int main()
{
char buf[256];
register int i;
FILE *f;
sampleFormat(PA_SAMPLE_S16BE,2); /* Valor por defecto, no hace falta llamar a esta función */
if(openRecord("prueba")) puts("error");
if(openPlay("prueba")) puts("error");
f=fopen("pruebasonido","w+b");
puts("Grabando");
for(i=0; i < 10000; ++i)
{
recordSound(buf,160);
fwrite(buf,1,160,f);
}
fclose(f);
closeRecord();
f=fopen("pruebasonido","rb");
puts("Reproduciendo");
for(i=0; i < 10000; ++i)
{
fread(buf,1,160,f);
playSound(buf,160);
}
fclose(f);
closePlay();
}
status_t AudioRecord::start()
{
status_t ret = NO_ERROR;
sp<ClientRecordThread> t = mClientRecordThread;
LOGV("start");
if (t != 0) {
if (t->exitPending()) {
if (t->requestExitAndWait() == WOULD_BLOCK) {
LOGE("AudioRecord::start called from thread");
return WOULD_BLOCK;
}
}
t->mLock.lock();
}
if (android_atomic_or(1, &mActive) == 0) {
ret = mAudioRecord->start();
if (ret == DEAD_OBJECT) {
LOGV("start() dead IAudioRecord: creating a new one");
ret = openRecord(mCblk->sampleRate, mFormat, mChannelCount,
mFrameCount, mFlags, getInput());
if (ret == NO_ERROR) {
ret = mAudioRecord->start();
}
}
if (ret == NO_ERROR) {
mNewPosition = mCblk->user + mUpdatePeriod;
mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
mCblk->waitTimeMs = 0;
if (t != 0) {
t->run("ClientRecordThread", THREAD_PRIORITY_AUDIO_CLIENT);
} else {
setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
}
} else {
LOGV("start() failed");
android_atomic_and(~1, &mActive);
}
}
if (t != 0) {
t->mLock.unlock();
}
return ret;
}
FMRadioSource::FMRadioSource()
: mInitCheck(NO_INIT),
mStarted(false),
mSessionId(AudioSystem::newAudioSessionId()) {
// get FM Radio RX input
audio_in_acoustics_t flags = (audio_in_acoustics_t)
(AUDIO_IN_ACOUSTICS_AGC_DISABLE |
AUDIO_IN_ACOUSTICS_NS_DISABLE |
AUDIO_IN_ACOUSTICS_TX_DISABLE );
audio_io_handle_t input = AudioSystem::getInput(AUDIO_SOURCE_FM_RADIO_RX,
kSampleRate,
kAudioFormat,
kChannelMask,
(audio_in_acoustics_t)flags,
mSessionId);
if (input == 0) {
ALOGE("Could not get audio input for FM Radio source");
mInitCheck = UNKNOWN_ERROR;
return;
}
// get frame count
int frameCount = 0;
status_t status = AudioRecord::getMinFrameCount(&frameCount, kSampleRate,
kAudioFormat, popcount(kChannelMask));
if (status != NO_ERROR) {
mInitCheck = status;
return;
}
// create the IAudioRecord
status = openRecord(frameCount, input);
if (status != NO_ERROR) {
mInitCheck = status;
return;
}
AudioSystem::acquireAudioSessionId(mSessionId);
mInitCheck = OK;
return;
}
status_t BnCameraRecordService::onTransact(uint32_t code, const Parcel& data,
Parcel* reply, uint32_t flags)
{
REPORT_FUNCTION();
switch (code) {
case INIT_RECORD: {
CHECK_INTERFACE(ICameraRecordService, data, reply);
uint32_t sampleRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
audio_channel_mask_t channelMask = data.readInt32();
reply->writeInt32(initRecord(sampleRate, format, channelMask));
return NO_ERROR;
} break;
case OPEN_RECORD: {
CHECK_INTERFACE(ICameraRecordService, data, reply);
uint32_t sampleRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
audio_channel_mask_t channelMask = data.readInt32();
size_t frameCount = data.readInt32();
pid_t tid = (pid_t) data.readInt32();
int sessionId = data.readInt32();
status_t status;
sp<IAudioRecord> record = openRecord(sampleRate, format, channelMask,
frameCount, tid, &sessionId, &status);
LOG_ALWAYS_FATAL_IF((record != 0) != (status == NO_ERROR));
reply->writeInt32(sessionId);
reply->writeInt32(status);
reply->writeStrongBinder(record->asBinder());
return NO_ERROR;
} break;
default:
return BBinder::onTransact(code, data, reply, flags);
}
return NO_ERROR;
}
status_t AudioRecord::set(
int inputSource,
uint32_t sampleRate,
int format,
uint32_t channels,
int frameCount,
uint32_t flags,
callback_t cbf,
void* user,
int notificationFrames,
bool threadCanCallJava)
{
LOGV("set(): sampleRate %d, channels %d, frameCount %d",sampleRate, channels, frameCount);
if (mAudioRecord != 0) {
return INVALID_OPERATION;
}
if (inputSource == AUDIO_SOURCE_DEFAULT) {
inputSource = AUDIO_SOURCE_MIC;
}
if (sampleRate == 0) {
sampleRate = DEFAULT_SAMPLE_RATE;
}
// these below should probably come from the audioFlinger too...
