package src.org.sipdroid.media;

import java.io.IOException;

import java.net.InetAddress;

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 src.org.sipdroid.net.RtpPacket;

import src.org.sipdroid.net.RtpSocket;

import src.org.sipdroid.net.SipdroidSocket;

import android.content.Context;

import android.preference.PreferenceManager;

import android.telephony.TelephonyManager;

public class RtpStreamSender extends Thread {

	/** Whether working in debug mode. */

	public static boolean DEBUG = true;

	RtpSocket rtp_socket = null;

	/** Payload type */

	int p_type;

	/** Number of frame per second */

	long frame_rate;

	int frame_size;

	/**

	 * Synchronization correction value, in milliseconds. It accellarates the

	 * sending rate respect to the nominal value, in order to compensate program

	 * latencies.

	 */

	 * @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);

	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();

		}

	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);

		}

	}

		if (rtp_socket == null)

		byte[] buffer = new byte[frame_size + 12];

		RtpPacket rtp_packet = new RtpPacket(buffer, 0);

		rtp_packet.setPayloadType(p_type);

		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);

		AudioRecord record = new AudioRecord(MediaRecorder.AudioSource.MIC, 8000, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, 

				AudioRecord.getMinBufferSize(8000, 

						AudioFormat.ENCODING_PCM_16BIT)*3/2);

		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();

			 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;

 				 }

 			 }

		rtp_socket.close();

		rtp_socket = null;

		if (!Sipdroid.release) System.out.println("RtpStreamSender: " + str);

}