--- /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);
+ }
+
+}
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