17 * You should have received a copy of the GNU General Public License |
17 * You should have received a copy of the GNU General Public License |
18 * along with this source code; if not, write to the Free Software |
18 * along with this source code; if not, write to the Free Software |
19 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA |
19 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA |
20 */ |
20 */ |
21 |
21 |
22 package src.org.sipdroid.media; |
22 package org.sipdroid.media; |
23 |
23 |
24 import java.io.IOException; |
24 import java.io.IOException; |
25 import java.io.InputStream; |
25 import java.io.InputStream; |
26 import java.net.InetAddress; |
26 import java.net.InetAddress; |
27 import java.util.Random; |
27 import java.util.Random; |
28 |
28 |
29 import org.sipdroid.sipua.UserAgent; |
29 |
30 import org.sipdroid.sipua.ui.Receiver; |
|
31 import org.sipdroid.sipua.ui.Settings; |
|
32 import org.sipdroid.sipua.ui.Sipdroid; |
|
33 import org.sipdroid.pjlib.Codec; |
30 import org.sipdroid.pjlib.Codec; |
34 |
31 |
35 import src.org.sipdroid.net.RtpPacket; |
32 import org.sipdroid.media.RtpStreamReceiver; |
36 import src.org.sipdroid.net.RtpSocket; |
33 import org.sipdroid.net.RtpPacket; |
37 import src.org.sipdroid.net.SipdroidSocket; |
34 import org.sipdroid.net.RtpSocket; |
|
35 import org.sipdroid.net.SipdroidSocket; |
|
36 |
|
37 import com.beem.project.beem.ui.Call; |
|
38 import com.beem.project.beem.utils.BeemConnectivity; |
38 |
39 |
39 import android.content.Context; |
40 import android.content.Context; |
40 import android.media.AudioFormat; |
41 import android.media.AudioFormat; |
41 import android.media.AudioManager; |
42 import android.media.AudioManager; |
42 import android.media.AudioRecord; |
43 import android.media.AudioRecord; |
43 import android.media.MediaRecorder; |
44 import android.media.MediaRecorder; |
44 import android.preference.PreferenceManager; |
45 import android.preference.PreferenceManager; |
|
46 import android.provider.Settings; |
45 import android.telephony.TelephonyManager; |
47 import android.telephony.TelephonyManager; |
46 |
48 |
47 /** |
49 /** |
48 * RtpStreamSender is a generic stream sender. It takes an InputStream and sends |
50 * RtpStreamSender is a generic stream sender. It takes an InputStream and sends |
49 * it through RTP. |
51 * it through RTP. |
50 */ |
52 */ |
51 public class RtpStreamSender extends Thread { |
53 public class RtpStreamSender extends Thread { |
52 /** Whether working in debug mode. */ |
54 /** Whether working in debug mode. */ |
53 public static boolean DEBUG = true; |
55 public static boolean DEBUG = true; |
54 |
56 |
55 /** The RtpSocket */ |
57 /** The RtpSocket */ |
56 RtpSocket rtp_socket = null; |
58 RtpSocket rtp_socket = null; |
57 |
59 |
58 /** Payload type */ |
60 /** Payload type */ |
59 int p_type; |
61 int p_type; |
60 |
62 |
61 /** Number of frame per second */ |
63 /** Number of frame per second */ |
62 long frame_rate; |
64 long frame_rate; |
63 |
65 |
64 /** Number of bytes per frame */ |
66 /** Number of bytes per frame */ |
65 int frame_size; |
67 int frame_size; |
66 |
68 |
67 /** |
69 /** |
68 * Whether it works synchronously with a local clock, or it it acts as slave |
70 * Whether it works synchronously with a local clock, or it it acts as slave |
69 * of the InputStream |
71 * of the InputStream |
70 */ |
72 */ |
71 boolean do_sync = true; |
73 boolean do_sync = true; |
72 |
74 |
73 /** |
75 /** |
74 * Synchronization correction value, in milliseconds. It accellarates the |
76 * Synchronization correction value, in milliseconds. It accellarates the |
75 * sending rate respect to the nominal value, in order to compensate program |
