How can I get packets data from AudioStream Android?

How can I get packets data from AudioStream Android?

By : Fahad Malik
Date : November 21 2020, 11:01 PM
it should still fix some issue From what I believe you are trying to do, you may want to consider looking into Visualizer. https://developer.android.com/reference/android/media/audiofx/Visualizer.html You can also see a sample here
code :

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Streaming audio from IP camera to android (audiostream.cgi)

Streaming audio from IP camera to android (audiostream.cgi)

By : user2457562
Date : March 29 2020, 07:55 AM
This might help you I have a IP camera. Which is able to stream sound according to api: , I've found solution for this
code :
    val socket = Socket(InetAddress.getByName(ip), port)
        val out = PrintWriter(socket.getOutputStream(), true)
        out.println("GET /audiostream.cgi?&loginuse=$login&loginpas=$pass&streamid=16 HTTP/0.9")
        out.println("Connection: Close")
        val inputStream = socket.getInputStream()
Android Chat VoIP with PTT without SIP - multicast, AudioStream, Audiogroup not working

Android Chat VoIP with PTT without SIP - multicast, AudioStream, Audiogroup not working

By : 也许你面容憔悴
Date : March 29 2020, 07:55 AM
will be helpful for those in need For who will have the same problem, I'll share what I discovered. At "http://developer.android.com/reference/android/net/rtp/AudioStream.html" I found that " the local port is assigned automatically to conform with RFC 3550"; it means that after the port is assigned in one android device, it must be sent to the the other device in some way (I did it manually to test). Then I tried firstly with an unicast IP address (using then the IP address of the device i.e. and it worked. When I simply exchanged the unicast address with a multicast address it didn't work. So my conclusion is that android.net.RTP doesn't work with multicast. I'll go to study JMF as suggested bt jaybers (thank you!!!) and in case of success I'll share the resul. I'm wondering why in official android site abou the package "android.net.rtp" is written "Provides APIs for RTP (Real-time Transport Protocol), allowing applications to manage on-demand or interactive data streaming. In particular, apps that provide VOIP, push-to-talk, conferencing, and audio streaming can use these APIs to initiate sessions and transmit or receive data streams over any available network." How is possible to do push to talk and conferencing without multicast? Anybody know how is possible to ask question to google about it? I had a look to support but I didn't find anything useful...
Android.net.rtp - AudioStream.join() - java.lang.NullPointerException

Android.net.rtp - AudioStream.join() - java.lang.NullPointerException

By : user3808544
Date : March 29 2020, 07:55 AM
With these it helps First you have to call myAudioStream.associate (InetAddress remoteAddress, int remotePort) before calling myAudioStream.join(myAudioGroup);
How to detect Android audiostream underflow/overflow?

How to detect Android audiostream underflow/overflow?

By : ImHungers
Date : March 29 2020, 07:55 AM
may help you . You might try using the MediaRecorder class instead of AudioRecord, which offers a few extra methods, including the ability to register an error callback method. I'm not sure if all versions of Android will call that in case of an underrun, though. If you get callbacks here, that would probably be the "cleaner" solution.
The other solution is that you can measure the time taken when polling for audio and determine if an underrun has occurred. Basically, you'd store the time in a local variable before calling read() and then again afterwards:
How to read incoming TCP data packets by packets in C# (i.e., without concatenation of packets)?

How to read incoming TCP data packets by packets in C# (i.e., without concatenation of packets)?

By : Amsalo Lagas
Date : March 29 2020, 07:55 AM
it should still fix some issue Unlike UDP, TCP is logically a stream API, not a packet API. While at some low levels it may be possible to access the raw packets, virtually all consumer APIs expose it as a stream, with no guarantee that receives will match sends in terms of the original packet layout. Instead, what is guaranteed is: same bytes, same order. This means that you need to implement some kind of "framing" protocol. For a text-based protocol this often means simply: lines - i.e. add a CR/LF/CRLF after each complete message. For binary protocols, this usually means some kind of length prefix before each complete message.
Note that ReadToEnd() is only usable if you expect all the data to come at once (instead of via a x/y/x/y/x/y conversation between the two endpoints); although personally I'd recommed avoiding StreamReader entirely when using sockets, and just use the raw socket API or stream API. It is easy for code like ReadToEnd to block forever. If this is a client that only expects a single socket, that might be OK, but for a server (with lots of sockets), it is terrible: a malicious client can lock a thread forever by simply sending "Hi" and not disconnecting. This will never satisfy ReadToEnd.
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