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authorMatthew Jordan <mjordan@digium.com>2014-07-20 22:06:33 +0000
committerMatthew Jordan <mjordan@digium.com>2014-07-20 22:06:33 +0000
commita2c912e9972c91973ea66902d217746133f96026 (patch)
tree50e01d14ba62950e3f78766d5ba435ba51ca327d /res/res_rtp_asterisk.c
parentb299052e203807c9a2111eb2cd919246d7589cb3 (diff)
media formats: re-architect handling of media for performance improvements
In the old times media formats were represented using a bit field. This was fast but had a few limitations. 1. Asterisk was limited in how many formats it could handle. 2. Formats, being a bit field, could not include any attribute information. A format was strictly its type, e.g., "this is ulaw". This was changed in Asterisk 10 (see https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for notes on that work) which led to the creation of the ast_format structure. This structure allowed Asterisk to handle attributes and bundle information with a format. Additionally, ast_format_cap was created to act as a container for multiple formats that, together, formed the capability of some entity. Another mechanism was added to allow logic to be registered which performed format attribute negotiation. Everywhere throughout the codebase Asterisk was changed to use this strategy. Unfortunately, in software, there is no free lunch. These new capabilities came at a cost. Performance analysis and profiling showed that we spend an inordinate amount of time comparing, copying, and generally manipulating formats and their related structures. Basic prototyping has shown that a reasonably large performance improvement could be made in this area. This patch is the result of that project, which overhauled the media format architecture and its usage in Asterisk to improve performance. Generally, the new philosophy for handling formats is as follows: * The ast_format structure is reference counted. This removed a large amount of the memory allocations and copying that was done in prior versions. * In order to prevent race conditions while keeping things performant, the ast_format structure is immutable by convention and lock-free. Violate this tenet at your peril! * Because formats are reference counted, codecs are also reference counted. The Asterisk core generally provides built-in codecs and caches the ast_format structures created to represent them. Generally, to prevent inordinate amounts of module reference bumping, codecs and formats can be added at run-time but cannot be removed. * All compatibility with the bit field representation of codecs/formats has been moved to a compatibility API. The primary user of this representation is chan_iax2, which must continue to maintain its bit-field usage of formats for interoperability concerns. * When a format is negotiated with attributes, or when a format cannot be represented by one of the cached formats, a new format object is created or cloned from an existing format. That format may have the same codec underlying it, but is a different format than a version of the format with different attributes or without attributes. * While formats are reference counted objects, the reference count maintained on the format should be manipulated with care. Formats are generally cached and will persist for the lifetime of Asterisk and do not explicitly need to have their lifetime modified. An exception to this is when the user of a format does not know where the format came from *and* the user may outlive the provider of the format. This occurs, for example, when a format is read from a channel: the channel may have a format with attributes (hence, non-cached) and the user of the format may last longer than the channel (if the reference to the channel is released prior to the format's reference). For more information on this work, see the API design notes: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite Finally, this work was the culmination of a large number of developer's efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the