diff --git a/jni/pjproject-android/.svn/pristine/f8/f8c0405be84da517713c762ea3f4fb62ad6065d7.svn-base b/jni/pjproject-android/.svn/pristine/f8/f8c0405be84da517713c762ea3f4fb62ad6065d7.svn-base
new file mode 100644
index 0000000..a77340b
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/f8/f8c0405be84da517713c762ea3f4fb62ad6065d7.svn-base
@@ -0,0 +1,949 @@
+/* $Id$ */
+/* 
+ * Copyright (C) 2012-2012 Teluu Inc. (http://www.teluu.com)
+ * Copyright (C) 2010-2012 Regis Montoya (aka r3gis - www.r3gis.fr)
+ *
+ * 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 
+ */
+/* This file is the implementation of Android OpenSL ES audio device.
+ * The original code was originally part of CSipSimple
+ * (http://code.google.com/p/csipsimple/) and was kindly donated
+ * by Regis Montoya.
+ */
+
+#include <pjmedia-audiodev/audiodev_imp.h>
+#include <pj/assert.h>
+#include <pj/log.h>
+#include <pj/os.h>
+#include <pj/string.h>
+#include <pjmedia/errno.h>
+
+#if defined(PJMEDIA_AUDIO_DEV_HAS_OPENSL) && PJMEDIA_AUDIO_DEV_HAS_OPENSL != 0
+
+#include <SLES/OpenSLES.h>
+
+#ifdef __ANDROID__
+    #include <SLES/OpenSLES_Android.h>
+    #include <SLES/OpenSLES_AndroidConfiguration.h>
+    #include <sys/system_properties.h>
+    #include <android/api-level.h>
+
+    #define W_SLBufferQueueItf SLAndroidSimpleBufferQueueItf
+    #define W_SLBufferQueueState SLAndroidSimpleBufferQueueState
+    #define W_SL_IID_BUFFERQUEUE SL_IID_ANDROIDSIMPLEBUFFERQUEUE
+#else
+    #define W_SLBufferQueueItf SLBufferQueueItf
+    #define W_SLBufferQueueState SLBufferQueueState
+    #define W_SL_IID_BUFFERQUEUE SL_IID_BUFFERQUEUE
+#endif
+
+#define THIS_FILE	"opensl_dev.c"
+#define DRIVER_NAME	"OpenSL"
+
+#define NUM_BUFFERS 2
+
+struct opensl_aud_factory
+{
+    pjmedia_aud_dev_factory  base;
+    pj_pool_factory         *pf;
+    pj_pool_t               *pool;
+    
+    SLObjectItf              engineObject;
+    SLEngineItf              engineEngine;
+    SLObjectItf              outputMixObject;
+};
+
+/*
+ * Sound stream descriptor.
+ * This struct may be used for both unidirectional or bidirectional sound
+ * streams.
+ */
+struct opensl_aud_stream
+{
+    pjmedia_aud_stream  base;
+    pj_pool_t          *pool;
+    pj_str_t            name;
+    pjmedia_dir         dir;
+    pjmedia_aud_param   param;
+    
+    void               *user_data;
+    pj_bool_t           quit_flag;
+    pjmedia_aud_rec_cb  rec_cb;
+    pjmedia_aud_play_cb play_cb;
+
+    pj_timestamp	play_timestamp;
+    pj_timestamp	rec_timestamp;
+    
+    pj_bool_t		rec_thread_initialized;
+    pj_thread_desc	rec_thread_desc;
+    pj_thread_t        *rec_thread;
+    
+    pj_bool_t		play_thread_initialized;
+    pj_thread_desc	play_thread_desc;
+    pj_thread_t        *play_thread;
+    
+    /* Player */
+    SLObjectItf         playerObj;
+    SLPlayItf           playerPlay;
+    SLVolumeItf         playerVol;
+    unsigned            playerBufferSize;
+    char               *playerBuffer[NUM_BUFFERS];
+    int                 playerBufIdx;
+    
+    /* Recorder */
+    SLObjectItf         recordObj;
+    SLRecordItf         recordRecord;
+    unsigned            recordBufferSize;
+    char               *recordBuffer[NUM_BUFFERS];
+    int                 recordBufIdx;
+
+    W_SLBufferQueueItf  playerBufQ;
+    W_SLBufferQueueItf  recordBufQ;
+};
+
+/* Factory prototypes */
+static pj_status_t opensl_init(pjmedia_aud_dev_factory *f);
+static pj_status_t opensl_destroy(pjmedia_aud_dev_factory *f);
+static pj_status_t opensl_refresh(pjmedia_aud_dev_factory *f);
+static unsigned opensl_get_dev_count(pjmedia_aud_dev_factory *f);
+static pj_status_t opensl_get_dev_info(pjmedia_aud_dev_factory *f,
+                                       unsigned index,
+                                       pjmedia_aud_dev_info *info);
+static pj_status_t opensl_default_param(pjmedia_aud_dev_factory *f,
+                                        unsigned index,
+                                        pjmedia_aud_param *param);
+static pj_status_t opensl_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);
+
+/* Stream prototypes */
+static pj_status_t strm_get_param(pjmedia_aud_stream *strm,
+                                  pjmedia_aud_param *param);
