Streaming audio to 64 kbps u-law client

We have a requirement to provide steaming audio to a client that only supports 64 kbps u-law. The audio source will be analag (stereo L/R). The client will use rtsp to setup the streaming session.
analog audio -> encoder -> streaming server -> client (rtsp/64 kbps u-law)
Will the Quicktime Broadcaster and Streaming Server satisfy this requirement?
Thanks,
frwillia

Hello, I also need to play audio from a server via http, but using bluetooth connections. I can play from the localhost using the emulator but i don't know how to do it from the phone using bluetooth.
Thanks.

Similar Messages

Streaming audio from clients to server

Hi folks,
I am looking for a method in which I can stream audio from a device mac/win to the server which is connected to the good sound system over the local network.
I have looked at pulseaudio and followed the directions in the wiki, but without any results
This was a linux to linux setup. But it seems to only get worse when trying to stream from mac/win (found these pages)
https://labitat.dk/wiki/Apple/Mac_OS_X#PulseAudio
http://gleamynode.net/articles/2228/
So what I am looking for here is either a thorough tutorial/directions or a plug n play solution.
Any inputs, links or directions are greatly appreciated, since I have spent this entire weekend researching it, and failed horribly every time

hmm its simple and it sort of works. Their is very little documentation as far as I can see.
I start server by commiting rsd -D -v
then stream some audio by cat sound.wav | rsdplay -s 127.0.0.1 --raw -r 44100 -c 2
It sounds very sluggish, and it also tells me that Only 8 or 16 bit WAVE files supported.
Also it looks like the last update was a OSS patch which was mentioned in the original thread over a year ago
Last edited by odp (2013-08-11 19:13:02)

Streaming audio data won't stop unless I turn off the phone for 5 minutes.

