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/*
* 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.
*/
// FIXME taken from OpenSLES_AndroidConfiguration.h
#define SL_ANDROID_KEY_PERFORMANCE_MODE ((const SLchar*) "androidPerformanceMode")
////////////////////////////////////////////
/// 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.
#define _USE_MATH_DEFINES
#include <cmath>
#include "sles.h"
#include "audio_utils/atomic.h"
#include <stdio.h>
#include <assert.h>
#include <unistd.h>
#include <string.h>
int slesInit(sles_data ** ppSles, int samplingRate, int frameCount, int micSource,
int performanceMode,
int testType, double frequency1, char* byteBufferPtr, int byteBufferLength,
short* loopbackTone, int maxRecordedLateCallbacks, int ignoreFirstFrames) {
int status = SLES_FAIL;
if (ppSles != NULL) {
sles_data * pSles = (sles_data*) malloc(sizeof(sles_data));
memset(pSles, 0, sizeof(sles_data));
SLES_PRINTF("pSles malloc %zu bytes at %p", sizeof(sles_data), pSles);
//__android_log_print(ANDROID_LOG_INFO, "sles_jni",
//"malloc %d bytes at %p", sizeof(sles_data), pSles);//Or ANDROID_LOG_INFO, ...
*ppSles = pSles;
if (pSles != NULL)
{
SLES_PRINTF("creating server. Sampling rate =%d, frame count = %d",
samplingRate, frameCount);
status = slesCreateServer(pSles, samplingRate, frameCount, micSource,
performanceMode, testType,
frequency1, byteBufferPtr, byteBufferLength, loopbackTone,
maxRecordedLateCallbacks, ignoreFirstFrames);
SLES_PRINTF("slesCreateServer =%d", status);
}
}
return status;
}
int slesDestroy(sles_data ** ppSles) {
int status = SLES_FAIL;
if (ppSles != NULL) {
slesDestroyServer(*ppSles);
if (*ppSles != NULL)
{
SLES_PRINTF("free memory at %p",*ppSles);
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, then we can read from this filled buffer
static void recorderCallback(SLAndroidSimpleBufferQueueItf caller __unused, void *context) {
sles_data *pSles = (sles_data*) context;
if (pSles != NULL) {
collectBufferPeriod(&pSles->recorderBufferStats, NULL /*fdpStats*/, &pSles->recorderTimeStamps,
pSles->expectedBufferPeriod);
//__android_log_print(ANDROID_LOG_INFO, "sles_jni", "in recorderCallback");
SLresult result;
//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]; //pSles->rxBuffers stores the data recorded
// Remove buffer from record queue
if (++pSles->rxFront > pSles->rxBufCount) {
pSles->rxFront = 0;
}
if (pSles->testType == TEST_TYPE_LATENCY) {
// Throw out first frames
if (pSles->ignoreFirstFrames) {
int framesToErase = pSles->ignoreFirstFrames;
if (framesToErase > (int) pSles->bufSizeInFrames) {
framesToErase = pSles->bufSizeInFrames;
}
pSles->ignoreFirstFrames -= framesToErase;
memset(buffer, 0, framesToErase * pSles->channels * sizeof(short));
}
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);
}
}
} else if (pSles->testType == TEST_TYPE_BUFFER_PERIOD) {
if (pSles->fifo2Buffer != NULL) {
ssize_t actual = byteBuffer_write(pSles, buffer, (size_t) pSles->bufSizeInFrames);
//FIXME should log errors using other methods instead of printing to terminal
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);
//__android_log_print(ANDROID_LOG_INFO, "recorderCallback", "recorder buffer size: %i",
// 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>");
} //pSles not null
}
// Write "count" amount of short from buffer to pSles->byteBufferPtr. This byteBuffer will read by
// java code.
