/*
* Copyright (C) 2015 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
////////////////////////////////////////////
/// Actual sles functions.
// Test program to record from default audio input and playback to default audio output.
// It will generate feedback (Larsen effect) if played through on-device speakers,
// or acts as a delay if played through headset.
#include "sles.h"
#include <stdio.h>
#include <stdlib.h>
#include <stddef.h>
#include <assert.h>
#include <pthread.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <unistd.h>
int slesInit(sles_data ** ppSles, int samplingRate, int frameCount, int micSource) {
int status = SLES_FAIL;
if (ppSles != NULL) {
sles_data * pSles = (sles_data*) calloc(1, sizeof (sles_data));
SLES_PRINTF("malloc %d bytes at %p",sizeof(sles_data), pSles);
*ppSles = pSles;
if (pSles != NULL)
{
SLES_PRINTF("creating server. Sampling rate =%d, frame count = %d",samplingRate,
frameCount);
status = slesCreateServer(pSles, samplingRate, frameCount, micSource);
SLES_PRINTF("slesCreateServer =%d",status);
}
}
return status;
}
int slesDestroy(sles_data ** ppSles) {
int status = SLES_FAIL;
if (ppSles != NULL) {
slesDestroyServer(*ppSles);
if (*ppSles != NULL)
{
free(*ppSles);
*ppSles = 0;
}
status = SLES_SUCCESS;
}
return status;
}
#define ASSERT_EQ(x, y) do { if ((x) == (y)) ; else { fprintf(stderr, "0x%x != 0x%x\n", \
(unsigned) (x), (unsigned) (y)); assert((x) == (y)); } } while (0)
// Called after audio recorder fills a buffer with data
static void recorderCallback(SLAndroidSimpleBufferQueueItf caller __unused, void *context) {
sles_data *pSles = (sles_data*) context;
if (pSles != NULL) {
SLresult result;
pthread_mutex_lock(&(pSles->mutex));
//ee SLES_PRINTF("<R");
// We should only be called when a recording buffer is done
assert(pSles->rxFront <= pSles->rxBufCount);
assert(pSles->rxRear <= pSles->rxBufCount);
assert(pSles->rxFront != pSles->rxRear);
char *buffer = pSles->rxBuffers[pSles->rxFront];
// Remove buffer from record queue
if (++pSles->rxFront > pSles->rxBufCount) {
pSles->rxFront = 0;
}
ssize_t actual = audio_utils_fifo_write(&(pSles->fifo), buffer,
(size_t) pSles->bufSizeInFrames);
if (actual != (ssize_t) pSles->bufSizeInFrames) {
write(1, "?", 1);
}
// This is called by a realtime (SCHED_FIFO) thread,
// and it is unsafe to do I/O as it could block for unbounded time.
// Flash filesystem is especially notorious for blocking.
if (pSles->fifo2Buffer != NULL) {
actual = audio_utils_fifo_write(&(pSles->fifo2), buffer,
(size_t) pSles->bufSizeInFrames);
if (actual != (ssize_t) pSles->bufSizeInFrames) {
write(1, "?", 1);
}
}
// Enqueue this same buffer for the recorder to fill again.
