Merge "Ensure MEDIA_INFO_BUFFERING_END is always sent"
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 7e66ac9..156c592 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -84,8 +84,8 @@
* more bytes than indicated by 'size' field and update 'size' if less bytes are
* read.
* - EVENT_OVERRUN: unused.
- * - EVENT_MARKER: pointer to an uin32_t containing the marker position in frames.
- * - EVENT_NEW_POS: pointer to an uin32_t containing the new position in frames.
+ * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames.
+ * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames.
*/
typedef void (*callback_t)(int event, void* user, void *info);
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 9d5813b..5060525 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -355,7 +355,6 @@
uint32_t AudioRecord::getSampleRate() const
{
- AutoMutex lock(mLock);
return mCblk->sampleRate;
}
@@ -363,6 +362,7 @@
{
if (mCbf == NULL) return INVALID_OPERATION;
+ AutoMutex lock(mLock);
mMarkerPosition = marker;
mMarkerReached = false;
@@ -373,6 +373,7 @@
{
if (marker == NULL) return BAD_VALUE;
+ AutoMutex lock(mLock);
*marker = mMarkerPosition;
return NO_ERROR;
@@ -384,6 +385,8 @@
uint32_t curPosition;
getPosition(&curPosition);
+
+ AutoMutex lock(mLock);
mNewPosition = curPosition + updatePeriod;
mUpdatePeriod = updatePeriod;
@@ -394,6 +397,7 @@
{
if (updatePeriod == NULL) return BAD_VALUE;
+ AutoMutex lock(mLock);
*updatePeriod = mUpdatePeriod;
return NO_ERROR;
@@ -434,6 +438,7 @@
pid_t tid = -1;
// FIXME see similar logic at AudioTrack
+ int originalSessionId = mSessionId;
sp<IAudioRecord> record = audioFlinger->openRecord(getpid(), input,
sampleRate, format,
channelMask,
@@ -442,6 +447,8 @@
tid,
&mSessionId,
&status);
+ ALOGE_IF(originalSessionId != 0 && mSessionId != originalSessionId,
+ "session ID changed from %d to %d", originalSessionId, mSessionId);
if (record == 0) {
ALOGE("AudioFlinger could not create record track, status: %d", status);
@@ -587,6 +594,7 @@
int AudioRecord::getSessionId() const
{
+ // no lock needed because session ID doesn't change after first set()
return mSessionId;
}
@@ -658,22 +666,31 @@
sp<IMemory> iMem = mCblkMemory;
audio_track_cblk_t* cblk = mCblk;
bool active = mActive;
+ uint32_t markerPosition = mMarkerPosition;
+ uint32_t newPosition = mNewPosition;
+ uint32_t user = cblk->user;
+ // determine whether a marker callback will be needed, while locked
+ bool needMarker = !mMarkerReached && (mMarkerPosition > 0) && (user >= mMarkerPosition);
+ if (needMarker) {
+ mMarkerReached = true;
+ }
+ // determine the number of new position callback(s) that will be needed, while locked
+ uint32_t updatePeriod = mUpdatePeriod;
+ uint32_t needNewPos = updatePeriod > 0 && user >= newPosition ?
+ ((user - newPosition) / updatePeriod) + 1 : 0;
+ mNewPosition = newPosition + updatePeriod * needNewPos;
mLock.unlock();
- // Manage marker callback
- if (!mMarkerReached && (mMarkerPosition > 0)) {
- if (cblk->user >= mMarkerPosition) {
- mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition);
- mMarkerReached = true;
- }
+ // perform marker callback, while unlocked
+ if (needMarker) {
+ mCbf(EVENT_MARKER, mUserData, &markerPosition);
}
- // Manage new position callback
- if (mUpdatePeriod > 0) {
- while (cblk->user >= mNewPosition) {
- mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
- mNewPosition += mUpdatePeriod;
- }
+ // perform new position callback(s), while unlocked
+ for (; needNewPos > 0; --needNewPos) {
+ uint32_t temp = newPosition;
+ mCbf(EVENT_NEW_POS, mUserData, &temp);
+ newPosition += updatePeriod;
}
do {
@@ -721,7 +738,7 @@
// otherwise we would have exited when obtainBuffer returned STOPPED earlier.