I have been testing the Curve 8330 regarding streaming audio from an icecast server. The address of the station is http://www.AirProgressive.org, click on "listen". (I have already run this by the icecast experts and they say it is a Blackberry "bug")
BLACKBERRY USERS! PLEASE BE AWARE THAT THE STREAM (dataflow) DOES NOT STOP WHEN YOU CLICK "STOP" ON THE MEDIA PLAYER! (and will continue for as long as you leave it on, and even if you turn off your BB for less than 5 Minutes.) I'm not sure if you are charged for these streams, but it seems that the battery is depleted rapidly if this continues without aborting it. See details of my testing below:
> Access icecast stream by clicking on the link on the website:
     http://www.airprogressive.org:8000/stream.m3u
> Stream consists of mp3 at 32kbps / 22050 Hz.
> Stock media player launches, stops at "Do you want to open or save the item."
    > No connections started yet (viewing "List Clients" page of icecast admin interface).
> Click "open", stream begins playing.
> Two connections are started instead of just one, always offset by 2 seconds.
    -- but sometimes only one connection starts.
> FYI, Icecast config: <client-timeout>30</client-timeout>
Here is the output from the "List Clients" page of Icecast2 Admin
IP
Seconds Connected
User Agent
Action
68.7.238.55
406
WinampMPEG/5.62, Ultravox/2.1
Kick
74.82.68.16
6
Mozilla/4.8 [en] (Windows NT 5.0; U)
Kick
74.82.68.16
4
Mozilla/4.8 [en] (Windows NT 5.0; U)
Kick
Note the last two entries with the same IP address. This is the Blackberry. Top entry is winamp player on PC for reference. Don't know why two listener connections are generated.
> Stream is "Stopped" (i.e. click stop)
> Stream is restarted (i.e. click Play)
> A new connection is started, this time with a new +1 IP address.
IP
Seconds Connected
User Agent
Action
68.7.238.55
687
WinampMPEG/5.62, Ultravox/2.1
Kick
74.82.68.16
287
Mozilla/4.8 [en] (Windows NT 5.0; U)
Kick
74.82.68.16
285
Mozilla/4.8 [en] (Windows NT 5.0; U)
Kick
74.82.68.17
8
Mozilla/4.8 [en] (Windows NT 5.0; U)
Kick
I can understand that the player may continue to communicate when "stopped" to fill the lookahead buffer, but when it is restarted, it should use the same connection and not start a new one.
> Click Stop.
> Click Play.
IP
Seconds Connected
User Agent
Action
68.7.238.55
803
WinampMPEG/5.62, Ultravox/2.1
Kick
74.82.68.16
403
Mozilla/4.8 [en] (Windows NT 5.0; U)
Kick
74.82.68.16
401
Mozilla/4.8 [en] (Windows NT 5.0; U)
Kick
74.82.68.17
124
Mozilla/4.8 [en] (Windows NT 5.0; U)
Kick
74.82.68.18
24
Mozilla/4.8 [en] (Windows NT 5.0; U)
Kick
> Yet another connection is started with a new "listener".
> Click "back" to exit from browser in Blackberry.
> All threads continue.
    --> I have tested this many times and the threads continue without being cleaned up indefinitely, even though the player is stopped and the stream is not "playing" on the handset.
> Turn off mobile network. Handset not communicating.
> Threads continue
> Handheld turned off.
IP
Seconds Connected
User Agent
Action
68.7.238.55
995
WinampMPEG/5.62, Ultravox/2.1
Kick
74.82.68.16
595
Mozilla/4.8 [en] (Windows NT 5.0; U)
Kick
74.82.68.16
593
Mozilla/4.8 [en] (Windows NT 5.0; U)
Kick
74.82.68.17
316
Mozilla/4.8 [en] (Windows NT 5.0; U)
Kick
74.82.68.18
216
Mozilla/4.8 [en] (Windows NT 5.0; U)
Kick
Finally, after over 300 seconds, the threads are finally cleaned up.
IP
Seconds Connected
User Agent
Action
68.7.238.55
1139
WinampMPEG/5.62, Ultravox/2.1
Kick
So, to stop the streaming server from communicating with the Blackberry handset, you must either turn it off or disable mobile network for at least 5 minutes.
It appears that the handset continues to transfer data (and therefore incur charges, if the user does not have an unlimited data plan) if the handset is not turned off or mobile network disabled for at least five minutes. It also runs down the battery much faster because the radio transmitter is actively being used, even if the user is not listening to the stream.
The Blackberry network has intervening servers that continue to receive data from the streaming server even if the handset is off. This behavior likely has utility to deal with the realities of the cellphone network, to allow the stream to be transferred from one cell to another, and deal with the possibility that the handset may be out of range for up to 5 minutes without "interruption." However, it is hard to turn off the stream.
QUESTIONS:
1. Is there a way to stop this sort of behavior using a configuration setting or something? It seems particularly stupid to start a new stream connection each time a STOP/START cycle is completed, allowing the old connections to continue streaming data (even if the data does not make it to the handset).
2. IS IT TRUE that the user will be "charged" for the data streams that are not disconnected?
3. Is there a way for a developer to send a signal to the intervening cellphone network to discontinue the orphan streams?
THANKS!!

No, I'm using something like this :
to start...
                   channel = new SoundChannel();
                                        soundPlay = new Sound();
                                        urlReq=new URLRequest;
                                        urlReq.url=mylUrl; // url of radio mp3
                                        soundPlay.load(urlReq);
                   channel = soundPlay.play();
to stop it...
                      channel.stop();
                      channel=null;
                      try{soundPlay.close();} catch (e:Error) {} // error or not, the sream won't stop.
Any idea ? Did you try with a NetStream object ?
Bernard

Bad sound quality streaming audio via Airport Express

I am streaming audio from iPhone to Apple Airport Express (Airplay). Via optical cable the signal goes to Cambridge DacMagic and then to Stereo Amplifier. Sound is not good, there's some kind of 'chattering'.
When using AirFoil instead of Airplay sound is much better, but then gaps appear from time to time.
Please tell me, what can be the reason for it? And how to solve?
Thanks in advance.