result = (*(pSles->recorderBufferQueue))->Enqueue(pSles->recorderBufferQueue, buffer,
pSles->bufSizeInBytes);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
// Update our model of the record queue
SLuint32 rxRearNext = pSles->rxRear+1;
if (rxRearNext > pSles->rxBufCount) {
rxRearNext = 0;
}
assert(rxRearNext != pSles->rxFront);
pSles->rxBuffers[pSles->rxRear] = buffer;
pSles->rxRear = rxRearNext;
//ee SLES_PRINTF("r>");
pthread_mutex_unlock(&(pSles->mutex));
} //pSles not null
}
// Called after audio player empties a buffer of data
static void playerCallback(SLBufferQueueItf caller __unused, void *context) {
sles_data *pSles = (sles_data*) context;
if (pSles != NULL) {
SLresult result;
pthread_mutex_lock(&(pSles->mutex));
//ee SLES_PRINTF("<P");
// Get the buffer that just finished playing
assert(pSles->txFront <= pSles->txBufCount);
assert(pSles->txRear <= pSles->txBufCount);
assert(pSles->txFront != pSles->txRear);
char *buffer = pSles->txBuffers[pSles->txFront];
if (++pSles->txFront > pSles->txBufCount) {
pSles->txFront = 0;
}
ssize_t actual = audio_utils_fifo_read(&(pSles->fifo), buffer, pSles->bufSizeInFrames);
if (actual != (ssize_t) pSles->bufSizeInFrames) {
write(1, "/", 1);
// on underrun from pipe, substitute silence
memset(buffer, 0, pSles->bufSizeInFrames * pSles->channels * sizeof(short));
}
if (pSles->injectImpulse == -1) {
// Experimentally, a single frame impulse was insufficient to trigger feedback.
// Also a Nyquist frequency signal was also insufficient, probably because
// the response of output and/or input path was not adequate at high frequencies.
// This short burst of a few cycles of square wave at Nyquist/4 was found to work well.
for (unsigned i = 0; i < pSles->bufSizeInFrames / 8; i += 8) {
for (int j = 0; j < 8; j++) {
for (unsigned k = 0; k < pSles->channels; k++) {
((short *)buffer)[(i+j)*pSles->channels+k] = j < 4 ? 0x7FFF : 0x8000;
}
}
}
pSles->injectImpulse = 0;
}
// Enqueue the filled buffer for playing
result = (*(pSles->playerBufferQueue))->Enqueue(pSles->playerBufferQueue, buffer,
pSles->bufSizeInBytes);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
// Update our model of the player queue
assert(pSles->txFront <= pSles->txBufCount);
assert(pSles->txRear <= pSles->txBufCount);
SLuint32 txRearNext = pSles->txRear+1;
if (txRearNext > pSles->txBufCount) {
txRearNext = 0;
}
assert(txRearNext != pSles->txFront);
pSles->txBuffers[pSles->txRear] = buffer;
pSles->txRear = txRearNext;
//ee SLES_PRINTF("p>");
pthread_mutex_unlock(&(pSles->mutex));
} //pSles not null
}
int slesCreateServer(sles_data *pSles, int samplingRate, int frameCount, int micSource) {
int status = SLES_FAIL;
if (pSles == NULL) {
return status;
}
// adb shell slesTest_feedback -r1 -t1 -s48000 -f240 -i300 -e3 -o/sdcard/log.wav
// r1 and t1 are the receive and transmit buffer counts, typically 1
// s is the sample rate, typically 48000 or 44100
// f is the frame count per buffer, typically 240 or 256
// i is the number of milliseconds before impulse. You may need to adjust this.