+static pj_status_t strm_get_cap(pjmedia_aud_stream *strm,
+                                pjmedia_aud_dev_cap cap,
+                                void *value);
+static pj_status_t strm_set_cap(pjmedia_aud_stream *strm,
+                                pjmedia_aud_dev_cap cap,
+                                const void *value);
+static pj_status_t strm_start(pjmedia_aud_stream *strm);
+static pj_status_t strm_stop(pjmedia_aud_stream *strm);
+static pj_status_t strm_destroy(pjmedia_aud_stream *strm);
+
+static pjmedia_aud_dev_factory_op opensl_op =
+{
+    &opensl_init,
+    &opensl_destroy,
+    &opensl_get_dev_count,
+    &opensl_get_dev_info,
+    &opensl_default_param,
+    &opensl_create_stream,
+    &opensl_refresh
+};
+
+static pjmedia_aud_stream_op opensl_strm_op =
+{
+    &strm_get_param,
+    &strm_get_cap,
+    &strm_set_cap,
+    &strm_start,
+    &strm_stop,
+    &strm_destroy
+};
+
+/* This callback is called every time a buffer finishes playing. */
+void bqPlayerCallback(W_SLBufferQueueItf bq, void *context)
+{
+    struct opensl_aud_stream *stream = (struct opensl_aud_stream*) context;
+    SLresult result;
+    int status;
+
+    pj_assert(context != NULL);
+    pj_assert(bq == stream->playerBufQ);
+
+    if (stream->play_thread_initialized == 0 || !pj_thread_is_registered())
+    {
+	pj_bzero(stream->play_thread_desc, sizeof(pj_thread_desc));
+	status = pj_thread_register("opensl_play", stream->play_thread_desc,
+				    &stream->play_thread);
+	stream->play_thread_initialized = 1;
+	PJ_LOG(5, (THIS_FILE, "Player thread started"));
+    }
+    
+    if (!stream->quit_flag) {
+        pjmedia_frame frame;
+        char * buf;
+        
+        frame.type = PJMEDIA_FRAME_TYPE_AUDIO;
+        frame.buf = buf = stream->playerBuffer[stream->playerBufIdx++];
+        frame.size = stream->playerBufferSize;
+        frame.timestamp.u64 = stream->play_timestamp.u64;
+        frame.bit_info = 0;
+        
+        status = (*stream->play_cb)(stream->user_data, &frame);
+        if (status != PJ_SUCCESS || frame.type != PJMEDIA_FRAME_TYPE_AUDIO)
+            pj_bzero(buf, stream->playerBufferSize);
+        
+        stream->play_timestamp.u64 += stream->param.samples_per_frame /
+                                      stream->param.channel_count;
+        
+        result = (*bq)->Enqueue(bq, buf, stream->playerBufferSize);
+        if (result != SL_RESULT_SUCCESS) {
+            PJ_LOG(3, (THIS_FILE, "Unable to enqueue next player buffer !!! %d",
+                                  result));
+        }
+        
+        stream->playerBufIdx %= NUM_BUFFERS;
+    }
+}
+
+/* This callback handler is called every time a buffer finishes recording */
+void bqRecorderCallback(W_SLBufferQueueItf bq, void *context)
+{
+    struct opensl_aud_stream *stream = (struct opensl_aud_stream*) context;
+    SLresult result;
+    int status;
+
+    pj_assert(context != NULL);
+    pj_assert(bq == stream->recordBufQ);
+
+    if (stream->rec_thread_initialized == 0 || !pj_thread_is_registered())
+    {
+	pj_bzero(stream->rec_thread_desc, sizeof(pj_thread_desc));
+	status = pj_thread_register("opensl_rec", stream->rec_thread_desc,
+				    &stream->rec_thread);
+	stream->rec_thread_initialized = 1;
+	PJ_LOG(5, (THIS_FILE, "Recorder thread started")); 
+    }
+    
+    if (!stream->quit_flag) {
+        pjmedia_frame frame;
+        char *buf;
+        
+        frame.type = PJMEDIA_FRAME_TYPE_AUDIO;
+        frame.buf = buf = stream->recordBuffer[stream->recordBufIdx++];
+        frame.size = stream->recordBufferSize;
+        frame.timestamp.u64 = stream->rec_timestamp.u64;
+        frame.bit_info = 0;
+        
+        status = (*stream->rec_cb)(stream->user_data, &frame);
+        
+        stream->rec_timestamp.u64 += stream->param.samples_per_frame /
+                                     stream->param.channel_count;
+        
+        /* And now enqueue next buffer */
+        result = (*bq)->Enqueue(bq, buf, stream->recordBufferSize);
+        if (result != SL_RESULT_SUCCESS) {
+            PJ_LOG(3, (THIS_FILE, "Unable to enqueue next record buffer !!! %d",
+                                  result));
+        }
+        
+        stream->recordBufIdx %= NUM_BUFFERS;
+    }
+}
+
+pj_status_t opensl_to_pj_error(SLresult code)