ALOGV("Overrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags);
if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) {
- mCbf(EVENT_OVERRUN, mUserData, 0);
+ mCbf(EVENT_OVERRUN, mUserData, NULL);
}
}
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index cc18a1d..163205e 100755
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -42,6 +42,7 @@
namespace android {
+static const int64_t kMinStreamableFileSizeInBytes = 5 * 1024 * 1024;
static const int64_t kMax32BitFileSize = 0x007fffffffLL;
static const uint8_t kNalUnitTypeSeqParamSet = 0x07;
static const uint8_t kNalUnitTypePicParamSet = 0x08;
@@ -487,8 +488,16 @@
CHECK_GT(mTimeScale, 0);
ALOGV("movie time scale: %d", mTimeScale);
- mStreamableFile = true;
- mWriteMoovBoxToMemory = false;
+ /*
+ * When the requested file size limit is small, the priority
+ * is to meet the file size limit requirement, rather than
+ * to make the file streamable.
+ */
+ mStreamableFile =
+ (mMaxFileSizeLimitBytes != 0 &&
+ mMaxFileSizeLimitBytes >= kMinStreamableFileSizeInBytes);
+
+ mWriteMoovBoxToMemory = mStreamableFile;
mMoovBoxBuffer = NULL;
mMoovBoxBufferOffset = 0;
@@ -504,11 +513,16 @@
mEstimatedMoovBoxSize = estimateMoovBoxSize(bitRate);
}
CHECK_GE(mEstimatedMoovBoxSize, 8);
- lseek64(mFd, mFreeBoxOffset, SEEK_SET);
- writeInt32(mEstimatedMoovBoxSize);
- write("free", 4);
+ if (mStreamableFile) {
+ // Reserve a 'free' box only for streamable file
+ lseek64(mFd, mFreeBoxOffset, SEEK_SET);
+ writeInt32(mEstimatedMoovBoxSize);
+ write("free", 4);
+ mMdatOffset = mFreeBoxOffset + mEstimatedMoovBoxSize;
+ } else {
+ mMdatOffset = mOffset;
+ }
- mMdatOffset = mFreeBoxOffset + mEstimatedMoovBoxSize;
mOffset = mMdatOffset;
lseek64(mFd, mMdatOffset, SEEK_SET);
if (mUse32BitOffset) {
@@ -689,7 +703,7 @@
lseek64(mFd, mOffset, SEEK_SET);
const off64_t moovOffset = mOffset;
- mWriteMoovBoxToMemory = true;
+ mWriteMoovBoxToMemory = mStreamableFile;
mMoovBoxBuffer = (uint8_t *) malloc(mEstimatedMoovBoxSize);
mMoovBoxBufferOffset = 0;
CHECK(mMoovBoxBuffer != NULL);
@@ -1062,6 +1076,10 @@
nTotalBytesEstimate += (*it)->getEstimatedTrackSizeBytes();
}
+ if (!mStreamableFile) {
+ // Add 1024 bytes as error tolerance
+ return nTotalBytesEstimate + 1024 >= mMaxFileSizeLimitBytes;
+ }
// Be conservative in the estimate: do not exceed 95% of
// the target file limit. For small target file size limit, though,
// this will not help.
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index f033080..f714cab 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -273,7 +273,7 @@
};
#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
-audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices)
+audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, audio_devices_t devices)
{
// if module is 0, the request comes from an old policy manager and we should load
// well known modules
@@ -299,7 +299,7 @@
return NULL;
}
-status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
+void AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -322,11 +322,10 @@
result.append(buffer);
}
write(fd, result.string(), result.size());
- return NO_ERROR;
}
-status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
+void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -339,10 +338,9 @@
(uint32_t)(mStandbyTimeInNsecs / 1000000));
result.append(buffer);
write(fd, result.string(), result.size());
- return NO_ERROR;
}
-status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
+void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -353,7 +351,6 @@
IPCThreadState::self()->getCallingUid());
result.append(buffer);
write(fd, result.string(), result.size());
- return NO_ERROR;
}
static bool tryLock(Mutex& mutex)
@@ -873,8 +870,7 @@
sp<RecordThread> thread = mRecordThreads.valueAt(i);
RecordThread::RecordTrack *track = thread->track();
if (track != NULL) {
- audio_devices_t device = (audio_devices_t)(
- thread->device() & AUDIO_DEVICE_IN_ALL);
+ audio_devices_t device = thread->device() & AUDIO_DEVICE_IN_ALL;
bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
thread->setEffectSuspended(FX_IID_AEC,
suspend,
@@ -1116,7 +1112,7 @@
// ----------------------------------------------------------------------------
AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
- uint32_t device, type_t type)
+ audio_devices_t device, type_t type)
: Thread(false),
mType(type),
mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0),
@@ -1124,7 +1120,7 @@
mChannelCount(0),
mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID),
mParamStatus(NO_ERROR),
- mStandby(false), mDevice((audio_devices_t) device), mId(id),
+ mStandby(false), mDevice(device), mId(id),
mDeathRecipient(new PMDeathRecipient(this))
{
}
@@ -1217,7 +1213,7 @@
mLock.unlock();
}
-status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
+void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -1274,10 +1270,9 @@
