Import pjproject-2.0.1

1 files changed, 1929 insertions, 0 deletions

diff --git a/pjmedia/src/pjmedia-audiodev/symb_aps_dev.cpp b/pjmedia/src/pjmedia-audiodev/symb_aps_dev.cpp
new file mode 100644
index 0000000..d2c6564
--- /dev/null
+++ b/pjmedia/src/pjmedia-audiodev/symb_aps_dev.cpp
@@ -0,0 +1,1929 @@
+/* $Id: symb_aps_dev.cpp 3841 2011-10-24 09:28:13Z ming $ */
+/* 
+ * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com)
+ * Copyright (C) 2003-2008 Benny Prijono <benny@prijono.org>
+ *
+ * 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 
+ */
+#include <pjmedia-audiodev/audiodev_imp.h>
+#include <pjmedia-audiodev/errno.h>
+#include <pjmedia/alaw_ulaw.h>
+#include <pjmedia/resample.h>
+#include <pjmedia/stereo.h>
+#include <pj/assert.h>
+#include <pj/log.h>
+#include <pj/math.h>
+#include <pj/os.h>
+#include <pj/string.h>
+
+#if PJMEDIA_AUDIO_DEV_HAS_SYMB_APS
+
+#include <e32msgqueue.h>
+#include <sounddevice.h>
+#include <APSClientSession.h>
+#include <pjmedia-codec/amr_helper.h>
+
+/* Pack/unpack G.729 frame of S60 DSP codec, taken from:  
+ * http://wiki.forum.nokia.com/index.php/TSS000776_-_Payload_conversion_for_G.729_audio_format
+ */
+#include "s60_g729_bitstream.h"
+
+
+#define THIS_FILE			"symb_aps_dev.c"
+#define BITS_PER_SAMPLE			16
+
+
+#if 1
+#   define TRACE_(st) PJ_LOG(3, st)
+#else
+#   define TRACE_(st)
+#endif
+
+
+/* App UID to open global APS queues to communicate with the APS server. */
+extern TPtrC APP_UID;
+
+/* APS G.711 frame length */
+static pj_uint8_t aps_g711_frame_len;
+
+
+/* APS factory */
+struct aps_factory
+{
+    pjmedia_aud_dev_factory	 base;
+    pj_pool_t			*pool;
+    pj_pool_factory		*pf;
+    pjmedia_aud_dev_info	 dev_info;
+};
+
+
+/* Forward declaration of CPjAudioEngine */
+class CPjAudioEngine;
+
+
+/* APS stream. */
+struct aps_stream
+{
+    // Base
+    pjmedia_aud_stream	 base;			/**< Base class.	*/
+    
+    // Pool
+    pj_pool_t		*pool;			/**< Memory pool.       */
+
+    // Common settings.
+    pjmedia_aud_param 	 param;			/**< Stream param.	*/
+    pjmedia_aud_rec_cb   rec_cb;		/**< Record callback.  	*/
+    pjmedia_aud_play_cb	 play_cb;		/**< Playback callback. */
+    void                *user_data;		/**< Application data.  */
+
+    // Audio engine
+    CPjAudioEngine	*engine;		/**< Internal engine.	*/
+
+    pj_timestamp  	 ts_play;		/**< Playback timestamp.*/
+    pj_timestamp	 ts_rec;		/**< Record timestamp.	*/
+
+    pj_int16_t		*play_buf;		/**< Playback buffer.	*/
+    pj_uint16_t		 play_buf_len;		/**< Playback buffer length. */
+    pj_uint16_t		 play_buf_start;	/**< Playback buffer start index. */
+    pj_int16_t		*rec_buf;		/**< Record buffer.	*/
+    pj_uint16_t		 rec_buf_len;		/**< Record buffer length. */
+    void                *strm_data;		/**< Stream data.	*/
+
+    /* Resampling is needed, in case audio device is opened with clock rate 
+     * other than 8kHz (only for PCM format).
+     */
+    pjmedia_resample	*play_resample;		/**< Resampler for playback. */
+    pjmedia_resample	*rec_resample;		/**< Resampler for recording */
+    pj_uint16_t		 resample_factor;	/**< Resample factor, requested
+						     clock rate / 8000	     */
+
+    /* When stream is working in PCM format, where the samples may need to be
+     * resampled from/to different clock rate and/or channel count, PCM buffer
+     * is needed to perform such resampling operations.
+     */
+    pj_int16_t		*pcm_buf;		/**< PCM buffer.	     */
+};
+
+
+/* Prototypes */
+static pj_status_t factory_init(pjmedia_aud_dev_factory *f);
+static pj_status_t factory_destroy(pjmedia_aud_dev_factory *f);
+static pj_status_t factory_refresh(pjmedia_aud_dev_factory *f);
+static unsigned    factory_get_dev_count(pjmedia_aud_dev_factory *f);
+static pj_status_t factory_get_dev_info(pjmedia_aud_dev_factory *f, 
+					unsigned index,
+					pjmedia_aud_dev_info *info);
+static pj_status_t factory_default_param(pjmedia_aud_dev_factory *f,
+					 unsigned index,
+					 pjmedia_aud_param *param);
+static pj_status_t factory_create_stream(pjmedia_aud_dev_factory *f,
+					 const pjmedia_aud_param *param,
+					 pjmedia_aud_rec_cb rec_cb,
+					 pjmedia_aud_play_cb play_cb,
+					 void *user_data,
+					 pjmedia_aud_stream **p_aud_strm);
+
+static pj_status_t stream_get_param(pjmedia_aud_stream *strm,
+				    pjmedia_aud_param *param);
+static pj_status_t stream_get_cap(pjmedia_aud_stream *strm,
+				  pjmedia_aud_dev_cap cap,
+				  void *value);
+static pj_status_t stream_set_cap(pjmedia_aud_stream *strm,
+				  pjmedia_aud_dev_cap cap,
+				  const void *value);
+static pj_status_t stream_start(pjmedia_aud_stream *strm);
+static pj_status_t stream_stop(pjmedia_aud_stream *strm);
+static pj_status_t stream_destroy(pjmedia_aud_stream *strm);
+
+
+/* Operations */
+static pjmedia_aud_dev_factory_op factory_op =
+{
+    &factory_init,
+    &factory_destroy,
+    &factory_get_dev_count,
+    &factory_get_dev_info,
+    &factory_default_param,
+    &factory_create_stream,
+    &factory_refresh
+};
+
+static pjmedia_aud_stream_op stream_op = 
+{
+    &stream_get_param,
+    &stream_get_cap,
+    &stream_set_cap,
+    &stream_start,
+    &stream_stop,
+    &stream_destroy
+};
+
+
+/****************************************************************************
+ * Internal APS Engine
+ */
+
+/*
+ * Utility: print sound device error
+ */
+static void snd_perror(const char *title, TInt rc)
+{
+    PJ_LOG(1,(THIS_FILE, "%s (error code=%d)", title, rc));
+}
+
+/*
+ * Utility: wait for specified time.
+ */
+static void snd_wait(unsigned ms) 
+{
+    TTime start, now;
+    
+    start.UniversalTime();
+    do {
+	pj_symbianos_poll(-1, ms);
+	now.UniversalTime();
+    } while (now.MicroSecondsFrom(start) < ms * 1000);
+}
+
+typedef void(*PjAudioCallback)(TAPSCommBuffer &buf, void *user_data);
+
+/**
+ * Abstract class for handler of callbacks from APS client.
+ */
+class MQueueHandlerObserver
+{
+public:
+    MQueueHandlerObserver(PjAudioCallback RecCb_, PjAudioCallback PlayCb_,
+			  void *UserData_)
+    : RecCb(RecCb_), PlayCb(PlayCb_), UserData(UserData_)
+    {}
+
+    virtual void InputStreamInitialized(const TInt aStatus) = 0;
+    virtual void OutputStreamInitialized(const TInt aStatus) = 0;
+    virtual void NotifyError(const TInt aError) = 0;
+
+public:
+    PjAudioCallback RecCb;
+    PjAudioCallback PlayCb;
+    void *UserData;
+};
+
+/**
+ * Handler for communication and data queue.
+ */
+class CQueueHandler : public CActive
+{
+public:
+    // Types of queue handler
+    enum TQueueHandlerType {
+        ERecordCommQueue,
+        EPlayCommQueue,
+        ERecordQueue,
+        EPlayQueue
+    };
+
+    // The order corresponds to the APS Server state, do not change!