+{
+    switch(code) {
+	case SL_RESULT_SUCCESS:
+            return PJ_SUCCESS;
+	case SL_RESULT_PRECONDITIONS_VIOLATED:
+	case SL_RESULT_PARAMETER_INVALID:
+	case SL_RESULT_CONTENT_CORRUPTED:
+	case SL_RESULT_FEATURE_UNSUPPORTED:
+            return PJMEDIA_EAUD_INVOP;
+	case SL_RESULT_MEMORY_FAILURE:
+	case SL_RESULT_BUFFER_INSUFFICIENT:
+            return PJ_ENOMEM;
+	case SL_RESULT_RESOURCE_ERROR:
+	case SL_RESULT_RESOURCE_LOST:
+	case SL_RESULT_CONTROL_LOST:
+            return PJMEDIA_EAUD_NOTREADY;
+	case SL_RESULT_CONTENT_UNSUPPORTED:
+            return PJ_ENOTSUP;
+	default:
+            return PJMEDIA_EAUD_ERR;
+    }
+}
+
+/* Init Android audio driver. */
+pjmedia_aud_dev_factory* pjmedia_opensl_factory(pj_pool_factory *pf)
+{
+    struct opensl_aud_factory *f;
+    pj_pool_t *pool;
+    
+    pool = pj_pool_create(pf, "opensles", 256, 256, NULL);
+    f = PJ_POOL_ZALLOC_T(pool, struct opensl_aud_factory);
+    f->pf = pf;
+    f->pool = pool;
+    f->base.op = &opensl_op;
+    
+    return &f->base;
+}
+
+/* API: Init factory */
+static pj_status_t opensl_init(pjmedia_aud_dev_factory *f)
+{
+    struct opensl_aud_factory *pa = (struct opensl_aud_factory*)f;
+    SLresult result;    
+    
+    /* Create engine */
+    result = slCreateEngine(&pa->engineObject, 0, NULL, 0, NULL, NULL);
+    if (result != SL_RESULT_SUCCESS) {
+        PJ_LOG(3, (THIS_FILE, "Cannot create engine %d ", result));
+        return opensl_to_pj_error(result);
+    }
+    
+    /* Realize the engine */
+    result = (*pa->engineObject)->Realize(pa->engineObject, SL_BOOLEAN_FALSE);
+    if (result != SL_RESULT_SUCCESS) {
+        PJ_LOG(3, (THIS_FILE, "Cannot realize engine"));
+        opensl_destroy(f);
+        return opensl_to_pj_error(result);
+    }
+    
+    /* Get the engine interface, which is needed in order to create
+     * other objects.
+     */
+    result = (*pa->engineObject)->GetInterface(pa->engineObject,
+                                               SL_IID_ENGINE,
+                                               &pa->engineEngine);
+    if (result != SL_RESULT_SUCCESS) {
+        PJ_LOG(3, (THIS_FILE, "Cannot get engine interface"));
+        opensl_destroy(f);
+        return opensl_to_pj_error(result);
+    }
+    
+    /* Create output mix */
+    result = (*pa->engineEngine)->CreateOutputMix(pa->engineEngine,
+                                                  &pa->outputMixObject,
+                                                  0, NULL, NULL);
+    if (result != SL_RESULT_SUCCESS) {
+        PJ_LOG(3, (THIS_FILE, "Cannot create output mix"));
+        opensl_destroy(f);
+        return opensl_to_pj_error(result);
+    }
+    
+    /* Realize the output mix */
+    result = (*pa->outputMixObject)->Realize(pa->outputMixObject,
+                                             SL_BOOLEAN_FALSE);
+    if (result != SL_RESULT_SUCCESS) {
+        PJ_LOG(3, (THIS_FILE, "Cannot realize output mix"));
+        opensl_destroy(f);
+        return opensl_to_pj_error(result);
+    }
+    
+    PJ_LOG(4,(THIS_FILE, "OpenSL sound library initialized"));
+    return PJ_SUCCESS;
+}
+
+/* API: Destroy factory */
+static pj_status_t opensl_destroy(pjmedia_aud_dev_factory *f)
+{
+    struct opensl_aud_factory *pa = (struct opensl_aud_factory*)f;
+    pj_pool_t *pool;
+    
+    PJ_LOG(4,(THIS_FILE, "OpenSL sound library shutting down.."));
+    
+    /* Destroy Output Mix object */
+    if (pa->outputMixObject) {
+        (*pa->outputMixObject)->Destroy(pa->outputMixObject);
+        pa->outputMixObject = NULL;
+    }
+    
+    /* Destroy engine object, and invalidate all associated interfaces */
+    if (pa->engineObject) {
+        (*pa->engineObject)->Destroy(pa->engineObject);
+        pa->engineObject = NULL;
+        pa->engineEngine = NULL;
+    }
+    
+    pool = pa->pool;
+    pa->pool = NULL;
+    pj_pool_release(pool);
+    
+    return PJ_SUCCESS;
+}
+
+/* API: refresh the list of devices */
+static pj_status_t opensl_refresh(pjmedia_aud_dev_factory *f)
+{
+    PJ_UNUSED_ARG(f);
+    return PJ_SUCCESS;
+}
+
+/* API: Get device count. */
+static unsigned opensl_get_dev_count(pjmedia_aud_dev_factory *f)
+{
+    PJ_UNUSED_ARG(f);
+    return 1;
+}
+
+/* API: Get device info. */