if (locked) {
mLock.unlock();
}
- return NO_ERROR;
}
-status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
+void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -1292,7 +1287,6 @@
chain->dump(fd, args);
}
}
- return NO_ERROR;
}
void AudioFlinger::ThreadBase::acquireWakeLock()
@@ -1500,7 +1494,7 @@
AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
audio_io_handle_t id,
- uint32_t device,
+ audio_devices_t device,
type_t type)
: ThreadBase(audioFlinger, id, device, type),
mMixBuffer(NULL), mSuspended(0), mBytesWritten(0),
@@ -1540,15 +1534,14 @@
delete [] mMixBuffer;
}
-status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
+void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
{
dumpInternals(fd, args);
dumpTracks(fd, args);
dumpEffectChains(fd, args);
- return NO_ERROR;
}
-status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
+void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -1596,11 +1589,9 @@
FastTrackUnderruns underruns = getFastTrackUnderruns(0);
fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n",
underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty);
-
- return NO_ERROR;
}
-status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
+void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -1624,8 +1615,6 @@
fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask);
dumpBase(fd, args);
-
- return NO_ERROR;
}
// Thread virtuals
@@ -2181,7 +2170,7 @@
// ----------------------------------------------------------------------------
AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
- audio_io_handle_t id, uint32_t device, type_t type)
+ audio_io_handle_t id, audio_devices_t device, type_t type)
: PlaybackThread(audioFlinger, output, id, device, type),
// mAudioMixer below
// mFastMixer below
@@ -2190,7 +2179,7 @@
// mPipeSink below
// mNormalSink below
{
- ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type);
+ ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type);
ALOGV("mSampleRate=%d, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, "
"mFrameCount=%d, mNormalFrameCount=%d",
mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,
@@ -3420,14 +3409,14 @@
#ifdef ADD_BATTERY_DATA
// when changing the audio output device, call addBatteryData to notify
// the change
- if ((int)mDevice != value) {
+ if (mDevice != value) {
uint32_t params = 0;
// check whether speaker is on
if (value & AUDIO_DEVICE_OUT_SPEAKER) {
params |= IMediaPlayerService::kBatteryDataSpeakerOn;
}
- int deviceWithoutSpeaker
+ audio_devices_t deviceWithoutSpeaker
= AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
// check if any other device (except speaker) is on
if (value & deviceWithoutSpeaker ) {
@@ -3442,7 +3431,7 @@
// forward device change to effects that have requested to be
// aware of attached audio device.
- mDevice = (audio_devices_t) value;
+ mDevice = value;
for (size_t i = 0; i < mEffectChains.size(); i++) {
mEffectChains[i]->setDevice_l(mDevice);
}
@@ -3499,7 +3488,7 @@
return reconfig;
}
-status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
+void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -3584,8 +3573,6 @@
AudioWatchdogDump wdCopy = mAudioWatchdogDump;
wdCopy.dump(fd);
}
-
- return NO_ERROR;
}
uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const
@@ -3611,7 +3598,7 @@
// ----------------------------------------------------------------------------
AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
- AudioStreamOut* output, audio_io_handle_t id, uint32_t device)
+ AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device)
: PlaybackThread(audioFlinger, output, id, device, DIRECT)
// mLeftVolFloat, mRightVolFloat
{
@@ -5900,7 +5887,7 @@
uint32_t sampleRate,
audio_channel_mask_t channelMask,
audio_io_handle_t id,
- uint32_t device) :
+ audio_devices_t device) :
ThreadBase(audioFlinger, id, device, RECORD),
mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),
// mRsmpInIndex and mInputBytes set by readInputParameters()
@@ -6179,8 +6166,8 @@
mTrack = track.get();
// disable AEC and NS if the device is a BT SCO headset supporting those pre processings
- bool suspend = audio_is_bluetooth_sco_device(
- (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff();
+ bool suspend = audio_is_bluetooth_sco_device(mDevice & AUDIO_DEVICE_IN_ALL) &&
+ mAudioFlinger->btNrecIsOff();
setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
}
@@ -6341,7 +6328,7 @@
return NAME_NOT_FOUND;
}
-status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
+void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -6375,8 +6362,6 @@
dumpBase(fd, args);
dumpEffectChains(fd, args);
-
- return NO_ERROR;
}
// AudioBufferProvider interface
@@ -6469,12 +6454,12 @@
// store input device and output device but do not forward output device to audio HAL.
