audio-streaming

The documentation for both of these methods are both very generic wherever I look. I would like to know what exactly I'm looking at with the returned arrays I'm getting from each method. For getByteTimeDomainData, what time period is covered with each pass? I believe most oscopes cover a 32 millisecond span for each pass. Is that what is covered here as well? For the actual element values themselves, the range seems to be 0 - 255. Is this equivalent to -1 - +1 volts? For getByteFrequencyData the frequencies covered is based on the sampling rate, so each index is an actual frequency, but what

Been banging my head against this problem all morning. I have setup a connection to a datasource which returns audio data (It is a recording device, so there is no set length on the data. the data just streams in. Like, if you would open a stream to a radio) and I have managed to receive all the packets of data in my code. Now I just need to play it. I want to play the data that is coming in, so I do not want to queue a few minutes or anything, I want to use the data I am recieving at that exact moment and play it. Now I been searching all morning finding different examples but none were

Currently I'm using NumPy to generate the WAV file from a NumPy array. I wonder if it's possible to play the NumPy array in realtime before it's actually written to the hard drive. All examples I found using PyAudio rely on writing the NumPy array to a WAV file first, but I'd like to have a preview function that just spits out the NumPy array to the audio output. Should be cross-platform, too. I'm using Python 3 (Anaconda distribution). This has worked! Thanks for help! def generate_sample(self, ob, preview): print("* Generating sample...") tone_out = array(ob, dtype=int16) if preview: print("

When playing a live stream using the HTTP Live Streaming method, is it possible read the current metadata (eg. Title and Artist)? This is for an iPhone radio app. m8labs Not sure that this question is still actual for its author, but may be it will help someone. After two days of pain I investigated that it's quite simple. Here is the code that works for me: AVPlayerItem* playerItem = [AVPlayerItem playerItemWithURL:[NSURL URLWithString:<here your http stream url>]]; [playerItem addObserver:self forKeyPath:@"timedMetadata" options:NSKeyValueObservingOptionNew context:nil]; AVPlayer* player = [

Description: I have following code for an Audio player. I can continue Audio playback from any duration by clicking on Progress Bar(between 0-to-mediaplayer.getDuration()). It is working perfectly for Audio playback. Problem in Audio Streaming: When I stream an Audio file from an internet server (say s3-bucket) it starts streaming correctly. But mediaPlayer.getDuration() and mediaPlayer.getCurrentPosition() return wrong values. At the beginning of streaming mediaPlayer.getCurrentPosition() returns 5 hours. Due to this I am not able to continue Audio streaming from a specified duration of

In Android, at the time of an incoming call, I want to receive it. Then, from my app, automatically play an audio file during a call and the other party should hear it. Is this possible? What you are talking about is not exactly possible with android. Android has no access to the in-call audio stream. Though i can give you a little bit idea about how to do it. first to intercept incoming call, you need to register a broadcast receiver, which is invoked whenever call is received public void onReceive(final Context context, Intent intent) { TelephonyManager telephonyManager = null;