+static pj_status_t opensl_get_dev_info(pjmedia_aud_dev_factory *f,
+                                       unsigned index,
+                                       pjmedia_aud_dev_info *info)
+{
+    PJ_UNUSED_ARG(f);
+
+    pj_bzero(info, sizeof(*info));
+    
+    pj_ansi_strcpy(info->name, "OpenSL ES Audio");
+    info->default_samples_per_sec = 8000;
+    info->caps = PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING;
+    info->input_count = 1;
+    info->output_count = 1;
+    
+    return PJ_SUCCESS;
+}
+
+/* API: fill in with default parameter. */
+static pj_status_t opensl_default_param(pjmedia_aud_dev_factory *f,
+                                        unsigned index,
+                                        pjmedia_aud_param *param)
+{
+    
+    pjmedia_aud_dev_info adi;
+    pj_status_t status;
+    
+    status = opensl_get_dev_info(f, index, &adi);
+    if (status != PJ_SUCCESS)
+	return status;
+    
+    pj_bzero(param, sizeof(*param));
+    if (adi.input_count && adi.output_count) {
+        param->dir = PJMEDIA_DIR_CAPTURE_PLAYBACK;
+        param->rec_id = index;
+        param->play_id = index;
+    } else if (adi.input_count) {
+        param->dir = PJMEDIA_DIR_CAPTURE;
+        param->rec_id = index;
+        param->play_id = PJMEDIA_AUD_INVALID_DEV;
+    } else if (adi.output_count) {
+        param->dir = PJMEDIA_DIR_PLAYBACK;
+        param->play_id = index;
+        param->rec_id = PJMEDIA_AUD_INVALID_DEV;
+    } else {
+        return PJMEDIA_EAUD_INVDEV;
+    }
+    
+    param->clock_rate = adi.default_samples_per_sec;
+    param->channel_count = 1;
+    param->samples_per_frame = adi.default_samples_per_sec * 20 / 1000;
+    param->bits_per_sample = 16;
+    param->input_latency_ms = PJMEDIA_SND_DEFAULT_REC_LATENCY;
+    param->output_latency_ms = PJMEDIA_SND_DEFAULT_PLAY_LATENCY;
+    
+    return PJ_SUCCESS;
+}
+
+/* API: create stream */
+static pj_status_t opensl_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)
+{
+    /* Audio sink for recorder and audio source for player */
+#ifdef __ANDROID__
+    SLDataLocator_AndroidSimpleBufferQueue loc_bq =
+        { SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, NUM_BUFFERS };
+#else
+    SLDataLocator_BufferQueue loc_bq =
+        { SL_DATALOCATOR_BUFFERQUEUE, NUM_BUFFERS };
+#endif
+    struct opensl_aud_factory *pa = (struct opensl_aud_factory*)f;
+    pj_pool_t *pool;
+    struct opensl_aud_stream *stream;
+    pj_status_t status = PJ_SUCCESS;
+    int i, bufferSize;
+    SLresult result;
+    SLDataFormat_PCM format_pcm;
+    
+    /* Only supports for mono channel for now */
+    PJ_ASSERT_RETURN(param->channel_count == 1, PJ_EINVAL);
+    PJ_ASSERT_RETURN(play_cb && rec_cb && p_aud_strm, PJ_EINVAL);
+
+    PJ_LOG(4,(THIS_FILE, "Creating OpenSL stream"));
+    
+    pool = pj_pool_create(pa->pf, "openslstrm", 1024, 1024, NULL);
+    if (!pool)
+        return PJ_ENOMEM;
+    
+    stream = PJ_POOL_ZALLOC_T(pool, struct opensl_aud_stream);
+    stream->pool = pool;
+    pj_strdup2_with_null(pool, &stream->name, "OpenSL");
+    stream->dir = PJMEDIA_DIR_CAPTURE_PLAYBACK;
+    pj_memcpy(&stream->param, param, sizeof(*param));
+    stream->user_data = user_data;
+    stream->rec_cb = rec_cb;
+    stream->play_cb = play_cb;
+    bufferSize = param->samples_per_frame * param->bits_per_sample / 8;
+
+    /* Configure audio PCM format */
+    format_pcm.formatType = SL_DATAFORMAT_PCM;
+    format_pcm.numChannels = param->channel_count;
+    /* Here samples per sec should be supported else we will get an error */
+    format_pcm.samplesPerSec  = (SLuint32) param->clock_rate * 1000;
+    format_pcm.bitsPerSample = (SLuint16) param->bits_per_sample;
+    format_pcm.containerSize = (SLuint16) param->bits_per_sample;
+    format_pcm.channelMask = SL_SPEAKER_FRONT_CENTER;
+    format_pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
+
+    if (stream->dir & PJMEDIA_DIR_PLAYBACK) {
+        /* Audio source */
+        SLDataSource audioSrc = {&loc_bq, &format_pcm};
+        /* Audio sink */
+        SLDataLocator_OutputMix loc_outmix = {SL_DATALOCATOR_OUTPUTMIX,
+                                              pa->outputMixObject};
+        SLDataSink audioSnk = {&loc_outmix, NULL};
+        /* Audio interface */
