Merge "Don't rely on the system locale for converting to/from bytes."
diff --git a/common/java/com/android/common/speech/LoggingEvents.java b/common/java/com/android/common/speech/LoggingEvents.java
index 3b3ecb8..1f3c6ef 100644
--- a/common/java/com/android/common/speech/LoggingEvents.java
+++ b/common/java/com/android/common/speech/LoggingEvents.java
@@ -117,6 +117,12 @@
public static final String EXTRA_N_BEST_CHOOSE_INDEX = "index"; // value should be int
public static final int TEXT_MODIFIED = 17;
+ public static final String EXTRA_TEXT_MODIFIED_LENGTH = "length"; // value should be int
+ public static final String EXTRA_TEXT_MODIFIED_TYPE = "type"; // value should be int below
+ public static final int TEXT_MODIFIED_TYPE_CHOOSE_SUGGESTION = 1;
+ public static final int TEXT_MODIFIED_TYPE_TYPING_DELETION = 2;
+ public static final int TEXT_MODIFIED_TYPE_TYPING_INSERTION = 3;
+ public static final int TEXT_MODIFIED_TYPE_TYPING_INSERTION_PUNCTUATION = 4;
public static final int INPUT_ENDED = 18;
diff --git a/include/utils/ResourceTypes.h b/include/utils/ResourceTypes.h
index 0e796dc..b701ce7 100644
--- a/include/utils/ResourceTypes.h
+++ b/include/utils/ResourceTypes.h
@@ -631,6 +631,8 @@
void restart();
+ const ResStringPool& getStrings() const;
+
event_code_t getEventType() const;
// Note, unlike XmlPullParser, the first call to next() will return
// START_TAG of the first element.
@@ -716,8 +718,6 @@
void uninit();
- const ResStringPool& getStrings() const;
-
private:
friend class ResXMLParser;
diff --git a/libs/audioflinger/Android.mk b/libs/audioflinger/Android.mk
index b68bfc1..870c0b8 100644
--- a/libs/audioflinger/Android.mk
+++ b/libs/audioflinger/Android.mk
@@ -24,7 +24,7 @@
LOCAL_SHARED_LIBRARIES := \
libcutils \
libutils \
- libbinder \
+ libbinder \
libmedia \
libhardware_legacy
@@ -85,7 +85,7 @@
LOCAL_SHARED_LIBRARIES := \
libcutils \
libutils \
- libbinder \
+ libbinder \
libmedia \
libhardware_legacy
@@ -114,9 +114,17 @@
endif
ifeq ($(TARGET_SIMULATOR),true)
- ifeq ($(HOST_OS),linux)
- LOCAL_LDLIBS += -lrt -lpthread
- endif
+ ifeq ($(HOST_OS),linux)
+ LOCAL_LDLIBS += -lrt -lpthread
+ endif
+endif
+
+ifeq ($(BOARD_USE_LVMX),true)
+ LOCAL_CFLAGS += -DLVMX
+ LOCAL_C_INCLUDES += vendor/nxp
+ LOCAL_STATIC_LIBRARIES += liblifevibes
+ LOCAL_SHARED_LIBRARIES += liblvmxservice
+# LOCAL_SHARED_LIBRARIES += liblvmxipc
endif
include $(BUILD_SHARED_LIBRARY)
diff --git a/libs/audioflinger/AudioFlinger.cpp b/libs/audioflinger/AudioFlinger.cpp
index 8089389..7902212 100644
--- a/libs/audioflinger/AudioFlinger.cpp
+++ b/libs/audioflinger/AudioFlinger.cpp
@@ -47,6 +47,10 @@
#include "A2dpAudioInterface.h"
#endif
+#ifdef LVMX
+#include "lifevibes.h"
+#endif
+
// ----------------------------------------------------------------------------
// the sim build doesn't have gettid
@@ -132,6 +136,9 @@
} else {
LOGE("Couldn't even initialize the stubbed audio hardware!");
}
+#ifdef LVMX
+ LifeVibes::init();
+#endif
}
AudioFlinger::~AudioFlinger()
@@ -411,6 +418,11 @@
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MODE;
status_t ret = mAudioHardware->setMode(mode);
+#ifdef LVMX
+ if (NO_ERROR == ret) {
+ LifeVibes::setMode(mode);
+ }
+#endif
mHardwareStatus = AUDIO_HW_IDLE;
return ret;
}
@@ -566,11 +578,37 @@
return PERMISSION_DENIED;
}
+#ifdef LVMX
+ AudioParameter param = AudioParameter(keyValuePairs);
+ LifeVibes::setParameters(ioHandle,keyValuePairs);
+ String8 key = String8(AudioParameter::keyRouting);
+ int device;
+ if (NO_ERROR != param.getInt(key, device)) {
+ device = -1;
+ }
+
+ key = String8(LifevibesTag);
+ String8 value;
+ int musicEnabled = -1;
+ if (NO_ERROR == param.get(key, value)) {
+ if (value == LifevibesEnable) {
+ musicEnabled = 1;
+ } else if (value == LifevibesDisable) {
+ musicEnabled = 0;
+ }
+ }
+#endif
+
// ioHandle == 0 means the parameters are global to the audio hardware interface
