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1 /* |
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2 * Copyright (C) 2009 The Sipdroid Open Source Project |
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3 * Copyright (C) 2005 Luca Veltri - University of Parma - Italy |
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4 * |
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5 * This file is part of Sipdroid (http://www.sipdroid.org) |
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6 * |
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7 * Sipdroid is free software; you can redistribute it and/or modify |
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8 * it under the terms of the GNU General Public License as published by |
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9 * the Free Software Foundation; either version 3 of the License, or |
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10 * (at your option) any later version. |
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11 * |
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12 * This source code is distributed in the hope that it will be useful, |
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13 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
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15 * GNU General Public License for more details. |
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16 * |
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17 * You should have received a copy of the GNU General Public License |
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18 * along with this source code; if not, write to the Free Software |
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19 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA |
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20 */ |
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21 |
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22 package org.sipdroid.media; |
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23 |
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24 import java.io.InputStream; |
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25 import java.net.DatagramPacket; |
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26 import java.util.Random; |
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27 |
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28 import jlibrtp.RTPSession; |
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29 import jlibrtp.RtpPkt; |
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30 |
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31 import org.sipdroid.media.codecs.Codec; |
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32 |
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33 import android.media.AudioFormat; |
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34 import android.media.AudioManager; |
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35 import android.media.AudioRecord; |
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36 import android.media.MediaRecorder; |
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37 |
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38 /** |
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39 * RtpStreamSender is a generic stream sender. It takes an InputStream and sends |
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40 * it through RTP. |
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41 */ |
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42 public class RtpStreamSender extends Thread{ |
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43 |
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44 private static final boolean DEBUG = true; |
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45 |
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46 /** The RtpSocket */ |
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47 //private RtpSocket rtp_socket = null; |
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48 private RTPSession rtpSession = null; |
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49 |
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50 /** Codec */ |
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51 private Codec codec; |
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52 |
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53 private int sampling_rate; |
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54 |
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55 /** Number of bytes per frame */ |
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56 private int frame_size; |
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57 |
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58 private int codec_frame_size; |
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59 |
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60 /** |
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61 * Whether it works synchronously with a local clock, or it it acts as slave |
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62 * of the InputStream |
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63 */ |
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64 boolean do_sync = true; |
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65 |
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66 int sync_adj = 0; |
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67 |
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68 /** Whether it is running */ |
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69 boolean running = false; |
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70 boolean muted = false; |
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71 |
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72 private int codec_divider; |
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73 |
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74 /** |
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75 * Constructs a RtpStreamSender. |
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76 * |
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77 */ |
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78 public RtpStreamSender(Codec co, RTPSession rtpSession) { |
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79 init(co, rtpSession); |
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80 } |
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81 |
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82 /** Inits the RtpStreamSender */ |
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83 private void init(Codec co, RTPSession rtpSession) { |
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84 this.rtpSession = rtpSession; |
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85 codec = co; |
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86 sampling_rate = codec.getInfo().samplingRate; |
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87 codec_frame_size = codec.getInfo().codecFrameSize; |
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88 codec_divider = codec.getInfo().rtpSampleDivider; |
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89 frame_size = 160 * codec_divider; |
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90 rtpSession.payloadType(codec.getInfo().rtpPayloadCode); |
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91 |
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92 this.do_sync = true; |
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93 } |
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94 |
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95 /** Sets the synchronization adjustment time (in milliseconds). */ |
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96 public void setSyncAdj(int millisecs) { |
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97 sync_adj = millisecs; |
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98 } |
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99 |
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100 /** Whether is running */ |
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101 public boolean isRunning() { |
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102 return running; |
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103 } |
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104 |
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105 public boolean mute() { |
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106 return muted = !muted; |
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107 } |
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108 |
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109 public static int delay = 0; |
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110 |
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111 /** Stops running */ |