+    enum TState {
+    	EAPSPlayerInitialize        = 1,
+    	EAPSRecorderInitialize      = 2,
+    	EAPSPlayData                = 3,
+    	EAPSRecordData              = 4,
+    	EAPSPlayerInitComplete      = 5,
+    	EAPSRecorderInitComplete    = 6
+    };
+
+    static CQueueHandler* NewL(MQueueHandlerObserver* aObserver,
+			       RMsgQueue<TAPSCommBuffer>* aQ,
+			       RMsgQueue<TAPSCommBuffer>* aWriteQ,
+			       TQueueHandlerType aType)
+    {
+	CQueueHandler* self = new (ELeave) CQueueHandler(aObserver, aQ, aWriteQ,
+							 aType);
+	CleanupStack::PushL(self);
+	self->ConstructL();
+	CleanupStack::Pop(self);
+	return self;
+    }
+
+    // Destructor
+    ~CQueueHandler() { Cancel(); }
+
+    // Start listening queue event
+    void Start() {
+	iQ->NotifyDataAvailable(iStatus);
+	SetActive();
+    }
+
+private:
+    // Constructor
+    CQueueHandler(MQueueHandlerObserver* aObserver,
+		  RMsgQueue<TAPSCommBuffer>* aQ,
+		  RMsgQueue<TAPSCommBuffer>* aWriteQ,
+		  TQueueHandlerType aType)
+	: CActive(CActive::EPriorityHigh),
+	  iQ(aQ), iWriteQ(aWriteQ), iObserver(aObserver), iType(aType)
+    {
+	CActiveScheduler::Add(this);
+
+	// use lower priority for comm queues
+	if ((iType == ERecordCommQueue) || (iType == EPlayCommQueue))
+	    SetPriority(CActive::EPriorityStandard);
+    }
+
+    // Second phase constructor
+    void ConstructL() {}
+
+    // Inherited from CActive
+    void DoCancel() { iQ->CancelDataAvailable(); }
+
+    void RunL() {
+	if (iStatus != KErrNone) {
+	    iObserver->NotifyError(iStatus.Int());
+	    return;
+        }
+
+	TAPSCommBuffer buffer;
+	TInt ret = iQ->Receive(buffer);
+
+	if (ret != KErrNone) {
+	    iObserver->NotifyError(ret);
+	    return;
+	}
+
+	switch (iType) {
+	case ERecordQueue:
+	    if (buffer.iCommand == EAPSRecordData) {
+		iObserver->RecCb(buffer, iObserver->UserData);
+	    } else {
+		iObserver->NotifyError(buffer.iStatus);
+	    }
+	    break;
+
+	// Callbacks from the APS main thread
+	case EPlayCommQueue:
+	    switch (buffer.iCommand) {
+		case EAPSPlayData:
+		    if (buffer.iStatus == KErrUnderflow) {
+			iObserver->PlayCb(buffer, iObserver->UserData);
+			iWriteQ->Send(buffer);
+		    }
+		    break;
+		case EAPSPlayerInitialize:
+		    iObserver->NotifyError(buffer.iStatus);
+		    break;
+		case EAPSPlayerInitComplete:
+		    iObserver->OutputStreamInitialized(buffer.iStatus);
+		    break;
+		case EAPSRecorderInitComplete:
+		    iObserver->InputStreamInitialized(buffer.iStatus);
+		    break;
+		default:
+		    iObserver->NotifyError(buffer.iStatus);
+		    break;
+	    }
+	    break;
+
+	// Callbacks from the APS recorder thread
+	case ERecordCommQueue:
+	    switch (buffer.iCommand) {
+		// The APS recorder thread will only report errors
+		// through this handler. All other callbacks will be
+		// sent from the APS main thread through EPlayCommQueue
+		case EAPSRecorderInitialize:
+		case EAPSRecordData:
+		default:
+		    iObserver->NotifyError(buffer.iStatus);
+		    break;
+	    }
+	    break;
+
+	default:
+	    break;
+        }
+
+        // issue next request
+        iQ->NotifyDataAvailable(iStatus);
+        SetActive();
+    }
+
+    TInt RunError(TInt) {
+	return 0;
+    }
+
+    // Data
+    RMsgQueue<TAPSCommBuffer>	*iQ;   // (not owned)
+    RMsgQueue<TAPSCommBuffer>	*iWriteQ;   // (not owned)
+    MQueueHandlerObserver	*iObserver; // (not owned)
+    TQueueHandlerType            iType;
+};
+
+/*
+ * Audio setting for CPjAudioEngine.
+ */
+class CPjAudioSetting
+{
+public:
+    TFourCC		 fourcc;
+    TAPSCodecMode	 mode;
+    TBool		 plc;
+    TBool		 vad;
+    TBool		 cng;
+    TBool		 loudspk;
+};
+
+/*
+ * Implementation: Symbian Input & Output Stream.
+ */
+class CPjAudioEngine : public CBase, MQueueHandlerObserver
+{
+public:
+    enum State
+    {
+	STATE_NULL,
+	STATE_INITIALIZING,
+	STATE_READY,
+	STATE_STREAMING,
+	STATE_PENDING_STOP
+    };
+
+    ~CPjAudioEngine();
+
+    static CPjAudioEngine *NewL(struct aps_stream *parent_strm,
+			        PjAudioCallback rec_cb,
+				PjAudioCallback play_cb,
+				void *user_data,
+				const CPjAudioSetting &setting);
+
+    TInt StartL();
+    void Stop();
+
+    TInt ActivateSpeaker(TBool active);
+    
+    TInt SetVolume(TInt vol) { return iSession.SetVolume(vol); }
+    TInt GetVolume() { return iSession.Volume(); }
+    TInt GetMaxVolume() { return iSession.MaxVolume(); }
+    
+    TInt SetGain(TInt gain) { return iSession.SetGain(gain); }
+    TInt GetGain() { return iSession.Gain(); }
+    TInt GetMaxGain() { return iSession.MaxGain(); }
+
+private:
+    CPjAudioEngine(struct aps_stream *parent_strm,
+		   PjAudioCallback rec_cb,
+		   PjAudioCallback play_cb,
+		   void *user_data,
+		   const CPjAudioSetting &setting);
+    void ConstructL();
+
+    TInt InitPlayL();
+    TInt InitRecL();
+    TInt StartStreamL();
+    void Deinit();
+
+    // Inherited from MQueueHandlerObserver
+    virtual void InputStreamInitialized(const TInt aStatus);
+    virtual void OutputStreamInitialized(const TInt aStatus);
+    virtual void NotifyError(const TInt aError);
+
+    TBool			 session_opened;
+    State			 state_;
+    struct aps_stream		*parentStrm_;
+    CPjAudioSetting		 setting_;
+
+    RAPSSession                  iSession;
+    TAPSInitSettings             iPlaySettings;
+    TAPSInitSettings             iRecSettings;
+
+    RMsgQueue<TAPSCommBuffer>    iReadQ;
+    RMsgQueue<TAPSCommBuffer>    iReadCommQ;
+    TBool			 readq_opened;
+    RMsgQueue<TAPSCommBuffer>    iWriteQ;
+    RMsgQueue<TAPSCommBuffer>    iWriteCommQ;
+    TBool			 writeq_opened;
+
+    CQueueHandler		*iPlayCommHandler;
+    CQueueHandler		*iRecCommHandler;
+    CQueueHandler		*iRecHandler;
+};
+
+
+CPjAudioEngine* CPjAudioEngine::NewL(struct aps_stream *parent_strm,
+				     PjAudioCallback rec_cb,
+				     PjAudioCallback play_cb,
+				     void *user_data,
+				     const CPjAudioSetting &setting)
+{
+    CPjAudioEngine* self = new (ELeave) CPjAudioEngine(parent_strm,
+						       rec_cb, play_cb,
+						       user_data,
+						       setting);
+    CleanupStack::PushL(self);
+    self->ConstructL();
+    CleanupStack::Pop(self);
+    return self;
+}
+
+CPjAudioEngine::CPjAudioEngine(struct aps_stream *parent_strm,
+			       PjAudioCallback rec_cb,
+			       PjAudioCallback play_cb,
+			       void *user_data,
+			       const CPjAudioSetting &setting)
+      : MQueueHandlerObserver(rec_cb, play_cb, user_data),
+        session_opened(EFalse),
+	state_(STATE_NULL),
+	parentStrm_(parent_strm),
+	setting_(setting),
+	readq_opened(EFalse),
+	writeq_opened(EFalse),
+	iPlayCommHandler(0),
+	iRecCommHandler(0),
+	iRecHandler(0)
+{
+}
+
+CPjAudioEngine::~CPjAudioEngine()
+{
+    Deinit();
+
+    TRACE_((THIS_FILE, "Sound device destroyed"));
+}
+
+TInt CPjAudioEngine::InitPlayL()
+{
+    TInt err = iSession.InitializePlayer(iPlaySettings);
+    if (err != KErrNone) {
+	Deinit();
+	snd_perror("Failed to initialize player", err);
+	return err;
+    }
+
+    // Open message queues for the output stream
+    TBuf<128> buf2 = iPlaySettings.iGlobal;
+    buf2.Append(_L("PlayQueue"));
+    TBuf<128> buf3 = iPlaySettings.iGlobal;
+    buf3.Append(_L("PlayCommQueue"));
+
+    while (iWriteQ.OpenGlobal(buf2))
+	User::After(10);
+    while (iWriteCommQ.OpenGlobal(buf3))
+	User::After(10);
+        
+    writeq_opened = ETrue;
+
+    // Construct message queue handler
+    iPlayCommHandler = CQueueHandler::NewL(this, &iWriteCommQ, &iWriteQ,
+					   CQueueHandler::EPlayCommQueue);
+
+    // Start observing APS callbacks on output stream message queue
+    iPlayCommHandler->Start();
+
+    return 0;
+}
+
+TInt CPjAudioEngine::InitRecL()
+{
+    // Initialize input stream device
+    TInt err = iSession.InitializeRecorder(iRecSettings);
+    if (err != KErrNone && err != KErrAlreadyExists) {
+	Deinit();
+	snd_perror("Failed to initialize recorder", err);
+	return err;
+    }
+
+    TBuf<128> buf1 = iRecSettings.iGlobal;
+    buf1.Append(_L("RecordQueue"));
+    TBuf<128> buf4 = iRecSettings.iGlobal;
+    buf4.Append(_L("RecordCommQueue"));
+
+    // Must wait for APS thread to finish creating message queues
+    // before we can open and use them.