if (format == 0) {
format = AudioSystem::PCM_16_BIT;
}
// validate parameters
if (!AudioSystem::isValidFormat(format)) {
LOGE("Invalid format");
return BAD_VALUE;
}
//hengai Fix:Record channels error
if(channels == 1) channels = 0x04;
else if(channels == 2) channels = 0x0c;
if (!AudioSystem::isInputChannel(channels)) {
return BAD_VALUE;
}
int channelCount = AudioSystem::popCount(channels);
audio_io_handle_t input = AudioSystem::getInput(inputSource,
sampleRate, format, channels, (AudioSystem::audio_in_acoustics)flags);
if (input == 0) {
LOGE("Could not get audio input for record source %d", inputSource);
return BAD_VALUE;
}
// validate framecount
size_t inputBuffSizeInBytes = -1;
if (AudioSystem::getInputBufferSize(sampleRate, format, channelCount, &inputBuffSizeInBytes)
!= NO_ERROR) {
LOGE("AudioSystem could not query the input buffer size.");
return NO_INIT;
}
if (inputBuffSizeInBytes == 0) {
LOGE("Recording parameters are not supported: sampleRate %d, channelCount %d, format %d",
sampleRate, channelCount, format);
return BAD_VALUE;
}
int frameSizeInBytes = channelCount * (format == AudioSystem::PCM_16_BIT ? 2 : 1);
if (AudioSystem::isLinearPCM(format)) {
frameSizeInBytes = channelCount * (format == AudioSystem::PCM_16_BIT ? sizeof(int16_t) : sizeof(int8_t));
} else {
frameSizeInBytes = sizeof(int8_t);
}
// We use 2* size of input buffer for ping pong use of record buffer.
int minFrameCount = 2 * inputBuffSizeInBytes / frameSizeInBytes;
LOGV("AudioRecord::set() minFrameCount = %d", minFrameCount);
if (frameCount == 0) {
frameCount = minFrameCount;
} else if (frameCount < minFrameCount) {
return BAD_VALUE;
}
if (notificationFrames == 0) {
notificationFrames = frameCount/2;
}
// create the IAudioRecord
status_t status = openRecord(sampleRate, format, channelCount,
frameCount, flags, input);
if (status != NO_ERROR) {
return status;
}
if (cbf != 0) {
mClientRecordThread = new ClientRecordThread(*this, threadCanCallJava);
if (mClientRecordThread == 0) {
return NO_INIT;
}
}
mStatus = NO_ERROR;
mFormat = format;
// Update buffer size in case it has been limited by AudioFlinger during track creation
mFrameCount = mCblk->frameCount;
mChannelCount = (uint8_t)channelCount;
mChannels = channels;
mActive = 0;
mCbf = cbf;
mNotificationFrames = notificationFrames;
mRemainingFrames = notificationFrames;
mUserData = user;
// TODO: add audio hardware input latency here
mLatency = (1000*mFrameCount) / sampleRate;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
mInputSource = (uint8_t)inputSource;
mFlags = flags;
mInput = input;
return NO_ERROR;
}
status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
{
int active;
status_t result;
audio_track_cblk_t* cblk = mCblk;
uint32_t framesReq = audioBuffer->frameCount;
uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS;
audioBuffer->frameCount = 0;
audioBuffer->size = 0;
uint32_t framesReady = cblk->framesReady();
if (framesReady == 0) {
cblk->lock.lock();
goto start_loop_here;
while (framesReady == 0) {
active = mActive;
if (UNLIKELY(!active)) {
cblk->lock.unlock();
return NO_MORE_BUFFERS;
}
if (UNLIKELY(!waitCount)) {
cblk->lock.unlock();
return WOULD_BLOCK;
}
result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
if (__builtin_expect(result!=NO_ERROR, false)) {
cblk->waitTimeMs += waitTimeMs;
if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
LOGW( "obtainBuffer timed out (is the CPU pegged?) "
"user=%08x, server=%08x", cblk->user, cblk->server);
cblk->lock.unlock();
result = mAudioRecord->start();
if (result == DEAD_OBJECT) {
LOGW("obtainBuffer() dead IAudioRecord: creating a new one");
result = openRecord(cblk->sampleRate, mFormat, mChannelCount,
mFrameCount, mFlags, getInput());
if (result == NO_ERROR) {
cblk = mCblk;
mAudioRecord->start();
}
}
cblk->lock.lock();
cblk->waitTimeMs = 0;
}
if (--waitCount == 0) {
cblk->lock.unlock();
return TIMED_OUT;
}
}
// read the server count again
start_loop_here:
framesReady = cblk->framesReady();
}
cblk->lock.unlock();
}
cblk->waitTimeMs = 0;
if (framesReq > framesReady) {
framesReq = framesReady;
}
uint32_t u = cblk->user;
uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
if (u + framesReq > bufferEnd) {
framesReq = bufferEnd - u;
}
audioBuffer->flags = 0;
audioBuffer->channelCount= mChannelCount;
audioBuffer->format = mFormat;
audioBuffer->frameCount = framesReq;
audioBuffer->size = framesReq*cblk->frameSize;
audioBuffer->raw = (int8_t*)cblk->buffer(u);
active = mActive;
return active ? status_t(NO_ERROR) : status_t(STOPPED);
}