Mr. Tessarex,
I use the Airport Exreme as router and Airport Express (2011, direct plug-in model)
Some facts I learned from the Airport-app on my iPhone:
Wireless clients from Airport Extreme:
iPhone 150Mb/s (802.11 a/n) connection: excellent
Acer laptop 144Mb/s (802.11 b/g/n) connection: excellent
The Airport Express:
240Mb/s (802.11 a/n) connection: good
I bought the song "Love Me Like You Do" (Ellie Goulding) via iTunes.
As you adviced me I tried to play it via iTunes, and streamed it from iPhone to Airport Express.
Especially listening to the bombastic part of the song (from 3:14). Sound was tolerable.
Playing the song from iPhone via Spotify: sound is awfull, totally overloaded.
Playing the song from laptop via Spotify (Airfoil): sound is good.
Playing the song from laptop via iTunes (Airplay): sound is good.
The song never plays 'Excellent' in any way. I Think the song was recorded 'over the top'.
Even when you listen to it via headphone, it sounds very messy, "on the edge...".
I am happy that most songs sound good. So far playing Spotify via laptop and Airfoil gives the best results.
According to the figures from the Airport-app, it seems I will never get optimal bitrate with my current system.
As amplifier I use the Luxman L505u, usually connected tot Airport Express via Cambridge Dacmagic,
but during testing via normal analog cable (without Dacmagic). So far I think the Dagmagic is not the problem.
What do you think, after reading my story?
Thanks in advance
Peter

Streaming audio-video on the mobile device

Please, I must do the streaming audio-video on the mobile device using a server standard.
I must to program the application for the server using j2se:in this case how i can send the file audio/video?This file have specifies classes?
For the client i must using the j2me:in this case how can receive the file audio and creating the player?
Thanks.

Hi, please help,
I have problem in this area also,
am intending to stream audio and video data to mobile phone using j2me for my bs project.,
is that true, midlets can only 'plays back' one file at time specified by url?
I want to store many files in database( e.g ms access), and call them to midlets, to make some how interactive and real,because i have tomcat server installed in my pc.
is this possible?
How can i call file stored beyond path of WTK ?
can i store audio/ video file in ms- access?
Please, please help !
thanks in advance.
Message was edited by:
nbkamani

Streaming audio to video UI component, any ways possible?

I'm looking into streaming audio from online radio sources our locally to the video component in interface builder, but I'm sure where to start. Any suggestions?

for a more generic approach -- since mpd requires you to put the media on the server and only plays certain defined filed types -- pulseaudio might be interesting, or even dlna/upnp? that last one for sure would be the most cross-platform solution. then each client could decide for itself which kind of audio to pass to the hub, be it mpd, radio, youtube etc.
interesing experiment you're doing here btw, i think more people will be interested in using a single small device that routes all their network audio to a stereo.
p.s., in case you want to stick to the classic mpd setup, there are windows clients available incluing a handy firefox plugin:
http://mpd.wikia.com/wiki/Windows_Compatibility
http://mpd.wikia.com/wiki/Clients

