rtp_get_current_formats(&newaudiortp, &peercapability, &peernoncodeccapability);
-	ast_rtp_get_current_formats(&newvideortp, &vpeercapability, &vpeernoncodeccapability);
+	ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability);
+	ast_rtp_get_current_formats(newvideortp, &vpeercapability, &vpeernoncodeccapability);
 
 	newjointcapability = p->capability & (peercapability | vpeercapability);
 	newpeercapability = (peercapability | vpeercapability);
@@ -4346,15 +4349,9 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 	p->peercapability = newpeercapability;		/* The other sides capability in latest offer */
 	p->noncodeccapability = newnoncodeccapability;	/* DTMF capabilities */
 
-	{
-		int i;
-		/* Copy payload types from source to destination */
-		for (i=0; i < MAX_RTP_PT; ++i) {
-			p->rtp->current_RTP_PT[i]= newaudiortp.current_RTP_PT[i];
-			if (p->vrtp) 
-				p->vrtp->current_RTP_PT[i]= newvideortp.current_RTP_PT[i];
-		}
-	}
+	ast_rtp_pt_copy(p->rtp, newaudiortp);
+	if (p->vrtp)
+		ast_rtp_pt_copy(p->vrtp, newvideortp);
 
 	if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) {
 		ast_clear_flag(&p->flags[0], SIP_DTMF);
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index 4410d7fb6..b7be53a1c 100644
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -80,59 +80,11 @@ struct rtpPayloadType {
 	int code;
 };
 
-/*! \brief RTP session description */
-struct ast_rtp {
-	int s;
-	char resp;
-	struct ast_frame f;
-	unsigned char rawdata[8192 + AST_FRIENDLY_OFFSET];
-	unsigned int ssrc;		/*!< Synchronization source, RFC 3550, page 10. */
-	unsigned int themssrc;		/*!< Their SSRC */
-	unsigned int rxssrc;
-	unsigned int lastts;
-	unsigned int lastdigitts;
-	unsigned int lastrxts;
-	unsigned int lastividtimestamp;
-	unsigned int lastovidtimestamp;
-	unsigned int lasteventseqn;
-	int lastrxseqno;                /*!< Last received sequence number */
-	unsigned short seedrxseqno;     /*!< What sequence number did they start with?*/
-	unsigned int seedrxts;          /*!< What RTP timestamp did they start with? */
-	unsigned int rxcount;           /*!< How many packets have we received? */
-	unsigned int rxoctetcount;      /*!< How many octets have we received? should be rxcount *160*/
-	unsigned int txcount;           /*!< How many packets have we sent? */
-	unsigned int txoctetcount;      /*!< How many octets have we sent? (txcount*160)*/
-	unsigned int cycles;            /*!< Shifted count of sequence number cycles */
-	double rxjitter;                /*!< Interarrival jitter at the moment */
-	double rxtransit;               /*!< Relative transit time for previous packet */
-	unsigned int lasteventendseqn;
-	int lasttxformat;
-	int lastrxformat;
-	int dtmfcount;
-	unsigned int dtmfduration;
-	int nat;
-	unsigned int flags;
-	struct sockaddr_in us;		/*!< Socket representation of the local endpoint. */
-	struct sockaddr_in them;	/*!< Socket representation of the remote endpoint. */
-	struct timeval rxcore;
-	struct timeval txcore;
-	double drxcore;                 /*!< The double representation of the first received packet */
-	struct timeval lastrx;          /*!< timeval when we last received a packet */
-	struct timeval dtmfmute;
-	struct ast_smoother *smoother;
-	int *ioid;
-	unsigned short seqno;		/*!< Sequence number, RFC 3550, page 13. */
-	unsigned short rxseqno;
-	struct sched_context *sched;
-	struct io_context *io;
-	void *data;
-	ast_rtp_callback callback;
-	struct rtpPayloadType current_RTP_PT[MAX_RTP_PT];
-	int rtp_lookup_code_cache_isAstFormat; /*!< a cache for the result of rtp_lookup_code(): */
-	int rtp_lookup_code_cache_code;
-	int rtp_lookup_code_cache_result;
-	struct ast_rtcp *rtcp;
-};
+/*!
+ * \brief Get the amount of space required to hold an RTP session
+ * \return number of bytes required
+ */
+size_t ast_rtp_alloc_size(void);
 