ssize_t byteBuffer_write(sles_data *pSles, char *buffer, size_t count) {
// bytebufferSize is in byte
int32_t rear; // rear should not exceed 2^31 - 1, or else overflow will happen
memcpy(&rear, (char *) (pSles->byteBufferPtr + pSles->byteBufferLength - 4), sizeof(rear));
size_t frameSize = pSles->channels * sizeof(short); // only one channel
int32_t maxLengthInShort = (pSles->byteBufferLength - 4) / frameSize;
// mask the upper bits to get the correct position in the pipe
int32_t tempRear = rear & (maxLengthInShort - 1);
size_t part1 = maxLengthInShort - tempRear;
if (part1 > count) {
part1 = count;
}
if (part1 > 0) {
memcpy(pSles->byteBufferPtr + (tempRear * frameSize), buffer,
part1 * frameSize);
size_t part2 = count - part1;
if (part2 > 0) {
memcpy(pSles->byteBufferPtr, (buffer + (part1 * frameSize)),
part2 * frameSize);
}
//TODO do we need something similar to the below function call?
//android_atomic_release_store(audio_utils_fifo_sum(fifo, fifo->mRear, availToWrite),
// &fifo->mRear);
}
// increase value of rear
int32_t* rear2 = (int32_t *) (pSles->byteBufferPtr + pSles->byteBufferLength - 4);
*rear2 += count;
return count;
}
// Calculate nanosecond difference between two timespec structs from clock_gettime(CLOCK_MONOTONIC)
// tv_sec [0, max time_t] , tv_nsec [0, 999999999]
int64_t diffInNano(struct timespec previousTime, struct timespec currentTime) {
return (int64_t) (currentTime.tv_sec - previousTime.tv_sec) * (int64_t) NANOS_PER_SECOND +
currentTime.tv_nsec - previousTime.tv_nsec;
}
// 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) {
collectBufferPeriod(&pSles->playerBufferStats, &pSles->recorderBufferStats /*fdpStats*/,
&pSles->playerTimeStamps, pSles->expectedBufferPeriod);
SLresult result;
//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;
}
if (pSles->testType == TEST_TYPE_LATENCY) {
// Jitter buffer should have strictly less than 2 buffers worth of data in it.
// This is to prevent the test itself from adding too much latency.
size_t discardedInputFrames = 0;
for (;;) {
size_t availToRead = audio_utils_fifo_availToRead(&pSles->fifo);
if (availToRead < pSles->bufSizeInFrames * 2) {
break;
}
ssize_t actual = audio_utils_fifo_read(&pSles->fifo, buffer, pSles->bufSizeInFrames);
if (actual > 0) {
discardedInputFrames += actual;
}
if (actual != (ssize_t) pSles->bufSizeInFrames) {
break;
}
}
if (discardedInputFrames > 0) {
if (pSles->totalDiscardedInputFrames > 0) {
__android_log_print(ANDROID_LOG_WARN, "sles_jni",
"Discarded an additional %zu input frames after a total of %zu input frames"
" had previously been discarded",
discardedInputFrames, pSles->totalDiscardedInputFrames);
}
pSles->totalDiscardedInputFrames += discardedInputFrames;
}
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) { // here we inject pulse
/*// 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 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;
}
}
}*/
//inject java generated tone
for (unsigned i = 0; i < pSles->bufSizeInFrames; ++i) {
for (unsigned k = 0; k < pSles->channels; ++k) {
((short *) buffer)[i * pSles->channels + k] = pSles->loopbackTone[i];
}
}
pSles->injectImpulse = 0;
pSles->totalDiscardedInputFrames = 0;
}
} else if (pSles->testType == TEST_TYPE_BUFFER_PERIOD) {
double twoPi = M_PI * 2;
int maxShort = 32767;
float amplitude = 0.8;
short value;
double phaseIncrement = pSles->frequency1 / pSles->sampleRate;
bool isGlitchEnabled = false;
for (unsigned i = 0; i < pSles->bufSizeInFrames; i++) {
value = (short) (sin(pSles->bufferTestPhase1) * maxShort * amplitude);
for (unsigned k = 0; k < pSles->channels; ++k) {
((short *) buffer)[i* pSles->channels + k] = value;
}
pSles->bufferTestPhase1 += twoPi * phaseIncrement;
// insert glitches if isGlitchEnabled == true, and insert it for every second
if (isGlitchEnabled && (pSles->count % pSles->sampleRate == 0)) {
pSles->bufferTestPhase1 += twoPi * phaseIncrement;
}
pSles->count++;
while (pSles->bufferTestPhase1 > twoPi) {
pSles->bufferTestPhase1 -= twoPi;
}
}
}
// 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;
} //pSles not null
}
// Used to set initial values for the bufferStats struct before values can be recorded.