work in the Asterisk core, chan_sip, and was an invaluable resource in peer reviews throughout this project. There were a substantial number of patches contributed during this work; the following issues/patch names simply reflect some of the work (and will cause the release scripts to give attribution to the individuals who work on them). Reviews: https://reviewboard.asterisk.org/r/3814 https://reviewboard.asterisk.org/r/3808 https://reviewboard.asterisk.org/r/3805 https://reviewboard.asterisk.org/r/3803 https://reviewboard.asterisk.org/r/3801 https://reviewboard.asterisk.org/r/3798 https://reviewboard.asterisk.org/r/3800 https://reviewboard.asterisk.org/r/3794 https://reviewboard.asterisk.org/r/3793 https://reviewboard.asterisk.org/r/3792 https://reviewboard.asterisk.org/r/3791 https://reviewboard.asterisk.org/r/3790 https://reviewboard.asterisk.org/r/3789 https://reviewboard.asterisk.org/r/3788 https://reviewboard.asterisk.org/r/3787 https://reviewboard.asterisk.org/r/3786 https://reviewboard.asterisk.org/r/3784 https://reviewboard.asterisk.org/r/3783 https://reviewboard.asterisk.org/r/3778 https://reviewboard.asterisk.org/r/3774 https://reviewboard.asterisk.org/r/3775 https://reviewboard.asterisk.org/r/3772 https://reviewboard.asterisk.org/r/3761 https://reviewboard.asterisk.org/r/3754 https://reviewboard.asterisk.org/r/3753 https://reviewboard.asterisk.org/r/3751 https://reviewboard.asterisk.org/r/3750 https://reviewboard.asterisk.org/r/3748 https://reviewboard.asterisk.org/r/3747 https://reviewboard.asterisk.org/r/3746 https://reviewboard.asterisk.org/r/3742 https://reviewboard.asterisk.org/r/3740 https://reviewboard.asterisk.org/r/3739 https://reviewboard.asterisk.org/r/3738 https://reviewboard.asterisk.org/r/3737 https://reviewboard.asterisk.org/r/3736 https://reviewboard.asterisk.org/r/3734 https://reviewboard.asterisk.org/r/3722 https://reviewboard.asterisk.org/r/3713 https://reviewboard.asterisk.org/r/3703 https://reviewboard.asterisk.org/r/3689 https://reviewboard.asterisk.org/r/3687 https://reviewboard.asterisk.org/r/3674 https://reviewboard.asterisk.org/r/3671 https://reviewboard.asterisk.org/r/3667 https://reviewboard.asterisk.org/r/3665 https://reviewboard.asterisk.org/r/3625 https://reviewboard.asterisk.org/r/3602 https://reviewboard.asterisk.org/r/3519 https://reviewboard.asterisk.org/r/3518 https://reviewboard.asterisk.org/r/3516 https://reviewboard.asterisk.org/r/3515 https://reviewboard.asterisk.org/r/3512 https://reviewboard.asterisk.org/r/3506 https://reviewboard.asterisk.org/r/3413 https://reviewboard.asterisk.org/r/3410 https://reviewboard.asterisk.org/r/3387 https://reviewboard.asterisk.org/r/3388 https://reviewboard.asterisk.org/r/3389 https://reviewboard.asterisk.org/r/3390 https://reviewboard.asterisk.org/r/3321 https://reviewboard.asterisk.org/r/3320 https://reviewboard.asterisk.org/r/3319 https://reviewboard.asterisk.org/r/3318 https://reviewboard.asterisk.org/r/3266 https://reviewboard.asterisk.org/r/3265 https://reviewboard.asterisk.org/r/3234 https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close Reported by: mjordan media_formats_translation_core.diff uploaded by kharwell (License 6464) rb3506.diff uploaded by mjordan (License 6283) media_format_app_file.diff uploaded by kharwell (License 6464) misc-2.diff uploaded by file (License 5000) chan_mild-3.diff uploaded by file (License 5000) chan_obscure.diff uploaded by file (License 5000) jingle.diff uploaded by file (License 5000) funcs.diff uploaded by file (License 5000) formats.