--- /dev/null Thu Jan 01 00:00:00 1970 +0000
+++ b/src/org/sipdroid/media/RtpStreamSender.java Fri Nov 20 19:29:42 2009 +0100
@@ -0,0 +1,220 @@
+/*
+ * 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.InputStream;
+import java.net.DatagramPacket;
+import java.util.Random;
+
+import jlibrtp.RTPSession;
+import jlibrtp.RtpPkt;
+
+import org.sipdroid.media.codecs.Codec;
+
+import android.media.AudioFormat;
+import android.media.AudioManager;
+import android.media.AudioRecord;
+import android.media.MediaRecorder;
+
+/**
+ * RtpStreamSender is a generic stream sender. It takes an InputStream and sends
+ * it through RTP.
+ */
+public class RtpStreamSender extends Thread{
+
+ private static final boolean DEBUG = true;
+
+ /** The RtpSocket */
+ //private RtpSocket rtp_socket = null;
+ private RTPSession rtpSession = null;
+
+ /** Codec */
+ private Codec codec;
+
+ private int sampling_rate;
+
+ /** Number of bytes per frame */
+ private int frame_size;
+
+ private int codec_frame_size;
+
+ /**
+ * Whether it works synchronously with a local clock, or it it acts as slave
+ * of the InputStream
+ */
+ boolean do_sync = true;
+
+ int sync_adj = 0;
+
+ /** Whether it is running */
+ boolean running = false;
+ boolean muted = false;
+
+ private int codec_divider;
+
+ /**
+ * Constructs a RtpStreamSender.
+ *
+ */
+ public RtpStreamSender(Codec co, RTPSession rtpSession) {
+ init(co, rtpSession);
+ }
+
+ /** Inits the RtpStreamSender */
+ private void init(Codec co, RTPSession rtpSession) {
+ this.rtpSession = rtpSession;
+ codec = co;
+ sampling_rate = codec.getInfo().samplingRate;
+ codec_frame_size = codec.getInfo().codecFrameSize;
+ codec_divider = codec.getInfo().rtpSampleDivider;
+ frame_size = 160 * codec_divider;
+ rtpSession.payloadType(codec.getInfo().rtpPayloadCode);
+
+ this.do_sync = true;
+ }
+
+ /** 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 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 int m;
+
+ /** Runs it in a new Thread. */
+ public void run() {
+ if (rtpSession == null)
+ return;
+ byte[] buffer = new byte[codec_frame_size + 12];
+ DatagramPacket packet = new DatagramPacket(buffer, codec_frame_size + 12);
+ RtpPkt pkt = new RtpPkt();
+ pkt.setRawPkt(buffer);
+ pkt.setPayloadType(codec.getInfo().rtpPayloadCode);
+ int seqn = 0;
+ long time = 0;
+ double p = 0;
+ running = true;
+ m = 1;
+
+ if (DEBUG)
+ println("Reading blocks of " + buffer.length + " bytes");
+
+ android.os.Process.setThreadPriority(android.os.Process.THREAD_PRIORITY_URGENT_AUDIO);
+
+ Codec.Context codecCtx = codec.initEncoder();
+
+ AudioRecord record = new AudioRecord(MediaRecorder.AudioSource.MIC, sampling_rate, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT,
+ AudioRecord.getMinBufferSize(sampling_rate,
+ AudioFormat.CHANNEL_CONFIGURATION_MONO,
+ AudioFormat.ENCODING_PCM_16BIT)*2);
+ record.startRecording();
+ short[] lin = new short[frame_size*11];
+ int num,ring = 0;
+ random = new Random();
+ while (running) {
+ 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 (nearend == 0)
+ p = 0.9*p + 0.1*s;
+ }
+ codec.encode(codecCtx, lin, ring%(frame_size*11), frame_size, buffer, 12);
+ ring += frame_size;
+ rtpSession.sendData(packet, pkt);
+ if (m == 2) {
+ rtpSession.sendData(packet, pkt);
+ println("retransmit");
+ }
+ seqn++;
+ time += num;
+ }
+ record.stop();
+ rtpSession = null;
+ codec.cleanEncoder(codecCtx);
+ if (DEBUG)
+ println("rtp sender terminated");
+ }
+
+ /** Debug output */
+ private static void println(String str) {
+ android.util.Log.d("DEBUG","RtpStreamSender: " + str);
+ }
+}