77 * sending rate respect to the nominal value, in order to compensate program |
76 * latencies. |
78 * latencies. |
77 */ |
79 */ |
78 int sync_adj = 0; |
80 int sync_adj = 0; |
79 |
81 |
80 /** Whether it is running */ |
82 /** Whether it is running */ |
81 boolean running = false; |
83 boolean running = false; |
82 boolean muted = false; |
84 boolean muted = false; |
83 |
85 |
84 /** |
86 /** |
85 * Constructs a RtpStreamSender. |
87 * Constructs a RtpStreamSender. |
86 * |
88 * |
87 * @param input_stream |
89 * @param input_stream |
88 * the stream to be sent |
90 * the stream to be sent |
89 * @param do_sync |
91 * @param do_sync |
90 * whether time synchronization must be performed by the |
92 * whether time synchronization must be performed by the |
91 * RtpStreamSender, or it is performed by the InputStream (e.g. |
93 * RtpStreamSender, or it is performed by the InputStream (e.g. |
92 * the system audio input) |
94 * the system audio input) |
93 * @param payload_type |
95 * @param payload_type |
94 * the payload type |
96 * the payload type |
95 * @param frame_rate |
97 * @param frame_rate |
96 * the frame rate, i.e. the number of frames that should be sent |
98 * the frame rate, i.e. the number of frames that should be sent |
97 * per second; it is used to calculate the nominal packet time |
99 * per second; it is used to calculate the nominal packet time |
98 * and,in case of do_sync==true, the next departure time |
100 * and,in case of do_sync==true, the next departure time |
99 * @param frame_size |
101 * @param frame_size |
100 * the size of the payload |
102 * the size of the payload |
101 * @param src_socket |
103 * @param src_socket |
102 * the socket used to send the RTP packet |
104 * the socket used to send the RTP packet |
103 * @param dest_addr |
105 * @param dest_addr |
104 * the destination address |
106 * the destination address |
105 * @param dest_port |
107 * @param dest_port |
106 * the destination port |
108 * the destination port |
107 */ |
109 */ |
108 public RtpStreamSender(boolean do_sync, |
110 public RtpStreamSender(boolean do_sync, |
109 int payload_type, long frame_rate, int frame_size, |
111 int payload_type, long frame_rate, int frame_size, |
110 SipdroidSocket src_socket, String dest_addr, int dest_port) { |
112 SipdroidSocket src_socket, String dest_addr, int dest_port) { |
111 init(do_sync, payload_type, frame_rate, frame_size, |
113 init(do_sync, payload_type, frame_rate, frame_size, |
112 src_socket, dest_addr, dest_port); |
114 src_socket, dest_addr, dest_port); |
113 } |
115 } |
114 |
116 |
115 /** Inits the RtpStreamSender */ |
117 /** Inits the RtpStreamSender */ |
116 private void init(boolean do_sync, |
118 private void init(boolean do_sync, |
117 int payload_type, long frame_rate, int frame_size, |
119 int payload_type, long frame_rate, int frame_size, |
118 SipdroidSocket src_socket, String dest_addr, |
120 SipdroidSocket src_socket, String dest_addr, |
119 int dest_port) { |
121 int dest_port) { |
120 this.p_type = payload_type; |
122 this.p_type = payload_type; |
121 this.frame_rate = frame_rate; |
123 this.frame_rate = frame_rate; |
122 this.frame_size = PreferenceManager.getDefaultSharedPreferences(Receiver.mContext).getString("server","").equals("pbxes.org")? |
124 this.frame_size = frame_size; |
123 (payload_type == 3?960:1024):frame_size; //15 |
125 this.do_sync = do_sync; |
124 this.do_sync = do_sync; |
126 try { |
|
127 rtp_socket = new RtpSocket(src_socket, InetAddress |
|
128 .getByName(dest_addr), dest_port); |
|
129 } catch (Exception e) { |
|
130 e.printStackTrace(); |
|
131 } |
|
132 } |
|
133 |
|
134 /** Sets the synchronization adjustment time (in milliseconds). */ |
