/*
* Copyright (C) 2010 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.
*/
// 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/OpenSLES.h>
#include <SLES/OpenSLES_Android.h>
#include <assert.h>
#include <pthread.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <unistd.h>
#include <audio_utils/fifo.h>
#include <audio_utils/sndfile.h>
#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)
// default values
static SLuint32 rxBufCount = 2; // -r#
static SLuint32 txBufCount = 2; // -t#
static SLuint32 bufSizeInFrames = 240; // -f#
static SLuint32 channels = 1; // -c#
static SLuint32 sampleRate = 48000; // -s#
static SLuint32 exitAfterSeconds = 60; // -e#
static SLuint32 freeBufCount = 0; // calculated
static SLuint32 bufSizeInBytes = 0; // calculated
// Storage area for the buffer queues
static char **rxBuffers;
static char **txBuffers;
static char **freeBuffers;
// Buffer indices
static SLuint32 rxFront; // oldest recording
static SLuint32 rxRear; // next to be recorded
static SLuint32 txFront; // oldest playing
static SLuint32 txRear; // next to be played
static SLuint32 freeFront; // oldest free
static SLuint32 freeRear; // next to be freed
static SLAndroidSimpleBufferQueueItf recorderBufferQueue;
static SLBufferQueueItf playerBufferQueue;
static pthread_mutex_t mutex = PTHREAD_MUTEX_INITIALIZER;
static audio_utils_fifo fifo;
static audio_utils_fifo fifo2;
static short *fifo2Buffer = NULL;
static int injectImpulse;
// Called after audio recorder fills a buffer with data
static void recorderCallback(SLAndroidSimpleBufferQueueItf caller __unused, void *context __unused)
{
SLresult result;
pthread_mutex_lock(&mutex);
// We should only be called when a recording buffer is done
assert(rxFront <= rxBufCount);
assert(rxRear <= rxBufCount);
assert(rxFront != rxRear);
char *buffer = rxBuffers[rxFront];
// Remove buffer from record queue
if (++rxFront > rxBufCount) {
rxFront = 0;
}
#if 1
ssize_t actual = audio_utils_fifo_write(&fifo, buffer, (size_t) bufSizeInFrames);
if (actual != (ssize_t) 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 (fifo2Buffer != NULL) {
actual = audio_utils_fifo_write(&fifo2, buffer, (size_t) bufSizeInFrames);
if (actual != (ssize_t) bufSizeInFrames) {
write(1, "?", 1);
}
}
// Enqueue this same buffer for the recorder to fill again.
result = (*recorderBufferQueue)->Enqueue(recorderBufferQueue, buffer, bufSizeInBytes);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
// Update our model of the record queue
SLuint32 rxRearNext = rxRear+1;
if (rxRearNext > rxBufCount) {
rxRearNext = 0;
}
assert(rxRearNext != rxFront);
rxBuffers[rxRear] = buffer;
rxRear = rxRearNext;
#else
// Enqueue the just-filled buffer for the player
result = (*playerBufferQueue)->Enqueue(playerBufferQueue, buffer, bufSizeInBytes);
if (SL_RESULT_SUCCESS == result) {
// There was room in the play queue, update our model of it
assert(txFront <= txBufCount);
assert(txRear <= txBufCount);
SLuint32 txRearNext = txRear+1;
if (txRearNext > txBufCount) {
txRearNext = 0;
}
assert(txRearNext != txFront);
txBuffers[txRear] = buffer;
txRear = txRearNext;
} else {
// Here if record has a filled buffer to play, but play queue is full.
assert(SL_RESULT_BUFFER_INSUFFICIENT == result);
write(1, "?", 1);
// We could either try again later, or discard. For now we discard and re-use buffer.
// Enqueue this same buffer for the recorder to fill again.
