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All in all WebRTC. A Complete Guide to enable Rich and High Quality of **Real-Time Voice Communication** on Android Platform. This repository involves a complete understanding, implementation and documentation related to WebRTC Audio Processing.

Java 0.75% Makefile 6.50% C++ 67.24% C 22.77% Python 1.08% Objective-C++ 0.85% Objective-C 0.12% Assembly 0.13% MATLAB 0.14% Shell 0.43%
webrtc audio-processing android android-ndk ndk-build voice-communication voice-quality

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android-audio-processing-using-webrtc's Issues

System.loadLibrary("soundtouch") cause fatal crash

System Information:
OS: Windows 10 64bit
IDE: Android Studio 3.6.2
JDK 1.8
NDK r21

Steps to Repro:
Downloaded git
On Sync Build I get following message in red color
17:00 Unsupported Modules Detected: Compilation is not supported for following modules: Android-Audio-Processing-Using-WebRTC. Unfortunately you can't have non-Gradle Java modules and Android-Gradle modules in one project.
On running the code, system crashes at runtime on calling System.loadLibrary("soundtouch") in JNISoundTouch class

Question - What am I doing something wrong here? Since I am a newbie to Android. How can I correct it.

If i want to change the SAMPLE_RATE from 16000 to 8000, how can i do?

If i want to change the SAMPLE_RATE from 16000 to 8000, how can i do?
I tried to change SAMPLE_RATE from 16000 to 8000, and change the code
private static final int SAMPLE_RATE = 16000;
private static final int BITS_PER_SAMPLE = 16;

   private static final int SAMPLE_RATE = 16000;

        soundtouch.setSampleRate(16000);
        soundtouch.setChannels(1);

to
private static final int SAMPLE_RATE = 8000;
private static final int BITS_PER_SAMPLE = 16;

   private static final int SAMPLE_RATE = 8000;

        soundtouch.setSampleRate(8000);
        soundtouch.setChannels(1);

then the voice is not right.

Which code should i change ?
would you please kindly help to give me a reply soon? Thanks in advance!

AudioProcessing.java line 145 may be some minor errors?

Thank you very much for the open source project, this helped me a lot。I analyze the source code,find that
_dataSocket = new DatagramSocket(Integer.parseInt(vm.getTargetPort()), localAddr);
maybe change to
_dataSocket = new DatagramSocket(PORT, localAddr);
Is that right?

Question about "MODE_IN_COMMUNICATION"

Hi @mail2chromium , there is a question:
If you set the MODE_IN_COMMUNICATION, you probably don't need the apm using webrtc.

AudioManager audioManager = (AudioManager) context.getSystemService(Context.AUDIO_SERVICE);
audioManager.setMode(AudioManager.MODE_IN_COMMUNICATION);

I found out if i set MODE_IN_COMMUNICATION, the voice is pretty, i didn't listen any echo.
if i don't set MODE_IN_COMMUNICATION, only use the apm using webrtc, the voice is not that good.
what is the diff?? btw, what the webrtc version compiled in this project?
would you please kindly help to give me a reply soon? Thanks in advance!

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