--- a/src/org/sipdroid/media/RtpStreamSender.java Sat Jan 23 21:48:58 2010 +0100
+++ b/src/org/sipdroid/media/RtpStreamSender.java Sat Jan 23 22:19:43 2010 +0100
@@ -19,42 +19,50 @@
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
-package org.sipdroid.media;
+package src.org.sipdroid.media;
+import java.io.IOException;
import java.io.InputStream;
-import java.net.DatagramPacket;
+import java.net.InetAddress;
import java.util.Random;
-import jlibrtp.RTPSession;
-import jlibrtp.RtpPkt;
+import org.sipdroid.sipua.UserAgent;
+import org.sipdroid.sipua.ui.Receiver;
+import org.sipdroid.sipua.ui.Settings;
+import org.sipdroid.sipua.ui.Sipdroid;
+import org.sipdroid.pjlib.Codec;
-import org.sipdroid.media.codecs.Codec;
+import src.org.sipdroid.net.RtpPacket;
+import src.org.sipdroid.net.RtpSocket;
+import src.org.sipdroid.net.SipdroidSocket;
+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.telephony.TelephonyManager;
/**
* 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;
+public class RtpStreamSender extends Thread {
+ /** Whether working in debug mode. */
+ public static boolean DEBUG = true;
/** The RtpSocket */
- private RTPSession rtpSession = null;
+ RtpSocket rtp_socket = null;
- /** Codec */
- private Codec codec;
+ /** Payload type */
+ int p_type;
- private int sampling_rate;
+ /** Number of frame per second */
+ long frame_rate;
/** Number of bytes per frame */
- private int frame_size;
-
- private int codec_frame_size;
+ int frame_size;
/**
* Whether it works synchronously with a local clock, or it it acts as slave
@@ -62,33 +70,64 @@
*/
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;
- private int codec_divider;
-
/**
* 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(Codec co, RTPSession rtpSession) {
- init(co, rtpSession);
+ 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(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;
+ 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 = PreferenceManager.getDefaultSharedPreferences(Receiver.mContext).getString("server","").equals("pbxes.org")?
+ (payload_type == 3?960:1024):frame_size; //15
+ this.do_sync = do_sync;
+ try {
+ rtp_socket = new RtpSocket(src_socket, InetAddress
+ .getByName(dest_addr), dest_port);
+ } catch (Exception e) {
+ if (!Sipdroid.release) e.printStackTrace();
+ }
}
/** Sets the synchronization adjustment time (in milliseconds). */
@@ -100,13 +139,13 @@
public boolean isRunning() {
return running;
}
-
+
public boolean mute() {
return muted = !muted;
}
public static int delay = 0;
-
+
/** Stops running */
public void halt() {
running = false;
@@ -115,11 +154,11 @@
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;
@@ -140,6 +179,43 @@
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;
@@ -153,21 +229,23 @@
lin[i+off+3] = ran;
}
}
-
+
public static int m;
-
+
/** Runs it in a new Thread. */
public void run() {
- if (rtpSession == null)
+ if (rtp_socket == 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);
+ 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) Receiver.mContext.getSystemService(Context.TELEPHONY_SERVICE);
+ boolean improve = PreferenceManager.getDefaultSharedPreferences(Receiver.mContext).getBoolean("improve",false);
+ boolean useGSM = !PreferenceManager.getDefaultSharedPreferences(Receiver.mContext).getString("compression","edge").equals("never");
+ int micgain = (int)(Settings.getMicGain()*10);
running = true;
m = 1;
@@ -175,45 +253,127 @@
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,
+ 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)*2);
- record.startRecording();
+ 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 = Receiver.mContext.getAssets().open("alerting");
+ } catch (IOException e2) {
+ if (!Sipdroid.release) e2.printStackTrace();
+ }
+ switch (p_type) {
+ case 3:
+ Codec.init();
+ break;
+ case 0:
+ case 8:
+ G711.init();
+ break;
+ }
+ record.startRecording();
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;
+ if (muted || Receiver.call_state == UserAgent.UA_STATE_HOLD) {
+ record.stop();
+ while (running && (muted || Receiver.call_state == UserAgent.UA_STATE_HOLD)) {
+ try {
+ sleep(1000);
+ } catch (InterruptedException e1) {
+ }
+ }
+ 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 (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 (Receiver.call_state != UserAgent.UA_STATE_INCALL && alerting != null) {
+ try {
+ if (alerting.available() < num)
+ alerting.reset();
+ alerting.read(buffer,12,num);
+ } catch (IOException e) {
+ if (!Sipdroid.release) 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);
+ if (m == 2)
+ rtp_socket.send(rtp_packet);
+ } catch (IOException e) {
+ }
+ time += frame_size;
+ if (improve && RtpStreamReceiver.good != 0 &&
+ RtpStreamReceiver.loss/RtpStreamReceiver.good > 0.01 &&
+ (Receiver.on_wlan || tm.getNetworkType() != TelephonyManager.NETWORK_TYPE_EDGE))
+ m = 2;
+ else
+ m = 1;
+ if (useGSM && p_type == 8 && !Receiver.on_wlan && tm.getNetworkType() == TelephonyManager.NETWORK_TYPE_EDGE) {
+ rtp_packet.setPayloadType(p_type = 3);
+ if (frame_size == 1024) {
+ frame_size = 960;
+ ring = 0;
+ }
+ }
}
record.stop();
- rtpSession = null;
- codec.cleanEncoder(codecCtx);
+
+ rtp_socket.close();
+ rtp_socket = null;
+
if (DEBUG)
println("rtp sender terminated");
}
/** Debug output */
private static void println(String str) {
- android.util.Log.d("DEBUG","RtpStreamSender: " + str);
+ if (!Sipdroid.release) System.out.println("RtpStreamSender: " + str);
}
-}
+
+}
\ No newline at end of file