result = (*recorderBufferQueue)->Enqueue(recorderBufferQueue, buffer, bufSizeInBytes);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
// Update our model of the record queue
SLuint32 rxRearNext = rxRear+1;
if (rxRearNext > rxBufCount) {
rxRearNext = 0;
}
assert(rxRearNext != rxFront);
rxBuffers[rxRear] = buffer;
rxRear = rxRearNext;
}
#endif
pthread_mutex_unlock(&mutex);
}
// Called after audio player empties a buffer of data
static void playerCallback(SLBufferQueueItf caller __unused, void *context __unused)
{
SLresult result;
pthread_mutex_lock(&mutex);
// Get the buffer that just finished playing
assert(txFront <= txBufCount);
assert(txRear <= txBufCount);
assert(txFront != txRear);
char *buffer = txBuffers[txFront];
if (++txFront > txBufCount) {
txFront = 0;
}
#if 1
ssize_t actual = audio_utils_fifo_read(&fifo, buffer, bufSizeInFrames);
if (actual != (ssize_t) bufSizeInFrames) {
write(1, "/", 1);
// on underrun from pipe, substitute silence
memset(buffer, 0, bufSizeInFrames * channels * sizeof(short));
}
if (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 < bufSizeInFrames / 8; i += 8) {
for (int j = 0; j < 8; j++) {
for (unsigned k = 0; k < channels; k++) {
((short *)buffer)[(i+j)*channels+k] = j < 4 ? 0x7FFF : 0x8000;
}
}
}
injectImpulse = 0;
}
// Enqueue the filled buffer for playing
result = (*playerBufferQueue)->Enqueue(playerBufferQueue, buffer, bufSizeInBytes);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
// Update our model of the player queue
assert(txFront <= txBufCount);
assert(txRear <= txBufCount);
SLuint32 txRearNext = txRear+1;
if (txRearNext > txBufCount) {
txRearNext = 0;
}
assert(txRearNext != txFront);
txBuffers[txRear] = buffer;
txRear = txRearNext;
#else
// First try to enqueue the free buffer for recording
result = (*recorderBufferQueue)->Enqueue(recorderBufferQueue, buffer, bufSizeInBytes);
if (SL_RESULT_SUCCESS == result) {
// There was room in the record queue, update our model of it
assert(rxFront <= rxBufCount);
assert(rxRear <= rxBufCount);
SLuint32 rxRearNext = rxRear+1;
if (rxRearNext > rxBufCount) {
rxRearNext = 0;
}
assert(rxRearNext != rxFront);
rxBuffers[rxRear] = buffer;
rxRear = rxRearNext;
} else {
// Here if record queue is full
assert(SL_RESULT_BUFFER_INSUFFICIENT == result);
// Instead enqueue the free buffer on the free queue
assert(freeFront <= freeBufCount);
assert(freeRear <= freeBufCount);
SLuint32 freeRearNext = freeRear+1;
if (freeRearNext > freeBufCount) {
freeRearNext = 0;
}
// There must always be room in the free queue
assert(freeRearNext != freeFront);
freeBuffers[freeRear] = buffer;
freeRear = freeRearNext;
}
#endif
pthread_mutex_unlock(&mutex);
}
// Main program
int main(int argc, char **argv)
{
const char *outFileName = NULL;
// process command-line options
int i;
for (i = 1; i < argc; ++i) {
char *arg = argv[i];
if (arg[0] != '-') {
break;
}
// -r# number of slots in receive buffer queue
if (!strncmp(arg, "-r", 2)) {
rxBufCount = atoi(&arg[2]);
if (rxBufCount < 1 || rxBufCount > 16) {
fprintf(stderr, "%s: unusual receive buffer queue size (%u buffers)\n", argv[0],
(unsigned) rxBufCount);
}
// -t# number of slots in transmit buffer queue
} else if (!strncmp(arg, "-t", 2)) {
txBufCount = atoi(&arg[2]);
if (txBufCount < 1 || txBufCount > 16) {
fprintf(stderr, "%s: unusual transmit buffer queue size (%u buffers)\n", argv[0],
(unsigned) txBufCount);
}
// -f# size of each buffer in frames
} else if (!strncmp(arg, "-f", 2)) {
bufSizeInFrames = atoi(&arg[2]);
if (bufSizeInFrames == 0) {
fprintf(stderr, "%s: unusual buffer size (%u frames)\n", argv[0],
(unsigned) bufSizeInFrames);
}
// -c1 mono or -c2 stereo
} else if (!strncmp(arg, "-c", 2)) {
channels = atoi(&arg[2]);
if (channels < 1 || channels > 2) {
fprintf(stderr, "%s: unusual channel count ignored (%u)\n", argv[0],
(unsigned) channels);
channels = 2;
}
// -s# sample rate in Hz
} else if (!strncmp(arg, "-s", 2)) {
sampleRate = atoi(&arg[2]);
switch (sampleRate) {
case 8000:
case 11025:
case 12000:
case 16000:
case 22050:
case 24000:
case 32000:
case 44100:
case 48000:
break;
default:
fprintf(stderr, "%s: unusual sample rate (%u Hz)\n", argv[0],
(unsigned) sampleRate);
break;
}
// -e# exit after this many seconds
} else if (!strncmp(arg, "-e", 2)) {
exitAfterSeconds = atoi(&arg[2]);
// -ofile log to output file also
} else if (!strncmp(arg, "-o", 2)) {
outFileName = &arg[2];
// -i# inject an impulse after # milliseconds
} else if (!strncmp(arg, "-i", 2)) {
injectImpulse = atoi(&arg[2]);
} else
fprintf(stderr, "%s: unknown option %s\n", argv[0], arg);
}
// no other arguments allowed
if (i < argc) {
fprintf(stderr, "usage: %s -r# -t# -f# -s# -c# -i# -ofile\n", argv[0]);
fprintf(stderr, " -r# receive buffer queue count for microphone input, default 1\n");
fprintf(stderr, " -t# transmit buffer queue count for speaker output, default 2\n");
fprintf(stderr, " -f# number of frames per buffer, default 512\n");
fprintf(stderr, " -s# sample rate in Hz, default 44100\n");
fprintf(stderr, " -c1 mono\n");
fprintf(stderr, " -c2 stereo, default\n");
fprintf(stderr, " -i# inject impulse after # milliseconds\n");
fprintf(stderr, " -ofile log input to specified .wav file also\n");
}
// compute total free buffers as -r plus -t
freeBufCount = rxBufCount + txBufCount;
// compute buffer size
bufSizeInBytes = channels * bufSizeInFrames * sizeof(short);
// Initialize free buffers
freeBuffers = (char **) calloc(freeBufCount+1, sizeof(char *));
unsigned j;
for (j = 0; j < freeBufCount; ++j) {
freeBuffers[j] = (char *) malloc(bufSizeInBytes);
}
freeFront = 0;
freeRear = freeBufCount;
freeBuffers[j] = NULL;
// Initialize record queue
rxBuffers = (char **) calloc(rxBufCount+1, sizeof(char *));
rxFront = 0;
rxRear = 0;
// Initialize play queue
txBuffers = (char **) calloc(txBufCount+1, sizeof(char *));
txFront = 0;
txRear = 0;
size_t frameSize = channels * sizeof(short);
#define FIFO_FRAMES 1024
short *fifoBuffer = new short[FIFO_FRAMES * channels];
audio_utils_fifo_init(&fifo, FIFO_FRAMES, frameSize, 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
fifo2Buffer = new short[FIFO2_FRAMES * channels];
audio_utils_fifo_init(&fifo2, FIFO2_FRAMES, frameSize, fifo2Buffer);
} else {
fprintf(stderr, "sf_open failed\n");
}
} else {
sndfile = NULL;
}
SLresult result;
// create engine
SLObjectItf engineObject;
result = slCreateEngine(&engineObject, 0, NULL, 0, NULL, NULL);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
result = (*engineObject)->Realize(engineObject, SL_BOOLEAN_FALSE);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
SLEngineItf engineEngine;
result = (*engineObject)->GetInterface(engineObject, SL_IID_ENGINE, &engineEngine);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
// create output mix
SLObjectItf outputmixObject;
result = (*engineEngine)->CreateOutputMix(engineEngine, &outputmixObject, 0, NULL, NULL);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
result = (*outputmixObject)->Realize(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 = txBufCount;
locator_outputmix.locatorType = SL_DATALOCATOR_OUTPUTMIX;
locator_outputmix.outputMix = outputmixObject;
pcm.formatType = SL_DATAFORMAT_PCM;
pcm.numChannels = channels;
pcm.samplesPerSec = sampleRate * 1000;
pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
pcm.containerSize = 16;
pcm.channelMask = 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;
SLObjectItf playerObject = NULL;
SLObjectItf recorderObject = NULL;
SLInterfaceID ids_tx[1] = {SL_IID_BUFFERQUEUE};
SLboolean flags_tx[1] = {SL_BOOLEAN_TRUE};
result = (*engineEngine)->CreateAudioPlayer(engineEngine, &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);
goto cleanup;
}
ASSERT_EQ(SL_RESULT_SUCCESS, result);
result = (*playerObject)->Realize(playerObject, SL_BOOLEAN_FALSE);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
SLPlayItf playerPlay;
result = (*playerObject)->GetInterface(playerObject, SL_IID_PLAY, &playerPlay);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