|
135 public void setSyncAdj(int millisecs) { |
|
136 sync_adj = millisecs; |
|
137 } |
|
138 |
|
139 /** Whether is running */ |
|
140 public boolean isRunning() { |
|
141 return running; |
|
142 } |
|
143 |
|
144 public boolean mute() { |
|
145 return muted = !muted; |
|
146 } |
|
147 |
|
148 public static int delay = 0; |
|
149 |
|
150 /** Stops running */ |
|
151 public void halt() { |
|
152 running = false; |
|
153 } |
|
154 |
|
155 Random random; |
|
156 double smin = 200,s; |
|
157 int nearend; |
|
158 |
|
159 void calc(short[] lin,int off,int len) { |
|
160 int i,j; |
|
161 double sm = 30000,r; |
|
162 |
|
163 for (i = 0; i < len; i += 5) { |
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164 j = lin[i+off]; |
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165 s = 0.03*Math.abs(j) + 0.97*s; |
|
166 if (s < sm) sm = s; |
|
167 if (s > smin) nearend = 3000/5; |
|
168 else if (nearend > 0) nearend--; |
|
169 } |
|
170 for (i = 0; i < len; i++) { |
|
171 j = lin[i+off]; |
|
172 if (j > 6550) |
|
173 lin[i+off] = 6550*5; |
|
174 else if (j < -6550) |
|
175 lin[i+off] = -6550*5; |
|
176 else |
|
177 lin[i+off] = (short)(j*5); |
|
178 } |
|
179 r = (double)len/100000; |
|
180 smin = sm*r + smin*(1-r); |
|
181 } |
|
182 |
|
183 void calc1(short[] lin,int off,int len) { |
|
184 int i,j; |
|
185 |
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186 for (i = 0; i < len; i++) { |
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187 j = lin[i+off]; |
|
188 lin[i+off] = (short)(j>>1); |
|
189 } |
|
190 } |
|
191 |
|
192 void calc5(short[] lin,int off,int len) { |
|
193 int i,j; |
|
194 |
|
195 for (i = 0; i < len; i++) { |
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196 j = lin[i+off]; |
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197 if (j > 16350) |
|
198 lin[i+off] = 16350<<1; |
|
199 else if (j < -16350) |
|
200 lin[i+off] = -16350<<1; |
|
201 else |
|
202 lin[i+off] = (short)(j<<1); |
|
203 } |
|
204 } |
|
205 |
|
206 void calc10(short[] lin,int off,int len) { |
|
207 int i,j; |
|
208 |
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209 for (i = 0; i < len; i++) { |
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210 j = lin[i+off]; |
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211 if (j > 8150) |
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212 lin[i+off] = 8150<<2; |
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213 else if (j < -8150) |
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214 lin[i+off] = -8150<<2; |
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215 else |
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216 lin[i+off] = (short)(j<<2); |
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217 } |
|
218 } |
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219 |
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220 void noise(short[] lin,int off,int len,double power) { |
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221 int i,r = (int)(power*2); |
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222 short ran; |
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223 |
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224 if (r == 0) r = 1; |
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225 for (i = 0; i < len; i += 4) { |
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226 ran = (short)(random.nextInt(r*2)-r); |
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227 lin[i+off] = ran; |
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228 lin[i+off+1] = ran; |
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229 lin[i+off+2] = ran; |