+    while (iReadQ.OpenGlobal(buf1))
+	User::After(10);
+    while (iReadCommQ.OpenGlobal(buf4))
+	User::After(10);
+
+    readq_opened = ETrue;
+
+    // Construct message queue handlers
+    iRecHandler = CQueueHandler::NewL(this, &iReadQ, NULL,
+				      CQueueHandler::ERecordQueue);
+    iRecCommHandler = CQueueHandler::NewL(this, &iReadCommQ, NULL,
+					  CQueueHandler::ERecordCommQueue);
+
+    // Start observing APS callbacks from on input stream message queue
+    iRecHandler->Start();
+    iRecCommHandler->Start();
+
+    return 0;
+}
+
+TInt CPjAudioEngine::StartL()
+{
+    if (state_ == STATE_READY)
+	return StartStreamL();
+
+    PJ_ASSERT_RETURN(state_ == STATE_NULL, PJMEDIA_EAUD_INVOP);
+    
+    if (!session_opened) {
+	TInt err = iSession.Connect();
+	if (err != KErrNone)
+	    return err;
+	session_opened = ETrue;
+    }
+
+    // Even if only capturer are opened, playback thread of APS Server need
+    // to be run(?). Since some messages will be delivered via play comm queue.
+    state_ = STATE_INITIALIZING;
+
+    return InitPlayL();
+}
+
+void CPjAudioEngine::Stop()
+{
+    if (state_ == STATE_STREAMING) {
+	iSession.Stop();
+	state_ = STATE_READY;
+	TRACE_((THIS_FILE, "Sound device stopped"));
+    } else if (state_ == STATE_INITIALIZING) {
+	// Initialization is on progress, so let's set the state to 
+	// STATE_PENDING_STOP to prevent it starting the stream.
+	state_ = STATE_PENDING_STOP;
+	
+	// Then wait until initialization done.
+	while (state_ != STATE_READY && state_ != STATE_NULL)
+	    pj_symbianos_poll(-1, 100);
+    }
+}
+
+void CPjAudioEngine::ConstructL()
+{
+    // Recorder settings
+    iRecSettings.iFourCC		= setting_.fourcc;
+    iRecSettings.iGlobal		= APP_UID;
+    iRecSettings.iPriority		= TMdaPriority(100);
+    iRecSettings.iPreference		= TMdaPriorityPreference(0x05210001);
+    iRecSettings.iSettings.iChannels	= EMMFMono;
+    iRecSettings.iSettings.iSampleRate	= EMMFSampleRate8000Hz;
+
+    // Player settings
+    iPlaySettings.iFourCC		= setting_.fourcc;
+    iPlaySettings.iGlobal		= APP_UID;
+    iPlaySettings.iPriority		= TMdaPriority(100);
+    iPlaySettings.iPreference		= TMdaPriorityPreference(0x05220001);
+    iPlaySettings.iSettings.iChannels	= EMMFMono;
+    iPlaySettings.iSettings.iSampleRate = EMMFSampleRate8000Hz;
+    iPlaySettings.iSettings.iVolume	= 0;
+
+    User::LeaveIfError(iSession.Connect());
+    session_opened = ETrue;
+}
+
+TInt CPjAudioEngine::StartStreamL()
+{
+    pj_assert(state_==STATE_READY || state_==STATE_INITIALIZING); 
+    
+    iSession.SetCng(setting_.cng);
+    iSession.SetVadMode(setting_.vad);
+    iSession.SetPlc(setting_.plc);
+    iSession.SetEncoderMode(setting_.mode);
+    iSession.SetDecoderMode(setting_.mode);
+    iSession.ActivateLoudspeaker(setting_.loudspk);
+
+    // Not only capture
+    if (parentStrm_->param.dir != PJMEDIA_DIR_CAPTURE) {
+	iSession.Write();
+	TRACE_((THIS_FILE, "Player started"));
+    }
+
+    // Not only playback
+    if (parentStrm_->param.dir != PJMEDIA_DIR_PLAYBACK) {
+	iSession.Read();
+	TRACE_((THIS_FILE, "Recorder started"));
+    }
+
+    state_ = STATE_STREAMING;
+    
+    return 0;
+}
+
+void CPjAudioEngine::Deinit()
+{
+    Stop();
+
+    delete iRecHandler;
+    delete iPlayCommHandler;
+    delete iRecCommHandler;
+
+    if (session_opened) {
+	enum { APS_CLOSE_WAIT_TIME = 200 }; /* in msecs */
+	
+	// On some devices, immediate closing after stopping may cause 
+	// APS server panic KERN-EXEC 0, so let's wait for sometime before
+	// closing the client session.
+	snd_wait(APS_CLOSE_WAIT_TIME);
+
+	iSession.Close();
+	session_opened = EFalse;
+    }
+
+    if (readq_opened) {
+	iReadQ.Close();
+	iReadCommQ.Close();
+	readq_opened = EFalse;
+    }
+
+    if (writeq_opened) {
+	iWriteQ.Close();
+	iWriteCommQ.Close();
+	writeq_opened = EFalse;
+    }
+
+    state_ = STATE_NULL;
+}
+
+void CPjAudioEngine::InputStreamInitialized(const TInt aStatus)
+{
+    TRACE_((THIS_FILE, "Recorder initialized, err=%d", aStatus));
+
+    if (aStatus == KErrNone) {
+	// Don't start the stream since Stop() has been requested. 
+	if (state_ != STATE_PENDING_STOP) {
+	    StartStreamL();
+	} else {
+	    state_ = STATE_READY;
+	}
+    } else {
+	Deinit();
+    }
+}
+
+void CPjAudioEngine::OutputStreamInitialized(const TInt aStatus)
+{
+    TRACE_((THIS_FILE, "Player initialized, err=%d", aStatus));
+
+    if (aStatus == KErrNone) {
+	if (parentStrm_->param.dir == PJMEDIA_DIR_PLAYBACK) {
+	    // Don't start the stream since Stop() has been requested.
+	    if (state_ != STATE_PENDING_STOP) {
+		StartStreamL();
+	    } else {
+		state_ = STATE_READY;
+	    }
+	} else
+	    InitRecL();
+    } else {
+	Deinit();
+    }
+}
+
+void CPjAudioEngine::NotifyError(const TInt aError)
+{
+    Deinit();
+    snd_perror("Error from CQueueHandler", aError);
+}
+
+TInt CPjAudioEngine::ActivateSpeaker(TBool active)
+{
+    if (state_ == STATE_READY || state_ == STATE_STREAMING) {
+        iSession.ActivateLoudspeaker(active);
+        TRACE_((THIS_FILE, "Loudspeaker turned %s", (active? "on":"off")));
+	return KErrNone;
+    }
+    return KErrNotReady;
+}
+
+/****************************************************************************
+ * Internal APS callbacks for PCM format
+ */
+
+static void RecCbPcm(TAPSCommBuffer &buf, void *user_data)
+{
+    struct aps_stream *strm = (struct aps_stream*) user_data;
+
+    /* Buffer has to contain normal speech. */
+    pj_assert(buf.iBuffer[0] == 1 && buf.iBuffer[1] == 0);
+
+    /* Detect the recorder G.711 frame size, player frame size will follow
+     * this recorder frame size.
+     */
+    if (aps_g711_frame_len == 0) {
+	aps_g711_frame_len = buf.iBuffer.Length() < 160? 80 : 160;
+	TRACE_((THIS_FILE, "Detected APS G.711 frame size = %u samples",
+		aps_g711_frame_len));
+    }
+
+    /* Decode APS buffer (coded in G.711) and put the PCM result into rec_buf.
+     * Whenever rec_buf is full, call parent stream callback.