void initBufferStats(bufferStats *stats) {
stats->buffer_period = new int[RANGE](); // initialized to zeros
stats->previous_time = {0,0};
stats->current_time = {0,0};
stats->buffer_count = 0;
stats->max_buffer_period = 0;
stats->measurement_count = 0;
stats->SDM = 0;
stats->var = 0;
}
// Called in the beginning of playerCallback() to collect the interval between each callback.
// fdpStats is either NULL or a pointer to the buffer statistics for the full-duplex partner.
void collectBufferPeriod(bufferStats *stats, bufferStats *fdpStats, callbackTimeStamps *timeStamps,
short expectedBufferPeriod) {
clock_gettime(CLOCK_MONOTONIC, &(stats->current_time));
if (timeStamps->startTime.tv_sec == 0 && timeStamps->startTime.tv_nsec == 0) {
timeStamps->startTime = stats->current_time;
}
(stats->buffer_count)++;
if (stats->previous_time.tv_sec != 0 && stats->buffer_count > BUFFER_PERIOD_DISCARD &&
(fdpStats == NULL || fdpStats->buffer_count > BUFFER_PERIOD_DISCARD_FULL_DUPLEX_PARTNER)) {
int64_t callbackDuration = diffInNano(stats->previous_time, stats->current_time);
bool outlier = updateBufferStats(stats, callbackDuration, expectedBufferPeriod);
//recording timestamps of buffer periods not at expected buffer period
if (outlier) {
int64_t timeStamp = diffInNano(timeStamps->startTime, stats->current_time);
recordTimeStamp(timeStamps, callbackDuration, timeStamp);
}
}
stats->previous_time = stats->current_time;
}
// Records an outlier given the duration in nanoseconds and the number of nanoseconds
// between it and the start of the test.
void recordTimeStamp(callbackTimeStamps *timeStamps,
int64_t callbackDuration, int64_t timeStamp) {
if (timeStamps->exceededCapacity) {
return;
}
//only marked as exceeded if attempting to record a late callback after arrays full
if (timeStamps->index == timeStamps->capacity){
timeStamps->exceededCapacity = true;
} else {
timeStamps->callbackDurations[timeStamps->index] =
(short) ((callbackDuration + NANOS_PER_MILLI - 1) / NANOS_PER_MILLI);
timeStamps->timeStampsMs[timeStamps->index] =
(int) ((timeStamp + NANOS_PER_MILLI - 1) / NANOS_PER_MILLI);
timeStamps->index++;
}
}
void atomicSetIfGreater(volatile int32_t *addr, int32_t val) {
// TODO: rewrite this to avoid the need for unbounded spinning
int32_t old;
do {
old = *addr;
if (val < old) return;
} while(!android_atomic_compare_exchange(&old, val, addr));
}
// Updates the stats being collected about buffer periods. Returns true if this is an outlier.