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) bridges.diff uploaded by file (License 5000) mf-codecs-2.diff uploaded by file (License 5000) mf-app_fax.diff uploaded by file (License 5000) mf-apps-3.diff uploaded by file (License 5000) media-formats-3.diff uploaded by file (License 5000) ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License 5909) rb3689.patch uploaded by mjordan (License 6283) ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283) mf-attributes-3.diff uploaded by file (License 5000) ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by coreyfarrell (License 5909) rb3800.patch uploaded by jrose (License 6182) chan_sip.diff uploaded by mjordan (License 6283) rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959 #close Tested by: sgriepentrog, mjordan, coreyfarrell sip_cleanup.diff uploaded by opticron (License 6273) chan_sip_caps.diff uploaded by mjordan (License 6283) rb3751.patch uploaded by coreyfarrell (License 5909) chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960 #close Tested by: opticron direct_media.diff uploaded by opticron (License 6273) pjsip-direct-media.diff uploaded by file (License 5000) format_cap_remove.diff uploaded by opticron (License 6273) media_format_fixes.diff uploaded by opticron (License 6273) chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966 #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti (License 5621) chan_dahdi.diff uploaded by file (License 5000) ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by rmudgett (License 5621) moh_cleanup.diff uploaded by opticron (License 6273) bridge_leak.diff uploaded by opticron (License 6273) translate.diff uploaded by file (License 5000) rb3795.patch uploaded by rmudgett (License 5621) tls_fix.diff uploaded by mjordan (License 6283) fax-mf-fix-2.diff uploaded by file (License 5000) rtp_transfer_stuff uploaded by mjordan (License 6283) rb3787.patch uploaded by rmudgett (License 5621) media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) format_cache_case_fix.diff uploaded by opticron (License 6273) rb3774.patch uploaded by rmudgett (License 5621) rb3775.patch uploaded by rmudgett (License 5621) rtp_engine_fix.diff uploaded by opticron (License 6273) rtp_crash_fix.diff uploaded by opticron (License 6273) rb3753.patch uploaded by mjordan (License 6283) rb3750.patch uploaded by mjordan (License 6283) rb3748.patch uploaded by rmudgett (License 5621) media_format_fixes.diff uploaded by opticron (License 6273) rb3740.patch uploaded by mjordan (License 6283) rb3739.patch uploaded by mjordan (License 6283) rb3734.patch uploaded by mjordan (License 6283) rb3689.patch uploaded by mjordan (License 6283) rb3674.patch uploaded by coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell (License 5909) rb3667.patch uploaded by coreyfarrell (License 5909) rb3665.patch uploaded by mjordan (License 6283) rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch uploaded by coreyfarrell (License 5909) format_compatibility-2.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res/res_rtp_asterisk.c')
-rw-r--r--res/res_rtp_asterisk.c187
1 files changed, 108 insertions, 79 deletions
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 7251e8f80..bd930295b 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -56,6 +56,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/stun.h"
#include "asterisk/pbx.h"
#include "asterisk/frame.h"
+#include "asterisk/format_cache.h"
#include "asterisk/channel.h"
#include "asterisk/acl.h"
#include "asterisk/config.h"
@@ -66,6 +67,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/unaligned.h"
#include "asterisk/module.h"
#include "asterisk/rtp_engine.h"
+#include "asterisk/smoother.h"
#include "asterisk/test.h"
#define MAX_TIMESTAMP_SKEW 640
@@ -216,8 +218,8 @@ struct ast_rtp {
unsigned int cycles; /*!< Shifted count of sequence number cycles */
double rxjitter; /*!< Interarrival jitter at the moment in seconds */
double rxtransit; /*!< Relative transit time for previous packet */
- struct ast_format lasttxformat;
- struct ast_format lastrxformat;
+ 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). */
@@ -1833,7 +1835,11 @@ static int rtp_sendto(struct ast_rtp_instance *instance, void *buf, size_t size,
static int rtp_get_rate(struct ast_format *format)
{
- return (format->id == AST_FORMAT_G722) ? 8000 : ast_format_rate(format);
+ /* For those wondering: due to a fluke in RFC publication, G.722 is advertised
+ * as having a sample rate of 8kHz, while implementations must know that its
+ * real rate is 16kHz. Seriously.