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230 lin[i+off+3] = ran; |
|
231 } |
|
232 } |
|
233 public static float getMicGain() { |
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234 if (Call.headset > 0) |
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235 return Float.valueOf(PreferenceManager.getDefaultSharedPreferences(Call.mContext).getString("hmicgain", "1.0")); |
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236 return Float.valueOf(PreferenceManager.getDefaultSharedPreferences(Call.mContext).getString("micgain", "0.25")); |
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237 } |
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238 |
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239 /** Runs it in a new Thread. */ |
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240 public void run() { |
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241 if (rtp_socket == null) |
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242 return; |
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243 byte[] buffer = new byte[frame_size + 12]; |
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244 RtpPacket rtp_packet = new RtpPacket(buffer, 0); |
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245 rtp_packet.setPayloadType(p_type); |
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246 int seqn = 0; |
|
247 long time = 0; |
|
248 double p = 0; |
|
249 TelephonyManager tm = (TelephonyManager) Call.mContext.getSystemService(Context.TELEPHONY_SERVICE); |
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250 boolean improve = PreferenceManager.getDefaultSharedPreferences(Call.mContext).getBoolean("improve",false); |
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251 boolean useGSM = !PreferenceManager.getDefaultSharedPreferences(Call.mContext).getString("compression","edge").equals("never"); |
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252 int micgain = (int)(getMicGain()*10); |
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253 running = true; |
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254 |
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255 if (DEBUG) |
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256 println("Reading blocks of " + buffer.length + " bytes"); |
|
257 |
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258 android.os.Process.setThreadPriority(android.os.Process.THREAD_PRIORITY_URGENT_AUDIO); |
|
259 AudioRecord record = new AudioRecord(MediaRecorder.AudioSource.MIC, 8000, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, |
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260 AudioRecord.getMinBufferSize(8000, |
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261 AudioFormat.CHANNEL_CONFIGURATION_MONO, |
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262 AudioFormat.ENCODING_PCM_16BIT)*3/2); |
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263 short[] lin = new short[frame_size*11]; |
|
264 int num,ring = 0; |
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265 random = new Random(); |
|
266 InputStream alerting = null; |
|
267 try { |
|
268 alerting = Call.mContext.getAssets().open("alerting"); |
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269 } catch (IOException e2) { |
|
270 e2.printStackTrace(); |
|
271 } |
|
272 switch (p_type) { |
|
273 case 3: |
|
274 Codec.init(); |
|
275 break; |
|
276 case 0: |
|
277 case 8: |
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278 G711.init(); |
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279 break; |
|
280 } |
|
281 record.startRecording(); |
|
282 while (running) { |