// e is number of seconds to record
// o is output .wav file name
// // default values
// SLuint32 rxBufCount = 1; // -r#
// SLuint32 txBufCount = 1; // -t#
// SLuint32 bufSizeInFrames = 240; // -f#
// SLuint32 channels = 1; // -c#
// SLuint32 sampleRate = 48000; // -s#
// SLuint32 exitAfterSeconds = 3; // -e#
// SLuint32 freeBufCount = 0; // calculated
// SLuint32 bufSizeInBytes = 0; // calculated
// int injectImpulse = 300; // -i#i
//
// // Storage area for the buffer queues
// char **rxBuffers;
// char **txBuffers;
// char **freeBuffers;
//
// // Buffer indices
// SLuint32 rxFront; // oldest recording
// SLuint32 rxRear; // next to be recorded
// SLuint32 txFront; // oldest playing
// SLuint32 txRear; // next to be played
// SLuint32 freeFront; // oldest free
// SLuint32 freeRear; // next to be freed
//
// audio_utils_fifo fifo; //(*)
// SLAndroidSimpleBufferQueueItf recorderBufferQueue;
// SLBufferQueueItf playerBufferQueue;
// default values
pSles->rxBufCount = 1; // -r#
pSles->txBufCount = 1; // -t#
pSles->bufSizeInFrames = frameCount;//240; // -f#
pSles->channels = 1; // -c#
pSles->sampleRate = samplingRate;//48000; // -s#
pSles->exitAfterSeconds = 3; // -e#
pSles->freeBufCount = 0; // calculated
pSles->bufSizeInBytes = 0; // calculated
pSles->injectImpulse = 300; // -i#i
// Storage area for the buffer queues
// char **rxBuffers;
// char **txBuffers;
// char **freeBuffers;
// Buffer indices
pSles->rxFront; // oldest recording
pSles->rxRear; // next to be recorded
pSles->txFront; // oldest playing
pSles->txRear; // next to be played
pSles->freeFront; // oldest free
pSles->freeRear; // next to be freed
pSles->fifo; //(*)
pSles->fifo2Buffer = NULL;
pSles->recorderBufferQueue;
pSles->playerBufferQueue;
// compute total free buffers as -r plus -t
pSles->freeBufCount = pSles->rxBufCount + pSles->txBufCount;
// compute buffer size
pSles->bufSizeInBytes = pSles->channels * pSles->bufSizeInFrames * sizeof(short);
// Initialize free buffers
pSles->freeBuffers = (char **) calloc(pSles->freeBufCount+1, sizeof(char *));
unsigned j;
for (j = 0; j < pSles->freeBufCount; ++j) {
pSles->freeBuffers[j] = (char *) malloc(pSles->bufSizeInBytes);
}
pSles->freeFront = 0;
pSles->freeRear = pSles->freeBufCount;
pSles->freeBuffers[j] = NULL;
// Initialize record queue
pSles->rxBuffers = (char **) calloc(pSles->rxBufCount+1, sizeof(char *));
pSles->rxFront = 0;
pSles->rxRear = 0;
// Initialize play queue
pSles->txBuffers = (char **) calloc(pSles->txBufCount+1, sizeof(char *));
pSles->txFront = 0;
pSles->txRear = 0;
size_t frameSize = pSles->channels * sizeof(short);
#define FIFO_FRAMES 1024
pSles->fifoBuffer = new short[FIFO_FRAMES * pSles->channels];
audio_utils_fifo_init(&(pSles->fifo), FIFO_FRAMES, frameSize, pSles->fifoBuffer);
// SNDFILE *sndfile;
// if (outFileName != NULL) {
// create .wav writer
// SF_INFO info;
// info.frames = 0;
// info.samplerate = sampleRate;
// info.channels = channels;
// info.format = SF_FORMAT_WAV | SF_FORMAT_PCM_16;
// sndfile = sf_open(outFileName, SFM_WRITE, &info);
// if (sndfile != NULL) {
#define FIFO2_FRAMES 65536
pSles->fifo2Buffer = new short[FIFO2_FRAMES * pSles->channels];
audio_utils_fifo_init(&(pSles->fifo2), FIFO2_FRAMES, frameSize, pSles->fifo2Buffer);
// } else {
// fprintf(stderr, "sf_open failed\n");
// }
// } else {
// sndfile = NULL;
// }
SLresult result;
// create engine
pSles->engineObject;
result = slCreateEngine(&(pSles->engineObject), 0, NULL, 0, NULL, NULL);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
result = (*(pSles->engineObject))->Realize(pSles->engineObject, SL_BOOLEAN_FALSE);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