+ */
+ return (ast_format_cmp(format, ast_format_g722) == AST_FORMAT_CMP_EQUAL) ? 8000 : (int)ast_format_get_sample_rate(format);
}
static unsigned int ast_rtcp_calc_interval(struct ast_rtp *rtp)
@@ -2184,6 +2190,10 @@ static int ast_rtp_new(struct ast_rtp_instance *instance,
rtp->dtlstimerid = -1;
#endif
+ rtp->f.subclass.format = ao2_bump(ast_format_none);
+ rtp->lastrxformat = ao2_bump(ast_format_none);
+ rtp->lasttxformat = ao2_bump(ast_format_none);
+
return 0;
}
@@ -2265,6 +2275,10 @@ static int ast_rtp_destroy(struct ast_rtp_instance *instance)
}
#endif
+ ao2_cleanup(rtp->lasttxformat);
+ ao2_cleanup(rtp->lastrxformat);
+ ao2_cleanup(rtp->f.subclass.format);
+
/* Destroy synchronization items */
ast_mutex_destroy(&rtp->lock);
ast_cond_destroy(&rtp->cond);
@@ -2444,7 +2458,7 @@ static int ast_rtp_dtmf_end_with_duration(struct ast_rtp_instance *instance, cha
rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
- if (duration > 0 && (measured_samples = duration * rtp_get_rate(&rtp->f.subclass.format) / 1000) > rtp->send_duration) {
+ if (duration > 0 && (measured_samples = duration * rtp_get_rate(rtp->f.subclass.format) / 1000) > rtp->send_duration) {
ast_debug(2, "Adjusting final end duration from %d to %u\n", rtp->send_duration, measured_samples);
rtp->send_duration = measured_samples;
}
@@ -2620,7 +2634,7 @@ static int ast_rtcp_write_report(struct ast_rtp_instance *instance, int sr)
int fraction_lost;
struct timeval dlsr = { 0, };
char bdata[512];
- int rate = rtp_get_rate(&rtp->f.subclass.format);
+ int rate = rtp_get_rate(rtp->f.subclass.format);
int ice;
int header_offset = 0;
struct ast_sockaddr remote_address = { {0,} };
@@ -2791,9 +2805,9 @@ static int ast_rtp_raw_write(struct ast_rtp_instance *instance, struct ast_frame
int pred, mark = 0;
unsigned int ms = calc_txstamp(rtp, &frame->delivery);
struct ast_sockaddr remote_address = { {0,} };
- int rate = rtp_get_rate(&frame->subclass.format) / 1000;
+ int rate = rtp_get_rate(frame->subclass.format) / 1000;
- if (frame->subclass.format.id == AST_FORMAT_G722) {
+ if (ast_format_cmp(frame->subclass.format, ast_format_g722) == AST_FORMAT_CMP_EQUAL) {
frame->samples /= 2;
}
@@ -2817,7 +2831,7 @@ static int ast_rtp_raw_write(struct ast_rtp_instance *instance, struct ast_frame
}
}
} else if (frame->frametype == AST_FRAME_VIDEO) {
- mark = ast_format_get_video_mark(&frame->subclass.format);
+ mark = frame->subclass.frame_ending;
pred = rtp->lastovidtimestamp + frame->samples;
/* Re-calculate last TS */
rtp->lastts = rtp->lastts + ms * 90;
@@ -2959,7 +2973,7 @@ static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *fr
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
struct ast_sockaddr remote_address = { {0,} };
- struct ast_format subclass;
+ struct ast_format *format;
int codec;
ast_rtp_instance_get_remote_address(instance, &remote_address);
@@ -3029,17 +3043,28 @@ static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *fr
}
/* Grab the subclass and look up the payload we are going to use */
- ast_format_copy(&subclass, &frame->subclass.format);
- if ((codec = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance), 1, &subclass, 0)) < 0) {
- ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(&frame->subclass.format));
+ codec = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance),
+ 1,
+ frame->subclass.format,
+ 0);
+ if (codec < 0) {
+ ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n",
+ ast_format_get_name(frame->subclass.format));
return -1;
}
- /* Oh dear, if the format changed we will have to set up a new smoother */
- if (ast_format_cmp(&rtp->lasttxformat, &subclass) == AST_FORMAT_CMP_NOT_EQUAL) {
- ast_debug(1, "Ooh, format changed from %s to %s\n", ast_getformatname(&rtp->lasttxformat), ast_getformatname(&subclass));
- rtp->lasttxformat = subclass;
- ast_format_copy(&rtp->lasttxformat, &subclass);