+#ifdef __ANDROID__
+        int numIface = 3;
+        const SLInterfaceID ids[3] = {SL_IID_BUFFERQUEUE,
+                                      SL_IID_VOLUME,
+                                      SL_IID_ANDROIDCONFIGURATION};
+        const SLboolean req[3] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE,
+                                  SL_BOOLEAN_TRUE};
+        SLAndroidConfigurationItf playerConfig;
+        SLint32 streamType = SL_ANDROID_STREAM_VOICE;
+#else
+        int numIface = 2;
+        const SLInterfaceID ids[2] = {SL_IID_BUFFERQUEUE,
+                                      SL_IID_VOLUME};
+        const SLboolean req[2] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
+#endif
+        
+        /* Create audio player */
+        result = (*pa->engineEngine)->CreateAudioPlayer(pa->engineEngine,
+                                                        &stream->playerObj,
+                                                        &audioSrc, &audioSnk,
+                                                        numIface, ids, req);
+        if (result != SL_RESULT_SUCCESS) {
+            PJ_LOG(3, (THIS_FILE, "Cannot create audio player: %d", result));
+            goto on_error;
+        }
+
+#ifdef __ANDROID__
+        /* Set Android configuration */
+        result = (*stream->playerObj)->GetInterface(stream->playerObj,
+                                                    SL_IID_ANDROIDCONFIGURATION,
+                                                    &playerConfig);
+        if (result == SL_RESULT_SUCCESS && playerConfig) {
+            result = (*playerConfig)->SetConfiguration(
+                         playerConfig, SL_ANDROID_KEY_STREAM_TYPE,
+                         &streamType, sizeof(SLint32));
+        }
+        if (result != SL_RESULT_SUCCESS) {
+            PJ_LOG(4, (THIS_FILE, "Warning: Unable to set android "
+                                  "player configuration"));
+        }
+#endif
+
+        /* Realize the player */
+        result = (*stream->playerObj)->Realize(stream->playerObj,
+                                               SL_BOOLEAN_FALSE);
+        if (result != SL_RESULT_SUCCESS) {
+            PJ_LOG(3, (THIS_FILE, "Cannot realize player : %d", result));
+            goto on_error;
+        }
+        
+        /* Get the play interface */
+        result = (*stream->playerObj)->GetInterface(stream->playerObj,
+                                                    SL_IID_PLAY,
+                                                    &stream->playerPlay);
+        if (result != SL_RESULT_SUCCESS) {
+            PJ_LOG(3, (THIS_FILE, "Cannot get play interface"));
+            goto on_error;
+        }
+        
+        /* Get the buffer queue interface */
+        result = (*stream->playerObj)->GetInterface(stream->playerObj,
+                                                    SL_IID_BUFFERQUEUE,
+                                                    &stream->playerBufQ);
+        if (result != SL_RESULT_SUCCESS) {
+            PJ_LOG(3, (THIS_FILE, "Cannot get buffer queue interface"));
+            goto on_error;
+        }
+        
+        /* Get the volume interface */
+        result = (*stream->playerObj)->GetInterface(stream->playerObj,
+                                                    SL_IID_VOLUME,
+                                                    &stream->playerVol);
+        
+        /* Register callback on the buffer queue */
+        result = (*stream->playerBufQ)->RegisterCallback(stream->playerBufQ,
+                                                         bqPlayerCallback,
+                                                         (void *)stream);
+        if (result != SL_RESULT_SUCCESS) {
+            PJ_LOG(3, (THIS_FILE, "Cannot register player callback"));
+            goto on_error;
+        }
+        
+        stream->playerBufferSize = bufferSize;
+        for (i = 0; i < NUM_BUFFERS; i++) {
+            stream->playerBuffer[i] = (char *)
+                                      pj_pool_alloc(stream->pool,
+                                                    stream->playerBufferSize);
+        }
+    }
+
+    if (stream->dir & PJMEDIA_DIR_CAPTURE) {
+        /* Audio source */
+        SLDataLocator_IODevice loc_dev = {SL_DATALOCATOR_IODEVICE,