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112 public void halt() { |
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113 running = false; |
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114 } |
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115 |
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116 Random random; |
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117 double smin = 200,s; |
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118 int nearend; |
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119 |
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120 void calc(short[] lin,int off,int len) { |
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121 int i,j; |
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122 double sm = 30000,r; |
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123 |
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124 for (i = 0; i < len; i += 5) { |
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125 j = lin[i+off]; |
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126 s = 0.03*Math.abs(j) + 0.97*s; |
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127 if (s < sm) sm = s; |
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128 if (s > smin) nearend = 3000/5; |
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129 else if (nearend > 0) nearend--; |
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130 } |
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131 for (i = 0; i < len; i++) { |
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132 j = lin[i+off]; |
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133 if (j > 6550) |
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134 lin[i+off] = 6550*5; |
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135 else if (j < -6550) |
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136 lin[i+off] = -6550*5; |
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137 else |
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138 lin[i+off] = (short)(j*5); |
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139 } |
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140 r = (double)len/100000; |
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141 smin = sm*r + smin*(1-r); |
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142 } |
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143 |
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144 void noise(short[] lin,int off,int len,double power) { |
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145 int i,r = (int)(power*2); |
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146 short ran; |
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147 |
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148 if (r == 0) r = 1; |
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149 for (i = 0; i < len; i += 4) { |
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150 ran = (short)(random.nextInt(r*2)-r); |
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151 lin[i+off] = ran; |
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152 lin[i+off+1] = ran; |
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153 lin[i+off+2] = ran; |
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154 lin[i+off+3] = ran; |
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155 } |
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156 } |
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157 |
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158 public static int m; |
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159 |
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160 /** Runs it in a new Thread. */ |
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161 public void run() { |
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162 if (rtpSession == null) |
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163 return; |
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164 byte[] buffer = new byte[codec_frame_size + 12]; |
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165 DatagramPacket packet = new DatagramPacket(buffer, codec_frame_size + 12); |
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166 RtpPkt pkt = new RtpPkt(); |
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167 pkt.setRawPkt(buffer); |
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168 pkt.setPayloadType(codec.getInfo().rtpPayloadCode); |
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169 int seqn = 0; |
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170 long time = 0; |
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171 double p = 0; |
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172 running = true; |
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173 m = 1; |
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174 |
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175 if (DEBUG) |
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176 println("Reading blocks of " + buffer.length + " bytes"); |
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177 |
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178 android.os.Process.setThreadPriority(android.os.Process.THREAD_PRIORITY_URGENT_AUDIO); |
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179 |
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180 Codec.Context codecCtx = codec.initEncoder(); |
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181 |
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182 AudioRecord record = new AudioRecord(MediaRecorder.AudioSource.MIC, sampling_rate, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, |
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183 AudioRecord.getMinBufferSize(sampling_rate, |
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184 AudioFormat.CHANNEL_CONFIGURATION_MONO, |
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185 AudioFormat.ENCODING_PCM_16BIT)*2); |
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186 record.startRecording(); |
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187 short[] lin = new short[frame_size*11]; |
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188 int num,ring = 0; |
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189 random = new Random(); |
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190 while (running) { |
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191 num = record.read(lin,(ring+delay)%(frame_size*11),frame_size); |
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192 if (RtpStreamReceiver.speakermode == AudioManager.MODE_NORMAL) { |
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193 calc(lin,(ring+delay)%(frame_size*11),num); |
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194 if (RtpStreamReceiver.nearend != 0) |
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195 noise(lin,(ring+delay)%(frame_size*11),num,p); |
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196 else if (nearend == 0) |
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197 p = 0.9*p + 0.1*s; |
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198 } |
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199 codec.encode(codecCtx, lin, ring%(frame_size*11), frame_size, buffer, 12); |
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200 ring += frame_size; |
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201 rtpSession.sendData(packet, pkt); |
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202 if (m == 2) { |
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203 rtpSession.sendData(packet, pkt); |
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204 println("retransmit"); |
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205 } |
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206 seqn++; |
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207 time += num; |
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208 } |
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209 record.stop(); |
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210 rtpSession = null; |
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211 codec.cleanEncoder(codecCtx); |
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212 if (DEBUG) |
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213 println("rtp sender terminated"); |
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214 } |
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215 |
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216 /** Debug output */ |
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217 private static void println(String str) { |
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218 android.util.Log.d("DEBUG","RtpStreamSender: " + str); |
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219 } |
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220 } |