+     */
+    unsigned samples_processed = 0;
+
+    while (samples_processed < aps_g711_frame_len) {
+	unsigned samples_to_process;
+	unsigned samples_req;
+
+	samples_to_process = aps_g711_frame_len - samples_processed;
+	samples_req = (strm->param.samples_per_frame /
+		       strm->param.channel_count /
+		       strm->resample_factor) -
+		      strm->rec_buf_len;
+	if (samples_to_process > samples_req)
+	    samples_to_process = samples_req;
+
+	pjmedia_ulaw_decode(&strm->rec_buf[strm->rec_buf_len],
+			    buf.iBuffer.Ptr() + 2 + samples_processed,
+			    samples_to_process);
+
+	strm->rec_buf_len += samples_to_process;
+	samples_processed += samples_to_process;
+
+	/* Buffer is full, time to call parent callback */
+	if (strm->rec_buf_len == strm->param.samples_per_frame / 
+				 strm->param.channel_count /
+				 strm->resample_factor) 
+	{
+	    pjmedia_frame f;
+
+	    /* Need to resample clock rate? */
+	    if (strm->rec_resample) {
+		unsigned resampled = 0;
+		
+		while (resampled < strm->rec_buf_len) {
+		    pjmedia_resample_run(strm->rec_resample, 
+				&strm->rec_buf[resampled],
+				strm->pcm_buf + 
+				resampled * strm->resample_factor);
+		    resampled += 80;
+		}
+		f.buf = strm->pcm_buf;
+	    } else {
+		f.buf = strm->rec_buf;
+	    }
+
+	    /* Need to convert channel count? */
+	    if (strm->param.channel_count != 1) {
+		pjmedia_convert_channel_1ton((pj_int16_t*)f.buf,
+					     (pj_int16_t*)f.buf,
+					     strm->param.channel_count,
+					     strm->param.samples_per_frame /
+					     strm->param.channel_count,
+					     0);
+	    }
+
+	    /* Call parent callback */
+	    f.type = PJMEDIA_FRAME_TYPE_AUDIO;
+	    f.size = strm->param.samples_per_frame << 1;
+	    strm->rec_cb(strm->user_data, &f);
+	    strm->rec_buf_len = 0;
+	}
+    }
+}
+
+static void PlayCbPcm(TAPSCommBuffer &buf, void *user_data)
+{
+    struct aps_stream *strm = (struct aps_stream*) user_data;
+    unsigned g711_frame_len = aps_g711_frame_len;
+
+    /* Init buffer attributes and header. */
+    buf.iCommand = CQueueHandler::EAPSPlayData;
+    buf.iStatus = 0;
+    buf.iBuffer.Zero();
+    buf.iBuffer.Append(1);
+    buf.iBuffer.Append(0);
+
+    /* Assume frame size is 10ms if frame size hasn't been known. */
+    if (g711_frame_len == 0)
+	g711_frame_len = 80;
+
+    /* Call parent stream callback to get PCM samples to play,
+     * encode the PCM samples into G.711 and put it into APS buffer.
+     */
+    unsigned samples_processed = 0;
+    
+    while (samples_processed < g711_frame_len) {
+	/* Need more samples to play, time to call parent callback */
+	if (strm->play_buf_len == 0) {
+	    pjmedia_frame f;
+	    unsigned samples_got;
+	    
+	    f.size = strm->param.samples_per_frame << 1;
+	    if (strm->play_resample || strm->param.channel_count != 1)
+		f.buf = strm->pcm_buf;
+	    else
+		f.buf = strm->play_buf;
+
+	    /* Call parent callback */
+	    strm->play_cb(strm->user_data, &f);
+	    if (f.type != PJMEDIA_FRAME_TYPE_AUDIO) {
+		pjmedia_zero_samples((pj_int16_t*)f.buf, 
+				     strm->param.samples_per_frame);
+	    }
+	    
+	    samples_got = strm->param.samples_per_frame / 
+			  strm->param.channel_count /
+			  strm->resample_factor;
+
+	    /* Need to convert channel count? */
+	    if (strm->param.channel_count != 1) {
+		pjmedia_convert_channel_nto1((pj_int16_t*)f.buf,
+					     (pj_int16_t*)f.buf,
+					     strm->param.channel_count,
+					     strm->param.samples_per_frame,
+					     PJ_FALSE,
+					     0);
+	    }
+
+	    /* Need to resample clock rate? */
+	    if (strm->play_resample) {
+		unsigned resampled = 0;
+		
+		while (resampled < samples_got) 
+		{
+		    pjmedia_resample_run(strm->play_resample, 
+				strm->pcm_buf + 
+				resampled * strm->resample_factor,
+				&strm->play_buf[resampled]);
+		    resampled += 80;
+		}
+	    }
+	    
+	    strm->play_buf_len = samples_got;
+	    strm->play_buf_start = 0;
+	}
+
+	unsigned tmp;
+
+	tmp = PJ_MIN(strm->play_buf_len, g711_frame_len - samples_processed);
+	pjmedia_ulaw_encode((pj_uint8_t*)&strm->play_buf[strm->play_buf_start],
+			    &strm->play_buf[strm->play_buf_start],
+			    tmp);
+	buf.iBuffer.Append((TUint8*)&strm->play_buf[strm->play_buf_start], tmp);
+	samples_processed += tmp;
+	strm->play_buf_len -= tmp;
+	strm->play_buf_start += tmp;
+    }
+}
+
+/****************************************************************************
+ * Internal APS callbacks for non-PCM format
+ */
+
+static void RecCb(TAPSCommBuffer &buf, void *user_data)
+{
+    struct aps_stream *strm = (struct aps_stream*) user_data;
+    pjmedia_frame_ext *frame = (pjmedia_frame_ext*) strm->rec_buf;
+    
+    switch(strm->param.ext_fmt.id) {
+    case PJMEDIA_FORMAT_AMR:
+	{
+	    const pj_uint8_t *p = (const pj_uint8_t*)buf.iBuffer.Ptr() + 1;
+	    unsigned len = buf.iBuffer.Length() - 1;
+	    
+	    pjmedia_frame_ext_append_subframe(frame, p, len << 3, 160);
+	    if (frame->samples_cnt == strm->param.samples_per_frame) {
+		frame->base.type = PJMEDIA_FRAME_TYPE_EXTENDED;
+		strm->rec_cb(strm->user_data, (pjmedia_frame*)frame);
+		frame->samples_cnt = 0;
+		frame->subframe_cnt = 0;
+	    }
+	}
+	break;
+	
+    case PJMEDIA_FORMAT_G729:
+	{
+	    /* Check if we got a normal or SID frame. */
+	    if (buf.iBuffer[0] != 0 || buf.iBuffer[1] != 0) {
+		enum { NORMAL_LEN = 22, SID_LEN = 8 };
+		TBitStream *bitstream = (TBitStream*)strm->strm_data;
+		unsigned src_len = buf.iBuffer.Length()- 2;
+		
+		pj_assert(src_len == NORMAL_LEN || src_len == SID_LEN);
+
+		const TDesC8& p = bitstream->CompressG729Frame(
+					    buf.iBuffer.Right(src_len), 
+					    src_len == SID_LEN);
+		
+		pjmedia_frame_ext_append_subframe(frame, p.Ptr(), 
+						  p.Length() << 3, 80);
+	    } else { /* We got null frame. */
+		pjmedia_frame_ext_append_subframe(frame, NULL, 0, 80);
+	    }
+	    
+	    if (frame->samples_cnt == strm->param.samples_per_frame) {
+		frame->base.type = PJMEDIA_FRAME_TYPE_EXTENDED;
+		strm->rec_cb(strm->user_data, (pjmedia_frame*)frame);
+		frame->samples_cnt = 0;
+		frame->subframe_cnt = 0;
+	    }
+	}
+	break;
+
+    case PJMEDIA_FORMAT_ILBC:
+	{
+	    unsigned samples_got;
+	    
+	    samples_got = strm->param.ext_fmt.bitrate == 15200? 160 : 240;
+	    
+	    /* Check if we got a normal frame. */
+	    if (buf.iBuffer[0] == 1 && buf.iBuffer[1] == 0) {
+		const pj_uint8_t *p = (const pj_uint8_t*)buf.iBuffer.Ptr() + 2;
+		unsigned len = buf.iBuffer.Length() - 2;
+		
+		pjmedia_frame_ext_append_subframe(frame, p, len << 3,
+						  samples_got);
+	    } else { /* We got null frame. */
+		pjmedia_frame_ext_append_subframe(frame, NULL, 0, samples_got);
+	    }
+	    
+	    if (frame->samples_cnt == strm->param.samples_per_frame) {
+		frame->base.type = PJMEDIA_FRAME_TYPE_EXTENDED;
+		strm->rec_cb(strm->user_data, (pjmedia_frame*)frame);
+		frame->samples_cnt = 0;
+		frame->subframe_cnt = 0;
+	    }
+	}
+	break;
+	
+    case PJMEDIA_FORMAT_PCMU:
+    case PJMEDIA_FORMAT_PCMA:
+	{
+	    unsigned samples_processed = 0;
+	    
+	    /* Make sure it is normal frame. */
+	    pj_assert(buf.iBuffer[0] == 1 && buf.iBuffer[1] == 0);
+
+	    /* Detect the recorder G.711 frame size, player frame size will 
+	     * follow this recorder frame size.
+	     */
+	    if (aps_g711_frame_len == 0) {
+		aps_g711_frame_len = buf.iBuffer.Length() < 160? 80 : 160;
+		TRACE_((THIS_FILE, "Detected APS G.711 frame size = %u samples",
+			aps_g711_frame_len));
+	    }
+	    
+	    /* Convert APS buffer format into pjmedia_frame_ext. Whenever 
+	     * samples count in the frame is equal to stream's samples per 
+	     * frame, call parent stream callback.