if (ioHandle == 0) {
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_SET_PARAMETER;
result = mAudioHardware->setParameters(keyValuePairs);
+#ifdef LVMX
+ if ((NO_ERROR == result) && (musicEnabled != -1)) {
+ LifeVibes::enableMusic((bool) musicEnabled);
+ }
+#endif
mHardwareStatus = AUDIO_HW_IDLE;
return result;
}
@@ -586,7 +624,13 @@
}
}
if (thread != NULL) {
- return thread->setParameters(keyValuePairs);
+ result = thread->setParameters(keyValuePairs);
+#ifdef LVMX
+ if ((NO_ERROR == result) && (device != -1)) {
+ LifeVibes::setDevice(LifeVibes::threadIdToAudioOutputType(thread->id()), device);
+ }
+#endif
+ return result;
}
return BAD_VALUE;
}
@@ -1058,12 +1102,24 @@
status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
{
+#ifdef LVMX
+ int audioOutputType = LifeVibes::getMixerType(mId, mType);
+ if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
+ LifeVibes::setMasterVolume(audioOutputType, value);
+ }
+#endif
mMasterVolume = value;
return NO_ERROR;
}
status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
{
+#ifdef LVMX
+ int audioOutputType = LifeVibes::getMixerType(mId, mType);
+ if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
+ LifeVibes::setMasterMute(audioOutputType, muted);
+ }
+#endif
mMasterMute = muted;
return NO_ERROR;
}
@@ -1080,12 +1136,24 @@
status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value)
{
+#ifdef LVMX
+ int audioOutputType = LifeVibes::getMixerType(mId, mType);
+ if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
+ LifeVibes::setStreamVolume(audioOutputType, stream, value);
+ }
+#endif
mStreamTypes[stream].volume = value;
return NO_ERROR;
}
status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted)
{
+#ifdef LVMX
+ int audioOutputType = LifeVibes::getMixerType(mId, mType);
+ if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
+ LifeVibes::setStreamMute(audioOutputType, stream, muted);
+ }
+#endif
mStreamTypes[stream].mute = muted;
return NO_ERROR;
}
@@ -1333,6 +1401,12 @@
mLastWriteTime = systemTime();
mInWrite = true;
mBytesWritten += mixBufferSize;
+#ifdef LVMX
+ int audioOutputType = LifeVibes::getMixerType(mId, mType);
+ if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
+ LifeVibes::process(audioOutputType, curBuf, mixBufferSize);
+ }
+#endif
int bytesWritten = (int)mOutput->write(curBuf, mixBufferSize);
if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
mNumWrites++;
@@ -1376,6 +1450,29 @@
uint32_t mixerStatus = MIXER_IDLE;
// find out which tracks need to be processed
size_t count = activeTracks.size();
+
+ float masterVolume = mMasterVolume;
+ bool masterMute = mMasterMute;
+
+#ifdef LVMX
+ bool tracksConnectedChanged = false;
+ bool stateChanged = false;
+
+ int audioOutputType = LifeVibes::getMixerType(mId, mType);
+ if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType))
+ {
+ int activeTypes = 0;
+ for (size_t i=0 ; i<count ; i++) {
+ sp<Track> t = activeTracks[i].promote();
+ if (t == 0) continue;
+ Track* const track = t.get();
+ int iTracktype=track->type();
+ activeTypes |= 1<<track->type();
+ }
+ LifeVibes::computeVolumes(audioOutputType, activeTypes, tracksConnectedChanged, stateChanged, masterVolume, masterMute);
+ }
+#endif
+
for (size_t i=0 ; i<count ; i++) {
sp<Track> t = activeTracks[i].promote();
if (t == 0) continue;
@@ -1393,15 +1490,27 @@
// compute volume for this track
int16_t left, right;
- if (track->isMuted() || mMasterMute || track->isPausing() ||
+ if (track->isMuted() || masterMute || track->isPausing() ||
mStreamTypes[track->type()].mute) {
left = right = 0;
if (track->isPausing()) {
track->setPaused();
}
} else {
+ // read original volumes with volume control
float typeVolume = mStreamTypes[track->type()].volume;
- float v = mMasterVolume * typeVolume;
+#ifdef LVMX
+ bool streamMute=false;
+ // read the volume from the LivesVibes audio engine.