// Note that status is ignored by the caller for output device
// (see AudioFlinger::setParameters()
- uint32_t /*audio_devices_t*/ newDevice = mDevice;
+ audio_devices_t newDevice = mDevice;
if (value & AUDIO_DEVICE_OUT_ALL) {
- newDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL);
+ newDevice &= ~(value & AUDIO_DEVICE_OUT_ALL);
status = BAD_VALUE;
} else {
- newDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL);
+ newDevice &= ~(value & AUDIO_DEVICE_IN_ALL);
// disable AEC and NS if the device is a BT SCO headset supporting those pre processings
if (mTrack != NULL) {
bool suspend = audio_is_bluetooth_sco_device(
@@ -6484,7 +6469,7 @@
}
}
newDevice |= value;
- mDevice = (audio_devices_t) newDevice; // since mDevice is read by other threads, only write to it once
+ mDevice = newDevice; // since mDevice is read by other threads, only write to it once
}
if (status == NO_ERROR) {
status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
@@ -6725,7 +6710,7 @@
ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
module,
- (pDevices != NULL) ? (int)*pDevices : 0,
+ (pDevices != NULL) ? *pDevices : 0,
config.sample_rate,
config.format,
config.channel_mask,
@@ -6985,7 +6970,7 @@
// Start record thread
// RecorThread require both input and output device indication to forward to audio
// pre processing modules
- uint32_t device = (*pDevices) | primaryOutputDevice_l();
+ audio_devices_t device = (*pDevices) | primaryOutputDevice_l();
thread = new RecordThread(this,
input,
reqSamplingRate,
@@ -7195,7 +7180,7 @@
return NULL;
}
-uint32_t AudioFlinger::primaryOutputDevice_l() const
+audio_devices_t AudioFlinger::primaryOutputDevice_l() const
{
PlaybackThread *thread = primaryPlaybackThread_l();
@@ -8534,14 +8519,14 @@
return status;
}
-status_t AudioFlinger::EffectModule::setDevice(uint32_t device)
+status_t AudioFlinger::EffectModule::setDevice(audio_devices_t device)
{
Mutex::Autolock _l(mLock);
status_t status = NO_ERROR;
if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
// audio pre processing modules on RecordThread can receive both output and
// input device indication in the same call
- uint32_t dev = device & AUDIO_DEVICE_OUT_ALL;
+ audio_devices_t dev = device & AUDIO_DEVICE_OUT_ALL;
if (dev) {
status_t cmdStatus;
uint32_t size = sizeof(status_t);
@@ -8626,7 +8611,7 @@
return enabled;
}
-status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
+void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -8706,8 +8691,6 @@
if (locked) {
mLock.unlock();
}
-
- return NO_ERROR;
}
// ----------------------------------------------------------------------------
@@ -9258,7 +9241,7 @@
}
// setDevice_l() must be called with PlaybackThread::mLock held
-void AudioFlinger::EffectChain::setDevice_l(uint32_t device)
+void AudioFlinger::EffectChain::setDevice_l(audio_devices_t device)
{
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
@@ -9333,7 +9316,7 @@
return hasControl;
}
-status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
+void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -9367,8 +9350,6 @@
if (locked) {
mLock.unlock();
}
-
- return NO_ERROR;
}
// must be called with ThreadBase::mLock held
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index fa1ad93..c9a3c3f 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -270,17 +270,17 @@
// RefBase
virtual void onFirstRef();
- audio_hw_device_t* findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices);
+ audio_hw_device_t* findSuitableHwDev_l(audio_module_handle_t module, audio_devices_t devices);
void purgeStaleEffects_l();
// standby delay for MIXER and DUPLICATING playback threads is read from property
// ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs
static nsecs_t mStandbyTimeInNsecs;
- // Internal dump utilites.
- status_t dumpPermissionDenial(int fd, const Vector<String16>& args);
- status_t dumpClients(int fd, const Vector<String16>& args);
- status_t dumpInternals(int fd, const Vector<String16>& args);
+ // Internal dump utilities.