IOS RTMP live stream audio

Hi guys hopefully some of you will be able to help me.
I am trying to stream audio from a live RTMP feed using the NetConnection and NetStream classes. I've managed to get my app running no problem on Android, however I am having some major difficulties getting it to play the audio back on iPad. Interestingly it works in the device emulators when debugging, however I'm assuming this is not really an accurate representation. I've tried streaming the RTMP in both AAC and MP3, but with no luck from either. I can verify through debug that it has connected to the stream, however I just get no audio playing.
Everything I've read about seems to suggest that this is possible on IOS as I'm only interested in audio and not video. Can anyone help?
Code sample below (it's quick and dirty! ).
THanks in advance!
<?xml version="1.0" encoding="utf-8"?>
<s:View xmlns:fx="http://ns.adobe.com/mxml/2009"
                    xmlns:s="library://ns.adobe.com/flex/spark" title="Audio" creationComplete="init()">
          <s:layout>
                    <s:VerticalLayout paddingLeft="10" paddingRight="10"
                                                              paddingTop="10" paddingBottom="10"/>
          </s:layout>
          <fx:Script>
                    <![CDATA[
                              import flash.media.Video;
                              import flash.net.NetConnection;
                              import flash.net.NetStream;
                              import mx.core.UIComponent;
                              private var vid:Video;
                              private var videoHolder:UIComponent;
                              private var nc:NetConnection;
                              private var defaultURL:String="[STREAM]";
                              private var streamName:String="[STREAMNAME]";
                              private var ns:NetStream;
                              private var msg:Boolean;
                              private var intervalMonitorBufferLengthEverySecond:uint;
                              private function init():void
                                        vid=new Video();
                                        vid.width=864;
                                        vid.height=576;
                                        vid.smoothing = true;
                                        //Attach the video to the stage
                                        videoHolder = new UIComponent();
                                        videoHolder.addChild(vid);
                                        addEventListener(SecurityErrorEvent.SECURITY_ERROR, onSecurityError);
                                        grpVideo.addElement(videoHolder);
                                        connect();
                              public function onSecurityError(e:SecurityError):void
                                        trace("Security error: ");
                              public function connect():void
                                        nc = new NetConnection();
                                        nc.client = this;
                                        nc.addEventListener(NetStatusEvent.NET_STATUS, netStatusHandler);
                                        nc.objectEncoding = flash.net.ObjectEncoding.AMF0;
                                        nc.connect(defaultURL);
                              public function netStatusHandler(e:NetStatusEvent):void
                                        switch (e.info.code) {
                                                  case "NetConnection.Connect.Success":
                                                            trace("audio - Connected successfully");
                                                            createNS();
                                                            break;
                                                  case "NetConnection.Connect.Closed":
                                                            trace("audio - Connection closed");
                                                            //connect();
                                                            break;
                                                  case "NetConnection.Connect.Failed":
                                                            trace("audio - Connection failed");
                                                            break;
                                                  case "NetConnection.Connect.Rejected":
                                                            trace("audio - Connection rejected");
                                                            break;
                                                  case "NetConnection.Connect.AppShutdown":
                                                            trace("audio - App shutdown");
                                                            break;
                                                  case "NetConnection.Connect.InvalidApp":
                                                            trace("audio - Connection invalid app");
                                                            break;
                                                  default:
                                                            trace("audio - " + e.info.code + "-" + e.info.description);
                                                            break;
                              public function createNS():void
                                        trace("Creating NetStream");
                                        ns=new NetStream(nc);
                                        //nc.call("FCSubscribe", null, "live_production"); // Only use this if your CDN requires it
                                        ns.addEventListener(NetStatusEvent.NET_STATUS, netStreamStatusHandler);
                                        vid.attachNetStream(ns);
                                        //Handle onMetaData and onCuePoint event callbacks: solution at http://tinyurl.com/mkadas
                                        //See another solution at http://www.adobe.com/devnet/flash/quickstart/metadata_cue_points/
                                        var infoClient:Object = new Object();
                                        infoClient.onMetaData = function oMD():void {};
                                        infoClient.onCuePoint = function oCP():void {};
                                        ns.client = infoClient;
                                        ns.bufferTime = 0;
                                        ns.play(streamName);
                                        ns.addEventListener(AsyncErrorEvent.ASYNC_ERROR, asyncErrorHandler);
                                        function asyncErrorHandler(event:AsyncErrorEvent):void {
                                                  trace(event.text);
                                        intervalMonitorBufferLengthEverySecond = setInterval(monPlayback, 1000);
                              public function netStreamStatusHandler(e:NetStatusEvent):void
                                        switch (e.info.code) {
                                                  case "NetStream.Buffer.Empty":
                                                            trace("audio - Buffer empty: ");
                                                            break;
                                                  case "NetStream.Buffer.Full":
                                                            trace("audio - Buffer full:");
                                                            break;
                                                  case "NetStream.Play.Start":
                                                            trace("audio - Play start:");
                                                            break;
                                                  default:
                                                            trace("audio - " + e.info.code + "-" + e.info.description);
                                                            break;
                              public function monPlayback():void {
                                        // Print current buffer length
                                        trace("audio - Buffer length: " + ns.bufferLength);
                                        trace("audio - FPS: " + ns.currentFPS);
                                        trace("audio - Live delay: " + ns.liveDelay);
                              public function onBWDone():void {
                                        //Do nothing
                              public function onFCSubscribe(info:Object):void {
                                        // Do nothing. Prevents error if connecting to CDN.
                              public function onFCUnsubscribe(info:Object):void {
                                        // Do nothing. Prevents error if connecting to CDN.
                    ]]>
          </fx:Script>
          <s:Group id="grpVideo">
          </s:Group>
</s:View>