+ /* Note that we do not increase the ref count here as this pointer
+ * will not be held by any thing explicitly. The format variable is
+ * merely a convenience reference to frame->subclass.format */
+ format = frame->subclass.format;
+ if (ast_format_cmp(rtp->lasttxformat, format) == AST_FORMAT_CMP_NOT_EQUAL) {
+ /* Oh dear, if the format changed we will have to set up a new smoother */
+ if (option_debug > 0) {
+ ast_debug(1, "Ooh, format changed from %s to %s\n",
+ ast_format_get_name(rtp->lasttxformat),
+ ast_format_get_name(frame->subclass.format));
+ }
+ ao2_replace(rtp->lasttxformat, format);
if (rtp->smoother) {
ast_smoother_free(rtp->smoother);
rtp->smoother = NULL;
@@ -3047,34 +3072,15 @@ static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *fr
}
/* If no smoother is present see if we have to set one up */
- if (!rtp->smoother) {
- struct ast_format_list fmt = ast_codec_pref_getsize(&ast_rtp_instance_get_codecs(instance)->pref, &subclass);
-
- switch (subclass.id) {
- case AST_FORMAT_SPEEX:
- case AST_FORMAT_SPEEX16:
- case AST_FORMAT_SPEEX32:
- case AST_FORMAT_SILK:
- case AST_FORMAT_CELT:
- case AST_FORMAT_G723_1:
- case AST_FORMAT_SIREN7:
- case AST_FORMAT_SIREN14:
- case AST_FORMAT_G719:
- /* Opus */
- case AST_FORMAT_OPUS:
- /* these are all frame-based codecs and cannot be safely run through
- a smoother */
- break;
- default:
- if (fmt.inc_ms) {
- if (!(rtp->smoother = ast_smoother_new((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms))) {
- ast_log(LOG_WARNING, "Unable to create smoother: format %s ms: %d len: %d\n", ast_getformatname(&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);
- }
- ast_debug(1, "Created smoother: format: %s ms: %d len: %d\n", ast_getformatname(&subclass), fmt.cur_ms, ((fmt.cur_ms * fmt.fr_len) / fmt.inc_ms));
+ if (!rtp->smoother && ast_format_can_be_smoothed(format)) {
+ unsigned int framing_ms = ast_rtp_codecs_get_framing(ast_rtp_instance_get_codecs(instance));
+
+ if (framing_ms) {
+ rtp->smoother = ast_smoother_new((framing_ms * ast_format_get_minimum_bytes(format)) / ast_format_get_minimum_ms(format));
+ if (!rtp->smoother) {
+ ast_log(LOG_WARNING, "Unable to create smoother: format %s ms: %u len: %u\n",
+ ast_format_get_name(format), framing_ms, ast_format_get_minimum_bytes(format));
+ return -1;
}
}
}
@@ -3122,7 +3128,7 @@ static void calc_rxstamp(struct timeval *tv, struct ast_rtp *rtp, unsigned int t
double d;
double dtv;
double prog;
- int rate = rtp_get_rate(&rtp->f.subclass.format);
+ int rate = rtp_get_rate(rtp->f.subclass.format);
double normdev_rxjitter_current;
if ((!rtp->rxcore.tv_sec && !rtp->rxcore.tv_usec) || mark) {
@@ -3277,7 +3283,7 @@ static void process_dtmf_rfc2833(struct ast_rtp_instance *instance, unsigned cha
rtp->dtmf_duration = new_duration;
rtp->resp = resp;
f = ast_frdup(create_dtmf_frame(instance, AST_FRAME_DTMF_END, 0));
- f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(&f->subclass.format)), ast_tv(0, 0));
+ f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass.format)), ast_tv(0, 0));
rtp->resp = 0;
rtp->dtmf_duration = rtp->dtmf_timeout = 0;
AST_LIST_INSERT_TAIL(frames, f, frame_list);
@@ -3308,7 +3314,7 @@ static void process_dtmf_rfc2833(struct ast_rtp_instance *instance, unsigned cha
if (rtp->resp && rtp->resp != resp) {
/* Another digit already began. End it */
f = ast_frdup(create_dtmf_frame(instance, AST_FRAME_DTMF_END, 0));
- f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(&f->subclass.format)), ast_tv(0, 0));
+ f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass.format)), ast_tv(0, 0));
rtp->resp = 0;
rtp->dtmf_duration = rtp->dtmf_timeout = 0;
AST_LIST_INSERT_TAIL(frames, f, frame_list);
@@ -3405,10 +3411,10 @@ static struct ast_frame *process_dtmf_cisco(struct ast_rtp_instance *instance, u
}
} else if ((rtp->resp == resp) && !power) {
