diff options
| author | tilghman <tilghman@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-11-06 22:51:48 +0000 |
|---|---|---|
| committer | tilghman <tilghman@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-11-06 22:51:48 +0000 |
| commit | 4b2fc9d3e738dbb2e7dd297de2f35e571e350691 (patch) | |
| tree | e611f1c508d4cbd9720858d06c92a4cfaa25dccb /main/rtp.c | |
| parent | fed34a6362c74a4f65bae5fb89bcae3f58db5f27 (diff) | |
Commit some cleanups to the format type code.
- Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits.
- Add a native slin16 type, so that 16kHz codecs can translate without losing resolution.
(This doesn't affect anything immediately, until another codec has wb support.)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89071 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'main/rtp.c')
| -rw-r--r-- | main/rtp.c | 10 |
1 files changed, 5 insertions, 5 deletions
diff --git a/main/rtp.c b/main/rtp.c index ef1cf0ddf..5ab4eb658 100644 --- a/main/rtp.c +++ b/main/rtp.c @@ -1571,7 +1571,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp) return f ? f : &ast_null_frame; } rtp->lastrxformat = rtp->f.subclass = rtpPT.code; - rtp->f.frametype = (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) ? AST_FRAME_VOICE : (rtp->f.subclass < AST_FORMAT_MAX_VIDEO) ? AST_FRAME_VIDEO : AST_FRAME_TEXT; + rtp->f.frametype = (rtp->f.subclass & AST_FORMAT_AUDIO_MASK) ? AST_FRAME_VOICE : (rtp->f.subclass & AST_FORMAT_VIDEO_MASK) ? AST_FRAME_VIDEO : AST_FRAME_TEXT; if (!rtp->lastrxts) rtp->lastrxts = timestamp; @@ -1586,7 +1586,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp) rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET; rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET; rtp->f.seqno = seqno; - if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) { + if (rtp->f.subclass & AST_FORMAT_AUDIO_MASK) { rtp->f.samples = ast_codec_get_samples(&rtp->f); if (rtp->f.subclass == AST_FORMAT_SLINEAR) ast_frame_byteswap_be(&rtp->f); @@ -1595,7 +1595,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp) rtp->f.has_timing_info = 1; rtp->f.ts = timestamp / 8; rtp->f.len = rtp->f.samples / 8; - } else if(rtp->f.subclass < AST_FORMAT_MAX_VIDEO) { + } else if(rtp->f.subclass & AST_FORMAT_VIDEO_MASK) { /* Video -- samples is # of samples vs. 90000 */ if (!rtp->lastividtimestamp) rtp->lastividtimestamp = timestamp; @@ -2958,7 +2958,7 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec ms = calc_txstamp(rtp, &f->delivery); /* Default prediction */ - if (f->subclass < AST_FORMAT_MAX_AUDIO) { + if (f->subclass & AST_FORMAT_AUDIO_MASK) { pred = rtp->lastts + f->samples; /* Re-calculate last TS */ @@ -2973,7 +2973,7 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec mark = 1; } } - } else if(f->subclass < AST_FORMAT_MAX_VIDEO) { + } else if(f->subclass & AST_FORMAT_VIDEO_MASK) { mark = f->subclass & 0x1; pred = rtp->lastovidtimestamp + f->samples; /* Re-calculate last TS */ |