 /*!
  * \brief Initializate a RTP session.
@@ -196,6 +148,10 @@ int ast_rtp_settos(struct ast_rtp *rtp, int tos);
 void ast_rtp_pt_clear(struct ast_rtp* rtp);
 /*! \brief Set payload types to defaults */
 void ast_rtp_pt_default(struct ast_rtp* rtp);
+
+/*! \brief Copy payload types between RTP structures */
+void ast_rtp_pt_copy(struct ast_rtp *dest, const struct ast_rtp *src);
+
 void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt);
 void ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
 			 char* mimeType, char* mimeSubtype);
diff --git a/rtp.c b/rtp.c
index c94fd15a2..829137492 100644
--- a/rtp.c
+++ b/rtp.c
@@ -90,6 +90,60 @@ static struct sockaddr_in rtcpdebugaddr;	/*!< Debug RTCP packets to/from this ho
 static int nochecksums = 0;
 #endif
 
+/*! \brief RTP session description */
+struct ast_rtp {
+	int s;
+	char resp;
+	struct ast_frame f;
+	unsigned char rawdata[8192 + AST_FRIENDLY_OFFSET];
+	unsigned int ssrc;		/*!< Synchronization source, RFC 3550, page 10. */
+	unsigned int themssrc;		/*!< Their SSRC */
+	unsigned int rxssrc;
+	unsigned int lastts;
+	unsigned int lastdigitts;
+	unsigned int lastrxts;
+	unsigned int lastividtimestamp;
+	unsigned int lastovidtimestamp;
+	unsigned int lasteventseqn;
+	int lastrxseqno;                /*!< Last received sequence number */
+	unsigned short seedrxseqno;     /*!< What sequence number did they start with?*/
+	unsigned int seedrxts;          /*!< What RTP timestamp did they start with? */
+	unsigned int rxcount;           /*!< How many packets have we received? */
+	unsigned int rxoctetcount;      /*!< How many octets have we received? should be rxcount *160*/
+	unsigned int txcount;           /*!< How many packets have we sent? */
+	unsigned int txoctetcount;      /*!< How many octets have we sent? (txcount*160)*/
+	unsigned int cycles;            /*!< Shifted count of sequence number cycles */
+	double rxjitter;                /*!< Interarrival jitter at the moment */
+	double rxtransit;               /*!< Relative transit time for previous packet */
+	unsigned int lasteventendseqn;
+	int lasttxformat;
+	int lastrxformat;
+	int dtmfcount;
+	unsigned int dtmfduration;
+	int nat;
+	unsigned int flags;
+	struct sockaddr_in us;		/*!< Socket representation of the local endpoint. */
+	struct sockaddr_in them;	/*!< Socket representation of the remote endpoint. */
+	struct timeval rxcore;
+	struct timeval txcore;
+	double drxcore;                 /*!< The double representation of the first received packet */
+	struct timeval lastrx;          /*!< timeval when we last received a packet */
+	struct timeval dtmfmute;
+	struct ast_smoother *smoother;
+	int *ioid;
+	unsigned short seqno;		/*!< Sequence number, RFC 3550, page 13. */
+	unsigned short rxseqno;
+	struct sched_context *sched;
+	struct io_context *io;
+	void *data;
+	ast_rtp_callback callback;
+	struct rtpPayloadType current_RTP_PT[MAX_RTP_PT];
+	int rtp_lookup_code_cache_isAstFormat; /*!< a cache for the result of rtp_lookup_code(): */
+	int rtp_lookup_code_cache_code;
+	int rtp_lookup_code_cache_result;
+	struct ast_rtcp *rtcp;
+};
+
 /* Forward declarations */
 static int ast_rtcp_write(void *data);
 static void timeval2ntp(struct timeval tv, unsigned int *msw, unsigned int *lsw);
@@ -303,6 +357,11 @@ static void stun_req_id(struct stun_header *req)
 		req->id.id[x] = ast_random();
 }
 
+size_t ast_rtp_alloc_size(void)
+{
+	return sizeof(struct ast_rtp);
+}
+
 void ast_rtp_stun_request(struct ast_rtp *rtp, struct sockaddr_in *suggestion, const char *username)
 {
 	struct stun_header *req;
@@ -1185,10 +1244,10 @@ void ast_rtp_pt_default(struct ast_rtp* rtp)
 	rtp->rtp_lookup_code_cache_result = 0;
 }
 
-static void ast_rtp_pt_copy(struct ast_rtp *dest, struct ast_rtp *src)
+void ast_rtp_pt_copy(struct ast_rtp *dest, const struct ast_rtp *src)
 {
-	int i;
-	/* Copy payload types from source to destination */
+	unsigned int i;
+
 	for (i=0; i < MAX_RTP_PT; ++i) {
 		dest->current_RTP_PT[i].isAstFormat = 
 			src->current_RTP_PT[i].isAstFormat;