bool updateBufferStats(bufferStats *stats, int64_t diff_in_nano, int expectedBufferPeriod) {
stats->measurement_count++;
// round up to nearest millisecond
int diff_in_milli = (int) ((diff_in_nano + NANOS_PER_MILLI - 1) / NANOS_PER_MILLI);
if (diff_in_milli > stats->max_buffer_period) {
stats->max_buffer_period = diff_in_milli;
}
// from 0 ms to 1000 ms, plus a sum of all instances > 1000ms
if (diff_in_milli >= (RANGE - 1)) {
(stats->buffer_period)[RANGE-1]++;
} else if (diff_in_milli >= 0) {
(stats->buffer_period)[diff_in_milli]++;
} else { // for diff_in_milli < 0
__android_log_print(ANDROID_LOG_INFO, "sles_player", "Having negative BufferPeriod.");
}
int64_t delta = diff_in_nano - (int64_t) expectedBufferPeriod * NANOS_PER_MILLI;
stats->SDM += delta * delta;
if (stats->measurement_count > 1) {
stats->var = stats->SDM / stats->measurement_count;
}
// check if the lateness is so bad that a systrace should be captured
// TODO: replace static threshold of lateness with a dynamic determination
if (diff_in_milli > expectedBufferPeriod + LATE_CALLBACK_CAPTURE_THRESHOLD) {
// TODO: log in a non-blocking way
//__android_log_print(ANDROID_LOG_INFO, "buffer_stats", "Callback late by %d ms",
// diff_in_milli - expectedBufferPeriod);
atomicSetIfGreater(&(stats->captureRank), diff_in_milli - expectedBufferPeriod);
}
return diff_in_milli > expectedBufferPeriod + LATE_CALLBACK_OUTLIER_THRESHOLD;
}
int slesCreateServer(sles_data *pSles, int samplingRate, int frameCount, int micSource,
int performanceMode,
int testType, double frequency1, char *byteBufferPtr, int byteBufferLength,
short *loopbackTone, int maxRecordedLateCallbacks, int ignoreFirstFrames) {
int status = SLES_FAIL;
if (pSles != NULL) {
// 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
pSles->totalDiscardedInputFrames = 0;
pSles->ignoreFirstFrames = ignoreFirstFrames;
// 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; //this fifo is for sending data to java code (to plot it)
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 *));
SLES_PRINTF(" calloc freeBuffers %zu bytes at %p",pSles->freeBufCount + 1,
pSles->freeBuffers);
unsigned j;
for (j = 0; j < pSles->freeBufCount; ++j) {
pSles->freeBuffers[j] = (char *) malloc(pSles->bufSizeInBytes);
SLES_PRINTF(" buff%d malloc %zu bytes at %p",j, pSles->bufSizeInBytes,
pSles->freeBuffers[j]);
}
pSles->freeFront = 0;
pSles->freeRear = pSles->freeBufCount;
pSles->freeBuffers[j] = NULL;
// Initialize record queue
pSles->rxBuffers = (char **) calloc(pSles->rxBufCount + 1, sizeof(char *));
SLES_PRINTF(" calloc rxBuffers %zu bytes at %p",pSles->rxBufCount + 1, pSles->rxBuffers);
pSles->rxFront = 0;
pSles->rxRear = 0;
// Initialize play queue
pSles->txBuffers = (char **) calloc(pSles->txBufCount + 1, sizeof(char *));
SLES_PRINTF(" calloc txBuffers %zu bytes at %p",pSles->txBufCount + 1, pSles->txBuffers);
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;
// }
initBufferStats(&pSles->recorderBufferStats);
initBufferStats(&pSles->playerBufferStats);
// init other variables needed for buffer test
pSles->testType = testType;
pSles->frequency1 = frequency1;
pSles->bufferTestPhase1 = 0;
pSles->count = 0;
pSles->byteBufferPtr = byteBufferPtr;
pSles->byteBufferLength = byteBufferLength;
//init loopback tone
pSles->loopbackTone = loopbackTone;
pSles->recorderTimeStamps = {
new int[maxRecordedLateCallbacks], //int* timeStampsMs
new short[maxRecordedLateCallbacks], //short* callbackDurations
0, //short index
{0,0}, //struct timespec startTime;
maxRecordedLateCallbacks, //int capacity
false //bool exceededCapacity
};
pSles->playerTimeStamps = {
new int[maxRecordedLateCallbacks], //int* timeStampsMs
new short[maxRecordedLateCallbacks], //short* callbackDurations;
0, //short index
{0,0}, //struct timespec startTime;
maxRecordedLateCallbacks, //int capacity
false //bool exceededCapacity
};
pSles->expectedBufferPeriod = (short) (
round(pSles->bufSizeInFrames * MILLIS_PER_SECOND / (float) pSles->sampleRate));
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[2] = {SL_IID_BUFFERQUEUE, SL_IID_ANDROIDCONFIGURATION};
SLboolean flags_tx[2] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
result = (*engineEngine)->CreateAudioPlayer(engineEngine, &(pSles->playerObject),
&audiosrc, &audiosnk, 2, 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);
{
/* Get the Android configuration interface which is explicit */
SLAndroidConfigurationItf configItf;
result = (*(pSles->playerObject))->GetInterface(pSles->playerObject,
SL_IID_ANDROIDCONFIGURATION, (void*)&configItf);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
/* Use the configuration interface to configure the player before it's realized */
if (performanceMode != -1) {
SLuint32 performanceMode32 = performanceMode;
result = (*configItf)->SetConfiguration(configItf, SL_ANDROID_KEY_PERFORMANCE_MODE,
&performanceMode32, sizeof(SLuint32));
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); //playerCallback is the name of callback function
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;
{ //why brackets here?
