diff options
| author | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-03-07 00:24:58 +0000 |
|---|---|---|
| committer | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-03-07 00:24:58 +0000 |
| commit | 5ffedddec1865ff45cede194076c3ff69b3f53ab (patch) | |
| tree | d3cc4eb0d3575328fd5fb7eb2fdb1743f78fadb2 /main | |
| parent | e2787803841f33a7dff2dfbc98b79dcbd058a551 (diff) | |
Merge changes from team/russell/g722-sillyness ...
Fix a number of other places where the number of samples in a G722 frame was
not properly handled because of various reasons.
main/rtp.c:
- When a G722 frame is read from the smoother, the number of samples in the
frame must be divided by 2 before being sent out over the network. Even
though G722 is 16 kHz, an error in some previous spec has made it so that
we have to list the number of samples such as if it was 8 kHz.
main/file.c:
- When scheduling the next time to expect a frame, take into account that the
format of the file we're reading from may not be 8 kHz.
codecs/codec_g722.c:
- When converting from G722 to slinear, g722_decode() expects its samples
parameter to be in the silly (real samples / 2) format. Make it so.
- When converting from slinear to G722, properly set the number of samples in
the frame to be the number of bytes of output * 2.
formats/format_pcm.c:
- This format module handles G722, among a number of other formats. However,
the read() and seek() functions did not account for the fact that G722 has
2 samples per byte.
(closes issue #12130, reported by rickross, patched by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106501 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'main')
| -rw-r--r-- | main/file.c | 6 | ||||
| -rw-r--r-- | main/rtp.c | 8 |
2 files changed, 11 insertions, 3 deletions
diff --git a/main/file.c b/main/file.c index fb684755e..1bec4fca4 100644 --- a/main/file.c +++ b/main/file.c @@ -669,7 +669,8 @@ static enum fsread_res ast_readaudio_callback(struct ast_filestream *s) ast_settimeout(s->owner, whennext, ast_fsread_audio, s); else #endif - s->owner->streamid = ast_sched_add(s->owner->sched, whennext / 8, ast_fsread_audio, s); + s->owner->streamid = ast_sched_add(s->owner->sched, + whennext / (ast_format_rate(s->fmt->format) / 1000), ast_fsread_audio, s); s->lasttimeout = whennext; return FSREAD_SUCCESS_NOSCHED; } @@ -713,7 +714,8 @@ static enum fsread_res ast_readvideo_callback(struct ast_filestream *s) } if (whennext != s->lasttimeout) { - s->owner->vstreamid = ast_sched_add(s->owner->sched, whennext / 8, + s->owner->vstreamid = ast_sched_add(s->owner->sched, + whennext / (ast_format_rate(s->fmt->format) / 1000), ast_fsread_video, s); s->lasttimeout = whennext; return FSREAD_SUCCESS_NOSCHED; diff --git a/main/rtp.c b/main/rtp.c index 433fa2c79..c449b8b17 100644 --- a/main/rtp.c +++ b/main/rtp.c @@ -3186,8 +3186,14 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f) ast_smoother_feed(rtp->smoother, _f); } - while ((f = ast_smoother_read(rtp->smoother)) && (f->data)) + while ((f = ast_smoother_read(rtp->smoother)) && (f->data)) { + if (f->subclass == AST_FORMAT_G722) { + /* G.722 is silllllllllllllly */ + f->samples /= 2; + } + ast_rtp_raw_write(rtp, f, codec); + } } else { /* Don't buffer outgoing frames; send them one-per-packet: */ if (_f->offset < hdrlen) |