status_t AudioRecord::set(
audio_source_t inputSource,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
callback_t cbf,
void* user,
uint32_t notificationFrames,
bool threadCanCallJava,
int sessionId,
transfer_type transferType,
audio_input_flags_t flags)
{
ALOGV("set(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
"notificationFrames %u, sessionId %d, transferType %d, flags %#x",
inputSource, sampleRate, format, channelMask, frameCount, notificationFrames,
sessionId, transferType, flags);
switch (transferType) {
case TRANSFER_DEFAULT:
if (cbf == NULL || threadCanCallJava) {
transferType = TRANSFER_SYNC;
} else {
transferType = TRANSFER_CALLBACK;
}
break;
case TRANSFER_CALLBACK:
if (cbf == NULL) {
ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL");
return BAD_VALUE;
}
break;
case TRANSFER_OBTAIN:
case TRANSFER_SYNC:
break;
default:
ALOGE("Invalid transfer type %d", transferType);
return BAD_VALUE;
}
mTransfer = transferType;
AutoMutex lock(mLock);
// invariant that mAudioRecord != 0 is true only after set() returns successfully
if (mAudioRecord != 0) {
ALOGE("Track already in use");
return INVALID_OPERATION;
}
// handle default values first.
if (inputSource == AUDIO_SOURCE_DEFAULT) {
inputSource = AUDIO_SOURCE_MIC;
}
mInputSource = inputSource;
if (sampleRate == 0) {
ALOGE("Invalid sample rate %u", sampleRate);
return BAD_VALUE;
}
mSampleRate = sampleRate;
// these below should probably come from the audioFlinger too...
if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
// validate parameters
if (!audio_is_valid_format(format)) {
ALOGE("Invalid format %#x", format);
return BAD_VALUE;
}
// Temporary restriction: AudioFlinger currently supports 16-bit PCM and compress formats only
if (format != AUDIO_FORMAT_PCM_16_BIT &&
!audio_is_compress_voip_format(format) &&
!audio_is_compress_capture_format(format)) {
ALOGE("Format %#x is not supported", format);
return BAD_VALUE;
}
mFormat = format;
if (!audio_is_input_channel(channelMask)) {
ALOGE("Invalid channel mask %#x", channelMask);
return BAD_VALUE;
}
mChannelMask = channelMask;
uint32_t channelCount = audio_channel_count_from_in_mask(channelMask);
mChannelCount = channelCount;
if (audio_is_linear_pcm(format)) {
mFrameSize = channelCount * audio_bytes_per_sample(format);
} else {
mFrameSize = sizeof(uint8_t);
}
// mFrameCount is initialized in openRecord_l
mReqFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
// mNotificationFramesAct is initialized in openRecord_l
if (sessionId == AUDIO_SESSION_ALLOCATE) {
mSessionId = AudioSystem::newAudioUniqueId();
} else {
mSessionId = sessionId;
}
ALOGV("set(): mSessionId %d", mSessionId);
mFlags = flags;
mCbf = cbf;
if (cbf != NULL) {
mAudioRecordThread = new AudioRecordThread(*this, threadCanCallJava);
mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);
}
// create the IAudioRecord
status_t status = openRecord_l(0 /*epoch*/);
if (status != NO_ERROR) {
if (mAudioRecordThread != 0) {
mAudioRecordThread->requestExit(); // see comment in AudioRecord.h
mAudioRecordThread->requestExitAndWait();
mAudioRecordThread.clear();
}
return status;
}
mStatus = NO_ERROR;
mActive = false;
mUserData = user;
// TODO: add audio hardware input latency here
mLatency = (1000*mFrameCount) / sampleRate;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
AudioSystem::acquireAudioSessionId(mSessionId, -1);
mSequence = 1;
mObservedSequence = mSequence;
mInOverrun = false;
return NO_ERROR;
}
// checks if the IO profile is compatible with specified parameters.
