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  • Video/Audio frame as input to OpenCore

    - by Vinay
    I am not able to use MediaPlayer/VideoView to make rtsp to work in Android. So I have created a client to interact with RTSP server, I have succeeded in doing this. I am able to get the video/audio frame from RTSP server (MySpace) in Android. Now I want to play the frames. I have searched OpenCore APIs to play the frames, but didn't get any APIs. My investigation: There is a class PlayerDriver.c It creates two sinks one audio and other video. handleSetVideoSurface handleSetAudioSink Two objects of type PVPlayerDataSinkPVMFNode are created. I suspect this class has a way to give the stream as input, but I am not getting the definition of this class. Can you suggest me is there any class I need to look into it?

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  • Dell Studio 17 with two headphone outs

    - by eych
    Got a Studio 17 with two headphone out jacks. It also has Soundblaster X-Fi Hi Def Audio Software enabled. Would it be possible to use one of the headphone outs as a monitor and the other as a master out with the Soundblaster sofware (for djing), or do you think the two jacks are just one output split into two?

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  • Automatic adaptive gain normalization

    - by Eduardo
    How may I normalize a (voice) audio mp3 or aac file with no loss, having the gain rised as much as possible with little distortion, so in a long conversation people that speak softer can have more gain for their voice and people that speak louder can have less gain?

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  • Make a radio-streaming PC pretend to be a mass-storage USB device

    - by monov
    I'm listening to a net radio on my PC I want the sound to go through my boombox cause it has nice speakers/amp The boombox has no "incoming" audio jack that just plays what comes over the wire However the boombox has a USB jack where you can put a thumbdrive with music. The question: How do I make the PC pretend to be a mass-storage device, and dynamically send all received audiodata to the boombox over a symmetric male USB cable? Failing that, at least tell me how to do it for local files (rather than streams). OS: Vista

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  • Python/Tkinter Audio Player

    - by Nicholas Quirk
    Hey everyone reading this, I've recently got into doing GUI development with Python. Tkinter seems like the easiest and most logical choice starting out. I did a little with wxPython but it was more sophisticated than what I needed. Anyway, I'm developing a media player. Right now it's a simple window with a button to load .wav files. The problem is I would like to implement a pause button now. But, when playing a audio file the GUI isn't accessible again (no buttons can be pushed) till the file is done playing. How can I make the GUI dynamic while an audio file is playing? I was thinking this maybe be because I'm using PyAudio, and their implementation doesn't allow this. Anyway, thanks for any advice before hand.

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  • Why does my 5.1 surround work in testing only?

    - by Mike Pateras
    I've got a 5.1 speaker setup. In both this SoundMax utility (I think it came with my motherboard), and in the Windows 7 sound test, all 5.1 speakers work properly, but that's the only time that I can get audio to come out of the rear speakers and the center channel. When playing games, video, music, etc., I only seem to get 2.1 speakers worth of sound, even though I've configured everything for 5.1 surround sound. How can I get my 5.1 surround sound working during actual use?

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  • Saving part of audio file (java)

    - by m159701m
    Hi evryone, While playing an audio file (.wav) I want , if I resort to ctrl+c , to stop the playback and save part of the audio file in a file called "file2.wav". Here's the thread I'd like to add to my code. Unfortunately it doesn't work at all. Thanks in advance class myThread extends Thread{ public void run(){ try { PipedOutputStream poStream = new PipedOutputStream(); PipedInputStream piStream = new PipedInputStream(); poStream.connect(piStream); File cutaudioFile = new File ("file2.wav"); AudioInputStream ais = new AudioInputStream(piStream, AudioFileFormat.Type.WAVE , cutaudioFile); poStream.write(ais,AudioFileFormat.Type.WAVE,cutaudioFile); }catch (Exception e){ e.printStackTrace(); } } // end run } // end myThread

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  • edit sound files

    - by doug
    I have an audio file with a very bad recording (recording was made in very noisy conditions, at a conference). My recorder was too far from the speakers. Do you have any idea about how to improve the sound quality? Do you know any good resource for that?

