mirror of
https://github.com/asterisk/asterisk.git
synced 2025-11-18 15:49:56 +00:00
Merge "res_rtp_asterisk: Remove some unused structure fields." into 16
This commit is contained in:
@@ -323,12 +323,10 @@ struct ast_rtp {
|
||||
unsigned int themssrc; /*!< Their SSRC */
|
||||
unsigned int themssrc_valid; /*!< True if their SSRC is available. */
|
||||
unsigned int lastts;
|
||||
unsigned int lastrxts;
|
||||
unsigned int lastividtimestamp;
|
||||
unsigned int lastovidtimestamp;
|
||||
unsigned int lastitexttimestamp;
|
||||
unsigned int lastotexttimestamp;
|
||||
unsigned int lasteventseqn;
|
||||
int lastrxseqno; /*!< Last received sequence number, from the network */
|
||||
int expectedrxseqno; /*!< Next expected sequence number, from the network */
|
||||
AST_VECTOR(, int) missing_seqno; /*!< A vector of sequence numbers we never received */
|
||||
@@ -345,10 +343,6 @@ struct ast_rtp {
|
||||
struct ast_format *lasttxformat;
|
||||
struct ast_format *lastrxformat;
|
||||
|
||||
int rtptimeout; /*!< RTP timeout time (negative or zero means disabled, negative value means temporarily disabled) */
|
||||
int rtpholdtimeout; /*!< RTP timeout when on hold (negative or zero means disabled, negative value means temporarily disabled). */
|
||||
int rtpkeepalive; /*!< Send RTP comfort noice packets for keepalive */
|
||||
|
||||
/* DTMF Reception Variables */
|
||||
char resp; /*!< The current digit being processed */
|
||||
unsigned int last_seqno; /*!< The last known sequence number for any DTMF packet */
|
||||
@@ -367,17 +361,11 @@ struct ast_rtp {
|
||||
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 ast_sched_context *sched;
|
||||
struct io_context *io;
|
||||
void *data;
|
||||
struct ast_rtcp *rtcp;
|
||||
struct ast_rtp *bridged; /*!< Who we are Packet bridged to */
|
||||
unsigned int asymmetric_codec; /*!< Indicate if asymmetric send/receive codecs are allowed */
|
||||
|
||||
struct ast_rtp_instance *bundled; /*!< The RTP instance we are bundled to */
|
||||
@@ -6456,7 +6444,6 @@ static struct ast_frame *ast_rtp_interpret(struct ast_rtp_instance *instance, st
|
||||
ast_codec_media_type2str(ast_format_get_type(rtp->f.subclass.format)));
|
||||
return &ast_null_frame;
|
||||
}
|
||||
rtp->rxseqno = seqno;
|
||||
|
||||
if (rtp->dtmf_timeout && rtp->dtmf_timeout < timestamp) {
|
||||
rtp->dtmf_timeout = 0;
|
||||
@@ -6472,8 +6459,6 @@ static struct ast_frame *ast_rtp_interpret(struct ast_rtp_instance *instance, st
|
||||
}
|
||||
}
|
||||
|
||||
rtp->lastrxts = timestamp;
|
||||
|
||||
rtp->f.src = "RTP";
|
||||
rtp->f.mallocd = 0;
|
||||
rtp->f.datalen = res - hdrlen;
|
||||
@@ -6863,7 +6848,6 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
|
||||
ast_sockaddr_copy(&rtp->rtcp->them, &addr);
|
||||
ast_sockaddr_set_port(&rtp->rtcp->them, ast_sockaddr_port(&addr) + 1);
|
||||
}
|
||||
rtp->rxseqno = 0;
|
||||
ast_set_flag(rtp, FLAG_NAT_ACTIVE);
|
||||
if (rtpdebug)
|
||||
ast_debug(0, "RTP NAT: Got audio from other end. Now sending to address %s\n",
|
||||
@@ -7385,8 +7369,6 @@ static void ast_rtp_remote_address_set(struct ast_rtp_instance *instance, struct
|
||||
ast_rtp_instance_set_remote_address(mapping->instance, addr);
|
||||
}
|
||||
|
||||
rtp->rxseqno = 0;
|
||||
|
||||
if (strictrtp && rtp->strict_rtp_state != STRICT_RTP_OPEN
|
||||
&& !ast_sockaddr_isnull(addr) && ast_sockaddr_cmp(addr, &rtp->strict_rtp_address)) {
|
||||
/* We only need to learn a new strict source address if we've been told the source is
|
||||
|
||||
Reference in New Issue
Block a user