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diff --git a/src/audio/audio_tx.h b/src/audio/audio_tx.h
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+/*
+    Copyright (C) 2005-2009  Michel de Boer <michel@twinklephone.com>
+
+    This program 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 2 of the License, or
+    (at your option) any later version.
+
+    This program 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 program; if not, write to the Free Software
+    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+*/
+
+#ifndef _AUDIO_TX_H
+#define _AUDIO_TX_H
+
+// Receive RTP and send audio to soundcard
+
+#include <map>
+#include <string>
+#include "audio_codecs.h"
+#include "audio_decoder.h"
+#include "audio_rx.h"
+#include "media_buffer.h"
+#include "rtp_telephone_event.h"
+#include "user.h"
+#include "threads/mutex.h"
+#include "gsm/inc/gsm.h"
+#include "audio_device.h"
+#include "twinkle_rtp_session.h"
+#include "twinkle_config.h"
+
+using namespace std;
+using namespace ost;
+
+// Forward declarations
+class t_audio_session;
+class t_line;
+
+class t_audio_tx {
+private:
+	// audio_session owning this audio transmitter
+	t_audio_session *audio_session;
+	
+	// User profile of user using the line
+	// This is a pointer to the user_config owned by a phone user.
+	// So this pointer should never be deleted.
+	t_user		*user_config;
+
+	// file descriptor audio capture device
+	t_audio_io		*playback_device;
+	t_twinkle_rtp_session	*rtp_session;
+
+	// Indicates if this transmitter is part of a 3-way conference
+	bool		is_3way;
+
+	// Indicates if this transmitter is the mixer in a 3-way conference.
+	// The mixer will mix this audio stream with the audio from the other
+	// party and send it to the soundcard.
+	// If the transmitter is part of a 3-way conference, but not the
+	// mixer, then it simply has to send its audio payload to the mixer.
+	bool		is_3way_mixer;
+
+	// Media buffer for media from the other far-end. This buffer is
+	// used by the mixer.
+	t_media_buffer	*media_3way_peer_tx;
+
+	// The peer audio transmitter in a 3-way conference
+	t_audio_tx	*peer_tx_3way;
+
+	// The audio receiver that needs input from this transmitter in
+	// a 3-way conference.
+	t_audio_rx	*peer_rx_3way;
+
+	// Buffer for mixing purposes in 3-way conference.
+	unsigned char	*mix_buf_3way;
+
+	// Mutex to protect 3-way resources
+	t_mutex		mtx_3way;
+
+	// Codec information
+	t_audio_codec	codec;
+	map<unsigned short, t_audio_codec> payload2codec;
+	unsigned short	ptime;	// in milliseconds
+	
+	// Sample rate of sound card.
+	// The sample rate of the sound card is set to the sample rate
+	// used for the initial codec. The far end may dynamically switch
+	// to a codec with another sample rate. This will not change the
+	// sample rate of the sound card! (capture and playback cannot
+	// be done at different sampling rates).
+	unsigned short	sc_sample_rate;
+	
+	// Mapping from codecs to decoders
+	map<t_audio_codec, t_audio_decoder *> map_audio_decoder;
+
+	// Buffer to store PCM samples of a received RTP packet
+	unsigned char	*sample_buf;
+
+	// Jitter buffer (PCM).
+	// jitter_buf_len indicates the number of bytes in the jitter buffer.
+	// At the start of playing the samples are stored in the jitter buffer.
+	// Once the buffer is full, all samples are copied to the memory of
+	// the soundcard. From that point the soundcard itself is the jitter
+	// buffer.
+	unsigned char	*jitter_buf;
+	unsigned short	jitter_buf_len;
+	bool		load_jitter_buf;
+
+	// Buffer to keep last played packets for concealment.
+	unsigned char	*conceal_buf[MAX_CONCEALMENT];
+	unsigned short	conceal_buflen[MAX_CONCEALMENT]; // length of packet
+	short		conceal_pos; // points to the oldest packet.
+	short		conceal_num; // number of retained packets.
+
+	unsigned short	soundcard_buf_size;
+
+	// Payload type for telephone-event payload.
+	// Some endpoints ignore the payload type that was sent in an
+	// outgoing INVITE and simply sends it with the payload type,
+	// they indicated in the 200 OK. Accept both payloads for
+	// interoperability.
+	int pt_telephone_event;
+	int pt_telephone_event_alt;
+
+	// Timestamp of previous DTMF tone
+	unsigned long	dtmf_previous_timestamp;
+
+	// Inidicates if the playing thread is running
+	volatile bool is_running;
+
+	// The thread exits when this indicator is set to true
+	volatile bool stop_running;
+
+	// Retain a packet (PCM encoded) for possible concealment.
+	void retain_for_concealment(unsigned char *buf, unsigned short len);
+
+	// Play last num packets again.
+	void conceal(short num);
+
+	// Erase concealment buffers.
+	void clear_conceal_buf(void);
+
+	// Play PCM encoded samples
+	// - only_3rd_party indicates if there is only 3rd_party audio available,
+	//   i.e. due to jitter, packet loss or silence suppression
+	void play_pcm(unsigned char *buf, unsigned short len, bool only_3rd_party = false);
+
+public:
+	// Create the audio transmitter
+	// _fd		file descriptor of capture device
+	// _rtp_session	RTP socket tp send the RTP stream
+	// _codec	audio codec to use
+	// _ptime	length of the audio packets in ms
+	// _ptime = 0 means use default ptime value for the codec
+	t_audio_tx(t_audio_session *_audio_session, t_audio_io *_playback_device,
+		   t_twinkle_rtp_session *_rtp_session,
+	           t_audio_codec _codec, 
+	           const map<unsigned short, t_audio_codec> &_payload2codec,
+	           unsigned short _ptime = 0);
+
+	~t_audio_tx();
+	
+	// Set the is running flag
+	void set_running(bool running);
+
+	void run(void);
+	
+	// Set the dynamic payload type for telephone events
+	void set_pt_telephone_event(int pt, int pt_alt);
+
+	// Get phone line belonging to this audio transmitter
+	t_line *get_line(void) const;
+
+	// Join this transmitter in a 3way conference.
+	// mixer indicates if this transmitter must be the mixer.
+	// - peer_tx is the peer transmitter in a 3-way
+	// - audio_rx is the audio receiver needing the output from this
+	//   transmitter for mixing.
+	void join_3way(bool mixer, t_audio_tx *peer_tx, t_audio_rx *peer_rx);
+
+	// Change the peer rx/tx (NULL to erase)
+	void set_peer_tx_3way(t_audio_tx *peer_tx);
+	void set_peer_rx_3way(t_audio_rx *peer_rx);
+
+	// Change the mixer role
+	void set_mixer_3way(bool mixer);
+
+	// Stop the 3-way conference and make it a 1-on-1 call again.
+	void stop_3way(void);
+
+	// Post media from the peer transmitter for a 3-way mixer.
+	void post_media_peer_tx_3way(unsigned char *media, int len, 
+		unsigned short peer_sample_rate);
+
+	// Returns if this transmitter is the mixer in a 3-way
+	bool get_is_3way_mixer(void) const;
+};
+
+#endif