Just an update on this for anyone coming along after me.
I've managed to get this working on MP3 but not AAC (I guess AAC just isn't supported?).
My problem was the buffertime. The docs seemed to indicate it shoudl be set to 0 for live streaming, however switching it to "1" solved my problem on the particular stream I was pointing at.
So essentially justchanged the above line:
                                        ns.bufferTime = 0;
to
                                        ns.bufferTime = 1;
Would be great to find out if anyone has gotten AAC working however...

Unwanted synchronization when streaming audio&video

hi,
i'm streaming audio and video, both of which are created with separate Processors. i'm playing them back on the client's side with separate players on separate ports. despite all this, when i pause the audio, the video pauses too!
any suggestions?
Thanks in advance :O)

This is suposed to happen if the streams received have the same CName. I guess if you are able to send two different streams with 2 different CNames they won't synchronize. But I still don't know how to do that... :)
Also, less useful but still a fact, if you receive the streams in two different VM's that won't happen either... :)
Bira

Streaming audio to an embedded Linux device

A friend and I are tossing around some ideas for a project, which would be a music-activated RGB lights controller. We have an ARM board with network connectivity, running Linux. What we want to be able to do is to stream audio from a laptop/desktop computer to the device, for processing. I'm not sure what software to use on the computer, for streaming, as well as client software on the embedded board. Here are my primary constraints:
1) The streaming software has to run on at least Windows/Linux (although Mac would be a plus), and has to either be part of, or be pluggable into, a music app that's suitable for using playlists and managing large libraries (e.g. something like Songbird, rather than VLC).
2) The client software needs to have as little overhead as possible, due to the limitations of the embedded board, and it needs to be CLI
3) We really don't care about the quality of the audio, so long as the waveforms look enough like the originals that the lights can actually correspond to something meaningful in the audio
4) Service discovery would be a plus, as we'd eventually like to just plug in the controller and go, without having to mess with IP addresses and such
Any suggestions?

for a more generic approach -- since mpd requires you to put the media on the server and only plays certain defined filed types -- pulseaudio might be interesting, or even dlna/upnp? that last one for sure would be the most cross-platform solution. then each client could decide for itself which kind of audio to pass to the hub, be it mpd, radio, youtube etc.
interesing experiment you're doing here btw, i think more people will be interested in using a single small device that routes all their network audio to a stereo.
p.s., in case you want to stick to the classic mpd setup, there are windows clients available incluing a handy firefox plugin:
http://mpd.wikia.com/wiki/Windows_Compatibility
http://mpd.wikia.com/wiki/Clients

Streaming audio from Logic 7.2

Has anyone found a good way to stream audio from Logic to another computer via the web? I'm tweaking a sound design mix for a client in another city and would love to be able to get them on the phone and have them be able to participate in real time. Where's the Rocket Network when you need it? : )Thanks.

2 solutions:
1. Old school: get a telephone coupler and send the mix down the phone line.
2. iChat: feed your mix into a small mixer, open a mic when you want to talk, and feed the mix into your "line input" on iChat (make sure it's selected in preferences) and do an audio chat.
DP800 G4/1 Ghz Ti PB 15" Mac OS X (10.4.5)

Streaming audio across the whole folio

We have a client who is interested in creating a "radio" function, where we would stream audio across the whole folio. The desired effect would be that audio would play continuously and uninterrupted on each page of each article.
I know that there is a way to play a local audio file across the whole folio. We already tested that and it should work just fine, but the client would like the audio to be streamed from the internet because we may not have the rights to embed the audio files.
Do you know if there is a provision to do something like this from Adobe DPS? We heard that it might be possible by using a set up similar to browser frames, but I was unable to find any documentation about that.
iPad only.

