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allow for packetization on rtp channel drivers, need to add

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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
This commit is contained in:
Matt O'Gorman
2006-09-18 23:32:57 +00:00
parent 9177d1143b
commit 465adf2bf1
6 changed files with 318 additions and 142 deletions
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140
main/rtp.c
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@@ -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;
}
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) {
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);
}
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) {
ast_log(LOG_WARNING, "Unable to create ILBC 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;
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);