+	     */
+	    while (samples_processed < aps_g711_frame_len) {
+		unsigned tmp;
+		const pj_uint8_t *pb = (const pj_uint8_t*)buf.iBuffer.Ptr() +
+				       2 + samples_processed;
+    
+		tmp = PJ_MIN(strm->param.samples_per_frame - frame->samples_cnt,
+			     aps_g711_frame_len - samples_processed);
+		
+		pjmedia_frame_ext_append_subframe(frame, pb, tmp << 3, tmp);
+		samples_processed += tmp;
+    
+		if (frame->samples_cnt == strm->param.samples_per_frame) {
+		    frame->base.type = PJMEDIA_FRAME_TYPE_EXTENDED;
+		    strm->rec_cb(strm->user_data, (pjmedia_frame*)frame);
+		    frame->samples_cnt = 0;
+		    frame->subframe_cnt = 0;
+		}
+	    }
+	}
+	break;
+	
+    default:
+	break;
+    }
+}
+
+static void PlayCb(TAPSCommBuffer &buf, void *user_data)
+{
+    struct aps_stream *strm = (struct aps_stream*) user_data;
+    pjmedia_frame_ext *frame = (pjmedia_frame_ext*) strm->play_buf;
+
+    /* Init buffer attributes and header. */
+    buf.iCommand = CQueueHandler::EAPSPlayData;
+    buf.iStatus = 0;
+    buf.iBuffer.Zero();
+
+    switch(strm->param.ext_fmt.id) {
+    case PJMEDIA_FORMAT_AMR:
+	{
+	    if (frame->samples_cnt == 0) {
+		frame->base.type = PJMEDIA_FRAME_TYPE_EXTENDED;
+		strm->play_cb(strm->user_data, (pjmedia_frame*)frame);
+		pj_assert(frame->base.type==PJMEDIA_FRAME_TYPE_EXTENDED ||
+			  frame->base.type==PJMEDIA_FRAME_TYPE_NONE);
+	    }
+
+	    if (frame->base.type == PJMEDIA_FRAME_TYPE_EXTENDED) { 
+		pjmedia_frame_ext_subframe *sf;
+		unsigned samples_cnt;
+		
+		sf = pjmedia_frame_ext_get_subframe(frame, 0);
+		samples_cnt = frame->samples_cnt / frame->subframe_cnt;
+		
+		if (sf->data && sf->bitlen) {
+		    /* AMR header for APS is one byte, the format (may be!):
+		     * 0xxxxy00, where xxxx:frame type, y:not sure. 
+		     */
+		    unsigned len = (sf->bitlen+7)>>3;
+		    enum {SID_FT = 8 };
+		    pj_uint8_t amr_header = 4, ft = SID_FT;
+
+		    if (len >= pjmedia_codec_amrnb_framelen[0])
+			ft = pjmedia_codec_amr_get_mode2(PJ_TRUE, len);
+		    
+		    amr_header |= ft << 3;
+		    buf.iBuffer.Append(amr_header);
+		    
+		    buf.iBuffer.Append((TUint8*)sf->data, len);
+		} else {
+		    enum {NO_DATA_FT = 15 };
+		    pj_uint8_t amr_header = 4 | (NO_DATA_FT << 3);
+
+		    buf.iBuffer.Append(amr_header);
+		}
+
+		pjmedia_frame_ext_pop_subframes(frame, 1);
+	    
+	    } else { /* PJMEDIA_FRAME_TYPE_NONE */
+		enum {NO_DATA_FT = 15 };
+		pj_uint8_t amr_header = 4 | (NO_DATA_FT << 3);
+
+		buf.iBuffer.Append(amr_header);
+
+		frame->samples_cnt = 0;
+		frame->subframe_cnt = 0;
+	    }
+	}
+	break;
+	
+    case PJMEDIA_FORMAT_G729:
+	{
+	    if (frame->samples_cnt == 0) {
+		frame->base.type = PJMEDIA_FRAME_TYPE_EXTENDED;
+		strm->play_cb(strm->user_data, (pjmedia_frame*)frame);
+		pj_assert(frame->base.type==PJMEDIA_FRAME_TYPE_EXTENDED ||
+			  frame->base.type==PJMEDIA_FRAME_TYPE_NONE);
+	    }
+
+	    if (frame->base.type == PJMEDIA_FRAME_TYPE_EXTENDED) { 
+		pjmedia_frame_ext_subframe *sf;
+		unsigned samples_cnt;
+		
+		sf = pjmedia_frame_ext_get_subframe(frame, 0);
+		samples_cnt = frame->samples_cnt / frame->subframe_cnt;
+		
+		if (sf->data && sf->bitlen) {
+		    enum { NORMAL_LEN = 10, SID_LEN = 2 };
+		    pj_bool_t sid_frame = ((sf->bitlen >> 3) == SID_LEN);
+		    TBitStream *bitstream = (TBitStream*)strm->strm_data;
+		    const TPtrC8 src(sf->data, sf->bitlen>>3);
+		    const TDesC8 &dst = bitstream->ExpandG729Frame(src,
+								   sid_frame); 
+		    if (sid_frame) {
+			buf.iBuffer.Append(2);
+			buf.iBuffer.Append(0);
+		    } else {
+			buf.iBuffer.Append(1);
+			buf.iBuffer.Append(0);
+		    }
+		    buf.iBuffer.Append(dst);
+		} else {
+		    buf.iBuffer.Append(2);
+		    buf.iBuffer.Append(0);
+		    buf.iBuffer.AppendFill(0, 22);
+		}
+
+		pjmedia_frame_ext_pop_subframes(frame, 1);
+	    
+	    } else { /* PJMEDIA_FRAME_TYPE_NONE */
+	        buf.iBuffer.Append(2);
+	        buf.iBuffer.Append(0);
+	        buf.iBuffer.AppendFill(0, 22);
+		
+		frame->samples_cnt = 0;
+		frame->subframe_cnt = 0;
+	    }
+	}
+	break;
+	
+    case PJMEDIA_FORMAT_ILBC:
+	{
+	    if (frame->samples_cnt == 0) {
+		frame->base.type = PJMEDIA_FRAME_TYPE_EXTENDED;
+		strm->play_cb(strm->user_data, (pjmedia_frame*)frame);
+		pj_assert(frame->base.type==PJMEDIA_FRAME_TYPE_EXTENDED ||
+			  frame->base.type==PJMEDIA_FRAME_TYPE_NONE);
+	    }
+
+	    if (frame->base.type == PJMEDIA_FRAME_TYPE_EXTENDED) { 
+		pjmedia_frame_ext_subframe *sf;
+		unsigned samples_cnt;
+		
+		sf = pjmedia_frame_ext_get_subframe(frame, 0);
+		samples_cnt = frame->samples_cnt / frame->subframe_cnt;
+		
+		pj_assert((strm->param.ext_fmt.bitrate == 15200 && 
+			   samples_cnt == 160) ||
+			  (strm->param.ext_fmt.bitrate != 15200 &&
+			   samples_cnt == 240));
+		
+		if (sf->data && sf->bitlen) {
+		    buf.iBuffer.Append(1);
+		    buf.iBuffer.Append(0);
+		    buf.iBuffer.Append((TUint8*)sf->data, sf->bitlen>>3);
+		} else {
+		    buf.iBuffer.Append(0);
+		    buf.iBuffer.Append(0);
+		}
+
+		pjmedia_frame_ext_pop_subframes(frame, 1);
+	    
+	    } else { /* PJMEDIA_FRAME_TYPE_NONE */
+		buf.iBuffer.Append(0);
+		buf.iBuffer.Append(0);
+		
+		frame->samples_cnt = 0;
+		frame->subframe_cnt = 0;
+	    }
+	}
+	break;
+	
+    case PJMEDIA_FORMAT_PCMU:
+    case PJMEDIA_FORMAT_PCMA:
+	{
+	    unsigned samples_ready = 0;
+	    unsigned samples_req = aps_g711_frame_len;
+	    
+	    /* Assume frame size is 10ms if frame size hasn't been known. */
+	    if (samples_req == 0)
+		samples_req = 80;
+	    
+	    buf.iBuffer.Append(1);
+	    buf.iBuffer.Append(0);
+	    
+	    /* Call parent stream callback to get samples to play. */
+	    while (samples_ready < samples_req) {
+		if (frame->samples_cnt == 0) {
+		    frame->base.type = PJMEDIA_FRAME_TYPE_EXTENDED;
+		    strm->play_cb(strm->user_data, (pjmedia_frame*)frame);
+		    pj_assert(frame->base.type==PJMEDIA_FRAME_TYPE_EXTENDED ||
+			      frame->base.type==PJMEDIA_FRAME_TYPE_NONE);
+		}
+    
+		if (frame->base.type == PJMEDIA_FRAME_TYPE_EXTENDED) { 
+		    pjmedia_frame_ext_subframe *sf;
+		    unsigned samples_cnt;
+		    
+		    sf = pjmedia_frame_ext_get_subframe(frame, 0);
+		    samples_cnt = frame->samples_cnt / frame->subframe_cnt;
+		    if (sf->data && sf->bitlen) {
+			buf.iBuffer.Append((TUint8*)sf->data, sf->bitlen>>3);
+		    } else {
+			pj_uint8_t silc;
+			silc = (strm->param.ext_fmt.id==PJMEDIA_FORMAT_PCMU)?
+				pjmedia_linear2ulaw(0) : pjmedia_linear2alaw(0);
+			buf.iBuffer.AppendFill(silc, samples_cnt);
+		    }
+		    samples_ready += samples_cnt;
+		    
+		    pjmedia_frame_ext_pop_subframes(frame, 1);
+		
+		} else { /* PJMEDIA_FRAME_TYPE_NONE */
+		    pj_uint8_t silc;
+		    
+		    silc = (strm->param.ext_fmt.id==PJMEDIA_FORMAT_PCMU)?
