src/org/sipdroid/media/RtpStreamSender.java
changeset 839 0e5b95573614
parent 836 2f2f5e24ac6a
child 1005 a2cad81f348b
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/src/org/sipdroid/media/RtpStreamSender.java	Sat Dec 25 17:01:00 2010 +0100
@@ -0,0 +1,376 @@
+/*
+ * Copyright (C) 2009 The Sipdroid Open Source Project
+ * Copyright (C) 2005 Luca Veltri - University of Parma - Italy
+ * 
+ * This file is part of Sipdroid (http://www.sipdroid.org)
+ * 
+ * Sipdroid is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ * 
+ * This source code is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ * 
+ * You should have received a copy of the GNU General Public License
+ * along with this source code; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+package org.sipdroid.media;
+
+import java.io.IOException;
+import java.io.InputStream;
+import java.net.InetAddress;
+import java.util.Random;
+
+
+import org.sipdroid.pjlib.Codec;
+
+import org.sipdroid.media.RtpStreamReceiver;
+import org.sipdroid.net.RtpPacket;
+import org.sipdroid.net.RtpSocket;
+import org.sipdroid.net.SipdroidSocket;
+
+import com.beem.project.beem.ui.Call;
+import com.beem.project.beem.utils.BeemConnectivity;
+
+import android.content.Context;
+import android.media.AudioFormat;
+import android.media.AudioManager;
+import android.media.AudioRecord;
+import android.media.MediaRecorder;
+import android.preference.PreferenceManager;
+import android.provider.Settings;
+import android.telephony.TelephonyManager;
+
+/**
+ * RtpStreamSender is a generic stream sender. It takes an InputStream and sends
+ * it through RTP.
+ */
+public class RtpStreamSender extends Thread {
+    /** Whether working in debug mode. */
+    public static boolean DEBUG = true;
+
+    /** The RtpSocket */
+    RtpSocket rtp_socket = null;
+
+    /** Payload type */
+    int p_type;
+
+    /** Number of frame per second */
+    long frame_rate;
+
+    /** Number of bytes per frame */
+    int frame_size;
+
+    /**
+     * Whether it works synchronously with a local clock, or it it acts as slave
+     * of the InputStream
+     */
+    boolean do_sync = true;
+
+    /**
+     * Synchronization correction value, in milliseconds. It accellarates the
+     * sending rate respect to the nominal value, in order to compensate program
+     * latencies.
+     */
+    int sync_adj = 0;
+
+    /** Whether it is running */
+    boolean running = false;
+    boolean muted = false;
+
+    /**
+     * Constructs a RtpStreamSender.
+     * 
+     * @param input_stream
+     *            the stream to be sent
+     * @param do_sync
+     *            whether time synchronization must be performed by the
+     *            RtpStreamSender, or it is performed by the InputStream (e.g.
+     *            the system audio input)
+     * @param payload_type
+     *            the payload type
+     * @param frame_rate
+     *            the frame rate, i.e. the number of frames that should be sent
+     *            per second; it is used to calculate the nominal packet time
+     *            and,in case of do_sync==true, the next departure time
+     * @param frame_size
+     *            the size of the payload
+     * @param src_socket
+     *            the socket used to send the RTP packet
+     * @param dest_addr
+     *            the destination address
+     * @param dest_port
+     *            the destination port
+     */
+    public RtpStreamSender(boolean do_sync,
+	int payload_type, long frame_rate, int frame_size,