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283 if (muted || Call.call_state == Call.UA_STATE_HOLD) { |
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284 record.stop(); |
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285 while (running && (muted || Call.call_state == Call.UA_STATE_HOLD)) { |
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286 try { |
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287 sleep(1000); |
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288 } catch (InterruptedException e1) { |
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289 e1.printStackTrace(); |
|
290 } |
|
291 } |
|
292 record.startRecording(); |
|
293 } |
|
294 num = record.read(lin,(ring+delay)%(frame_size*11),frame_size); |
|
295 |
|
296 if (RtpStreamReceiver.speakermode == AudioManager.MODE_NORMAL) { |
|
297 calc(lin,(ring+delay)%(frame_size*11),num); |
|
298 if (RtpStreamReceiver.nearend != 0) |
|
299 noise(lin,(ring+delay)%(frame_size*11),num,p); |
|
300 else if (RtpStreamReceiver.nearend == 0) |
|
301 p = 0.9*p + 0.1*s; |
|
302 } else switch (micgain) { |
|
303 case 1: |
|
304 calc1(lin,(ring+delay)%(frame_size*11),num); |
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305 break; |
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306 case 5: |
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307 calc5(lin,(ring+delay)%(frame_size*11),num); |
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308 break; |
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309 case 10: |
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310 calc10(lin,(ring+delay)%(frame_size*11),num); |
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311 break; |
|
312 } |
|
313 if (Call.call_state != Call.UA_STATE_INCALL && alerting != null) { |
125 try { |
314 try { |
126 rtp_socket = new RtpSocket(src_socket, InetAddress |
315 if (alerting.available() < num) |
127 .getByName(dest_addr), dest_port); |
316 alerting.reset(); |
128 } catch (Exception e) { |
317 alerting.read(buffer,12,num); |
129 if (!Sipdroid.release) e.printStackTrace(); |
318 } catch (IOException e) { |
130 } |
319 e.printStackTrace(); |
131 } |
320 } |
132 |
321 switch (p_type) {// have to add ulaw case? |
133 /** Sets the synchronization adjustment time (in milliseconds). */ |
322 case 3: |
134 public void setSyncAdj(int millisecs) { |
323 G711.alaw2linear(buffer, lin, num); |
135 sync_adj = millisecs; |
324 num = Codec.encode(lin, 0, buffer, num); |
136 } |
325 break; |
137 |
326 case 0: |
138 /** Whether is running */ |
327 G711.alaw2linear(buffer, lin, num); |
139 public boolean isRunning() { |
328 G711.linear2ulaw(lin, 0, buffer, num); |
140 return running; |
329 break; |
141 } |
330 } |
142 |
331 } else { |
143 public boolean mute() { |
|
144 return muted = !muted; |
|
145 } |
|
146 |
|
147 public static int delay = 0; |
|
148 |
|
149 /** Stops running */ |
|
150 public void halt() { |
|
151 running = false; |
|
152 } |
|
153 |
|
154 Random random; |
|
155 double smin = 200,s; |
|
156 int nearend; |
|
157 |
|
158 void calc(short[] lin,int off,int len) { |
|
159 int i,j; |
|
160 double sm = 30000,r; |
|
161 |
|
162 for (i = 0; i < len; i += 5) { |
|
163 j = lin[i+off]; |
|
164 s = 0.03*Math.abs(j) + 0.97*s; |
|
165 if (s < sm) sm = s; |
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166 if (s > smin) nearend = 3000/5; |
|
167 else if (nearend > 0) nearend--; |
|
168 } |
|
169 for (i = 0; i < len; i++) { |
|
170 j = lin[i+off]; |
|
171 if (j > 6550) |
|
172 lin[i+off] = 6550*5; |
|
173 else if (j < -6550) |
|
174 lin[i+off] = -6550*5; |
|
175 else |
|
176 lin[i+off] = (short)(j*5); |
|
177 } |
|
178 r = (double)len/100000; |
|
179 smin = sm*r + smin*(1-r); |
|
180 } |