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  • ruby on rails: audio/mp3 content header download

    - by bandhunt
    How do you set the headers for downloads in ruby/rails? In php I'd set the header for an mp3 download like this: header("Content-Transfer-Encoding: binary"); header("Content-type: audio/mp3"); header("Content-Disposition: attachment; filename=\"$songname.mp3\""); header("Content-Length: " . $size); @readfile("http://example.com/12345.mp3"); Seems like there should be an easy should an easy solution. I did find this: response.headers['Content-type'] = 'Content-type: audio/mp3' But I'm not sure how/where the readfile would come into play and other headers. Thanks!

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  • Is it possible to prevent the win7 sleep state while using spotify?

    - by Skadlig
    Does anyone know if there is a way to prevent windows 7 to go to sleep while using Spotify? I have read the answers in this question but if it's possible I'd rather not resort to start a third party program like insomnia every time I want to listen to music. So are there a setting or a registry entry buried somewhere deep in windows that allows you to do this? Either for a group like "all audio" or for specific programs?

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  • Is there an equivalent to Airfoil available for Linux?

    - by Chris Adams
    Hi there, I just stumbled across this Airfoil on Mac OS X, which lets me 'throw' music from my laptop to any other linux machine, iPhone or other device hooked up to a better set of speakers than my laptop - here's the page below: http://www.rogueamoeba.com/airfoil/mac/ What tools would I use to recreate this functionality on a linux box - are there any nice GUI interfaces to something like pulse audio (which is what I imagine I'd use ) for doing this? Thanks

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  • Stereo mix causing inaudible input in XP?

    - by jsnlxndrlv
    I'm trying to record some of my work process in Camtasia using the stereo mix input. However, my headset microphone is quiet at the best of times, and no matter what combination of input and output volume settings I use, my voice is consistently inaudible. Know of a good fix for this? Should I be looking for a decent audio mixer--and if so, do you know of one that's as effective as it is free?

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  • Converting audio files(.3gp) to video with Album cover and uploading to YouTube

    - by Samuh
    I have an audio file in .3gp format on my Android device which I wish to upload to YouTube. I know that YouTube is a video upload site and that I need to convert this sound file to video. I just want an image to display all the time the audio is playing. Google tells me there are number of tools that can help me. But I want to do this via java code from my Android device. Please help. Thanks.

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  • Windows 7 sound control is not flexible.

    - by jon
    I would like to be able to listen to music on both of two audio output devices, but Windows 7 seems to only allow me to select one or the other as the Default device. When device A is the Default device, device B is muted; and vice versa. This seems to be stunningly inflexible. Since Windows 7 is unable to do this, can anyone recommend any add-on software that would control the hardware more flexibly and thoughtfully?

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  • Audio Player Royalty Free Music (dynamic audios )?

    - by Surya sasidhar
    hi, I am using Royalty Free Music player for playing the audio. ya it is playing perfect but i need to play it dynamically, i mean the audio will come from database how can i write the code for that. This is the royalty free music code..... var so = new SWFObject("playerSingle.swf", "mymovie", "192", "67", "7", "#FFFFFF"); so.addVariable("autoPlay", "yes"); so.addVariable("soundPath","song.mp3"); so.addVariable("overColor","#000044") so.addVariable("playerSkin","1") so.write("flashPlayer"); this above code is written in source code with in the script tag, then how can i write for dynamic audios please help me thanking you and this is the link for that site.. http://www.premiumbeat.com/flash_resources/free_flash_music_player/single_track_flash_mp3_player.php

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  • Use an iPhone as a bluetooth headset for a mac?

    - by Phillip Oldham
    Is there any way, such as an iPhone app, that will let me connect my iPhone to my iMac via bluetooth, so that the iMac pushes all audio through the iPhone? Specifically, what I'm looking to do is be able to watch movies on my iMac with the sound being played through my iPhone & in turn the ear buds.