We don't support authoring streaming audio and video using the DPS authoring tools at the moment. There's no way to do this with a web overlay as the overlay will get removed from memory once your reader moves away to another article.
Neil

Streaming Audio using JMF

I am trying to create a application that uses streaming audio and would be an internet radio with a server that will stream songs and a client that can listen to the stream. How can I stream audio, specifically mp3 files, using the JMF?

I am in a similar project.
I found JMF does not provide enough support for stream.
I can't split a stream into some packets and transfer them to another peer.

Is it possible to stream audio to mobiles over RTP

I am researching streaming audio to mobile phones. To set the context - a scenario would be the sending of a wav file arising from text-to-speech conversion being streamed to a specific mobile phone engaged in the dialogue. Using the MMAPI I have created a Player and can play wav files using the Manager.createrPlayer("http://blah:8080/test.wav"). My understanding is that http based on tcp is not suitable for streaming. Far better to use RTP over UDP. I have done this for desktop clients. However, my question is can i use the RTP protocol to stream wav files to mobile devices. I think the answer is no since it is not a supported protocol as identified by the Manager.getSupportedProtocols("audio/x-wav") method. The supported protocols returned are http and capture. Neither does it work in my dummy test. I get the following error if i try to create the player with an rtp protocol spec in the media locator, as in Manager.createPlayer("rtp://192.9.206.164:49150/audio");:
javax.microedition.media.MediaException: Cannot create a DataSource for: rtp://192.9.206.164:49150/audio
     at javax.microedition.media.Manager.createDataSource(+92)
     at javax.microedition.media.Manager.createPlayer(+93)
     at MobileClient$ConnectionManager.run(+110)
Comments most welcome please.

that protocol is not requiered to be implemented by Sun's j2me specification. So it's up to the vendor whether he implements it or not. My expierence shows, that most vendors just implement the minimum requierements and don't spend any more dollar on anything not requiered.
Since not even a socket connection is requiered to be implemented, you not even get the chance to implement an own 'transmission' protocol that might fit your needs.
So you're only chance will be if a vendor decides to implement any device specific api.

Streaming audio, web hosting, scripts for streaming?

Hello all,
Back several years ago, it seemed all the big web hosts
featured "streaming
audio and video" as a feature of their hosting plans at some
dollar amount. You
know, once you elevated to the $19.95 a month plan at XYZ
host, your website
could stream audio or video.
I don't see that anywhere in web hosts monthly plans any
longer at any dollar
amount. Anyway I have a client who sells her own music, and I
have been trying
to find a web host that supports streaming audio, mp3, etc. I
checked with 1
and 1 (webhost) and they responded as follows:
"Basically our eshop can support streaming but it would be
through
scripts only. So, it would be in your end to do those
matters."
I have no idea what that actually means. And I realize I
could just send the
guy a note and ask him, but I figured I would get a more
comprehensive and
valuable answer here since I use Dreamweaver. By "script" are
they talking
about something like Flash Audio Kit? I am confused by the
whole thing now. I
thought it was as simple as looking for a hosting plan that
supported streaming.
I use Dreamweaver 8, Windows, if that matters for this
question.
Thanks a million!
Tim

Thank you for responding but I finally got it to work.
By the way it was http://wm1.uwsp.edu/90fm ... go trivia...

Fix streaming audio dropouts with Airport Express (811g)?

Every time I try playing streaming audio with just an external audio system connected via Airport Express (811g) I get frequent dropouts no matter what kbps setting of the source. Looks like it is not being buffered properly. However, if I choose Multiple Speakers and include Computer, everything is fine. Actually, if I choose Multiple Speakers, include Computer and let it run this way for 5-10 minutes, I can then turn off Computer and everything is fine. Obviously I have a work-around, but is there a better solution?

I experience a similar drop - only w/streaming audio URLs I've opened in the "Advanced" option of iTunes' Menu; I don't experience drops when audio is streamed from iTunes Radio listings. Just want to thank you for the work-around info: it works and until reading your post if I wanted streaming audio sound from remote speakersI had to disconnect the mac w/my music from Airport and connect it directly to modem!
Thanks (sorry don't have insight for dismissing "work-around")!

Streaming audio to 64 kbps u-law client

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