search for: format_s

I have a situation where I have a number of fields and a rating for the field. So, in a DRY way, this works fine: <%= [''abc'', ''def'', ''klm''].each do |e| %> <tr><td><%= @result[e] %></td> <td> <%= @result[e+''rating''] %></td> </tr> <% end %> Now, I

...vsroot/icecast/src/format.c,v retrieving revision 1.20 diff -u -p -r1.20 format.c --- src/format.c 15 Mar 2003 02:10:17 -0000 1.20 +++ src/format.c 2 Jul 2003 23:17:53 -0000 @@ -1,3 +1,4 @@ +/* -*- c-basic-offset: 4; -*- */ /* format.c ** ** format plugin implementation @@ -101,7 +102,11 @@ void format_send_general_headers(format_ http_var_t *var; avl_node *node; int bytes; + char *client_is_icy; + /* if the client uses icy headers, we'll respond in kind */ + client_is_icy = httpp_getvar(client->parser, "icy-metadata"); + /* iterate through source...

...m/bug_view_page.php?bug_id=0002381 These include app_chanspy, the ability to spy on ANY bridged call taking place inside asterisk. NOT just ZAP as with ZapScan/Barge. Native format_* files being used for moh. Reload enabled res_musiconhold. format_mp3.c that produces SLNR output to asterisk, format_slinear.c for raw headerless audio, format_base65_wav_gsm.c aka wav49 held in a base64 containers(it can read and playback from these .b64 files) All this is thanks to my employer asterlink.com and anthm. So everyone please test and provide feedback. Thanks, Brian Asterlink.com PS: More to come at...

Oh -- then I did not understand the documentation. fallback-mount This optional value specifies a mountpoint that clients are automatically moved to if the source shuts down or is not streaming at the time a listener connects. Only one can be listed in each mount and should refer to another mountpoint on the same server that is streaming in the _same streaming format_. I was thinking that

Hi guys, I'm writing an application that will sniff audio streams going across the network and allow a seperate client application to listen into one of these streams. I was looking into ways to distribute the captured stream media to a client for listening, and icecast came to mind. Icecast, as I understand it, is usually used for streaming a few data sources to MANY clients. As I want

reflum, On Wed, 2015-06-24 at 06:55 -0700, that Jack Elliott wrote: > Oh -- then I did not understand the documentation. > > fallback-mount > This optional value specifies a mountpoint that clients are > automatically moved to if the source shuts down or is not streaming at > the time a listener connects. Only one can be listed in each mount and > should refer to another

...vsroot/icecast/src/format.c,v retrieving revision 1.20 diff -u -p -r1.20 format.c --- src/format.c 15 Mar 2003 02:10:17 -0000 1.20 +++ src/format.c 2 Jul 2003 23:45:51 -0000 @@ -1,3 +1,4 @@ +/* -*- c-basic-offset: 4; -*- */ /* format.c ** ** format plugin implementation @@ -109,8 +110,12 @@ void format_send_general_headers(format_ var = (http_var_t *)node->key; if (strcasecmp(var->name, "ice-password") && (!strncasecmp("ice-", var->name, 4) || - !strncasecmp("icy-", var->name, 4))) { -...

...> of speex support for it anywhere (including the source). Looking through > the source, it looks like such support would be somewhat trivial for > someone that knew what they were doing. Would I be correct in my > assumption that the easiest way to do this would be to write a > format_speex.c and .h file from format_vorbis.c/.h that conforms to the > Speex in Ogg encapsulation spec at > http://www.speex.org/manual2/node7.html#SECTION00073000000000000000 ? > It also seems that libshout would require some slight modifications > (e.x. a new FORMAT_ specified for SPE...

Hello, Running one asterisk server with below details. Only SIP to SIP calls. No real time configuration, no recording, no voicemail, no IVR, no codec translation. Average CPU load varies between 4 to 30 for 150 to 200 concurrent calls and we start getting problem in call quality like delay in connectivity, voice breakage etc.... Hardware: 2 Physical processor Intel(R) Xeon(R) CPU

I finally managed to get back to working on this patch. To get some feedback I've linked to it and posted a few comments on it. http://mediacast1.com/~karl/single-q.diff My previous patch (client write) I sent to the list needs to be applied first, there was no feedback to it. http://www.xiph.org/archives/icecast-dev/1153.html (looks like it still applies to svn, if not then let me know)

...the available modules for sound files: root@asterisk1 modules]# cd /usr/lib/asterisk/modules/ root@asterisk1 modules]# ls -1 format_* format_au.so format_g723.so format_g726.so format_g729.so format_gsm.so format_h263.so format_ilbc.so format_jpeg.so format_mp3.so format_pcm_alaw.so format_pcm.so format_sln.so format_vox.so format_wav_gsm.so format_wav.so So I imagine I can use .au and .mp3 format for audio files. That's what I want. But I cannot find now the appropriate format to specify in voicemail.conf format directive. I set format=mp3 first and format=au But when the voicemail prompt...

Icecast allows using a fallback mount file which the listener will hear if the main mount stops. It is specified in icecast.xml as this example: <fallback-mount>/example2.ogg</fallback-mount> It is necessary that the file have the same bitrate and same format (single or two-channels) as the main mount. We were at a music festival last weekend and used butt (Broadcast Using This

relaying a shoutcast stream via icecast2 is pretty much unlistenable -- it constantly stutters. for example, shoutcast relay -- http://205.188.234.1:8030/listen.pls icecast2 relay -- http://liveradio.indymedia.org:8001/sf-hifi.mp3.m3u --mark --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a

Greetings. I'm trying to recreate in R some regression models I've done in SAS, but I'm not getting the same results. My advisor suspects this may be due to differences in precision between R and SAS. Does anyone know where I can find specifications for R's type double? (It doesn't seem to be in the R Language Definition.) Thanks in advance for any help anyone can

I 100% get that we might not like the decisions clang-format is making, but how does one overcome this when adding new code? The pre-merge checks enforce clang-formatting before commit and that's a common review comment anyway for those who didn't join the pre-merge checking group. I'm just wondering are we not all following the same guidelines? Concerns of clang-format not being good