diff options
| author | markster <markster@f38db490-d61c-443f-a65b-d21fe96a405b> | 2003-04-15 04:36:52 +0000 |
|---|---|---|
| committer | markster <markster@f38db490-d61c-443f-a65b-d21fe96a405b> | 2003-04-15 04:36:52 +0000 |
| commit | 9fa1f6563a431a73d7eb0c8a284e4cf419cbc2fa (patch) | |
| tree | 468bcd1944564d1b84b5a9d46b2eb437fe3c73a3 /rtp.c | |
| parent | 0cc1bf692bcfc7f3f9ffef9d62c6621cd9ab9f4a (diff) | |
Add iLBC codec
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@852 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'rtp.c')
| -rwxr-xr-x | rtp.c | 21 |
1 files changed, 21 insertions, 0 deletions
@@ -30,6 +30,7 @@ #include <asterisk/logger.h> #include <asterisk/options.h> #include <asterisk/channel.h> +#include <asterisk/acl.h> #include <asterisk/channel_pvt.h> #define TYPE_SILENCE 0x2 @@ -332,6 +333,9 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp) case AST_FORMAT_GSM: rtp->f.samples = 160 * (rtp->f.datalen / 33); break; + case AST_FORMAT_ILBC: + rtp->f.samples = 240 * (rtp->f.datalen / 52); + break; case AST_FORMAT_ADPCM: rtp->f.samples = rtp->f.datalen * 2; break; @@ -370,6 +374,7 @@ static struct { {{1, AST_FORMAT_LPC10}, "audio", "LPC"}, {{1, AST_FORMAT_G729A}, "audio", "G729"}, {{1, AST_FORMAT_SPEEX}, "audio", "SPEEX"}, + {{1, AST_FORMAT_ILBC}, "audio", "iLBC"}, {{0, AST_RTP_DTMF}, "audio", "telephone-event"}, {{0, AST_RTP_CN}, "audio", "CN"}, {{1, AST_FORMAT_JPEG}, "video", "JPEG"}, @@ -401,6 +406,7 @@ static struct rtpPayloadType static_RTP_PT[MAX_RTP_PT] = { [34] = {1, AST_FORMAT_H263}, [101] = {0, AST_RTP_DTMF}, [110] = {1, AST_FORMAT_SPEEX}, + [120] = {1, AST_FORMAT_ILBC}, }; void ast_rtp_pt_clear(struct ast_rtp* rtp) @@ -710,6 +716,9 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec case AST_FORMAT_GSM: pred = rtp->lastts + (f->datalen * 160 / 33); break; + case AST_FORMAT_ILBC: + pred = rtp->lastts + (f->datalen * 240 / 52); + break; case AST_FORMAT_G723_1: pred = rtp->lastts + g723_samples(f->data, f->datalen); break; @@ -822,6 +831,18 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_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(52); + } + 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 %d packets\n", _f->subclass); // fall through to... |