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;
//SL_ANDROID_RECORDING_PRESET_CAMCORDER;//SL_ANDROID_RECORDING_PRESET_NONE;
/* 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);
}
if (performanceMode != -1) {
SLuint32 performanceMode32 = performanceMode;
result = (*configItf)->SetConfiguration(configItf, SL_ANDROID_KEY_PERFORMANCE_MODE,
&performanceMode32, 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);
int xx = 1;
}
return status;
}
// Read data from fifo2Buffer and store into pSamples.
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) {
SLresult result;
for (int i = 0; i < 10; i++) {
usleep(100000); // sleep for 0.1s
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) {
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 buffers
if (NULL != pSles->freeBuffers) {
for (unsigned j = 0; j < pSles->freeBufCount; ++j) {
if (NULL != pSles->freeBuffers[j]) {
SLES_PRINTF(" free buff%d at %p",j, pSles->freeBuffers[j]);
free (pSles->freeBuffers[j]);
}
}
SLES_PRINTF(" free freeBuffers at %p", pSles->freeBuffers);
free(pSles->freeBuffers);
} else {
SLES_PRINTF(" freeBuffers NULL, no need to free");
}
if (NULL != pSles->rxBuffers) {
SLES_PRINTF(" free rxBuffers at %p", pSles->rxBuffers);
free(pSles->rxBuffers);
} else {
SLES_PRINTF(" rxBuffers NULL, no need to free");
}
if (NULL != pSles->txBuffers) {
SLES_PRINTF(" free txBuffers at %p", pSles->txBuffers);
free(pSles->txBuffers);
} else {
SLES_PRINTF(" txBuffers NULL, no need to free");
}
status = SLES_SUCCESS;
}
SLES_PRINTF("End slesDestroyServer: status = %d", status);
return status;
}
int* slesGetRecorderBufferPeriod(sles_data *pSles) {
return pSles->recorderBufferStats.buffer_period;
}
int slesGetRecorderMaxBufferPeriod(sles_data *pSles) {
return pSles->recorderBufferStats.max_buffer_period;
}
int64_t slesGetRecorderVarianceBufferPeriod(sles_data *pSles) {
return pSles->recorderBufferStats.var;
}
int* slesGetPlayerBufferPeriod(sles_data *pSles) {
return pSles->playerBufferStats.buffer_period;
}
int slesGetPlayerMaxBufferPeriod(sles_data *pSles) {
return pSles->playerBufferStats.max_buffer_period;
}
int64_t slesGetPlayerVarianceBufferPeriod(sles_data *pSles) {
return pSles->playerBufferStats.var;
}
int slesGetCaptureRank(sles_data *pSles) {
// clear the capture flags since they're being handled now
int recorderRank = android_atomic_exchange(0, &pSles->recorderBufferStats.captureRank);
int playerRank = android_atomic_exchange(0, &pSles->playerBufferStats.captureRank);
if (recorderRank > playerRank) {
return recorderRank;
} else {
return playerRank;
}
}