android - 使用 android.net.rtp

Question

我正在尝试使用 android.net.rtp 在我的平板电脑上传输音频 rtp 数据包接口(interface)。在接收数据包之前，我想测试一下 android.net.rtp.AudioGroup

• 我正在以 MODE SEND_ONLY 模式创建 AudioStream。
• 在 MODE_ECHO_SUPPRESSION 中创建音频组
• 然后音频流加入音频组。

AudioGroup 不应该从麦克风接收，在扬声器上播放吗？我不关心它发送数据包。我只想通过音频组中的扬声器功能测试麦克风录音和播放。

在 MANIFEST 中，我确实设置了使用 Internet、Record_Audio、modify_audio_settings(用于 mode_in_communication)的权限

代码如下。 打包 rtp.stream;

    import android.app.Activity;
    import android.os.Bundle;
    import android.media.AudioManager;
    import android.net.rtp.*;

    import android.util.Log;

    import java.net.*;

    public class Rtpstream3_2 extends Activity {
        /** Called when the activity is first created. */
        @Override
        public void onCreate(Bundle savedInstanceState) {
            super.onCreate(savedInstanceState);
            setContentView(R.layout.main);
            // AudioManager
            AudioManager audioManager = (AudioManager) getSystemService(AUDIO_SERVICE);
            audioManager.setMode(AudioManager.MODE_IN_COMMUNICATION);

            // Initialize AudioStream and set codec
            AudioStream inRtpStream = null;
            try {
                inRtpStream = new AudioStream(createInet(127, 0, 0, 1));
            } catch (SocketException e) {
                Log.d("Quit", "Socket Error");
                System.exit(1);
            }
            inRtpStream.setMode(RtpStream.MODE_SEND_ONLY);
            inRtpStream.setCodec(AudioCodec.PCMU);
            inRtpStream.associate(createInet(10,2,0,165), 17222);

            // Initialize an AudioGroup and attach an AudioStream
            AudioGroup main_grp = new AudioGroup();
            main_grp.setMode(AudioGroup.MODE_ECHO_SUPPRESSION);
            inRtpStream.join(main_grp);
            Log.d("Log"," Group joined"+inRtpStream.getLocalPort());
        }

        private InetAddress createInet(int b1, int b2, int b3, int b4) {
            InetAddress addr = null;
            try {
                addr = InetAddress.getByAddress(new byte[] {(byte)b1, (byte)b2,
                                                            (byte)b4, (byte)b3
                                                            });
            } 
            catch (UnknownHostException e) {
                Log.d("Error", "Cannot create Inet address");
                System.exit(1);
            }
            return addr;
        }
    }

当我在运行 Android 3.2 版本的平板电脑上运行此程序时，我根本听不到自己的声音。我在 logcat 中看到一条错误消息，告诉我 AudioGroup 无法从 AudioRecord(mic) 中读取。就在这之前，我看到一条来自 AudioPolicyManager 的消息，它说输入已经开始。但目前没有其他应用程序使用麦克风。