SLEngineItf engineEngine;
result = (*(pSles->engineObject))->GetInterface(pSles->engineObject, SL_IID_ENGINE,
&engineEngine);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
// create output mix
pSles->outputmixObject;
result = (*engineEngine)->CreateOutputMix(engineEngine, &(pSles->outputmixObject), 0, NULL,
NULL);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
result = (*(pSles->outputmixObject))->Realize(pSles->outputmixObject, SL_BOOLEAN_FALSE);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
// create an audio player with buffer queue source and output mix sink
SLDataSource audiosrc;
SLDataSink audiosnk;
SLDataFormat_PCM pcm;
SLDataLocator_OutputMix locator_outputmix;
SLDataLocator_BufferQueue locator_bufferqueue_tx;
locator_bufferqueue_tx.locatorType = SL_DATALOCATOR_BUFFERQUEUE;
locator_bufferqueue_tx.numBuffers = pSles->txBufCount;
locator_outputmix.locatorType = SL_DATALOCATOR_OUTPUTMIX;
locator_outputmix.outputMix = pSles->outputmixObject;
pcm.formatType = SL_DATAFORMAT_PCM;
pcm.numChannels = pSles->channels;
pcm.samplesPerSec = pSles->sampleRate * 1000;
pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
pcm.containerSize = 16;
pcm.channelMask = pSles->channels == 1 ? SL_SPEAKER_FRONT_CENTER :
(SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT);
pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
audiosrc.pLocator = &locator_bufferqueue_tx;
audiosrc.pFormat = &pcm;
audiosnk.pLocator = &locator_outputmix;
audiosnk.pFormat = NULL;
pSles->playerObject = NULL;
pSles->recorderObject = NULL;
SLInterfaceID ids_tx[1] = {SL_IID_BUFFERQUEUE};
SLboolean flags_tx[1] = {SL_BOOLEAN_TRUE};
result = (*engineEngine)->CreateAudioPlayer(engineEngine, &(pSles->playerObject),
&audiosrc, &audiosnk, 1, ids_tx, flags_tx);
if (SL_RESULT_CONTENT_UNSUPPORTED == result) {
fprintf(stderr, "Could not create audio player (result %x), check sample rate\n",
result);
SLES_PRINTF("ERROR: Could not create audio player (result %x), check sample rate\n",
result);
goto cleanup;
}
ASSERT_EQ(SL_RESULT_SUCCESS, result);
result = (*(pSles->playerObject))->Realize(pSles->playerObject, SL_BOOLEAN_FALSE);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
SLPlayItf playerPlay;
result = (*(pSles->playerObject))->GetInterface(pSles->playerObject, SL_IID_PLAY,
&playerPlay);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
result = (*(pSles->playerObject))->GetInterface(pSles->playerObject, SL_IID_BUFFERQUEUE,
&(pSles->playerBufferQueue));
ASSERT_EQ(SL_RESULT_SUCCESS, result);
result = (*(pSles->playerBufferQueue))->RegisterCallback(pSles->playerBufferQueue,
playerCallback, pSles);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
// Enqueue some zero buffers for the player
for (j = 0; j < pSles->txBufCount; ++j) {
// allocate a free buffer
assert(pSles->freeFront != pSles->freeRear);
char *buffer = pSles->freeBuffers[pSles->freeFront];
if (++pSles->freeFront > pSles->freeBufCount) {
pSles->freeFront = 0;
}
// put on play queue
SLuint32 txRearNext = pSles->txRear + 1;
if (txRearNext > pSles->txBufCount) {
txRearNext = 0;
}
assert(txRearNext != pSles->txFront);
pSles->txBuffers[pSles->txRear] = buffer;
pSles->txRear = txRearNext;
result = (*(pSles->playerBufferQueue))->Enqueue(pSles->playerBufferQueue,
buffer, pSles->bufSizeInBytes);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
}
result = (*playerPlay)->SetPlayState(playerPlay, SL_PLAYSTATE_PLAYING);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
// Create an audio recorder with microphone device source and buffer queue sink.
// The buffer queue as sink is an Android-specific extension.