+ if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType))
+ {
+ LifeVibes::getStreamVolumes(audioOutputType, track->type(), &typeVolume, &streamMute);
+ if (streamMute) {
+ typeVolume = 0;
+ }
+ }
+#endif
+ float v = masterVolume * typeVolume;
float v_clamped = v * cblk->volume[0];
if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
left = int16_t(v_clamped);
@@ -1427,7 +1536,13 @@
// do not apply ramp
param = AudioMixer::RAMP_VOLUME;
}
-
+#ifdef LVMX
+ if ( tracksConnectedChanged || stateChanged )
+ {
+ // only do the ramp when the volume is changed by the user / application
+ param = AudioMixer::VOLUME;
+ }
+#endif
mAudioMixer->setParameter(param, AudioMixer::VOLUME0, left);
mAudioMixer->setParameter(param, AudioMixer::VOLUME1, right);
mAudioMixer->setParameter(
@@ -3652,6 +3767,18 @@
} else {
thread = new MixerThread(this, output, ++mNextThreadId);
LOGV("openOutput() created mixer output: ID %d thread %p", mNextThreadId, thread);
+
+#ifdef LVMX
+ unsigned bitsPerSample =
+ (format == AudioSystem::PCM_16_BIT) ? 16 :
+ ((format == AudioSystem::PCM_8_BIT) ? 8 : 0);
+ unsigned channelCount = (channels == AudioSystem::CHANNEL_OUT_STEREO) ? 2 : 1;
+ int audioOutputType = LifeVibes::threadIdToAudioOutputType(thread->id());
+
+ LifeVibes::init_aot(audioOutputType, samplingRate, bitsPerSample, channelCount);
+ LifeVibes::setDevice(audioOutputType, *pDevices);
+#endif
+
}
mPlaybackThreads.add(mNextThreadId, thread);
diff --git a/libs/audioflinger/AudioHardwareStub.cpp b/libs/audioflinger/AudioHardwareStub.cpp
index ae215d1..d481150 100644
--- a/libs/audioflinger/AudioHardwareStub.cpp
+++ b/libs/audioflinger/AudioHardwareStub.cpp
@@ -166,7 +166,7 @@
// ----------------------------------------------------------------------------
status_t AudioStreamInStub::set(int *pFormat, uint32_t *pChannels, uint32_t *pRate,
- AudioSystem::audio_in_acoustics acoustics)
+ AudioSystem::audio_in_acoustics acoustics)
{
return NO_ERROR;
}
diff --git a/libs/audioflinger/AudioPolicyManagerBase.cpp b/libs/audioflinger/AudioPolicyManagerBase.cpp
index cfcc3ea..2b0a6c8 100644
--- a/libs/audioflinger/AudioPolicyManagerBase.cpp
+++ b/libs/audioflinger/AudioPolicyManagerBase.cpp
@@ -344,6 +344,7 @@
{
LOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState);
+ bool forceVolumeReeval = false;
switch(usage) {
case AudioSystem::FOR_COMMUNICATION:
if (config != AudioSystem::FORCE_SPEAKER && config != AudioSystem::FORCE_BT_SCO &&
@@ -374,6 +375,7 @@
config != AudioSystem::FORCE_BT_DESK_DOCK && config != AudioSystem::FORCE_WIRED_ACCESSORY) {
LOGW("setForceUse() invalid config %d for FOR_DOCK", config);
}
+ forceVolumeReeval = true;
mForceUse[usage] = config;
break;
default:
@@ -388,6 +390,9 @@
#endif
updateDeviceForStrategy();
setOutputDevice(mHardwareOutput, newDevice);
+ if (forceVolumeReeval) {
+ applyStreamVolumes(mHardwareOutput, newDevice);
+ }
}
AudioSystem::forced_config AudioPolicyManagerBase::getForceUse(AudioSystem::force_use usage)
diff --git a/libs/audioflinger/AudioResampler.h b/libs/audioflinger/AudioResampler.h
index 39656c0..2dfac76 100644
--- a/libs/audioflinger/AudioResampler.h
+++ b/libs/audioflinger/AudioResampler.h
@@ -76,8 +76,8 @@
int32_t mInSampleRate;
AudioBufferProvider::Buffer mBuffer;
union {
- int16_t mVolume[2];
- uint32_t mVolumeRL;
+ int16_t mVolume[2];
+ uint32_t mVolumeRL;
};
int16_t mTargetVolume[2];
format mFormat;
diff --git a/libs/utils/ResourceTypes.cpp b/libs/utils/ResourceTypes.cpp
index 38d8412..7e0f881 100644
--- a/libs/utils/ResourceTypes.cpp
+++ b/libs/utils/ResourceTypes.cpp
@@ -625,6 +625,10 @@
mCurNode = NULL;
mEventCode = mTree.mError == NO_ERROR ? START_DOCUMENT : BAD_DOCUMENT;
}
+const ResStringPool& ResXMLParser::getStrings() const
+{
+ return mTree.mStrings;
+}
ResXMLParser::event_code_t ResXMLParser::getEventType() const
{
@@ -1149,11 +1153,6 @@
restart();
}
-const ResStringPool& ResXMLTree::getStrings() const
-{
- return mStrings;
-}
-
status_t ResXMLTree::validateNode(const ResXMLTree_node* node) const
{
const uint16_t eventCode = dtohs(node->header.type);