// Sampling rate, format and channel mask must be specified in order to
// get a valid a match
bool IOProfile::isCompatibleProfile(audio_devices_t device,
String8 address,
uint32_t samplingRate,
uint32_t *updatedSamplingRate,
audio_format_t format,
audio_format_t *updatedFormat,
audio_channel_mask_t channelMask,
audio_channel_mask_t *updatedChannelMask,
uint32_t flags) const
{
const bool isPlaybackThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SOURCE;
const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
ALOG_ASSERT(isPlaybackThread != isRecordThread);
if (device != AUDIO_DEVICE_NONE) {
// just check types if multiple devices are selected
if (popcount(device & ~AUDIO_DEVICE_BIT_IN) > 1) {
if ((mSupportedDevices.types() & device) != device) {
return false;
}
} else if (mSupportedDevices.getDevice(device, address) == 0) {
return false;
}
}
if (samplingRate == 0) {
return false;
}
uint32_t myUpdatedSamplingRate = samplingRate;
if (isPlaybackThread && checkExactSamplingRate(samplingRate) != NO_ERROR) {
return false;
}
if (isRecordThread && checkCompatibleSamplingRate(samplingRate, &myUpdatedSamplingRate) !=
NO_ERROR) {
return false;
}
if (!audio_is_valid_format(format)) {
return false;
}
if (isPlaybackThread && checkExactFormat(format) != NO_ERROR) {
return false;
}
audio_format_t myUpdatedFormat = format;
if (isRecordThread && checkCompatibleFormat(format, &myUpdatedFormat) != NO_ERROR) {
return false;
}
if (isPlaybackThread && (!audio_is_output_channel(channelMask) ||
checkExactChannelMask(channelMask) != NO_ERROR)) {
return false;
}
audio_channel_mask_t myUpdatedChannelMask = channelMask;
if (isRecordThread && (!audio_is_input_channel(channelMask) ||
checkCompatibleChannelMask(channelMask, &myUpdatedChannelMask) != NO_ERROR)) {
return false;
}
if (isPlaybackThread && (mFlags & flags) != flags) {
return false;
}
// The only input flag that is allowed to be different is the fast flag.
// An existing fast stream is compatible with a normal track request.
// An existing normal stream is compatible with a fast track request,
// but the fast request will be denied by AudioFlinger and converted to normal track.
if (isRecordThread && ((mFlags ^ flags) &
~AUDIO_INPUT_FLAG_FAST)) {
return false;
}
if (updatedSamplingRate != NULL) {
*updatedSamplingRate = myUpdatedSamplingRate;
}
if (updatedFormat != NULL) {
*updatedFormat = myUpdatedFormat;
}
if (updatedChannelMask != NULL) {
*updatedChannelMask = myUpdatedChannelMask;
}
return true;
}
status_t AudioRecord::set(
audio_source_t inputSource,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
callback_t cbf,
void* user,
uint32_t notificationFrames,
bool threadCanCallJava,
audio_session_t sessionId,
transfer_type transferType,
audio_input_flags_t flags,
int uid,
pid_t pid,
const audio_attributes_t* pAttributes)
{
ALOGV("set(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
"notificationFrames %u, sessionId %d, transferType %d, flags %#x, opPackageName %s "
"uid %d, pid %d",
inputSource, sampleRate, format, channelMask, frameCount, notificationFrames,
sessionId, transferType, flags, String8(mOpPackageName).string(), uid, pid);
switch (transferType) {
case TRANSFER_DEFAULT:
if (cbf == NULL || threadCanCallJava) {
transferType = TRANSFER_SYNC;
} else {
transferType = TRANSFER_CALLBACK;
}
break;
case TRANSFER_CALLBACK:
if (cbf == NULL) {
ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL");
return BAD_VALUE;
}
break;
case TRANSFER_OBTAIN:
case TRANSFER_SYNC:
break;
default:
ALOGE("Invalid transfer type %d", transferType);
return BAD_VALUE;
}
mTransfer = transferType;
// invariant that mAudioRecord != 0 is true only after set() returns successfully
if (mAudioRecord != 0) {
ALOGE("Track already in use");
return INVALID_OPERATION;
}
if (pAttributes == NULL) {
memset(&mAttributes, 0, sizeof(audio_attributes_t));
mAttributes.source = inputSource;
} else {
// stream type shouldn't be looked at, this track has audio attributes
memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
ALOGV("Building AudioRecord with attributes: source=%d flags=0x%x tags=[%s]",
mAttributes.source, mAttributes.flags, mAttributes.tags);
}
mSampleRate = sampleRate;
// these below should probably come from the audioFlinger too...