+	SipdroidSocket src_socket, String dest_addr, int dest_port) {
+	init(do_sync, payload_type, frame_rate, frame_size,
+	    src_socket, dest_addr, dest_port);
+    }
+
+    /** Inits the RtpStreamSender */
+    private void init(boolean do_sync,
+	int payload_type, long frame_rate, int frame_size,
+	SipdroidSocket src_socket, String dest_addr,
+	int dest_port) {
+	this.p_type = payload_type;
+	this.frame_rate = frame_rate;
+	this.frame_size = frame_size;
+	this.do_sync = do_sync;
+	try {
+	    rtp_socket = new RtpSocket(src_socket, InetAddress
+		.getByName(dest_addr), dest_port);
+	} catch (Exception e) {
+	    e.printStackTrace();
+	}
+    }
+
+    /** Sets the synchronization adjustment time (in milliseconds). */
+    public void setSyncAdj(int millisecs) {
+	sync_adj = millisecs;
+    }
+
+    /** Whether is running */
+    public boolean isRunning() {
+	return running;
+    }
+
+    public boolean mute() {
+	return muted = !muted;
+    }
+
+    public static int delay = 0;
+
+    /** Stops running */
+    public void halt() {
+	running = false;
+    }
+
+    Random random;
+    double smin = 200,s;
+    int nearend;
+
+    void calc(short[] lin,int off,int len) {
+	int i,j;
+	double sm = 30000,r;
+
+	for (i = 0; i < len; i += 5) {
+	    j = lin[i+off];
+	    s = 0.03*Math.abs(j) + 0.97*s;
+	    if (s < sm) sm = s;
+	    if (s > smin) nearend = 3000/5;
+	    else if (nearend > 0) nearend--;
+	}
+	for (i = 0; i < len; i++) {
+	    j = lin[i+off];
+	    if (j > 6550)
+		lin[i+off] = 6550*5;
+	    else if (j < -6550)
+		lin[i+off] = -6550*5;
+	    else
+		lin[i+off] = (short)(j*5);
+	}
+	r = (double)len/100000;
+	smin = sm*r + smin*(1-r);
+    }
+
+    void calc1(short[] lin,int off,int len) {
+	int i,j;
+
+	for (i = 0; i < len; i++) {
+	    j = lin[i+off];
+	    lin[i+off] = (short)(j>>1);
+	}
+    }
+
+    void calc5(short[] lin,int off,int len) {
+	int i,j;
+
+	for (i = 0; i < len; i++) {
+	    j = lin[i+off];
+	    if (j > 16350)
+		lin[i+off] = 16350<<1;
+	    else if (j < -16350)
+		lin[i+off] = -16350<<1;
+	    else
+		lin[i+off] = (short)(j<<1);
+	}
+    }
+
+    void calc10(short[] lin,int off,int len) {
+	int i,j;
+
+	for (i = 0; i < len; i++) {
+	    j = lin[i+off];
+	    if (j > 8150)
+		lin[i+off] = 8150<<2;
+	    else if (j < -8150)
+		lin[i+off] = -8150<<2;
+	    else
+		lin[i+off] = (short)(j<<2);
+	}
+    }
+
+    void noise(short[] lin,int off,int len,double power) {
+	int i,r = (int)(power*2);
+	short ran;
+
+	if (r == 0) r = 1;
+	for (i = 0; i < len; i += 4) {
+	    ran = (short)(random.nextInt(r*2)-r);
+	    lin[i+off] = ran;
+	    lin[i+off+1] = ran;
+	    lin[i+off+2] = ran;
+	    lin[i+off+3] = ran;
+	}
+    }
+    public static float getMicGain() {
+	if (Call.headset > 0)
+	    return Float.valueOf(PreferenceManager.getDefaultSharedPreferences(Call.mContext).getString("hmicgain", "1.0"));
+	return Float.valueOf(PreferenceManager.getDefaultSharedPreferences(Call.mContext).getString("micgain", "0.25"));
+    }
+
+    /** Runs it in a new Thread. */
+    public void run() {
+	if (rtp_socket == null)
+	    return;
+	byte[] buffer = new byte[frame_size + 12];
+	RtpPacket rtp_packet = new RtpPacket(buffer, 0);
+	rtp_packet.setPayloadType(p_type);
+	int seqn = 0;
+	long time = 0;
+	double p = 0;
+	TelephonyManager tm = (TelephonyManager) Call.mContext.getSystemService(Context.TELEPHONY_SERVICE);
+	boolean improve = PreferenceManager.getDefaultSharedPreferences(Call.mContext).getBoolean("improve",false);
+	boolean useGSM = !PreferenceManager.getDefaultSharedPreferences(Call.mContext).getString("compression","edge").equals("never");