有人以前见过类似的东西吗？任何想法都将不胜感激。

下面是日志的摘录。

    03-15 20:06:00.820: I/AudioService(286):  AudioFocus  requestAudioFocus() from AudioFocus_For_Phone_Ring_And_Calls
    03-15 20:06:00.820: D/AudioHardwareMot(6337): setMode(IN_COMMUNICATION)
    03-15 20:06:00.820: D/AudioHardwareMot(6337): Putting streams to standby
    03-15 20:06:00.820: D/AudioHardwareMot(6337): AudioStreamOutMot::standby called
    03-15 20:06:00.820: D/AudioHardwareMot(6337): Input 0x849f8 entering standby
    03-15 20:06:00.820: D/AcousticsModule(6337): Acoustics close stub called.
    03-15 20:06:00.820: D/AudioHardwareMot(6337): setMode(): mode is MODE_IN_COMMUNICATION. Calling enableecns
    03-15 20:06:00.820: D/AudioPostProcessor(6337): Reading ecns param file from /system/bin
    03-15 20:06:00.820: D/AudioPostProcessor(6337): enableEcns() called: found param file. mEcnsEnabled value =0
    03-15 20:06:00.820: D/AudioPostProcessor(6337): enableEcns(true)
    03-15 20:06:00.820: **D/AudioHardwareInterface(6337): setMode(IN_COMMUNICATION)**
    03-15 20:06:00.820: D/AudioHardwareMot(6337): setMode END
    03-15 20:06:00.820: D/AudioHardwareMot(6337): Capture session will read from ECNS thread
    03-15 20:06:00.820: D/AudioHardwareMot(6337): AudioStreamOutMot::setParameters() routing=8
    **03-15 20:06:00.820: D/AudioHardwareMot(6337): Entering doALSAInputRouting
    03-15 20:06:00.820: D/AudioHardwareMot(6337): Enter doALSAOutputRouting**
    03-15 20:06:00.828: D/AudioHardwareMot(6337): Output gain set
    03-15 20:06:00.828: D/AudioHardwareMot(6337): doRouting(): Calling mapaccy with accy 9, mNrec 1
    03-15 20:06:00.828: D/AudioHardwareMot(6337): doRouting devs: stereo 0, mono 0, input 0. Chose speaker None (gain 0xb) mic None (gain 0x1c1c)
    03-15 20:06:00.828: I/dalvikvm(7385): threadid=1: recursive native library load attempt (/system/lib/librtp_jni.so)
    **03-15 20:06:00.835: D/AudioGroup(7385): stream[38] is configured as PCMU 8kHz 20ms mode 1
    03-15 20:06:00.835: D/AudioGroup(7385): stream[42] is configured as RAW 8kHz 32ms mode 0
    03-15 20:06:00.835: D/AudioGroup(7385): stream[42] joins group[41]
    03-15 20:06:00.835: D/AudioGroup(7385): group[41] switches from mode 0 to 3
    03-15 20:06:00.835: D/AudioGroup(7385): stream[38] joins group[41]**
    03-15 20:06:00.835: D/Log(7385):  Group joined55442
    03-15 20:06:00.835: D/AudioHardwareMot(6337): Output latency, using cached value = 99
    03-15 20:06:00.835: D/AudioGroup(7385): reported frame count: output 789, input 800
    03-15 20:06:00.835: D/AudioGroup(7385): adjusted frame count: output 789, input 800
    03-15 20:06:00.835: D/AudioHardwareMot(6337): Output latency, using cached value = 99
    03-15 20:06:00.843: D/AudioHardwareMot(6337): Output latency, using cached value = 99
    03-15 20:06:00.843: D/AudioHardwareMot(6337): AudioHardwareMot::openInputStream enter
    03-15 20:06:00.843: D/AudioHardwareMot(6337): AudioStreamInMot::set(0xf9e8, 40000, 1, 10, 8000)
    03-15 20:06:00.843: D/Omap4ALSA(6337): open called for devices 00040000 in mode 3...
    03-15 20:06:00.851: I/gralloc(416): Unregistered ID: 10 handle: 0x92a7b0 size: 1280 x 800 fmt: 5 usage: 0x1300
    03-15 20:06:00.851: D/Omap4ALSA(6337): setCodecDefaultControls
    03-15 20:06:00.851: I/Omap4ALSA(6337): **Initialized ALSA CAPTURE device hw:0,1**
    03-15 20:06:00.851: D/AcousticsModule(6337): Acoustics set_params stub called with 0.
    03-15 20:06:00.851: D/AcousticsModule(6337): Acoustics close stub called.
    03-15 20:06:00.851: I/gralloc(416): Unregistered ID: 9 handle: 0x94f0e8 size: 1280 x 800 fmt: 5 usage: 0x1300
    03-15 20:06:00.851: D/AudioPostProcessor(6337): **Disabling beamformer due to unsupported sample rate**
    03-15 20:06:00.851: D/AudioHardwareMot(6337): Input bufSize from ALSA = 352
    03-15 20:06:00.851: D/AudioHardwareMot(6337): Output latency, using cached value = 99
    03-15 20:06:00.859: D/AudioGroup(7385): latency: output 197, input 100
    03-15 20:06:00.859: W/AudioPolicyManagerBase(6337): startInput() input 1559 failed: other input already started
    03-15 20:06:00.882: E/AudioGroup(7385): **cannot read from AudioRecord**
    03-15 20:06:00.902: W/AudioPolicyManagerBase(89): startInput() input 10711 failed: other input already started

如有任何指点，我们将不胜感激。

Answer 1

你是否交换了 byte[] 的顺序？它应该是:new byte[] {b1, b2, b3, b4} 对吧？我希望这有帮助。我也在尝试为 android 制作一个 rtp 客户端。

private InetAddress createInet(int b1, int b2, int b3, int b4) {
InetAddress addr = null;
try {
     addr = InetAddress.getByAddress(new byte[] {(byte)b1, (byte)b2, (byte)b4, (byte)b3
    });
}