f = create_dtmf_frame(instance, AST_FRAME_DTMF_END, ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_DTMF_COMPENSATE));
- f->samples = rtp->dtmfsamples * (rtp->lastrxformat.id ? (rtp_get_rate(&rtp->lastrxformat) / 1000) : 8);
+ f->samples = rtp->dtmfsamples * (rtp_get_rate(rtp->lastrxformat) / 1000);
rtp->resp = 0;
} else if (rtp->resp == resp) {
- rtp->dtmfsamples += 20 * (rtp->lastrxformat.id ? (rtp_get_rate(&rtp->lastrxformat) / 1000) : 8);
+ rtp->dtmfsamples += 20 * (rtp_get_rate(rtp->lastrxformat) / 1000);
}
rtp->dtmf_timeout = 0;
@@ -3424,7 +3430,8 @@ static struct ast_frame *process_cn_rfc3389(struct ast_rtp_instance *instance, u
totally help us out becuase we don't have an engine to keep it going and we are not
guaranteed to have it every 20ms or anything */
if (rtpdebug) {
- ast_debug(0, "- RTP 3389 Comfort noise event: Level %d (len = %d)\n", (int) rtp->lastrxformat.id, len);
+ ast_debug(0, "- RTP 3389 Comfort noise event: Format %s (len = %d)\n",
+ ast_format_get_name(rtp->lastrxformat), len);
}
if (ast_test_flag(rtp, FLAG_3389_WARNING)) {
@@ -3796,7 +3803,7 @@ static int bridge_p2p_rtp_write(struct ast_rtp_instance *instance, unsigned int
struct ast_rtp_instance *instance1 = ast_rtp_instance_get_bridged(instance);
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance), *bridged = ast_rtp_instance_get_data(instance1);
int res = 0, payload = 0, bridged_payload = 0, mark;
- struct ast_rtp_payload_type payload_type;
+ RAII_VAR(struct ast_rtp_payload_type *, payload_type, NULL, ao2_cleanup);
int reconstruct = ntohl(rtpheader[0]);
struct ast_sockaddr remote_address = { {0,} };
int ice;
@@ -3806,10 +3813,13 @@ static int bridge_p2p_rtp_write(struct ast_rtp_instance *instance, unsigned int
mark = (((reconstruct & 0x800000) >> 23) != 0);
/* Check what the payload value should be */
- payload_type = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(instance), payload);
+ payload_type = ast_rtp_codecs_get_payload(ast_rtp_instance_get_codecs(instance), payload);
+ if (!payload_type) {
+ return -1;
+ }
/* Otherwise adjust bridged payload to match */
- bridged_payload = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance1), payload_type.asterisk_format, &payload_type.format, payload_type.rtp_code);
+ bridged_payload = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance1), payload_type->asterisk_format, payload_type->format, payload_type->rtp_code);
/* If no codec could be matched between instance and instance1, then somehow things were made incompatible while we were still bridged. Bail. */
if (bridged_payload < 0) {
@@ -3880,7 +3890,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
struct ast_sockaddr addr;
int res, hdrlen = 12, version, payloadtype, padding, mark, ext, cc, prev_seqno;
unsigned int *rtpheader = (unsigned int*)(rtp->rawdata + AST_FRIENDLY_OFFSET), seqno, ssrc, timestamp;
- struct ast_rtp_payload_type payload;
+ RAII_VAR(struct ast_rtp_payload_type *, payload, NULL, ao2_cleanup);
struct ast_sockaddr remote_address = { {0,} };
struct frame_list frames;
@@ -4120,20 +4130,20 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
payloadtype, seqno, timestamp,res - hdrlen);
}
- payload = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(instance), payloadtype);
+ payload = ast_rtp_codecs_get_payload(ast_rtp_instance_get_codecs(instance), payloadtype);
/* If the payload is not actually an Asterisk one but a special one pass it off to the respective handler */
- if (!payload.asterisk_format) {
+ if (!payload->asterisk_format) {
struct ast_frame *f = NULL;
- if (payload.rtp_code == AST_RTP_DTMF) {
+ if (payload->rtp_code == AST_RTP_DTMF) {
/* process_dtmf_rfc2833 may need to return multiple frames. We do this
* by passing the pointer to the frame list to it so that the method
* can append frames to the list as needed.