+	int micgain = (int)(getMicGain()*10);
+	running = true;
+
+	if (DEBUG)
+	    println("Reading blocks of " + buffer.length + " bytes");
+
+	android.os.Process.setThreadPriority(android.os.Process.THREAD_PRIORITY_URGENT_AUDIO);
+	AudioRecord record = new AudioRecord(MediaRecorder.AudioSource.MIC, 8000, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, 
+	    AudioRecord.getMinBufferSize(8000, 
+		AudioFormat.CHANNEL_CONFIGURATION_MONO, 
+		AudioFormat.ENCODING_PCM_16BIT)*3/2);
+	short[] lin = new short[frame_size*11];
+	int num,ring = 0;
+	random = new Random();
+	InputStream alerting = null;
+	try {
+	    alerting = Call.mContext.getAssets().open("alerting");
+	} catch (IOException e2) {
+	    e2.printStackTrace();
+	}
+	switch (p_type) {
+	    case 3:
+		Codec.init();
+		break;
+	    case 0:
+	    case 8:
+		G711.init();
+		break;
+	}
+	record.startRecording();
+	while (running) {
+	    if (muted || Call.call_state == Call.UA_STATE_HOLD) {
+		record.stop();
+		while (running && (muted || Call.call_state == Call.UA_STATE_HOLD)) {
+		    try {
+			sleep(1000);
+		    } catch (InterruptedException e1) {
+			e1.printStackTrace();
+		    }
+		}
+		record.startRecording();
+	    }
+	    num = record.read(lin,(ring+delay)%(frame_size*11),frame_size);
+
+	    if (RtpStreamReceiver.speakermode == AudioManager.MODE_NORMAL) {
+		calc(lin,(ring+delay)%(frame_size*11),num);
+		if (RtpStreamReceiver.nearend != 0)
+		    noise(lin,(ring+delay)%(frame_size*11),num,p);
+		else if (RtpStreamReceiver.nearend == 0)
+		    p = 0.9*p + 0.1*s;
+	    } else switch (micgain) {
+		case 1:
+		    calc1(lin,(ring+delay)%(frame_size*11),num);
+		    break;
+		case 5:
+		    calc5(lin,(ring+delay)%(frame_size*11),num);
+		    break;
+		case 10:
+		    calc10(lin,(ring+delay)%(frame_size*11),num);
+		    break;
+	    }
+	    if (Call.call_state != Call.UA_STATE_INCALL && alerting != null) {
+		try {
+		    if (alerting.available() < num)
+			alerting.reset();
+		    alerting.read(buffer,12,num);
+		} catch (IOException e) {
+		    e.printStackTrace();
+		}
+		switch (p_type) {// have to add ulaw case?
+		    case 3:
+			G711.alaw2linear(buffer, lin, num);
+			num = Codec.encode(lin, 0, buffer, num);
+			break;
+		    case 0:
+			G711.alaw2linear(buffer, lin, num);
+			G711.linear2ulaw(lin, 0, buffer, num);
+			break;
+		}
+	    } else {
+		switch (p_type) {
+		    case 3:
+			num = Codec.encode(lin, ring%(frame_size*11), buffer, num);
+			break;
+		    case 0:
+			G711.linear2ulaw(lin, ring%(frame_size*11), buffer, num);
+			break;
+		    case 8:
+			G711.linear2alaw(lin, ring%(frame_size*11), buffer, num);
+			break;
+		}
+	    }
+	    ring += frame_size;
+	    rtp_packet.setSequenceNumber(seqn++);
+	    rtp_packet.setTimestamp(time);
+	    rtp_packet.setPayloadLength(num);
+	    try {
+		rtp_socket.send(rtp_packet);
+	    } catch (IOException e) {
+		e.printStackTrace();
+	    }
+	    time += frame_size;
+	    /*if (useGSM && p_type == 8 && !BeemConnectivity.isWifi(Call.mContext) && tm.getNetworkType() == TelephonyManager.NETWORK_TYPE_EDGE) {
+		rtp_packet.setPayloadType(p_type = 3);
+		if (frame_size == 1024) {
+		    frame_size = 960;
+		    ring = 0;
+		}
+	    }*/
+	}
+	record.stop();
+
+	rtp_socket.close();
+	rtp_socket = null;
+
+	if (DEBUG)
+	    println("rtp sender terminated");
+    }
+
+    /** Debug output */
+    private static void println(String str) {
+	System.out.println("RtpStreamSender: " + str);
+    }
+
+}
\ No newline at end of file