SLDataLocator_IODevice locator_iodevice;
SLDataLocator_AndroidSimpleBufferQueue locator_bufferqueue_rx;
locator_iodevice.locatorType = SL_DATALOCATOR_IODEVICE;
locator_iodevice.deviceType = SL_IODEVICE_AUDIOINPUT;
locator_iodevice.deviceID = SL_DEFAULTDEVICEID_AUDIOINPUT;
locator_iodevice.device = NULL;
audiosrc.pLocator = &locator_iodevice;
audiosrc.pFormat = NULL;
locator_bufferqueue_rx.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE;
locator_bufferqueue_rx.numBuffers = pSles->rxBufCount;
audiosnk.pLocator = &locator_bufferqueue_rx;
audiosnk.pFormat = &pcm;
{
SLInterfaceID ids_rx[2] = {SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
SL_IID_ANDROIDCONFIGURATION};
SLboolean flags_rx[2] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
result = (*engineEngine)->CreateAudioRecorder(engineEngine, &(pSles->recorderObject),
&audiosrc, &audiosnk, 2, ids_rx, flags_rx);
if (SL_RESULT_SUCCESS != result) {
fprintf(stderr, "Could not create audio recorder (result %x), "
"check sample rate and channel count\n", result);
status = SLES_FAIL;
SLES_PRINTF("ERROR: Could not create audio recorder (result %x), "
"check sample rate and channel count\n", result);
goto cleanup;
}
}
ASSERT_EQ(SL_RESULT_SUCCESS, result);
{
/* Get the Android configuration interface which is explicit */
SLAndroidConfigurationItf configItf;
result = (*(pSles->recorderObject))->GetInterface(pSles->recorderObject,
SL_IID_ANDROIDCONFIGURATION, (void*)&configItf);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
SLuint32 presetValue = micSource;
/* Use the configuration interface to configure the recorder before it's realized */
if (presetValue != SL_ANDROID_RECORDING_PRESET_NONE) {
result = (*configItf)->SetConfiguration(configItf, SL_ANDROID_KEY_RECORDING_PRESET,
&presetValue, sizeof(SLuint32));
ASSERT_EQ(SL_RESULT_SUCCESS, result);
}
}
result = (*(pSles->recorderObject))->Realize(pSles->recorderObject, SL_BOOLEAN_FALSE);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
SLRecordItf recorderRecord;
result = (*(pSles->recorderObject))->GetInterface(pSles->recorderObject, SL_IID_RECORD,
&recorderRecord);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
result = (*(pSles->recorderObject))->GetInterface(pSles->recorderObject,
SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &(pSles->recorderBufferQueue));
ASSERT_EQ(SL_RESULT_SUCCESS, result);
result = (*(pSles->recorderBufferQueue))->RegisterCallback(pSles->recorderBufferQueue,
recorderCallback, pSles);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
// Enqueue some empty buffers for the recorder
for (j = 0; j < pSles->rxBufCount; ++j) {
// allocate a free buffer
assert(pSles->freeFront != pSles->freeRear);
char *buffer = pSles->freeBuffers[pSles->freeFront];
if (++pSles->freeFront > pSles->freeBufCount) {
pSles->freeFront = 0;
}
// put on record queue
SLuint32 rxRearNext = pSles->rxRear + 1;
if (rxRearNext > pSles->rxBufCount) {
rxRearNext = 0;
}
assert(rxRearNext != pSles->rxFront);
pSles->rxBuffers[pSles->rxRear] = buffer;
pSles->rxRear = rxRearNext;
result = (*(pSles->recorderBufferQueue))->Enqueue(pSles->recorderBufferQueue,
buffer, pSles->bufSizeInBytes);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
}
// Kick off the recorder
result = (*recorderRecord)->SetRecordState(recorderRecord, SL_RECORDSTATE_RECORDING);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
// Tear down the objects and exit
status = SLES_SUCCESS;
cleanup:
SLES_PRINTF("Finished initialization with status: %d", status);