*/
process_dtmf_rfc2833(instance, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp, &addr, payloadtype, mark, &frames);
- } else if (payload.rtp_code == AST_RTP_CISCO_DTMF) {
+ } else if (payload->rtp_code == AST_RTP_CISCO_DTMF) {
f = process_dtmf_cisco(instance, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp, &addr, payloadtype, mark);
- } else if (payload.rtp_code == AST_RTP_CN) {
+ } else if (payload->rtp_code == AST_RTP_CN) {
f = process_cn_rfc3389(instance, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp, &addr, payloadtype, mark);
} else {
ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n",
@@ -4153,10 +4163,25 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
return &ast_null_frame;
}
- ast_format_copy(&rtp->lastrxformat, &payload.format);
- ast_format_copy(&rtp->f.subclass.format, &payload.format);
- rtp->f.frametype = (AST_FORMAT_GET_TYPE(rtp->f.subclass.format.id) == AST_FORMAT_TYPE_AUDIO) ? AST_FRAME_VOICE : (AST_FORMAT_GET_TYPE(rtp->f.subclass.format.id) == AST_FORMAT_TYPE_VIDEO) ? AST_FRAME_VIDEO : AST_FRAME_TEXT;
-
+ ao2_replace(rtp->lastrxformat, payload->format);
+ ao2_replace(rtp->f.subclass.format, payload->format);
+ switch (ast_format_get_type(rtp->f.subclass.format)) {
+ case AST_MEDIA_TYPE_AUDIO:
+ rtp->f.frametype = AST_FRAME_VOICE;
+ break;
+ case AST_MEDIA_TYPE_VIDEO:
+ rtp->f.frametype = AST_FRAME_VIDEO;
+ break;
+ case AST_MEDIA_TYPE_TEXT:
+ rtp->f.frametype = AST_FRAME_TEXT;
+ break;
+ case AST_MEDIA_TYPE_IMAGE:
+ /* Fall through */
+ default:
+ ast_log(LOG_WARNING, "Unknown or unsupported media type: %s\n",
+ 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) {
@@ -4165,7 +4190,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
if (rtp->resp) {
struct ast_frame *f;
f = create_dtmf_frame(instance, AST_FRAME_DTMF_END, 0);
- f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(&f->subclass.format)), ast_tv(0, 0));
+ f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass.format)), ast_tv(0, 0));
rtp->resp = 0;
rtp->dtmf_timeout = rtp->dtmf_duration = 0;
AST_LIST_INSERT_TAIL(&frames, f, frame_list);
@@ -4182,7 +4207,9 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
rtp->f.seqno = seqno;
- if (rtp->f.subclass.format.id == AST_FORMAT_T140 && (int)seqno - (prev_seqno+1) > 0 && (int)seqno - (prev_seqno+1) < 10) {
+ if ((ast_format_cmp(rtp->f.subclass.format, ast_format_t140) == AST_FORMAT_CMP_EQUAL)
+ && ((int)seqno - (prev_seqno + 1) > 0)
+ && ((int)seqno - (prev_seqno + 1) < 10)) {
unsigned char *data = rtp->f.data.ptr;
memmove(rtp->f.data.ptr+3, rtp->f.data.ptr, rtp->f.datalen);
@@ -4192,7 +4219,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
*data = 0xBD;
}
- if (rtp->f.subclass.format.id == AST_FORMAT_T140RED) {
+ if (ast_format_cmp(rtp->f.subclass.format, ast_format_t140_red) == AST_FORMAT_CMP_EQUAL) {
unsigned char *data = rtp->f.data.ptr;
unsigned char *header_end;
int num_generations;