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  • Comapring pitches with digital audio

    - by user2250569
    I work on application which will compare musical notes with digital audio. My first idea was analyzes wav file (or sound in real-time) with some polyphonic pitch algorithms and gets notes and chords from this file and subsequently compared with notes in dataset. I went through a lot of pages and it seems to be a lot of hard work because existing implementations and algorithms are mainly/only focus on monophonic sound. Now, I got the idea to do this in the opposite way. In dataset I have for example note: A4 or better example chord: A4 B4 H4. And my idea is make some wave (or whatever I don't know what) from this note or chord and then compared with piece of digital audio. Is this good idea? Is it better/harder solution? If yes can you recommend me how to do it?

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  • Problem with waveOutWrite and waveOutGetPosition deadlock

    - by MusiGenesis
    I'm working on an app that plays audio continuously using the waveOut... API from winmm.dll. The app uses "leapfrog" buffers, which are basically a bunch of arrays of samples that you dump into the audio queue. Windows plays them seamlessly in sequence, and as each buffer completes Windows calls a callback function. Inside this function, I load the next set of samples into the buffer, process them however, and then dump the buffer back into the audio queue. In this way, the audio plays indefinitely. For animation purposes, I'm trying to incorporate waveOutGetPosition into the application (since the "buffer done" callbacks are irregular enough to cause jerky animation). waveOutGetPosition returns the current position of playback, so it's hyper-precise. The problem is that in my application, making calls to waveOutGetPosition eventually causes the application to lock up - the sound stops and the call never returns. I've boiled things down to a simple app that demonstrates the problem. You can run the app here: http://www.musigenesis.com/SO/waveOut%20demo.exe If you just hear a tiny bit of piano over and over, it's working. It's just meant to demonstrate the problem. The source code for this project is here: http://www.musigenesis.com/SO/WaveOutDemo.zip The first button runs the app in leapfrog mode without making the calls to waveOutGetPosition. If you click this, the app will play forever without breaking (the X button will close it and shut it off). The second button starts the leapfrogger and also starts a forms timer that calls the waveOutGetPosition and displays the current position. Click this and the app will run for a short while and then lock up. On my laptop, it usually locks up in 15-30 seconds; at most it's taken a minute. I have no idea how to fix this, so any help or suggestions would be most welcome. I've found very few posts on this issue, but it seems that there is a potential deadlock, either from multiple calls to waveOutGetPosition or from calls to that and waveOutWrite that occur at the same time. It's possible that I'm calling this too frequently for the system to handle.

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  • i2s0: transmitter underrun (0)

    - by tbarbe
    were doing some audio stuff and I keep seeing this in the Organizer Console. Sun May 2 20:16:48 unknown kernel[0] : i2s0: transmitter underrun (0) Are these transmitter underruns bad? I think its just when were shutting down audio input...but could a few of these cause some issues later on?

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  • iPhone SDK: How to record voices with ambient noise supression?

    - by Harkonian
    Can anyone point me in the right direction on how I would minimize ambient noise while recording someone speaking using the iPhone SDK Core Audio? I'm guessing a band-pass filter that eliminates any frequencies above and below the human vocal range might work. I have no idea how I would implement band filters on audio in the SDK though. The optimum solution would be one that eliminates the noise from the stream before it is written to memory/disk. Some sample code would be appreciated.

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  • iPhone: CPU power to do DSP/Fourier transform/frequency domain?

    - by mahboudz
    I want to analyze MIC audio on an ongoing basis (not just a snipper or prerecorded sample), and display frequency graph and filter out certain aspects of the audio. Is the iPhone powerful enough for that? I suspect the answer is a yes, given the Google and iPhone voice recognition, Shazaam and other music recognition apps, and guitar tuner apps out there. However, I don't know what limitations I'll have to deal with. Anyone play around with this area?

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