if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
// validate parameters
// AudioFlinger capture only supports linear PCM
if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) {
ALOGE("Format %#x is not linear pcm", format);
return BAD_VALUE;
}
mFormat = format;
if (!audio_is_input_channel(channelMask)) {
ALOGE("Invalid channel mask %#x", channelMask);
return BAD_VALUE;
}
mChannelMask = channelMask;
uint32_t channelCount = audio_channel_count_from_in_mask(channelMask);
mChannelCount = channelCount;
if (audio_is_linear_pcm(format)) {
mFrameSize = channelCount * audio_bytes_per_sample(format);
} else {
mFrameSize = sizeof(uint8_t);
}
// mFrameCount is initialized in openRecord_l
mReqFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
// mNotificationFramesAct is initialized in openRecord_l
if (sessionId == AUDIO_SESSION_ALLOCATE) {
mSessionId = (audio_session_t) AudioSystem::newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
} else {
mSessionId = sessionId;
}
ALOGV("set(): mSessionId %d", mSessionId);
int callingpid = IPCThreadState::self()->getCallingPid();
int mypid = getpid();
if (uid == -1 || (callingpid != mypid)) {
mClientUid = IPCThreadState::self()->getCallingUid();
} else {
mClientUid = uid;
}
if (pid == -1 || (callingpid != mypid)) {
mClientPid = callingpid;
} else {
mClientPid = pid;
}
mOrigFlags = mFlags = flags;
mCbf = cbf;
if (cbf != NULL) {
mAudioRecordThread = new AudioRecordThread(*this, threadCanCallJava);
mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);
// thread begins in paused state, and will not reference us until start()
}
// create the IAudioRecord
status_t status = openRecord_l(0 /*epoch*/, mOpPackageName);
if (status != NO_ERROR) {
if (mAudioRecordThread != 0) {
mAudioRecordThread->requestExit(); // see comment in AudioRecord.h
mAudioRecordThread->requestExitAndWait();
mAudioRecordThread.clear();
}
return status;
}
mStatus = NO_ERROR;
mUserData = user;
// TODO: add audio hardware input latency here
mLatency = (1000 * mFrameCount) / mSampleRate;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
AudioSystem::acquireAudioSessionId(mSessionId, -1);
mSequence = 1;
mObservedSequence = mSequence;
mInOverrun = false;
mFramesRead = 0;
mFramesReadServerOffset = 0;
return NO_ERROR;
}
status_t AudioRecord::set(
int inputSource,
uint32_t sampleRate,
int format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
callback_t cbf,
void* user,
int notificationFrames,
bool threadCanCallJava,
int sessionId)
{
LOGV("set(): sampleRate %d, channelMask %d, frameCount %d",sampleRate, channelMask, frameCount);
AutoMutex lock(mLock);
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 = AUDIO_FORMAT_PCM_16_BIT;
}
// validate parameters
if (!audio_is_valid_format(format)) {
LOGE("Invalid format");
return BAD_VALUE;
}
if (!audio_is_input_channel(channelMask)) {
return BAD_VALUE;
}
int channelCount = popcount(channelMask);
if (sessionId == 0 ) {
mSessionId = AudioSystem::newAudioSessionId();
} else {
mSessionId = sessionId;
}
LOGV("set(): mSessionId %d", mSessionId);
audio_io_handle_t input = AudioSystem::getInput(inputSource,
sampleRate,
format,
channelMask,
(audio_in_acoustics_t)flags,
mSessionId);
if (input == 0) {
LOGE("Could not get audio input for record source %d", inputSource);
return BAD_VALUE;
}
// validate framecount
int minFrameCount = 0;
status_t status = getMinFrameCount(&minFrameCount, sampleRate, format, channelCount);
if (status != NO_ERROR) {
return status;
}
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 = openRecord_l(sampleRate, format, channelMask,
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;
mChannelMask = channelMask;
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;
AudioSystem::acquireAudioSessionId(mSessionId);
return NO_ERROR;
}
status_t AudioRecord::set(
audio_source_t inputSource,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
int frameCountInt,
callback_t cbf,
void* user,
int notificationFrames,
bool threadCanCallJava,
int sessionId,
transfer_type transferType,
audio_input_flags_t flags)
{
ALOGV("sampleRate %u, channelMask %#x, format %d", sampleRate, channelMask, format);
ALOGV("inputSource %d", inputSource);
switch (transferType) {
case TRANSFER_DEFAULT:
if (cbf == NULL || threadCanCallJava) {
transferType = TRANSFER_SYNC;
} else {
transferType = TRANSFER_CALLBACK;
}
break;
case TRANSFER_CALLBACK:
if (cbf == NULL) {
ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL");
return BAD_VALUE;
}
break;
case TRANSFER_OBTAIN:
case TRANSFER_SYNC:
break;
default:
ALOGE("Invalid transfer type %d", transferType);
return BAD_VALUE;
}
mTransfer = transferType;
// FIXME "int" here is legacy and will be replaced by size_t later
if (frameCountInt < 0) {
ALOGE("Invalid frame count %d", frameCountInt);
return BAD_VALUE;
}
size_t frameCount = frameCountInt;
ALOGV("set(): sampleRate %u, channelMask %#x, frameCount %u", sampleRate, channelMask,
frameCount);
AutoMutex lock(mLock);
if (mAudioRecord != 0) {
ALOGE("Track already in use");
return INVALID_OPERATION;
}
if (inputSource == AUDIO_SOURCE_DEFAULT) {
inputSource = AUDIO_SOURCE_MIC;
}
mInputSource = inputSource;
if (sampleRate == 0) {
ALOGE("Invalid sample rate %u", sampleRate);
return BAD_VALUE;
}
mSampleRate = sampleRate;
// these below should probably come from the audioFlinger too...