+			    pjmedia_linear2ulaw(0) : pjmedia_linear2alaw(0);
+		    buf.iBuffer.AppendFill(silc, samples_req - samples_ready);
+
+		    samples_ready = samples_req;
+		    frame->samples_cnt = 0;
+		    frame->subframe_cnt = 0;
+		}
+	    }
+	}
+	break;
+	
+    default:
+	break;
+    }
+}
+
+
+/****************************************************************************
+ * Factory operations
+ */
+
+/*
+ * C compatible declaration of APS factory.
+ */
+PJ_BEGIN_DECL
+PJ_DECL(pjmedia_aud_dev_factory*) pjmedia_aps_factory(pj_pool_factory *pf);
+PJ_END_DECL
+
+/*
+ * Init APS audio driver.
+ */
+PJ_DEF(pjmedia_aud_dev_factory*) pjmedia_aps_factory(pj_pool_factory *pf)
+{
+    struct aps_factory *f;
+    pj_pool_t *pool;
+
+    pool = pj_pool_create(pf, "APS", 1000, 1000, NULL);
+    f = PJ_POOL_ZALLOC_T(pool, struct aps_factory);
+    f->pf = pf;
+    f->pool = pool;
+    f->base.op = &factory_op;
+
+    return &f->base;
+}
+
+/* API: init factory */
+static pj_status_t factory_init(pjmedia_aud_dev_factory *f)
+{
+    struct aps_factory *af = (struct aps_factory*)f;
+
+    pj_ansi_strcpy(af->dev_info.name, "S60 APS");
+    af->dev_info.default_samples_per_sec = 8000;
+    af->dev_info.caps = PJMEDIA_AUD_DEV_CAP_EXT_FORMAT |
+			//PJMEDIA_AUD_DEV_CAP_INPUT_VOLUME_SETTING |
+			PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING |
+			PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE |
+			PJMEDIA_AUD_DEV_CAP_VAD |
+			PJMEDIA_AUD_DEV_CAP_CNG;
+    af->dev_info.routes = PJMEDIA_AUD_DEV_ROUTE_EARPIECE | 
+			  PJMEDIA_AUD_DEV_ROUTE_LOUDSPEAKER;
+    af->dev_info.input_count = 1;
+    af->dev_info.output_count = 1;
+
+    /* Enumerate codecs by trying to initialize each codec and examining
+     * the error code. Consider the following:
+     * - not possible to reinitialize the same APS session with 
+     *   different settings,
+     * - closing APS session and trying to immediately reconnect may fail,
+     *   clients should wait ~5s before attempting to reconnect.
+     */
+
+    unsigned i, fmt_cnt = 0;
+    pj_bool_t g711_supported = PJ_FALSE;
+
+    /* Do not change the order! */
+    TFourCC fourcc[] = {
+	TFourCC(KMCPFourCCIdAMRNB),
+	TFourCC(KMCPFourCCIdG711),
+	TFourCC(KMCPFourCCIdG729),
+	TFourCC(KMCPFourCCIdILBC)
+    };
+
+    for (i = 0; i < PJ_ARRAY_SIZE(fourcc); ++i) {
+	pj_bool_t supported = PJ_FALSE;
+	unsigned retry_cnt = 0;
+	enum { MAX_RETRY = 3 }; 
+
+#if (PJMEDIA_AUDIO_DEV_SYMB_APS_DETECTS_CODEC == 0)
+	/* Codec detection is disabled */
+	supported = PJ_TRUE;
+#elif (PJMEDIA_AUDIO_DEV_SYMB_APS_DETECTS_CODEC == 1)
+	/* Minimal codec detection, AMR-NB and G.711 only */
+	if (i > 1) {
+	    /* If G.711 has been checked, skip G.729 and iLBC checks */
+	    retry_cnt = MAX_RETRY;
+	    supported = g711_supported;
+	}
+#endif
+	
+	while (!supported && ++retry_cnt <= MAX_RETRY) {
+	    RAPSSession iSession;
+	    TAPSInitSettings iPlaySettings;
+	    TAPSInitSettings iRecSettings;
+	    TInt err;
+
+	    // Recorder settings
+	    iRecSettings.iGlobal		= APP_UID;
+	    iRecSettings.iPriority		= TMdaPriority(100);
+	    iRecSettings.iPreference		= TMdaPriorityPreference(0x05210001);
+	    iRecSettings.iSettings.iChannels	= EMMFMono;
+	    iRecSettings.iSettings.iSampleRate	= EMMFSampleRate8000Hz;
+
+	    // Player settings
+	    iPlaySettings.iGlobal		= APP_UID;
+	    iPlaySettings.iPriority		= TMdaPriority(100);
+	    iPlaySettings.iPreference		= TMdaPriorityPreference(0x05220001);
+	    iPlaySettings.iSettings.iChannels	= EMMFMono;
+	    iPlaySettings.iSettings.iSampleRate = EMMFSampleRate8000Hz;
+
+	    iRecSettings.iFourCC = iPlaySettings.iFourCC = fourcc[i];
+
+	    err = iSession.Connect();
+	    if (err == KErrNone)
+		err = iSession.InitializePlayer(iPlaySettings);
+	    if (err == KErrNone)
+		err = iSession.InitializeRecorder(iRecSettings);
+	    
+	    // On some devices, immediate closing causes APS Server panic,
+	    // e.g: N95, so let's just wait for some time before closing.
+	    enum { APS_CLOSE_WAIT_TIME = 200 }; /* in msecs */
+	    snd_wait(APS_CLOSE_WAIT_TIME);
+	    
+	    iSession.Close();
+
+	    if (err == KErrNone) {
+		/* All fine, stop retyring */
+		supported = PJ_TRUE;
+	    }  else if (err == KErrAlreadyExists && retry_cnt < MAX_RETRY) {
+		/* Seems that the previous session is still arround,
+		 * let's wait before retrying.
+		 */
+		enum { RETRY_WAIT_TIME = 3000 }; /* in msecs */
+		snd_wait(RETRY_WAIT_TIME);
+	    } else {
+		/* Seems that this format is not supported */
+		retry_cnt = MAX_RETRY;
+	    }
+	}
+
+	if (supported) {
+	    switch(i) {
+	    case 0: /* AMRNB */
+		af->dev_info.ext_fmt[fmt_cnt].id = PJMEDIA_FORMAT_AMR;
+		af->dev_info.ext_fmt[fmt_cnt].bitrate = 7400;
+		af->dev_info.ext_fmt[fmt_cnt].vad = PJ_TRUE;
+		++fmt_cnt;
+		break;
+	    case 1: /* G.711 */
+		af->dev_info.ext_fmt[fmt_cnt].id = PJMEDIA_FORMAT_PCMU;
+		af->dev_info.ext_fmt[fmt_cnt].bitrate = 64000;
+		af->dev_info.ext_fmt[fmt_cnt].vad = PJ_FALSE;
+		++fmt_cnt;
+		af->dev_info.ext_fmt[fmt_cnt].id = PJMEDIA_FORMAT_PCMA;
+		af->dev_info.ext_fmt[fmt_cnt].bitrate = 64000;
+		af->dev_info.ext_fmt[fmt_cnt].vad = PJ_FALSE;
+		++fmt_cnt;
+		g711_supported = PJ_TRUE;
+		break;
+	    case 2: /* G.729 */
+		af->dev_info.ext_fmt[fmt_cnt].id = PJMEDIA_FORMAT_G729;
+		af->dev_info.ext_fmt[fmt_cnt].bitrate = 8000;
+		af->dev_info.ext_fmt[fmt_cnt].vad = PJ_FALSE;
+		++fmt_cnt;
+		break;
+	    case 3: /* iLBC */
+		af->dev_info.ext_fmt[fmt_cnt].id = PJMEDIA_FORMAT_ILBC;
+		af->dev_info.ext_fmt[fmt_cnt].bitrate = 13333;
+		af->dev_info.ext_fmt[fmt_cnt].vad = PJ_TRUE;
+		++fmt_cnt;
+		break;
+	    }
+	}
+    }
+    
+    af->dev_info.ext_fmt_cnt = fmt_cnt;
+
+    PJ_LOG(4, (THIS_FILE, "APS initialized"));
+
+    return PJ_SUCCESS;
+}
+
+/* API: destroy factory */
+static pj_status_t factory_destroy(pjmedia_aud_dev_factory *f)
+{
+    struct aps_factory *af = (struct aps_factory*)f;
+    pj_pool_t *pool = af->pool;
+
+    af->pool = NULL;
+    pj_pool_release(pool);
+
+    PJ_LOG(4, (THIS_FILE, "APS destroyed"));
+    
+    return PJ_SUCCESS;
+}
+
+/* API: refresh the device list */
+static pj_status_t factory_refresh(pjmedia_aud_dev_factory *f)
+{
+    PJ_UNUSED_ARG(f);
+    return PJ_ENOTSUP;
+}
+
+/* API: get number of devices */
+static unsigned factory_get_dev_count(pjmedia_aud_dev_factory *f)
+{
+    PJ_UNUSED_ARG(f);
+    return 1;
+}
+
+/* API: get device info */
+static pj_status_t factory_get_dev_info(pjmedia_aud_dev_factory *f, 
+					unsigned index,
+					pjmedia_aud_dev_info *info)
+{
+    struct aps_factory *af = (struct aps_factory*)f;
