kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-11-02 03:48:02 +00:00
Code Issues Packages Projects Releases Wiki Activity

Reduce memory consumption significantly for users of the RTP engine API by storing only the payloads present and in use instead of every possible one.

Browse Source 
Review: https://reviewboard.asterisk.org/r/2052/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2012-08-07 13:07:58 +00:00
parent 5c4578f4ad
commit 15e41c7542
5 changed files with 237 additions and 90 deletions
Download Patch File Download Diff File Expand all files Collapse all files

48
channels/chan_sip.c
View File

@@ -9450,7 +9450,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
int peernoncodeccapability = 0, vpeernoncodeccapability = 0, tpeernoncodeccapability = 0;
struct ast_rtp_codecs *newaudiortp = NULL, *newvideortp = NULL, *newtextrtp = NULL;
struct ast_rtp_codecs newaudiortp = { 0, }, newvideortp = { 0, }, newtextrtp = { 0, };
struct ast_format_cap *newjointcapability = ast_format_cap_alloc_nolock(); /* Negotiated capability */
struct ast_format_cap *newpeercapability = ast_format_cap_alloc_nolock();
int newnoncodeccapability;
@@ -9487,8 +9487,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
goto process_sdp_cleanup;
}
if (!(newaudiortp = ast_calloc(1, sizeof(*newaudiortp))) || !(newvideortp = ast_calloc(1, sizeof(*newvideortp))) ||
!(newtextrtp = ast_calloc(1, sizeof(*newtextrtp)))) {
if (ast_rtp_codecs_payloads_initialize(&newaudiortp) || ast_rtp_codecs_payloads_initialize(&newvideortp) ||
ast_rtp_codecs_payloads_initialize(&newtextrtp)) {
res = -1;
goto process_sdp_cleanup;
}
@@ -9532,11 +9532,11 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
if (process_sdp_a_sendonly(value, &sendonly)) {
processed = TRUE;
}
else if (process_sdp_a_audio(value, p, newaudiortp, &last_rtpmap_codec))
else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec))
processed = TRUE;
else if (process_sdp_a_video(value, p, newvideortp, &last_rtpmap_codec))
else if (process_sdp_a_video(value, p, &newvideortp, &last_rtpmap_codec))
processed = TRUE;
else if (process_sdp_a_text(value, p, newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec))
else if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec))
processed = TRUE;
else if (process_sdp_a_image(value, p))
processed = TRUE;
@@ -9650,7 +9650,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_verbose("Found RTP audio format %d\n", codec);
}
ast_rtp_codecs_payloads_set_m_type(newaudiortp, NULL, codec);
ast_rtp_codecs_payloads_set_m_type(&newaudiortp, NULL, codec);
}
} else {
ast_log(LOG_WARNING, "Rejecting audio media offer due to invalid or unsupported syntax: %s\n", m);
@@ -9722,7 +9722,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
if (debug) {
ast_verbose("Found RTP video format %d\n", codec);
}
ast_rtp_codecs_payloads_set_m_type(newvideortp, NULL, codec);
ast_rtp_codecs_payloads_set_m_type(&newvideortp, NULL, codec);
}
} else {
ast_log(LOG_WARNING, "Rejecting video media offer due to invalid or unsupported syntax: %s\n", m);
@@ -9786,7 +9786,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
if (debug) {
ast_verbose("Found RTP text format %d\n", codec);
}
ast_rtp_codecs_payloads_set_m_type(newtextrtp, NULL, codec);
ast_rtp_codecs_payloads_set_m_type(&newtextrtp, NULL, codec);
}
} else {
ast_log(LOG_WARNING, "Rejecting text stream offer due to invalid or unsupported syntax: %s\n", m);
@@ -9904,7 +9904,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
} else if (!processed_crypto && process_crypto(p, p->rtp, &p->srtp, value)) {
processed_crypto = TRUE;
processed = TRUE;
} else if (process_sdp_a_audio(value, p, newaudiortp, &last_rtpmap_codec)) {
} else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec)) {
processed = TRUE;
}
}
@@ -9915,7 +9915,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
} else if (!processed_crypto && process_crypto(p, p->vrtp, &p->vsrtp, value)) {
processed_crypto = TRUE;
processed = TRUE;
} else if (process_sdp_a_video(value, p, newvideortp, &last_rtpmap_codec)) {
} else if (process_sdp_a_video(value, p, &newvideortp, &last_rtpmap_codec)) {
processed = TRUE;
}
}
@@ -9923,7 +9923,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
else if (text) {
if (process_sdp_a_ice(value, p, p->trtp)) {
processed = TRUE;
} if (process_sdp_a_text(value, p, newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec)) {
} if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec)) {
processed = TRUE;
} else if (!processed_crypto && process_crypto(p, p->trtp, &p->tsrtp, value)) {
processed_crypto = TRUE;
@@ -9996,9 +9996,9 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
}
/* Now gather all of the codecs that we are asked for: */
ast_rtp_codecs_payload_formats(newaudiortp, peercapability, &peernoncodeccapability);
ast_rtp_codecs_payload_formats(newvideortp, vpeercapability, &vpeernoncodeccapability);
ast_rtp_codecs_payload_formats(newtextrtp, tpeercapability, &tpeernoncodeccapability);
ast_rtp_codecs_payload_formats(&newaudiortp, peercapability, &peernoncodeccapability);
ast_rtp_codecs_payload_formats(&newvideortp, vpeercapability, &vpeernoncodeccapability);
ast_rtp_codecs_payload_formats(&newtextrtp, tpeercapability, &tpeernoncodeccapability);
ast_format_cap_append(newpeercapability, peercapability);
ast_format_cap_append(newpeercapability, vpeercapability);
@@ -10061,7 +10061,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_sockaddr_stringify(sa));
}
ast_rtp_codecs_payloads_copy(newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp);
ast_rtp_codecs_payloads_copy(&newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp);
/* Ensure RTCP is enabled since it may be inactive
if we're coming back from a T.38 session */
ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1);
@@ -10108,7 +10108,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_verbose("Peer video RTP is at port %s\n",
ast_sockaddr_stringify(vsa));
}
ast_rtp_codecs_payloads_copy(newvideortp, ast_rtp_instance_get_codecs(p->vrtp), p->vrtp);
ast_rtp_codecs_payloads_copy(&newvideortp, ast_rtp_instance_get_codecs(p->vrtp), p->vrtp);
} else {
ast_rtp_instance_stop(p->vrtp);
if (debug)
@@ -10132,7 +10132,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
} else {
p->red = 0;
}
ast_rtp_codecs_payloads_copy(newtextrtp, ast_rtp_instance_get_codecs(p->trtp), p->trtp);
ast_rtp_codecs_payloads_copy(&newtextrtp, ast_rtp_instance_get_codecs(p->trtp), p->trtp);
} else {
ast_rtp_instance_stop(p->trtp);
if (debug)
@@ -10250,15 +10250,9 @@ process_sdp_cleanup:
if (res) {
offered_media_list_destroy(p);
}
if (newtextrtp) {
ast_free(newtextrtp);
}
if (newvideortp) {
ast_free(newvideortp);
}
if (newaudiortp) {
ast_free(newaudiortp);
}
ast_rtp_codecs_payloads_destroy(&newtextrtp);
ast_rtp_codecs_payloads_destroy(&newvideortp);
ast_rtp_codecs_payloads_destroy(&newaudiortp);
ast_format_cap_destroy(peercapability);
ast_format_cap_destroy(vpeercapability);
ast_format_cap_destroy(tpeercapability);