+                                          SL_IODEVICE_AUDIOINPUT,
+                                          SL_DEFAULTDEVICEID_AUDIOINPUT,
+                                          NULL};
+        SLDataSource audioSrc = {&loc_dev, NULL};
+        /* Audio sink */
+        SLDataSink audioSnk = {&loc_bq, &format_pcm};
+        /* Audio interface */
+#ifdef __ANDROID__
+        int numIface = 2;
+        const SLInterfaceID ids[2] = {W_SL_IID_BUFFERQUEUE,
+                                      SL_IID_ANDROIDCONFIGURATION};
+        const SLboolean req[2] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
+        SLAndroidConfigurationItf recorderConfig;
+#else
+        int numIface = 1;
+        const SLInterfaceID ids[1] = {W_SL_IID_BUFFERQUEUE};
+        const SLboolean req[1] = {SL_BOOLEAN_TRUE};
+#endif
+        
+        /* Create audio recorder
+         * (requires the RECORD_AUDIO permission)
+         */
+        result = (*pa->engineEngine)->CreateAudioRecorder(pa->engineEngine,
+                                                          &stream->recordObj,
+                                                          &audioSrc, &audioSnk,
+                                                          numIface, ids, req);
+        if (result != SL_RESULT_SUCCESS) {
+            PJ_LOG(3, (THIS_FILE, "Cannot create recorder: %d", result));
+            goto on_error;
+        }
+
+#ifdef __ANDROID__
+        /* Set Android configuration */
+        result = (*stream->recordObj)->GetInterface(stream->recordObj,
+                                                    SL_IID_ANDROIDCONFIGURATION,
+                                                    &recorderConfig);
+        if (result == SL_RESULT_SUCCESS) {
+            SLint32 streamType = SL_ANDROID_RECORDING_PRESET_GENERIC;
+#if __ANDROID_API__ >= 14
+            char sdk_version[PROP_VALUE_MAX];
+            pj_str_t pj_sdk_version;
+            int sdk_v;
+
+            __system_property_get("ro.build.version.sdk", sdk_version);
+            pj_sdk_version = pj_str(sdk_version);
+            sdk_v = pj_strtoul(&pj_sdk_version);
+            if (sdk_v >= 14)
+                streamType = SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION;
+            PJ_LOG(4, (THIS_FILE, "Recording stream type %d, SDK : %d",
+                                  streamType, sdk_v));
+#endif
+            result = (*recorderConfig)->SetConfiguration(
+                         recorderConfig, SL_ANDROID_KEY_RECORDING_PRESET,
+                         &streamType, sizeof(SLint32));
+        }
+        if (result != SL_RESULT_SUCCESS) {
+            PJ_LOG(4, (THIS_FILE, "Warning: Unable to set android "
+                                  "recorder configuration"));
+        }
+#endif
+        
+        /* Realize the recorder */
+        result = (*stream->recordObj)->Realize(stream->recordObj,
+                                               SL_BOOLEAN_FALSE);
+        if (result != SL_RESULT_SUCCESS) {
+            PJ_LOG(3, (THIS_FILE, "Cannot realize recorder : %d", result));
+            goto on_error;
+        }
+        
+        /* Get the record interface */
+        result = (*stream->recordObj)->GetInterface(stream->recordObj,
+                                                    SL_IID_RECORD,
+                                                    &stream->recordRecord);
+        if (result != SL_RESULT_SUCCESS) {
+            PJ_LOG(3, (THIS_FILE, "Cannot get record interface"));
+            goto on_error;
+        }
+        
+        /* Get the buffer queue interface */
+        result = (*stream->recordObj)->GetInterface(
+                     stream->recordObj, W_SL_IID_BUFFERQUEUE,
+                     &stream->recordBufQ);
+        if (result != SL_RESULT_SUCCESS) {
+            PJ_LOG(3, (THIS_FILE, "Cannot get recorder buffer queue iface"));
+            goto on_error;
+        }
+        
+        /* Register callback on the buffer queue */
+        result = (*stream->recordBufQ)->RegisterCallback(stream->recordBufQ,
+                                                         bqRecorderCallback, 
+                                                         (void *) stream);
+        if (result != SL_RESULT_SUCCESS) {
