Hello. I am using javasound and have an AudioInputStream of format PCM_SIGNED 8000.0 Hz, 16 bit, stereo, 4 bytes/frame, little-endian Using AudioSystem.getAudioInputStream(target_format, original_stream) produces an 'IllegalArgumentException: Unsupported Conversion' when the target_format is PCM_SIGNED 8000.0 Hz, 16 bit, mono, 2 bytes...
I am using Adobe Flash Media Live Encoder to stream live video to a video streaming server. The webcam is in our office pointed out the window. Thankfully, Flash Media Live Encoder has a checkbox to un-include audio. I am wondering how I can push a recorded message to the audio ( or music ). Is there any way I can play a recording an...
I'm planning on building an application where audio media is going to be streamed to the mobile phone for the user to listen. The targets are smartphones: iPhone/Blackberry/Android/(J2ME ?). I see that streaming on iPhone has to be done with HTTP Live streaming, but I don't see it supported by other platforms. Should I broadcast the s...
Is there a way to stream audio and video over the internet using Python Web Programming and not Flash? ...
Is it possible to play a .m3u file streamed from a remote location through Flash Player in a browser? I have a player that loads and plays .mp3 files but also want to be able to play .m3u files. I have looked at the as3plsreader on google code but I think this is only for AIR and desktop files. anyone tried this or know where I should ...
We offer a streaming player for a number of our clients, who are responsible for their providing us with their own audio streams. We have written a very simple flash player that can play all of the streams that we support (icecast/shoutcast/live365/mp3 over http/etc). Unfortunately, we have found that when listening, our player sometime...
Hello, hope you can help. I am recording audio from a microphone and streaming it live across a network. The quality of the samples is 11025hz, 8 bit, mono. Although there is a small delay (1 second), it works great. What I need help with is I am trying to now implement noise reduction and compression, to make the audio quieter and use...
Hi All, We are storing sound from mic to pc via sound forge. We would like to broadcast the sound which comes from the mic to the pc as live streaming audio. Basically a person speaks in a mic, we like to give it as live stream audio. The web-site is hosted on yahoo server. Can you please let me know in what are the ways we can achi...
I've been able to implement HTTP pseudostreaming for MP4/F4V files in Flash through an Apache module and by following varous tutorials, such as http://www.code-shop.com/2007/10/18/h264-pseudo-streaming http://flowplayer.org/plugins/streaming/pseudostreaming.html This is fine. I now want to do the same with a custom-built Flash audio pl...
Hi, I'm confuse with the RTP timestamp. With a PCMU file 8000hz 20ms it is 160Bytes. But, how can I calculate it for a L16Mono file with 44100Hz and also 20ms. VLC says the timestamp is 2205 and the packet size is 4410. Can anybody help me with this calculation? All regards St. ...
I need to be able to play MP3/AAC audio with a custom-built Flash player, embedded in a web page using my standard HTTP server. The loadSound() method seems to work fine for this, but I need to be able to skip the MP3 to unbuffered regions of the audio timeline and to start it streaming/downloading from there. Anyone know how I can acc...
Hi , I wrote simple audio playback application on c++ via latest DirectX SDK. On PC it working fine (Windows xp, Vista, Windows 7), but when I execute application on laptop it produce only silence. What differnce between audio on PC and laptop? ...
I am trying to stream a MP3 over the LAN using VLC Player(1.0.5.0), RTP multicast. On observation with wireshark in the receiving side,it interprets all the below fields: padding, extension, contributing source identifier count as 0, which means there are no additional byte after the first 12 bytes of fixed RTP header. But I could see ...
Hello, Can we stream live audio which is in WAV format from iPhone to server? ...
Hi there, The application that I am trying to develop for andriod, records frames at 48Khz (PCM 16bits & mono) and sends them to the network. Also, there is an incoming stream of audio at 8Khz. So, I receive 8Khz sampled frames and play them (my AudioTrack object is set to 8Khz), but when playing them, everything works but the latency i...
I am hoping to make an example using the developer build of chrome and being able to use subsonic to stream a binary audio file. So far I have not had any luck though. Granted my next option will be try to load in the audio files into windowStorage and toss some magic dust on them. Does anybody know a way to stream a audio file to the ...
I need some advice on what is the best practice to build an online audio library with ASP.Net + C# + MSSQL. Some key requirements are as follows: The audio files are stored in database in binary type On the web UI, user can click on an audio icon to play the audio. The audio will be very short, a few secs max, so there is no need to pr...
I need to play an RTSP audio stream on an Android phone, linked to from a web-page. What's the simplest way to achieve this? I hope this will be very simple but unfortunately I don't have a phone on which to test. ...
Hi, I have written a voice streaming application in iPhone using AudioQue. At the audio recording starts I initiated the network connection and pass the instance of NSAudioOutStream to AudioInputCallback using inUserData reference. void AudioInputCallback( void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer, ...
I'm planning to stream my usb tv tuner via specific protocol (ex: RTMP) to the media server (ex: Red5) that possible to control its channel from client (ex: flash application). Is there any ideas how do I get it working out? another protocol would be okay (RTMP, RTP, RTSP, or maybe using TCP & UDP socket programming). thank you. ...