+
+    PJ_ASSERT_RETURN(index == 0, PJMEDIA_EAUD_INVDEV);
+
+    pj_memcpy(info, &af->dev_info, sizeof(*info));
+
+    return PJ_SUCCESS;
+}
+
+/* API: create default device parameter */
+static pj_status_t factory_default_param(pjmedia_aud_dev_factory *f,
+					 unsigned index,
+					 pjmedia_aud_param *param)
+{
+    struct aps_factory *af = (struct aps_factory*)f;
+
+    PJ_ASSERT_RETURN(index == 0, PJMEDIA_EAUD_INVDEV);
+
+    pj_bzero(param, sizeof(*param));
+    param->dir = PJMEDIA_DIR_CAPTURE_PLAYBACK;
+    param->rec_id = index;
+    param->play_id = index;
+    param->clock_rate = af->dev_info.default_samples_per_sec;
+    param->channel_count = 1;
+    param->samples_per_frame = af->dev_info.default_samples_per_sec * 20 / 1000;
+    param->bits_per_sample = BITS_PER_SAMPLE;
+    param->flags = PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE;
+    param->output_route = PJMEDIA_AUD_DEV_ROUTE_EARPIECE;
+
+    return PJ_SUCCESS;
+}
+
+
+/* API: create stream */
+static pj_status_t factory_create_stream(pjmedia_aud_dev_factory *f,
+					 const pjmedia_aud_param *param,
+					 pjmedia_aud_rec_cb rec_cb,
+					 pjmedia_aud_play_cb play_cb,
+					 void *user_data,
+					 pjmedia_aud_stream **p_aud_strm)
+{
+    struct aps_factory *af = (struct aps_factory*)f;
+    pj_pool_t *pool;
+    struct aps_stream *strm;
+
+    CPjAudioSetting aps_setting;
+    PjAudioCallback aps_rec_cb;
+    PjAudioCallback aps_play_cb;
+
+    /* Can only support 16bits per sample */
+    PJ_ASSERT_RETURN(param->bits_per_sample == BITS_PER_SAMPLE, PJ_EINVAL);
+
+    /* Supported clock rates:
+     * - for non-PCM format: 8kHz  
+     * - for PCM format: 8kHz and 16kHz  
+     */
+    PJ_ASSERT_RETURN(param->clock_rate == 8000 ||
+		     (param->clock_rate == 16000 && 
+		      param->ext_fmt.id == PJMEDIA_FORMAT_L16),
+		     PJ_EINVAL);
+
+    /* Supported channels number:
+     * - for non-PCM format: mono
+     * - for PCM format: mono and stereo  
+     */
+    PJ_ASSERT_RETURN(param->channel_count == 1 || 
+		     (param->channel_count == 2 &&
+		      param->ext_fmt.id == PJMEDIA_FORMAT_L16),
+		     PJ_EINVAL);
+
+    /* Create and Initialize stream descriptor */
+    pool = pj_pool_create(af->pf, "aps-dev", 1000, 1000, NULL);
+    PJ_ASSERT_RETURN(pool, PJ_ENOMEM);
+
+    strm = PJ_POOL_ZALLOC_T(pool, struct aps_stream);
+    strm->pool = pool;
+    strm->param = *param;
+
+    if (strm->param.flags & PJMEDIA_AUD_DEV_CAP_EXT_FORMAT == 0)
+	strm->param.ext_fmt.id = PJMEDIA_FORMAT_L16;
+	
+    /* Set audio engine fourcc. */
+    switch(strm->param.ext_fmt.id) {
+    case PJMEDIA_FORMAT_L16:
+    case PJMEDIA_FORMAT_PCMU:
+    case PJMEDIA_FORMAT_PCMA:
+	aps_setting.fourcc = TFourCC(KMCPFourCCIdG711);
+	break;
+    case PJMEDIA_FORMAT_AMR:
+	aps_setting.fourcc = TFourCC(KMCPFourCCIdAMRNB);
+	break;
+    case PJMEDIA_FORMAT_G729:
+	aps_setting.fourcc = TFourCC(KMCPFourCCIdG729);
+	break;
+    case PJMEDIA_FORMAT_ILBC:
+	aps_setting.fourcc = TFourCC(KMCPFourCCIdILBC);
+	break;
+    default:
+	aps_setting.fourcc = 0;
+	break;
+    }
+
+    /* Set audio engine mode. */
+    if (strm->param.ext_fmt.id == PJMEDIA_FORMAT_AMR)
+    {
+	aps_setting.mode = (TAPSCodecMode)strm->param.ext_fmt.bitrate;
+    } 
+    else if (strm->param.ext_fmt.id == PJMEDIA_FORMAT_PCMU ||
+	     strm->param.ext_fmt.id == PJMEDIA_FORMAT_L16 ||
+	    (strm->param.ext_fmt.id == PJMEDIA_FORMAT_ILBC  &&
+	     strm->param.ext_fmt.bitrate != 15200))
+    {
+	aps_setting.mode = EULawOr30ms;
+    } 
+    else if (strm->param.ext_fmt.id == PJMEDIA_FORMAT_PCMA ||
+	    (strm->param.ext_fmt.id == PJMEDIA_FORMAT_ILBC &&
+	     strm->param.ext_fmt.bitrate == 15200))
+    {
+	aps_setting.mode = EALawOr20ms;
+    }
+
+    /* Disable VAD on L16, G711, and also G729 (G729's VAD potentially 
+     * causes noise?).
+     */
+    if (strm->param.ext_fmt.id == PJMEDIA_FORMAT_PCMU ||
+	strm->param.ext_fmt.id == PJMEDIA_FORMAT_PCMA ||
+	strm->param.ext_fmt.id == PJMEDIA_FORMAT_L16 ||
+	strm->param.ext_fmt.id == PJMEDIA_FORMAT_G729)
+    {
+	aps_setting.vad = EFalse;
+    } else {
+	aps_setting.vad = strm->param.ext_fmt.vad;
+    }
+    
+    /* Set other audio engine attributes. */
+    aps_setting.plc = strm->param.plc_enabled;
+    aps_setting.cng = aps_setting.vad;
+    aps_setting.loudspk = 
+		strm->param.output_route==PJMEDIA_AUD_DEV_ROUTE_LOUDSPEAKER;
+
+    /* Set audio engine callbacks. */
+    if (strm->param.ext_fmt.id == PJMEDIA_FORMAT_L16) {
+	aps_play_cb = &PlayCbPcm;
+	aps_rec_cb  = &RecCbPcm;
+    } else {
+	aps_play_cb = &PlayCb;
+	aps_rec_cb  = &RecCb;
+    }
+
+    strm->rec_cb = rec_cb;
+    strm->play_cb = play_cb;
+    strm->user_data = user_data;
+    strm->resample_factor = strm->param.clock_rate / 8000;
+
+    /* play_buf size is samples per frame scaled in to 8kHz mono. */
+    strm->play_buf = (pj_int16_t*)pj_pool_zalloc(
+					pool, 
+					(strm->param.samples_per_frame / 
+					strm->resample_factor /
+					strm->param.channel_count) << 1);
+    strm->play_buf_len = 0;
+    strm->play_buf_start = 0;
+
+    /* rec_buf size is samples per frame scaled in to 8kHz mono. */
+    strm->rec_buf  = (pj_int16_t*)pj_pool_zalloc(
+					pool, 
+					(strm->param.samples_per_frame / 
+					strm->resample_factor /
+					strm->param.channel_count) << 1);
+    strm->rec_buf_len = 0;
+
+    if (strm->param.ext_fmt.id == PJMEDIA_FORMAT_G729) {
+	TBitStream *g729_bitstream = new TBitStream;
+	
+	PJ_ASSERT_RETURN(g729_bitstream, PJ_ENOMEM);
+	strm->strm_data = (void*)g729_bitstream;
+    }
+	
+    /* Init resampler when format is PCM and clock rate is not 8kHz */
+    if (strm->param.clock_rate != 8000 && 
+	strm->param.ext_fmt.id == PJMEDIA_FORMAT_L16)
+    {
+	pj_status_t status;
+	
+	if (strm->param.dir & PJMEDIA_DIR_CAPTURE) {
+	    /* Create resample for recorder */
+	    status = pjmedia_resample_create( pool, PJ_TRUE, PJ_FALSE, 1, 
+					      8000,
+					      strm->param.clock_rate,
+					      80,
+					      &strm->rec_resample);
+	    if (status != PJ_SUCCESS)
+		return status;
+	}
+    
+	if (strm->param.dir & PJMEDIA_DIR_PLAYBACK) {
+	    /* Create resample for player */
+	    status = pjmedia_resample_create( pool, PJ_TRUE, PJ_FALSE, 1, 
+					      strm->param.clock_rate,
+					      8000,
+					      80 * strm->resample_factor,
+					      &strm->play_resample);
+	    if (status != PJ_SUCCESS)
+		return status;
+	}
+    }
+
+    /* Create PCM buffer, when the clock rate is not 8kHz or not mono */
+    if (strm->param.ext_fmt.id == PJMEDIA_FORMAT_L16 &&
+	(strm->resample_factor > 1 || strm->param.channel_count != 1)) 