+            PJ_LOG(3, (THIS_FILE, "Cannot register recorder callback"));
+            goto on_error;
+        }
+        
+        stream->recordBufferSize = bufferSize;
+        for (i = 0; i < NUM_BUFFERS; i++) {
+            stream->recordBuffer[i] = (char *)
+                                      pj_pool_alloc(stream->pool,
+                                                    stream->recordBufferSize);
+        }
+
+    }
+    
+    if (param->flags & PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING) {
+	strm_set_cap(&stream->base, PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING,
+                     &param->output_vol);
+    }
+    
+    /* Done */
+    stream->base.op = &opensl_strm_op;
+    *p_aud_strm = &stream->base;
+    return PJ_SUCCESS;
+    
+on_error:
+    strm_destroy(&stream->base);
+    return status;
+}
+
+/* API: Get stream parameters */
+static pj_status_t strm_get_param(pjmedia_aud_stream *s,
+                                  pjmedia_aud_param *pi)
+{
+    struct opensl_aud_stream *strm = (struct opensl_aud_stream*)s;
+    PJ_ASSERT_RETURN(strm && pi, PJ_EINVAL);
+    pj_memcpy(pi, &strm->param, sizeof(*pi));
+
+    if (strm_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 strm_get_cap(pjmedia_aud_stream *s,
+                                pjmedia_aud_dev_cap cap,
+                                void *pval)
+{
+    struct opensl_aud_stream *strm = (struct opensl_aud_stream*)s;    
+    pj_status_t status = PJMEDIA_EAUD_INVCAP;
+    
+    PJ_ASSERT_RETURN(s && pval, PJ_EINVAL);
+    
+    if (cap==PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING &&
+	(strm->param.dir & PJMEDIA_DIR_PLAYBACK))
+    {
+        if (strm->playerVol) {
+            SLresult res;
+            SLmillibel vol, mvol;
+            
+            res = (*strm->playerVol)->GetMaxVolumeLevel(strm->playerVol,
+                                                        &mvol);
+            if (res == SL_RESULT_SUCCESS) {
+                res = (*strm->playerVol)->GetVolumeLevel(strm->playerVol,
+                                                         &vol);
+                if (res == SL_RESULT_SUCCESS) {
+                    *(int *)pval = ((int)vol - SL_MILLIBEL_MIN) * 100 /
+                                   ((int)mvol - SL_MILLIBEL_MIN);
+                    return PJ_SUCCESS;
+                }
+            }
+        }
+    }
+    
+    return status;
+}
+
+/* API: set capability */
+static pj_status_t strm_set_cap(pjmedia_aud_stream *s,
+                                pjmedia_aud_dev_cap cap,
+                                const void *value)
+{
+    struct opensl_aud_stream *strm = (struct opensl_aud_stream*)s;
+    
+    PJ_ASSERT_RETURN(s && value, PJ_EINVAL);
+
+    if (cap==PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING &&
+	(strm->param.dir & PJMEDIA_DIR_PLAYBACK))
+    {
+        if (strm->playerVol) {
+            SLresult res;
+            SLmillibel vol, mvol;
+            
+            res = (*strm->playerVol)->GetMaxVolumeLevel(strm->playerVol,
+                                                        &mvol);
+            if (res == SL_RESULT_SUCCESS) {
+                vol = (SLmillibel)(*(int *)value *
+                      ((int)mvol - SL_MILLIBEL_MIN) / 100 + SL_MILLIBEL_MIN);
+                res = (*strm->playerVol)->SetVolumeLevel(strm->playerVol,
+                                                         vol);
+                if (res == SL_RESULT_SUCCESS)
+                    return PJ_SUCCESS;
+            }
+        }
+    }
+
+    return PJMEDIA_EAUD_INVCAP;
+}
+
+/* API: start stream. */
+static pj_status_t strm_start(pjmedia_aud_stream *s)
+{
+    struct opensl_aud_stream *stream = (struct opensl_aud_stream*)s;
+    int i;
+    SLresult result = SL_RESULT_SUCCESS;
+    
+    PJ_LOG(4, (THIS_FILE, "Starting %s stream..", stream->name.ptr));
+    stream->quit_flag = 0;
+
+    if (stream->recordBufQ && stream->recordRecord) {
+        /* Enqueue an empty buffer to be filled by the recorder
+         * (for streaming recording, we need to enqueue at least 2 empty
+         * buffers to start things off)
+         */
+        for (i = 0; i < NUM_BUFFERS; i++) {
+            result = (*stream->recordBufQ)->Enqueue(stream->recordBufQ,
