All geek questions in one place

Using android.net.rtp

I am trying to transfer rtp audio packets to my tablet using android.net.rtp api. Before receiving the packages, I want to check the version of android.net.rtp.AudioGroup

  • I am creating an AudioStream in MODE SEND_ONLY.
  • Create an AudioGroup in MODE_ECHO_SUPPRESSION
  • The audio stream then joins the Audiogroup.

Shouldn't I get an AudioGroup from a microphone, play a speaker? I am not interested in sending packages. I only want to test the recording from the microphone and play through the speaker functionality in the audio group.

In MANIFEST, I have permissions set to use Internet, Record_Audio, modify_audio_settings (for mode_in_communication)

The code is as follows. rtp.stream package;

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; } }

When I run this on my tablet version of Android 3.2 , I don’t hear at all. I see an error message in logcat telling me that AudioGroup cannot read AudioRecord (mic). Right before that, I see a message from AudioPolicyManager saying that there is already an initial input. But there is currently no other application using a microphone.

Does anyone see something like this before? Any thoughts at all would be appreciated.

The following is an excerpt from the magazine.

  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

Any pointers at all would be appreciated.

+4
source share
2 answers

Have you reordered byte[] ? It should be: new byte[] {b1, b2, b3, b4} right? Hope this helps. I am also trying to make an rtp client for android.

 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 }); }
+1
source

I think you should install a speaker!

Perhaps you can use the following method:

 audioManager.setSpeakerphoneOn(true);
+1
source

Source: https://habr.com/ru/post/1401696/

More articles:


All Articles