BigEndian / LittleEndian and wave representation
Can someone tell me why big endian / little endian can affect the wave representation of an audio signal? ...
Can someone tell me why big endian / little endian can affect the wave representation of an audio signal? ...
I am building a site that stores a few mp3 audio files in mySQL. At the moment only 5, so efficiency is not an issue at all. CUrrently, I am still trying to make a cross browser/platform solution that will not require other file formats. I have a PHP script that retrieves the mp3 binary data and sends the appropriate header to the brow...
According to this Mozilla article on Ogg media, media works more seamlessly in the browser with an X-Content-Duration header, giving the length in seconds of the piece. Assuming that I have that length stored somewhere (certainly in a database, perhaps also in the filename itself (video-file-name.XXX.ogv, where XXX is the time in second...
I tried the Audio module at http://www.drupal.org/project/audio but I'm looking for alternatives if better ones exist. My problem with the Audio module is that the current release (and the past 5 releases) seem to have all been released as unsable. The second problem is that the player itself that plays the audio is not showing when ...
I read the following question and answer http://stackoverflow.com/questions/1995589/html5-audio-safari-live-broadcast-vs-not Still unclear how to resolve the issue in my situation: The audio files (mp3) I am working with are stored in mySQL and pulled into the browser by a PHP script that sends the Content-length and Content-type heade...
I need your help. I'd like to use the following code since I'm developing an audio tool (for .wav files only) whose main feature is displaying a signal waveform. Big Endian as well as Little endian are elements I can't help dealing with. Am I right in thinking that the following code tackles the problem in a way that: - if the audio file...
For the life of me I can figure out how to do a simple fast forward and rewind on the iPhone for an audio player using HTML5. All the Apple documentation says that the playbackRate should be used for this. I have watched the WWDC 2009 and 2010 sessions on this and downloaded all the example source code I can find but I still can't figure...
Hi, Amazingly, in bash, you can do $ curl http://mp3.streampower.be/radio1-high.mp3 > test.mp3 with an audiostream and then ^C out, and you will have a working mp3 file, leading me to believe I could have this automated in python, but I can't find how. If i just do file('python.mp3', 'w').write(urllib2.urlopen("http://mp3.streampo...
how to process an audio file using python ...
Trying to make a tiny app which plays an audio file, and I have designed 5 vertical progressbars for monitoring the left, middle and right channels of the audio file (the middle just being an average between the left and right channels) just for a fancy look. Can someone help me read the audio levels(decibles or whatever) of the audio fi...
Appears that java's sound API's work well for single streams, and even for setting the input from the microphone, but not for setting the master volume level in Vista/Windows 7. refs: http://stackoverflow.com/questions/1727511/java-sound-api-to-access-the-system-master-volume-control-in-vista-and-win-7 http://stackoverflow.com/questio...
Please bear with me as my understanding of audio codec is limited. I have this audio source from a IPCAM (through a htto//... CGI interface). I am trying to write several client programs to play this audio source on Windows, MAC, as well as Android phone. The audio is encoded in G711 (PCM ulaw). Do I need to decode the PCM audio data...
Hi there, I need to convert audio files to mp3 using ffmpeg. When i write the command as ffmpeg -i audio.ogg -acodec mp3 newfile.mp3, I get the error: FFmpeg version 0.5.2, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 ...
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 have a need to play some sporadic sound effects in the background of an iPhone app. From everything I've read and experienced with iOS 4, I can keep my app running in the background as long as I am running GPS by specifying "location" as a background mode. That actually works. But at times I want to play a sound effect...in other wo...
I'd like to take a set of images and a sound track and use that to form a basic video slideshow using gstreamer. There seems to be a lot of documentation and examples of basic gstreamer usage like playing a video or audio file, or even transcoding and the like. But I can't seem to find anything particularly useful for, I suppose, video ...
I'm unable to record audio: Having a problem with this. Error: (in Eclipse) The method setParameters(String) is undefined for the type MediaRecorder. This is the piece of code which causes the above error recorder.setParameters(String.format( "audio-param-sampling-rate=%d", 8000/*mProfile.mAudioSamplingRate*/)...
Hi, I appreciate what you do guys in this great site. I want to know if there are any way to play Sound list clips with HTML5. This is a audio code in HTML5: <audio controls="controls" autoplay="autoplay"> <source src="test1.mp3" type="audio/mpeg" /> </audio> it is work with me but what i want is how can i play many sound list exa...
I'm allowing the user to push buttons to play sounds via a SoundPool. Is it possible to record whatever the SoundPool is playing so that the user can record a sequence of sounds? ...
I've been hunting bugs for the past three days and I've kind of given up. I've plowed through all the samples in the OpenAL SDK and the Vorbis examples, but to no avail, so I hope someone can help me. The problem: I record audio using OpenAL, and for debug reasons i output it to C:/out.wav which i can then play with any audio player of...