+    {
+	strm->pcm_buf = (pj_int16_t*)pj_pool_zalloc(pool, 
+					strm->param.samples_per_frame << 1);
+    }
+
+    
+    /* Create the audio engine. */
+    TRAPD(err, strm->engine = CPjAudioEngine::NewL(strm,
+						   aps_rec_cb, aps_play_cb,
+						   strm, aps_setting));
+    if (err != KErrNone) {
+    	pj_pool_release(pool);
+	return PJ_RETURN_OS_ERROR(err);
+    }
+
+    /* Apply output volume setting if specified */
+    if (param->flags & PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING) {
+	stream_set_cap(&strm->base, PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING, 
+		       &param->output_vol);
+    }
+
+    /* Done */
+    strm->base.op = &stream_op;
+    *p_aud_strm = &strm->base;
+
+    return PJ_SUCCESS;
+}
+
+/* API: Get stream info. */
+static pj_status_t stream_get_param(pjmedia_aud_stream *s,
+				    pjmedia_aud_param *pi)
+{
+    struct aps_stream *strm = (struct aps_stream*)s;
+
+    PJ_ASSERT_RETURN(strm && pi, PJ_EINVAL);
+
+    pj_memcpy(pi, &strm->param, sizeof(*pi));
+
+    /* Update the output volume setting */
+    if (stream_get_cap(s, PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING,
+		       &pi->output_vol) == PJ_SUCCESS)
+    {
+	pi->flags |= PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING;
+    }
+    
+    return PJ_SUCCESS;
+}
+
+/* API: get capability */
+static pj_status_t stream_get_cap(pjmedia_aud_stream *s,
+				  pjmedia_aud_dev_cap cap,
+				  void *pval)
+{
+    struct aps_stream *strm = (struct aps_stream*)s;
+    pj_status_t status = PJ_ENOTSUP;
+
+    PJ_ASSERT_RETURN(s && pval, PJ_EINVAL);
+
+    switch (cap) {
+    case PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE: 
+	if (strm->param.dir & PJMEDIA_DIR_PLAYBACK) {
+	    *(pjmedia_aud_dev_route*)pval = strm->param.output_route;
+	    status = PJ_SUCCESS;
+	}
+	break;
+    
+    /* There is a case that GetMaxGain() stucks, e.g: in N95. */ 
+    /*
+    case PJMEDIA_AUD_DEV_CAP_INPUT_VOLUME_SETTING:
+	if (strm->param.dir & PJMEDIA_DIR_CAPTURE) {
+	    PJ_ASSERT_RETURN(strm->engine, PJ_EINVAL);
+	    
+	    TInt max_gain = strm->engine->GetMaxGain();
+	    TInt gain = strm->engine->GetGain();
+	    
+	    if (max_gain > 0 && gain >= 0) {
+		*(unsigned*)pval = gain * 100 / max_gain; 
+		status = PJ_SUCCESS;
+	    } else {
+		status = PJMEDIA_EAUD_NOTREADY;
+	    }
+	}
+	break;
+    */
+
+    case PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING:
+	if (strm->param.dir & PJMEDIA_DIR_PLAYBACK) {
+	    PJ_ASSERT_RETURN(strm->engine, PJ_EINVAL);
+	    
+	    TInt max_vol = strm->engine->GetMaxVolume();
+	    TInt vol = strm->engine->GetVolume();
+	    
+	    if (max_vol > 0 && vol >= 0) {
+		*(unsigned*)pval = vol * 100 / max_vol; 
+		status = PJ_SUCCESS;
+	    } else {
+		status = PJMEDIA_EAUD_NOTREADY;
+	    }
+	}
+	break;
+    default:
+	break;
+    }
+    
+    return status;
+}
+
+/* API: set capability */
+static pj_status_t stream_set_cap(pjmedia_aud_stream *s,
+				  pjmedia_aud_dev_cap cap,
+				  const void *pval)
+{
+    struct aps_stream *strm = (struct aps_stream*)s;
+    pj_status_t status = PJ_ENOTSUP;
+
+    PJ_ASSERT_RETURN(s && pval, PJ_EINVAL);
+
+    switch (cap) {
+    case PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE: 
+	if (strm->param.dir & PJMEDIA_DIR_PLAYBACK) {
+	    pjmedia_aud_dev_route r = *(const pjmedia_aud_dev_route*)pval;
+	    TInt err;
+
+	    PJ_ASSERT_RETURN(strm->engine, PJ_EINVAL);
+	    
+	    switch (r) {
+	    case PJMEDIA_AUD_DEV_ROUTE_DEFAULT:
+	    case PJMEDIA_AUD_DEV_ROUTE_EARPIECE:
+		err = strm->engine->ActivateSpeaker(EFalse);
+		status = (err==KErrNone)? PJ_SUCCESS:PJ_RETURN_OS_ERROR(err);
+		break;
+	    case PJMEDIA_AUD_DEV_ROUTE_LOUDSPEAKER:
+		err = strm->engine->ActivateSpeaker(ETrue);
+		status = (err==KErrNone)? PJ_SUCCESS:PJ_RETURN_OS_ERROR(err);
+		break;
+	    default:
+		status = PJ_EINVAL;
+		break;
+	    }
+	    if (status == PJ_SUCCESS)
+		strm->param.output_route = r; 
+	}
+	break;
+
+    /* There is a case that GetMaxGain() stucks, e.g: in N95. */ 
+    /*
+    case PJMEDIA_AUD_DEV_CAP_INPUT_VOLUME_SETTING:
+	if (strm->param.dir & PJMEDIA_DIR_CAPTURE) {
+	    PJ_ASSERT_RETURN(strm->engine, PJ_EINVAL);
+	    
+	    TInt max_gain = strm->engine->GetMaxGain();
+	    if (max_gain > 0) {
+		TInt gain, err;
+		
+		gain = *(unsigned*)pval * max_gain / 100;
+		err = strm->engine->SetGain(gain);
+		status = (err==KErrNone)? PJ_SUCCESS:PJ_RETURN_OS_ERROR(err);
+	    } else {
+		status = PJMEDIA_EAUD_NOTREADY;
+	    }
+	    if (status == PJ_SUCCESS)
+		strm->param.input_vol = *(unsigned*)pval;
+	}
+	break;
+    */
+
+    case PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING:
+	if (strm->param.dir & PJMEDIA_DIR_PLAYBACK) {
+	    PJ_ASSERT_RETURN(strm->engine, PJ_EINVAL);
+	    
+	    TInt max_vol = strm->engine->GetMaxVolume();
+	    if (max_vol > 0) {
+		TInt vol, err;
+		
+		vol = *(unsigned*)pval * max_vol / 100;
+		err = strm->engine->SetVolume(vol);
+		status = (err==KErrNone)? PJ_SUCCESS:PJ_RETURN_OS_ERROR(err);
+	    } else {
+		status = PJMEDIA_EAUD_NOTREADY;
+	    }
+	    if (status == PJ_SUCCESS)
+		strm->param.output_vol = *(unsigned*)pval;
+	}
+	break;
+    default:
+	break;
+    }
+    
+    return status;
+}
+
+/* API: Start stream. */
+static pj_status_t stream_start(pjmedia_aud_stream *strm)
+{
+    struct aps_stream *stream = (struct aps_stream*)strm;
+
+    PJ_ASSERT_RETURN(stream, PJ_EINVAL);
+
+    if (stream->engine) {
+	TInt err = stream->engine->StartL();
+    	if (err != KErrNone)
+    	    return PJ_RETURN_OS_ERROR(err);
+    }
+
+    return PJ_SUCCESS;
+}
+
+/* API: Stop stream. */
+static pj_status_t stream_stop(pjmedia_aud_stream *strm)
+{
+    struct aps_stream *stream = (struct aps_stream*)strm;
+
+    PJ_ASSERT_RETURN(stream, PJ_EINVAL);
+
+    if (stream->engine) {
+    	stream->engine->Stop();
+    }
+
+    return PJ_SUCCESS;
+}
+
+
+/* API: Destroy stream. */
+static pj_status_t stream_destroy(pjmedia_aud_stream *strm)
+{
+    struct aps_stream *stream = (struct aps_stream*)strm;
+
+    PJ_ASSERT_RETURN(stream, PJ_EINVAL);
+
+    stream_stop(strm);
+
+    delete stream->engine;
+    stream->engine = NULL;
+
+    if (stream->param.ext_fmt.id == PJMEDIA_FORMAT_G729) {
+	TBitStream *g729_bitstream = (TBitStream*)stream->strm_data;
+	stream->strm_data = NULL;
+	delete g729_bitstream;
+    }
+
+    pj_pool_t *pool;
+    pool = stream->pool;
+    if (pool) {
+    	stream->pool = NULL;
+    	pj_pool_release(pool);
+    }
+
+    return PJ_SUCCESS;
+}
+
+#endif // PJMEDIA_AUDIO_DEV_HAS_SYMB_APS
+