+                                                stream->recordBuffer[i],
+                                                stream->recordBufferSize);
+            /* The most likely other result is SL_RESULT_BUFFER_INSUFFICIENT,
+             * which for this code would indicate a programming error
+             */
+            pj_assert(result == SL_RESULT_SUCCESS);
+        }
+        
+        result = (*stream->recordRecord)->SetRecordState(
+                     stream->recordRecord, SL_RECORDSTATE_RECORDING);
+        if (result != SL_RESULT_SUCCESS) {
+            PJ_LOG(3, (THIS_FILE, "Cannot start recorder"));
+            goto on_error;
+        }
+    }
+    
+    if (stream->playerPlay && stream->playerBufQ) {
+        /* Set the player's state to playing */
+        result = (*stream->playerPlay)->SetPlayState(stream->playerPlay,
+                                                     SL_PLAYSTATE_PLAYING);
+        if (result != SL_RESULT_SUCCESS) {
+            PJ_LOG(3, (THIS_FILE, "Cannot start player"));
+            goto on_error;
+        }
+        
+        for (i = 0; i < NUM_BUFFERS; i++) {
+            pj_bzero(stream->playerBuffer[i], stream->playerBufferSize/100);
+            result = (*stream->playerBufQ)->Enqueue(stream->playerBufQ,
+                                                stream->playerBuffer[i],
+                                                stream->playerBufferSize/100);
+            pj_assert(result == SL_RESULT_SUCCESS);
+        }
+    }
+    
+    PJ_LOG(4, (THIS_FILE, "%s stream started", stream->name.ptr));
+    return PJ_SUCCESS;
+    
+on_error:
+    if (result != SL_RESULT_SUCCESS)
+        strm_stop(&stream->base);
+    return opensl_to_pj_error(result);
+}
+
+/* API: stop stream. */
+static pj_status_t strm_stop(pjmedia_aud_stream *s)
+{
+    struct opensl_aud_stream *stream = (struct opensl_aud_stream*)s;
+    
+    if (stream->quit_flag)
+        return PJ_SUCCESS;
+    
+    PJ_LOG(4, (THIS_FILE, "Stopping stream"));
+    
+    stream->quit_flag = 1;    
+    
+    if (stream->recordBufQ && stream->recordRecord) {
+        /* Stop recording and clear buffer queue */
+        (*stream->recordRecord)->SetRecordState(stream->recordRecord,
+                                                  SL_RECORDSTATE_STOPPED);
+        (*stream->recordBufQ)->Clear(stream->recordBufQ);
+    }
+
+    if (stream->playerBufQ && stream->playerPlay) {
+        /* Wait until the PCM data is done playing, the buffer queue callback
+         * will continue to queue buffers until the entire PCM data has been
+         * played. This is indicated by waiting for the count member of the
+         * SLBufferQueueState to go to zero.
+         */
+/*      
+        SLresult result;
+        W_SLBufferQueueState state;
+
+        result = (*stream->playerBufQ)->GetState(stream->playerBufQ, &state);
+        while (state.count) {
+            (*stream->playerBufQ)->GetState(stream->playerBufQ, &state);
+        } */
+        /* Stop player */
+        (*stream->playerPlay)->SetPlayState(stream->playerPlay,
+                                            SL_PLAYSTATE_STOPPED);
+    }
+
+    PJ_LOG(4,(THIS_FILE, "OpenSL stream stopped"));
+    
+    return PJ_SUCCESS;
+    
+}
+
+/* API: destroy stream. */
+static pj_status_t strm_destroy(pjmedia_aud_stream *s)
+{    
+    struct opensl_aud_stream *stream = (struct opensl_aud_stream*)s;
+    
+    /* Stop the stream */
+    strm_stop(s);
+    
+    if (stream->playerObj) {
+        /* Destroy the player */
+        (*stream->playerObj)->Destroy(stream->playerObj);
+        /* Invalidate all associated interfaces */
+        stream->playerObj = NULL;
+        stream->playerPlay = NULL;
+        stream->playerBufQ = NULL;
+        stream->playerVol = NULL;
+    }
+    
+    if (stream->recordObj) {
+        /* Destroy the recorder */
+        (*stream->recordObj)->Destroy(stream->recordObj);
+        /* Invalidate all associated interfaces */
+        stream->recordObj = NULL;
+        stream->recordRecord = NULL;
+        stream->recordBufQ = NULL;
+    }
+    
+    pj_pool_release(stream->pool);
+    PJ_LOG(4, (THIS_FILE, "OpenSL stream destroyed"));
+    
+    return PJ_SUCCESS;
+}
+
+#endif	/* PJMEDIA_AUDIO_DEV_HAS_OPENSL */