return status;
}
int slesProcessNext(sles_data *pSles, double *pSamples, long maxSamples) {
//int status = SLES_FAIL;
SLES_PRINTF("slesProcessNext: pSles = %p, currentSample: %p, maxSamples = %ld", pSles,
pSamples, maxSamples);
int samplesRead = 0;
int currentSample = 0;
double *pCurrentSample = pSamples;
int maxValue = 32768;
if (pSles == NULL) {
return samplesRead;
}
SLresult result;
for (int i = 0; i < 10; i++) {
usleep(100000);
if (pSles->fifo2Buffer != NULL) {
for (;;) {
short buffer[pSles->bufSizeInFrames * pSles->channels];
ssize_t actual = audio_utils_fifo_read(&(pSles->fifo2), buffer,
pSles->bufSizeInFrames);
if (actual <= 0)
break;
{
for (int jj =0; jj<actual && currentSample < maxSamples; jj++) {
*(pCurrentSample++) = ((double)buffer[jj])/maxValue;
currentSample++;
}
}
samplesRead +=actual;
}
}
if (pSles->injectImpulse > 0) {
if (pSles->injectImpulse <= 100) {
pSles->injectImpulse = -1;
write(1, "I", 1);
} else {
if ((pSles->injectImpulse % 1000) < 100) {
write(1, "i", 1);
}
pSles->injectImpulse -= 100;
}
} else if (i == 9) {
write(1, ".", 1);
}
}
SLBufferQueueState playerBQState;
result = (*(pSles->playerBufferQueue))->GetState(pSles->playerBufferQueue,
&playerBQState);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
SLAndroidSimpleBufferQueueState recorderBQState;
result = (*(pSles->recorderBufferQueue))->GetState(pSles->recorderBufferQueue,
&recorderBQState);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
SLES_PRINTF("End of slesProcessNext: pSles = %p, samplesRead = %d, maxSamples= %ld", pSles,
samplesRead, maxSamples);
return samplesRead;
}
int slesDestroyServer(sles_data *pSles) {
int status = SLES_FAIL;
SLES_PRINTF("Start slesDestroyServer: pSles = %p", pSles);
if (pSles == NULL) {
return status;
}
if (NULL != pSles->playerObject) {
SLES_PRINTF("stopping player...");
SLPlayItf playerPlay;
SLresult result = (*(pSles->playerObject))->GetInterface(pSles->playerObject,
SL_IID_PLAY, &playerPlay);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
//stop player and recorder if they exist
result = (*playerPlay)->SetPlayState(playerPlay, SL_PLAYSTATE_STOPPED);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
}
if (NULL != pSles->recorderObject) {
SLES_PRINTF("stopping recorder...");
SLRecordItf recorderRecord;
SLresult result = (*(pSles->recorderObject))->GetInterface(pSles->recorderObject,
SL_IID_RECORD, &recorderRecord);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
result = (*recorderRecord)->SetRecordState(recorderRecord, SL_RECORDSTATE_STOPPED);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
}
usleep(1000);
audio_utils_fifo_deinit(&(pSles->fifo));
delete[] pSles->fifoBuffer;
SLES_PRINTF("slesDestroyServer 2");
// if (sndfile != NULL) {
audio_utils_fifo_deinit(&(pSles->fifo2));
delete[] pSles->fifo2Buffer;
SLES_PRINTF("slesDestroyServer 3");
// sf_close(sndfile);
// }
if (NULL != pSles->playerObject) {
(*(pSles->playerObject))->Destroy(pSles->playerObject);
}
SLES_PRINTF("slesDestroyServer 4");
if (NULL != pSles->recorderObject) {
(*(pSles->recorderObject))->Destroy(pSles->recorderObject);
}
SLES_PRINTF("slesDestroyServer 5");
(*(pSles->outputmixObject))->Destroy(pSles->outputmixObject);
SLES_PRINTF("slesDestroyServer 6");
(*(pSles->engineObject))->Destroy(pSles->engineObject);
SLES_PRINTF("slesDestroyServer 7");
// free(pSles);
// pSles=NULL;
status = SLES_SUCCESS;
SLES_PRINTF("End slesDestroyServer: status = %d", status);
return status;
}