+ void dumpPermissionDenial(int fd, const Vector<String16>& args);
+ void dumpClients(int fd, const Vector<String16>& args);
+ void dumpInternals(int fd, const Vector<String16>& args);
// --- Client ---
class Client : public RefBase {
@@ -352,11 +352,11 @@
RECORD // Thread class is RecordThread
};
- ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, uint32_t device, type_t type);
+ ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, audio_devices_t device, type_t type);
virtual ~ThreadBase();
- status_t dumpBase(int fd, const Vector<String16>& args);
- status_t dumpEffectChains(int fd, const Vector<String16>& args);
+ void dumpBase(int fd, const Vector<String16>& args);
+ void dumpEffectChains(int fd, const Vector<String16>& args);
void clearPowerManager();
@@ -963,10 +963,10 @@
}; // end of OutputTrack
PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
- audio_io_handle_t id, uint32_t device, type_t type);
+ audio_io_handle_t id, audio_devices_t device, type_t type);
virtual ~PlaybackThread();
- status_t dump(int fd, const Vector<String16>& args);
+ void dump(int fd, const Vector<String16>& args);
// Thread virtuals
virtual status_t readyToRun();
@@ -1115,8 +1115,8 @@
void readOutputParameters();
- virtual status_t dumpInternals(int fd, const Vector<String16>& args);
- status_t dumpTracks(int fd, const Vector<String16>& args);
+ virtual void dumpInternals(int fd, const Vector<String16>& args);
+ void dumpTracks(int fd, const Vector<String16>& args);
SortedVector< sp<Track> > mTracks;
// mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread
@@ -1185,14 +1185,14 @@
MixerThread (const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
audio_io_handle_t id,
- uint32_t device,
+ audio_devices_t device,
type_t type = MIXER);
virtual ~MixerThread();
// Thread virtuals
virtual bool checkForNewParameters_l();
- virtual status_t dumpInternals(int fd, const Vector<String16>& args);
+ virtual void dumpInternals(int fd, const Vector<String16>& args);
protected:
virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
@@ -1240,7 +1240,7 @@
public:
DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
- audio_io_handle_t id, uint32_t device);
+ audio_io_handle_t id, audio_devices_t device);
virtual ~DirectOutputThread();
// Thread virtuals
@@ -1329,7 +1329,7 @@
bool reRegister);
// return thread associated with primary hardware device, or NULL
PlaybackThread *primaryPlaybackThread_l() const;
- uint32_t primaryOutputDevice_l() const;
+ audio_devices_t primaryOutputDevice_l() const;
sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId);
@@ -1407,7 +1407,7 @@
uint32_t sampleRate,
audio_channel_mask_t channelMask,
audio_io_handle_t id,
- uint32_t device);
+ audio_devices_t device);
virtual ~RecordThread();
// Thread
@@ -1433,7 +1433,7 @@
AudioSystem::sync_event_t event,
int triggerSession);
void stop(RecordTrack* recordTrack);
- status_t dump(int fd, const Vector<String16>& args);
+ void dump(int fd, const Vector<String16>& args);
AudioStreamIn* clearInput();
virtual audio_stream_t* stream() const;
@@ -1575,7 +1575,7 @@
effect_descriptor_t& desc() { return mDescriptor; }
wp<EffectChain>& chain() { return mChain; }
- status_t setDevice(uint32_t device);
+ status_t setDevice(audio_devices_t device);
status_t setVolume(uint32_t *left, uint32_t *right, bool controller);
status_t setMode(audio_mode_t mode);
status_t start();
@@ -1591,7 +1591,7 @@
void lock() { mLock.lock(); }
void unlock() { mLock.unlock(); }
- status_t dump(int fd, const Vector<String16>& args);
+ void dump(int fd, const Vector<String16>& args);
protected:
friend class AudioFlinger; // for mHandles
@@ -1738,7 +1738,7 @@
sp<EffectModule> getEffectFromId_l(int id);
sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type);
bool setVolume_l(uint32_t *left, uint32_t *right);
- void setDevice_l(uint32_t device);
+ void setDevice_l(audio_devices_t device);
void setMode_l(audio_mode_t mode);
void setInBuffer(int16_t *buffer, bool ownsBuffer = false) {
@@ -1779,7 +1779,7 @@
void clearInputBuffer();
- status_t dump(int fd, const Vector<String16>& args);
+ void dump(int fd, const Vector<String16>& args);
protected:
friend class AudioFlinger; // for mThread, mEffects