Merge "stingray: update kernel / wifi prebuilt"
diff --git a/AndroidBoard.mk b/AndroidBoard.mk
index d16e172..9678291 100644
--- a/AndroidBoard.mk
+++ b/AndroidBoard.mk
@@ -49,6 +49,10 @@
 include $(COMMON_DIR)/TegraBoard.mk
 endif
 
-include $(LOCAL_PATH)/ril/Android.mk
+subdir_makefiles:= \
+	$(LOCAL_PATH)/ril/Android.mk \
+	$(LOCAL_PATH)/libaudio/Android.mk
+
+include $(subdir_makefiles)
 
 -include vendor/moto/stingray/AndroidBoardVendor.mk
diff --git a/BoardConfig.mk b/BoardConfig.mk
index 754d91a..25492c4 100755
--- a/BoardConfig.mk
+++ b/BoardConfig.mk
@@ -7,6 +7,9 @@
 ARCH_ARM_HAVE_TLS_REGISTER := true
 
 TARGET_USERIMAGES_USE_EXT4 := true
+BOARD_SYSTEMIMAGE_PARTITION_SIZE := 134217728
+BOARD_USERDATAIMAGE_PARTITION_SIZE := 16777216
+BOARD_FLASH_BLOCK_SIZE := 4096
 
