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