result = (*playerObject)->GetInterface(playerObject, SL_IID_BUFFERQUEUE, &playerBufferQueue);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
result = (*playerBufferQueue)->RegisterCallback(playerBufferQueue, playerCallback, NULL);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
// Enqueue some zero buffers for the player
for (j = 0; j < txBufCount; ++j) {
// allocate a free buffer
assert(freeFront != freeRear);
char *buffer = freeBuffers[freeFront];
if (++freeFront > freeBufCount) {
freeFront = 0;
}
// put on play queue
SLuint32 txRearNext = txRear + 1;
if (txRearNext > txBufCount) {
txRearNext = 0;
}
assert(txRearNext != txFront);
txBuffers[txRear] = buffer;
txRear = txRearNext;
result = (*playerBufferQueue)->Enqueue(playerBufferQueue,
buffer, 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 = rxBufCount;
audiosnk.pLocator = &locator_bufferqueue_rx;
audiosnk.pFormat = &pcm;
{
SLInterfaceID ids_rx[1] = {SL_IID_ANDROIDSIMPLEBUFFERQUEUE};
SLboolean flags_rx[1] = {SL_BOOLEAN_TRUE};
result = (*engineEngine)->CreateAudioRecorder(engineEngine, &recorderObject, &audiosrc,
&audiosnk, 1, 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);
goto cleanup;
}
}
ASSERT_EQ(SL_RESULT_SUCCESS, result);
result = (*recorderObject)->Realize(recorderObject, SL_BOOLEAN_FALSE);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
SLRecordItf recorderRecord;
result = (*recorderObject)->GetInterface(recorderObject, SL_IID_RECORD, &recorderRecord);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
result = (*recorderObject)->GetInterface(recorderObject, SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
&recorderBufferQueue);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
result = (*recorderBufferQueue)->RegisterCallback(recorderBufferQueue, recorderCallback, NULL);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
// Enqueue some empty buffers for the recorder
for (j = 0; j < rxBufCount; ++j) {
// allocate a free buffer
assert(freeFront != freeRear);
char *buffer = freeBuffers[freeFront];
if (++freeFront > freeBufCount) {
freeFront = 0;
}
// put on record queue
SLuint32 rxRearNext = rxRear + 1;
if (rxRearNext > rxBufCount) {
rxRearNext = 0;
}
assert(rxRearNext != rxFront);
rxBuffers[rxRear] = buffer;
rxRear = rxRearNext;
result = (*recorderBufferQueue)->Enqueue(recorderBufferQueue,
buffer, bufSizeInBytes);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
}
// Kick off the recorder
result = (*recorderRecord)->SetRecordState(recorderRecord, SL_RECORDSTATE_RECORDING);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
#if 0
// give recorder a head start so that the pipe is initially filled
sleep(1);
#endif
// Wait patiently
do {
for (int i = 0; i < 10; i++) {
usleep(100000);
if (fifo2Buffer != NULL) {
for (;;) {
short buffer[bufSizeInFrames * channels];
ssize_t actual = audio_utils_fifo_read(&fifo2, buffer, bufSizeInFrames);
if (actual <= 0)
break;
(void) sf_writef_short(sndfile, buffer, (sf_count_t) actual);
}
}
if (injectImpulse > 0) {
if (injectImpulse <= 100) {
injectImpulse = -1;
write(1, "I", 1);
} else {
if ((injectImpulse % 1000) < 100) {
write(1, "i", 1);
}
injectImpulse -= 100;
}
} else if (i == 9) {
write(1, ".", 1);
}
}
SLBufferQueueState playerBQState;
result = (*playerBufferQueue)->GetState(playerBufferQueue, &playerBQState);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
SLAndroidSimpleBufferQueueState recorderBQState;
result = (*recorderBufferQueue)->GetState(recorderBufferQueue, &recorderBQState);
ASSERT_EQ(SL_RESULT_SUCCESS, result);
} while (--exitAfterSeconds);
// Tear down the objects and exit
cleanup:
audio_utils_fifo_deinit(&fifo);
delete[] fifoBuffer;
if (sndfile != NULL) {
audio_utils_fifo_deinit(&fifo2);
delete[] fifo2Buffer;
sf_close(sndfile);
}
if (NULL != playerObject) {
(*playerObject)->Destroy(playerObject);
}
if (NULL != recorderObject) {
(*recorderObject)->Destroy(recorderObject);
}
(*outputmixObject)->Destroy(outputmixObject);
(*engineObject)->Destroy(engineObject);
return EXIT_SUCCESS;
}