@@ -4201,7 +4228,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
int diff =(int)seqno - (prev_seqno+1); /* if diff = 0, no drop*/
int x;
- ast_format_set(&rtp->f.subclass.format, AST_FORMAT_T140, 0);
+ ao2_replace(rtp->f.subclass.format, ast_format_t140);
header_end = memchr(data, ((*data) & 0x7f), rtp->f.datalen);
if (header_end == NULL) {
return AST_LIST_FIRST(&frames) ? AST_LIST_FIRST(&frames) : &ast_null_frame;
@@ -4239,17 +4266,17 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
}
}
- if (AST_FORMAT_GET_TYPE(rtp->f.subclass.format.id) == AST_FORMAT_TYPE_AUDIO) {
- rtp->f.samples = ast_codec_get_samples(&rtp->f);
- if (ast_format_is_slinear(&rtp->f.subclass.format)) {
+ if (ast_format_get_type(rtp->f.subclass.format) == AST_MEDIA_TYPE_AUDIO) {
+ rtp->f.samples = ast_codec_samples_count(&rtp->f);
+ if (ast_format_cache_is_slinear(rtp->f.subclass.format)) {
ast_frame_byteswap_be(&rtp->f);
}
calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
/* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */
ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO);
- rtp->f.ts = timestamp / (rtp_get_rate(&rtp->f.subclass.format) / 1000);
- rtp->f.len = rtp->f.samples / ((ast_format_rate(&rtp->f.subclass.format) / 1000));
- } else if (AST_FORMAT_GET_TYPE(rtp->f.subclass.format.id) == AST_FORMAT_TYPE_VIDEO) {
+ rtp->f.ts = timestamp / (rtp_get_rate(rtp->f.subclass.format) / 1000);
+ rtp->f.len = rtp->f.samples / ((ast_format_get_sample_rate(rtp->f.subclass.format) / 1000));
+ } else if (ast_format_get_type(rtp->f.subclass.format) == AST_MEDIA_TYPE_VIDEO) {
/* Video -- samples is # of samples vs. 90000 */
if (!rtp->lastividtimestamp)
rtp->lastividtimestamp = timestamp;
@@ -4258,10 +4285,8 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
rtp->f.delivery.tv_sec = 0;
rtp->f.delivery.tv_usec = 0;
/* Pass the RTP marker bit as bit */
- if (mark) {
- ast_format_set_video_mark(&rtp->f.subclass.format);
- }
- } else {
+ rtp->f.subclass.frame_ending = mark;
+ } else if (ast_format_get_type(rtp->f.subclass.format) == AST_MEDIA_TYPE_TEXT) {
/* TEXT -- samples is # of samples vs. 1000 */
if (!rtp->lastitexttimestamp)
rtp->lastitexttimestamp = timestamp;
@@ -4269,6 +4294,10 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
rtp->lastitexttimestamp = timestamp;
rtp->f.delivery.tv_sec = 0;
rtp->f.delivery.tv_usec = 0;
+ } else {
+ ast_log(LOG_WARNING, "Unknown or unsupported media type: %s\n",
+ ast_codec_media_type2str(ast_format_get_type(rtp->f.subclass.format)));
+ return &ast_null_frame;;
}
AST_LIST_INSERT_TAIL(&frames, &rtp->f, frame_list);
@@ -4413,7 +4442,7 @@ static int rtp_red_init(struct ast_rtp_instance *instance, int buffer_time, int
}
rtp->red->t140.frametype = AST_FRAME_TEXT;
- ast_format_set(&rtp->red->t140.subclass.format, AST_FORMAT_T140RED, 0);
+ ao2_replace(rtp->red->t140.subclass.format, ast_format_t140_red);
rtp->red->t140.data.ptr = &rtp->red->buf_data;
rtp->red->t140.ts = 0;