similar to: STREAM FILE question

Trying to redirect to -user... On Tue, 7 Jul 2009, Bryant Zimmerman wrote: > Hey guys I posted this earlier and did not get any responses. You posted what appear[s|ed] to be a user question to the dev list. I did reply (on June 3), but I may have mis-understood. > I am working on some AGI development that requires control of audio file > playback. The control stream file is working

I highly recommend the open source Jinzora project (http://www.jinzora.org/). It has merged with and extended the capabilities of the older NetJuke project. It's geared towards authenticated, on-demand playback and supports many formats, ogg included. I've been thoroughly impressed with the quality of the package, it's GUI options, and its robustness. I personally use icecast2 to

Hi While testing I found a solution to my problem. I don't understand it maybe someone here can explain it. In my script, if I call a Playback just before my stream file then everything works ok. Without the playback then the digits are not captured I will playback a silence to patch my scripts. Patrick =========================== Hi I have a strange problem. I use the agi command

On Mon, Jun 6, 2011 at 10:10 AM, Steve Edwards <asterisk.org at sedwards.com>wrote: > On Mon, Jun 6, 2011 at 2:26 AM, Steve Edwards <asterisk.org at sedwards.com> >> wrote: >> >> I strongly suggest using an existing library for the language of your >> choice. >> > > On Mon, 6 Jun 2011, A E [Gmail] wrote: > > Copy that. Not planning to

Hi, We've run into an interesting (to us) problem with SpeechBackground. Inside a AGI script, we're playing some extended audio?basically, like a podcast?and we want playback to stop if and only if the speech recognized matches something in our grammar. If there's speech that doesn't match, we just want to go right on playing. (We're using LumenVox as our speech recognition

Hello. Is this normal. Or does it mean there is a problem ? ------------------------- stop now Beginning asterisk shutdown.... Executing last minute cleanups == Destroying any remaining musiconhold processes Yuck! Error in buffer handling...: Connection reset by peer Yuck! Error in buffer handling...: Broken pipe Yuck! Error in buffer handling...: Broken pipe Asterisk cleanly ending (0).

Hello, I'm currently trying to implement streaming off ogg-files out of an archive file using SDL and SDL_mixer for playback. Therefore i create a FILE * filepointer, place it to the beginning of an ogg file inside the archive and hand over the filepointer to ov_open. ( command: ov_open(fp, &music->vf, NULL, 0) ). This works fine so far. The problem is: if i put many oggs together in

On 10/11/2017 09:49 PM, Michael S. Tsirkin wrote: > On Wed, Oct 11, 2017 at 02:03:20PM +0800, Wei Wang wrote: >> On 10/10/2017 11:15 PM, Michael S. Tsirkin wrote: >>> On Mon, Oct 02, 2017 at 04:38:01PM +0000, Wang, Wei W wrote: >>>> On Sunday, October 1, 2017 11:19 AM, Michael S. Tsirkin wrote: >>>>> On Sat, Sep 30, 2017 at 12:05:54PM +0800, Wei Wang

I'll toss in here and augment with Wade's remarks. Absolutely this is not an optimal use of icecast but may actually result in reduced complexity for deployment for people who need to stream both dynamic and static content. For example, if we have a live streaming site but we want to put some sample static content in the profile of the band who will be streamed later - going ahead and

On Jun 4, 2008, at 5:25 AM, Thomas B. Ruecker wrote: > H. Wade Minter schrieb: >> I'm in a situation where I have some MP3 files I'd like to stream on- >> demand, but for various reasons can't just make the MP3 files >> themselves available. >> >> Is there a way, using icecast, to set up a server that would just >> bind >> to a port, and

Thank you Philipp, Your answers are very clear. We understand the difference between the wall clock time (server side) and the playback time. Just one more point. You wrote that the "into file" is sent to the listener before the listener is attached to the actual stream. If the server would take 20 seconds to send the "into file" and if the listener only remains connected 5

On Thu, May 18, 2006 at 05:31:00PM +0000, Georg Holzmann wrote: > Hallo! > > >>So if a user wants to hear record A, then a stream should be opened > >>(maybe a new mount point ?) and only this user should be allowed to hear > >>the stream ... > >>Is this possible with icecast ? > > > >It probably would. > >(Aber man kann sich auch

On 22 Jul 2016, at 20:39, Damien Sykes-Lindley wrote: > Hi there, Hello! > […] > I then tried accessing the stream directly from the url and found to > my surprise that the stream was working. Since my browser has stream > playback support, it started playing straight away. If your browser is able to play the stream, Icecast works and if it does not with a different player,

Hi there, I am relatively new in the field of server administration, so please do bear with me if my question has an answer which seems blatantly obvious. I recently installed the latest Icecast onto my machine and read the documentation in order to find out how to configure the server. Both the information in the docs, and even the example config file, seemed reasonablly self-explanatory. After

Yes it's the minimum, as you can't decode half a frame. In Speex frames are 20ms which should be low enough anyway. Jean-Marc Le ven 16/05/2003 à 04:03, Robert M. a écrit : > Here is my situation: > > I have large Speex encoded stream which I would like to playback on wave > out device. I could decode whole stream to pcm and play it back as pcm > audio, but this

On Wed, Oct 11, 2017 at 02:03:20PM +0800, Wei Wang wrote: > On 10/10/2017 11:15 PM, Michael S. Tsirkin wrote: > > On Mon, Oct 02, 2017 at 04:38:01PM +0000, Wang, Wei W wrote: > > > On Sunday, October 1, 2017 11:19 AM, Michael S. Tsirkin wrote: > > > > On Sat, Sep 30, 2017 at 12:05:54PM +0800, Wei Wang wrote: > > > > > +static void ctrlq_send_cmd(struct

On Wed, Oct 11, 2017 at 02:03:20PM +0800, Wei Wang wrote: > On 10/10/2017 11:15 PM, Michael S. Tsirkin wrote: > > On Mon, Oct 02, 2017 at 04:38:01PM +0000, Wang, Wei W wrote: > > > On Sunday, October 1, 2017 11:19 AM, Michael S. Tsirkin wrote: > > > > On Sat, Sep 30, 2017 at 12:05:54PM +0800, Wei Wang wrote: > > > > > +static void ctrlq_send_cmd(struct

Hello, I'm new to this list, and pretty much also to the codec universe. In a computer game I'm working on, there is an 3D TV-screen onto which I project the frames of a movie. The movie is of course stored as theora, no audio as it's not needed yet. I have it working so far, extracting a frame and uploading it to the graphics card when the scene is drawn. The example file

> > > $ gdb /path/to/ices /path/to/coredump > > > > > #0 0x402b708d in III_dequantize_sample () from > > #/usr/local/lib/libmp3lame.so.0 1 0x402d0bc0 in ispow () from > > #/usr/local/lib/libmp3lame.so.0 > > Cannot access memory at address 0x1 > > Could this be an alpha-version of LAME? Switching to the latest > stable version (3.93-1 afair)

The sound file is played correctly for the first time, then when rewind to the initial position. then copy PCM to buffer, the OpenAL report an error. It seems like the OpenAL doesn't recognize the PCM data. The OpenAL error number : AL_INVALID_VALUE 0xA003 void Buffer::PCMData (ALuint id, ALenum eFormat, ALvoid *data, ALsizei size, ALsizei freq) { // Copy