|
181 |
|
182 void calc1(short[] lin,int off,int len) { |
|
183 int i,j; |
|
184 |
|
185 for (i = 0; i < len; i++) { |
|
186 j = lin[i+off]; |
|
187 lin[i+off] = (short)(j>>1); |
|
188 } |
|
189 } |
|
190 |
|
191 void calc5(short[] lin,int off,int len) { |
|
192 int i,j; |
|
193 |
|
194 for (i = 0; i < len; i++) { |
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195 j = lin[i+off]; |
|
196 if (j > 16350) |
|
197 lin[i+off] = 16350<<1; |
|
198 else if (j < -16350) |
|
199 lin[i+off] = -16350<<1; |
|
200 else |
|
201 lin[i+off] = (short)(j<<1); |
|
202 } |
|
203 } |
|
204 |
|
205 void calc10(short[] lin,int off,int len) { |
|
206 int i,j; |
|
207 |
|
208 for (i = 0; i < len; i++) { |
|
209 j = lin[i+off]; |
|
210 if (j > 8150) |
|
211 lin[i+off] = 8150<<2; |
|
212 else if (j < -8150) |
|
213 lin[i+off] = -8150<<2; |
|
214 else |
|
215 lin[i+off] = (short)(j<<2); |
|
216 } |
|
217 } |
|
218 |
|
219 void noise(short[] lin,int off,int len,double power) { |
|
220 int i,r = (int)(power*2); |
|
221 short ran; |
|
222 |
|
223 if (r == 0) r = 1; |
|
224 for (i = 0; i < len; i += 4) { |
|
225 ran = (short)(random.nextInt(r*2)-r); |
|
226 lin[i+off] = ran; |
|
227 lin[i+off+1] = ran; |
|
228 lin[i+off+2] = ran; |
|
229 lin[i+off+3] = ran; |
|
230 } |
|
231 } |
|
232 |
|
233 public static int m; |
|
234 |
|
235 /** Runs it in a new Thread. */ |
|
236 public void run() { |
|
237 if (rtp_socket == null) |
|
238 return; |
|
239 byte[] buffer = new byte[frame_size + 12]; |
|
240 RtpPacket rtp_packet = new RtpPacket(buffer, 0); |
|
241 rtp_packet.setPayloadType(p_type); |
|
242 int seqn = 0; |
|
243 long time = 0; |
|
244 double p = 0; |
|
245 TelephonyManager tm = (TelephonyManager) Receiver.mContext.getSystemService(Context.TELEPHONY_SERVICE); |
|
246 boolean improve = PreferenceManager.getDefaultSharedPreferences(Receiver.mContext).getBoolean("improve",false); |
|
247 boolean useGSM = !PreferenceManager.getDefaultSharedPreferences(Receiver.mContext).getString("compression","edge").equals("never"); |
|
248 int micgain = (int)(Settings.getMicGain()*10); |
|
249 running = true; |
|
250 m = 1; |
|
251 |
|
252 if (DEBUG) |
|
253 println("Reading blocks of " + buffer.length + " bytes"); |
|
254 |
|
255 android.os.Process.setThreadPriority(android.os.Process.THREAD_PRIORITY_URGENT_AUDIO); |
|
256 AudioRecord record = new AudioRecord(MediaRecorder.AudioSource.MIC, 8000, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, |
|
257 AudioRecord.getMinBufferSize(8000, |
|
258 AudioFormat.CHANNEL_CONFIGURATION_MONO, |
|
259 AudioFormat.ENCODING_PCM_16BIT)*3/2); |
|
260 short[] lin = new short[frame_size*11]; |
|
261 int num,ring = 0; |
|
262 random = new Random(); |
|
263 InputStream alerting = null; |
|
264 try { |
|
265 alerting = Receiver.mContext.getAssets().open("alerting"); |
|
266 } catch (IOException e2) { |
|
267 if (!Sipdroid.release) e2.printStackTrace(); |
|
268 } |
|
269 switch (p_type) { |
332 switch (p_type) { |
270 case 3: |
333 case 3: |
271 Codec.init(); |
334 num = Codec.encode(lin, ring%(frame_size*11), buffer, num); |
272 break; |
335 break; |
273 case 0: |
336 case 0: |
274 case 8: |
337 G711.linear2ulaw(lin, ring%(frame_size*11), buffer, num); |
275 G711.init(); |
338 break; |
276 break; |
339 case 8: |
277 } |
340 G711.linear2alaw(lin, ring%(frame_size*11), buffer, num); |
278 record.startRecording(); |
341 break; |
279 while (running) { |
342 } |
280 if (muted || Receiver.call_state == UserAgent.UA_STATE_HOLD) { |
343 } |
281 record.stop(); |
344 ring += frame_size; |
282 while (running && (muted || Receiver.call_state == UserAgent.UA_STATE_HOLD)) { |
345 rtp_packet.setSequenceNumber(seqn++); |
283 try { |
