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Ant Media Server supports RTMP, RTSP, MP4, HLS, WebRTC, Adaptive Streaming, etc.
There is a newer version: 2.11.3
/*
 *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

package org.webrtc.audio;

/**
 * This class contains static functions to query sample rate and input/output audio buffer sizes.
 */
class WebRtcAudioManager {
  private static final String TAG = "WebRtcAudioManagerExternal";

  //TODO: We may use different sample rate?
  private static final int DEFAULT_SAMPLE_RATE_HZ = 16000;

  // Default audio data format is PCM 16 bit per sample.
  // Guaranteed to be supported by all devices.
  private static final int BITS_PER_SAMPLE = 16;

  private static final int DEFAULT_FRAME_PER_BUFFER = 256;

  //@CalledByNative
  static Object getAudioManager(Object context) {
    //return (AudioManager) context.getSystemService(Context.AUDIO_SERVICE);
    return null;
  }

  //@CalledByNative
  static int getOutputBufferSize(
      Object context, Object audioManager, int sampleRate, int numberOfOutputChannels) {
////    return isLowLatencyOutputSupported(context)
//        ? getLowLatencyFramesPerBuffer(audioManager)
//        : getMinOutputFrameSize(sampleRate, numberOfOutputChannels);
        
    return DEFAULT_FRAME_PER_BUFFER;
  }

  //@CalledByNative
  static int getInputBufferSize(
		  Object context, Object audioManager, int sampleRate, int numberOfInputChannels) {
//    return isLowLatencyInputSupported(context)
//        ? getLowLatencyFramesPerBuffer(audioManager)
//        : getMinInputFrameSize(sampleRate, numberOfInputChannels);
	  return DEFAULT_FRAME_PER_BUFFER;
  }

//  private static boolean isLowLatencyOutputSupported(Context context) {
//    return context.getPackageManager().hasSystemFeature(PackageManager.FEATURE_AUDIO_LOW_LATENCY);
//  }

//  private static boolean isLowLatencyInputSupported(Context context) {
//    // TODO(henrika): investigate if some sort of device list is needed here
//    // as well. The NDK doc states that: "As of API level 21, lower latency
//    // audio input is supported on select devices. To take advantage of this
//    // feature, first confirm that lower latency output is available".
//    return Build.VERSION.SDK_INT >= 21 && isLowLatencyOutputSupported(context);
//  }

  /**
   * Returns the native input/output sample rate for this device's output stream.
   */
  //@CalledByNative
  static int getSampleRate(Object audioManager) {
    // Override this if we're running on an old emulator image which only
    // supports 8 kHz and doesn't support PROPERTY_OUTPUT_SAMPLE_RATE.
//    if (WebRtcAudioUtils.runningOnEmulator()) {
//      Logging.d(TAG, "Running emulator, overriding sample rate to 8 kHz.");
//      return 8000;
//    }
//    // Deliver best possible estimate based on default Android AudioManager APIs.
//    final int sampleRateHz = getSampleRateForApiLevel(audioManager);
//    Logging.d(TAG, "Sample rate is set to " + sampleRateHz + " Hz");
//    return sampleRateHz;
    return DEFAULT_SAMPLE_RATE_HZ;
  }

//  private static int getSampleRateForApiLevel(AudioManager audioManager) {
//    if (Build.VERSION.SDK_INT < 17) {
//      return DEFAULT_SAMPLE_RATE_HZ;
//    }
//    String sampleRateString = audioManager.getProperty(AudioManager.PROPERTY_OUTPUT_SAMPLE_RATE);
//    return (sampleRateString == null) ? DEFAULT_SAMPLE_RATE_HZ : Integer.parseInt(sampleRateString);
//  }
//
//  // Returns the native output buffer size for low-latency output streams.
//  private static int getLowLatencyFramesPerBuffer(AudioManager audioManager) {
//    if (Build.VERSION.SDK_INT < 17) {
//      return DEFAULT_FRAME_PER_BUFFER;
//    }
//    String framesPerBuffer =
//        audioManager.getProperty(AudioManager.PROPERTY_OUTPUT_FRAMES_PER_BUFFER);
//    return framesPerBuffer == null ? DEFAULT_FRAME_PER_BUFFER : Integer.parseInt(framesPerBuffer);
//  }
//
//  // Returns the minimum output buffer size for Java based audio (AudioTrack).
//  // This size can also be used for OpenSL ES implementations on devices that
//  // lacks support of low-latency output.
//  private static int getMinOutputFrameSize(int sampleRateInHz, int numChannels) {
//    final int bytesPerFrame = numChannels * (BITS_PER_SAMPLE / 8);
//    final int channelConfig =
//        (numChannels == 1 ? AudioFormat.CHANNEL_OUT_MONO : AudioFormat.CHANNEL_OUT_STEREO);
//    return AudioTrack.getMinBufferSize(
//               sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT)
//        / bytesPerFrame;
//  }
//
//  // Returns the minimum input buffer size for Java based audio (AudioRecord).
//  // This size can calso be used for OpenSL ES implementations on devices that
//  // lacks support of low-latency input.
//  private static int getMinInputFrameSize(int sampleRateInHz, int numChannels) {
//    final int bytesPerFrame = numChannels * (BITS_PER_SAMPLE / 8);
//    final int channelConfig =
//        (numChannels == 1 ? AudioFormat.CHANNEL_IN_MONO : AudioFormat.CHANNEL_IN_STEREO);
//    return AudioRecord.getMinBufferSize(
//               sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT)
//        / bytesPerFrame;
//  }
}