if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
// validate parameters
if (!audio_is_valid_format(format)) {
ALOGE("Invalid format %d", format);
return BAD_VALUE;
}
#if defined(QCOM_HARDWARE) && !defined(QCOM_DIRECTTRACK)
if (format != AUDIO_FORMAT_PCM_16_BIT &&
!audio_is_compress_voip_format(format) &&
!audio_is_compress_capture_format(format)) {
#else
#ifndef QCOM_DIRECTTRACK
// Temporary restriction: AudioFlinger currently supports 16-bit PCM only
if (format != AUDIO_FORMAT_PCM_16_BIT) {
#endif
#endif
#ifndef QCOM_DIRECTTRACK
ALOGE("Format %d is not supported", format);
return BAD_VALUE;
}
#endif
mFormat = format;
if (!audio_is_input_channel(channelMask)) {
ALOGE("Invalid channel mask %#x", channelMask);
return BAD_VALUE;
}
mChannelMask = channelMask;
uint32_t channelCount = popcount(channelMask);
mChannelCount = channelCount;
#ifdef QCOM_DIRECTTRACK
mFrameSize = frameSize();
size_t inputBuffSizeInBytes = -1;
status_t status = AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &inputBuffSizeInBytes);
if (status != NO_ERROR) {
ALOGE("AudioSystem could not query the input buffer size; status %d", status);
return NO_INIT;
}
if (inputBuffSizeInBytes == 0) {
ALOGE("Unsupported configuration: sampleRate %u, format %d, channelMask %#x",
sampleRate, format, channelMask);
return BAD_VALUE;
}
int minFrameCount = (inputBuffSizeInBytes * 2)/mFrameSize;
#else
// Assumes audio_is_linear_pcm(format), else sizeof(uint8_t)
#ifdef QCOM_HARDWARE
if (audio_is_linear_pcm(format))
mFrameSize = channelCount * audio_bytes_per_sample(format);
else
mFrameSize = sizeof(uint8_t);
#else
mFrameSize = channelCount * audio_bytes_per_sample(format);
#endif
// validate framecount
size_t minFrameCount = 0;
status_t status = AudioRecord::getMinFrameCount(&minFrameCount,
sampleRate, format, channelMask);
if (status != NO_ERROR) {
ALOGE("getMinFrameCount() failed; status %d", status);
return status;
}
#endif
ALOGV("AudioRecord::set() minFrameCount = %d", minFrameCount);
if (frameCount == 0) {
frameCount = minFrameCount;
} else if (frameCount < minFrameCount) {
ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount);
return BAD_VALUE;
}
mFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
mNotificationFramesAct = 0;
if (sessionId == 0 ) {
mSessionId = AudioSystem::newAudioSessionId();
} else {
mSessionId = sessionId;
}
ALOGV("set(): mSessionId %d", mSessionId);
mFlags = flags;
// create the IAudioRecord
status = openRecord_l(0 /*epoch*/);
if (status) {
return status;
}
if (cbf != NULL) {
mAudioRecordThread = new AudioRecordThread(*this, threadCanCallJava);
mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);
}
mStatus = NO_ERROR;
// Update buffer size in case it has been limited by AudioFlinger during track creation
mFrameCount = mCblk->frameCount_;
mActive = false;
mCbf = cbf;
mRefreshRemaining = true;
mUserData = user;
// TODO: add audio hardware input latency here
mLatency = (1000*mFrameCount) / sampleRate;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
AudioSystem::acquireAudioSessionId(mSessionId);
mSequence = 1;
mObservedSequence = mSequence;
mInOverrun = false;
return NO_ERROR;
}
#ifdef QCOM_DIRECTTRACK
audio_source_t AudioRecord::inputSource() const
{
return mInputSource;
}
#endif
// -------------------------------------------------------------------------
status_t AudioRecord::start(AudioSystem::sync_event_t event, int triggerSession)
{
ALOGV("start, sync event %d trigger session %d", event, triggerSession);
AutoMutex lock(mLock);
if (mActive) {
return NO_ERROR;
}
// reset current position as seen by client to 0
mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition());
mNewPosition = mProxy->getPosition() + mUpdatePeriod;
int32_t flags = android_atomic_acquire_load(&mCblk->mFlags);
status_t status = NO_ERROR;
if (!(flags & CBLK_INVALID)) {
ALOGV("mAudioRecord->start()");
status = mAudioRecord->start(event, triggerSession);
if (status == DEAD_OBJECT) {
flags |= CBLK_INVALID;
}
}
if (flags & CBLK_INVALID) {
status = restoreRecord_l("start");
}
if (status != NO_ERROR) {
ALOGE("start() status %d", status);
} else {
mActive = true;
sp<AudioRecordThread> t = mAudioRecordThread;
if (t != 0) {
t->resume();
} else {
mPreviousPriority = getpriority(PRIO_PROCESS, 0);
get_sched_policy(0, &mPreviousSchedulingGroup);
androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
}
}
return status;
}