346 rtp_packet.setTimestamp(time); |
284 sleep(1000); |
347 rtp_packet.setPayloadLength(num); |
285 } catch (InterruptedException e1) { |
348 try { |
286 } |
349 rtp_socket.send(rtp_packet); |
287 } |
350 } catch (IOException e) { |
288 record.startRecording(); |
351 e.printStackTrace(); |
289 } |
352 } |
290 num = record.read(lin,(ring+delay)%(frame_size*11),frame_size); |
353 time += frame_size; |
291 |
354 if (useGSM && p_type == 8 && !BeemConnectivity.isWifi(Call.mContext) && tm.getNetworkType() == TelephonyManager.NETWORK_TYPE_EDGE) { |
292 if (RtpStreamReceiver.speakermode == AudioManager.MODE_NORMAL) { |
355 rtp_packet.setPayloadType(p_type = 3); |
293 calc(lin,(ring+delay)%(frame_size*11),num); |
356 if (frame_size == 1024) { |
294 if (RtpStreamReceiver.nearend != 0) |
357 frame_size = 960; |
295 noise(lin,(ring+delay)%(frame_size*11),num,p); |
358 ring = 0; |
296 else if (nearend == 0) |
359 } |
297 p = 0.9*p + 0.1*s; |
360 } |
298 } else switch (micgain) { |
361 } |
299 case 1: |
362 record.stop(); |
300 calc1(lin,(ring+delay)%(frame_size*11),num); |
363 |
301 break; |
364 rtp_socket.close(); |
302 case 5: |
365 rtp_socket = null; |
303 calc5(lin,(ring+delay)%(frame_size*11),num); |
366 |
304 break; |
367 if (DEBUG) |
305 case 10: |
368 println("rtp sender terminated"); |
306 calc10(lin,(ring+delay)%(frame_size*11),num); |
369 } |
307 break; |
370 |
308 } |
371 /** Debug output */ |
309 if (Receiver.call_state != UserAgent.UA_STATE_INCALL && alerting != null) { |
372 private static void println(String str) { |
310 try { |
373 System.out.println("RtpStreamSender: " + str); |
311 if (alerting.available() < num) |
374 } |
312 alerting.reset(); |
|
313 alerting.read(buffer,12,num); |
|
314 } catch (IOException e) { |
|
315 if (!Sipdroid.release) e.printStackTrace(); |
|
316 } |
|
317 switch (p_type) {// have to add ulaw case? |
|
318 case 3: |
|
319 G711.alaw2linear(buffer, lin, num); |
|
320 num = Codec.encode(lin, 0, buffer, num); |
|
321 break; |
|
322 case 0: |
|
323 G711.alaw2linear(buffer, lin, num); |
|
324 G711.linear2ulaw(lin, 0, buffer, num); |
|
325 break; |
|
326 } |
|
327 } else { |
|
328 switch (p_type) { |
|
329 case 3: |
|
330 num = Codec.encode(lin, ring%(frame_size*11), buffer, num); |
|
331 break; |
|
332 case 0: |
|
333 G711.linear2ulaw(lin, ring%(frame_size*11), buffer, num); |
|
334 break; |
|
335 case 8: |
|
336 G711.linear2alaw(lin, ring%(frame_size*11), buffer, num); |
|
337 break; |
|
338 } |
|
339 } |
|
340 ring += frame_size; |
|
341 rtp_packet.setSequenceNumber(seqn++); |
|
342 rtp_packet.setTimestamp(time); |
|
343 rtp_packet.setPayloadLength(num); |
|
344 try { |
|
345 rtp_socket.send(rtp_packet); |
|
346 if (m == 2) |
|
347 rtp_socket.send(rtp_packet); |
|
348 } catch (IOException e) { |
|
349 } |
|
350 time += frame_size; |
|
351 if (improve && RtpStreamReceiver.good != 0 && |
|
352 RtpStreamReceiver.loss/RtpStreamReceiver.good > 0.01 && |
|
353 (Receiver.on_wlan || tm.getNetworkType() != TelephonyManager.NETWORK_TYPE_EDGE)) |
|
354 m = 2; |
|
355 else |
|
356 m = 1; |
|
357 if (useGSM && p_type == 8 && !Receiver.on_wlan && tm.getNetworkType() == TelephonyManager.NETWORK_TYPE_EDGE) { |
|
358 rtp_packet.setPayloadType(p_type = 3); |
|
359 if (frame_size == 1024) { |
|
360 frame_size = 960; |
|
361 ring = 0; |
|
362 } |
|
363 } |
|
364 } |
|
365 record.stop(); |
|
366 |
|
367 rtp_socket.close(); |
|
368 rtp_socket = null; |
|
369 |
|
370 if (DEBUG) |
|
371 println("rtp sender terminated"); |
|
372 } |
|
373 |
|
374 /** Debug output */ |
|
375 private static void println(String str) { |
|
376 if (!Sipdroid.release) System.out.println("RtpStreamSender: " + str); |
|
377 } |
|
378 |
375 |
379 } |
376 } |