 # Wifi related defines
 BOARD_WPA_SUPPLICANT_DRIVER := WEXT
diff --git a/libaudio/Android.mk b/libaudio/Android.mk
new file mode 100644
index 0000000..9dca5bb
--- /dev/null
+++ b/libaudio/Android.mk
@@ -0,0 +1,56 @@
+ifneq ($(BUILD_TINY_ANDROID),true)
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:=               \
+    AudioPolicyManager.cpp
+
+LOCAL_SHARED_LIBRARIES := \
+    libcutils \
+    libutils \
+    libmedia
+
+LOCAL_STATIC_LIBRARIES := libaudiopolicybase
+
+LOCAL_MODULE:= libaudiopolicy
+
+ifeq ($(BOARD_HAVE_BLUETOOTH),true)
+  LOCAL_CFLAGS += -DWITH_A2DP
+endif
+
+include $(BUILD_SHARED_LIBRARY)
+
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE := libaudio
+
+LOCAL_SHARED_LIBRARIES := \
+    libcutils \
+    libutils \
+    libmedia \
+    libhardware_legacy
+
+ifeq ($TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
+LOCAL_LDLIBS += -ldl
+endif
+
+ifneq ($(TARGET_SIMULATOR),true)
+LOCAL_SHARED_LIBRARIES += libdl
+endif
+
+LOCAL_SRC_FILES += AudioHardware.cpp
+
+LOCAL_CFLAGS += -fno-short-enums
+
+LOCAL_STATIC_LIBRARIES += libaudiointerface
+ifeq ($(BOARD_HAVE_BLUETOOTH),true)
+  LOCAL_SHARED_LIBRARIES += liba2dp
+endif
+
+include $(BUILD_SHARED_LIBRARY)
+
+endif # not BUILD_TINY_ANDROID
+
diff --git a/libaudio/AudioHardware.cpp b/libaudio/AudioHardware.cpp
new file mode 100644
index 0000000..16e455c
--- /dev/null
+++ b/libaudio/AudioHardware.cpp
@@ -0,0 +1,941 @@
+/*
+** Copyright 2008, 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.
+*/
+
+#include <math.h>
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "AudioHardwareTegra"
+#include <utils/Log.h>
+#include <utils/String8.h>
+
+#include <stdio.h>
+#include <unistd.h>
+#include <sys/ioctl.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <dlfcn.h>
+#include <fcntl.h>
+
+// hardware specific functions
+
+#include "AudioHardware.h"
+#include <media/AudioRecord.h>
+
+namespace android {
+const uint32_t AudioHardware::inputSamplingRates[] = {
+        8000, 16000, 44100
+};
+// ----------------------------------------------------------------------------
+
+AudioHardware::AudioHardware() :
+    mInit(false), mMicMute(true), mBluetoothNrec(true), mBluetoothId(0),
+    mOutput(0), mCurSndDevice(-1)
+{
+    int fd = ::open("/dev/audio_ctl", O_RDWR);
+    if (fd >= 0) {
+        if (::ioctl(fd, CPCAP_AUDIO_OUT_GET_OUTPUT, &mCurSndDevice))
+            LOGE("Could not open audio_ctl: %s", strerror(errno));
+        else
+            LOGI("current output device: %d", mCurSndDevice);
+    }
+    mInit = true;
+}
+
+AudioHardware::~AudioHardware()
+{
+    for (size_t index = 0; index < mInputs.size(); index++) {
+        closeInputStream((AudioStreamIn*)mInputs[index]);
+    }
+    mInputs.clear();
+    closeOutputStream((AudioStreamOut*)mOutput);
+    mInit = false;
+}
+
+status_t AudioHardware::initCheck()
+{
+    return mInit ? NO_ERROR : NO_INIT;
+}
+
+AudioStreamOut* AudioHardware::openOutputStream(
+        uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
+{
+    { // scope for the lock
+        Mutex::Autolock lock(mLock);
+
+        // only one output stream allowed
+        if (mOutput) {
+            if (status) {
+                *status = INVALID_OPERATION;
+            }
+            return 0;
+        }
+
+        // create new output stream
+        AudioStreamOutTegra* out = new AudioStreamOutTegra();
+        status_t lStatus = out->set(this, devices, format, channels, sampleRate);
+        if (status) {
+            *status = lStatus;
+        }
+        if (lStatus == NO_ERROR) {
+            mOutput = out;
+        } else {
+            delete out;
+        }
+    }
+    return mOutput;
+}
+
+void AudioHardware::closeOutputStream(AudioStreamOut* out) {
+    Mutex::Autolock lock(mLock);
+    if (mOutput == 0 || mOutput != out) {
+        LOGW("Attempt to close invalid output stream");
+    }
+    else {
+        delete mOutput;
+        mOutput = 0;
+    }
+}
+
+AudioStreamIn* AudioHardware::openInputStream(
+        uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status,
+        AudioSystem::audio_in_acoustics acoustic_flags)
+{
+    // check for valid input source
+    if (!AudioSystem::isInputDevice((AudioSystem::audio_devices)devices)) {
+        return 0;
+    }
+
+    mLock.lock();
+
+    AudioStreamInTegra* in = new AudioStreamInTegra();
+    status_t lStatus = in->set(this, devices, format, channels, sampleRate, acoustic_flags);
+    if (status) {
+        *status = lStatus;
+    }
+    if (lStatus != NO_ERROR) {
+        mLock.unlock();
+        delete in;
+        return 0;
+    }
+
+    mInputs.add(in);
+    mLock.unlock();
+
+    return in;
+}
+
+void AudioHardware::closeInputStream(AudioStreamIn* in) {
+    Mutex::Autolock lock(mLock);
+
+    ssize_t index = mInputs.indexOf((AudioStreamInTegra *)in);
+    if (index < 0) {
+        LOGW("Attempt to close invalid input stream");
+    } else {
+        mLock.unlock();
+        delete mInputs[index];
+        mLock.lock();
+        mInputs.removeAt(index);
+    }
+}
+
+status_t AudioHardware::setMode(int mode)
+{
+    status_t status = AudioHardwareBase::setMode(mode);
+    if (status == NO_ERROR) {
+        // make sure that doAudioRouteOrMute() is called by doRouting()
+        // even if the new device selected is the same as current one.
+        clearCurDevice();
+    }
+    return status;
+}
+
+bool AudioHardware::checkOutputStandby()
+{
+    if (mOutput)
+        if (!mOutput->checkStandby())
+            return false;
+
+    return true;
+}
+
+status_t AudioHardware::setMicMute(bool state)
+{
+    Mutex::Autolock lock(mLock);
+    return setMicMute_nosync(state);
+}
+
+// always call with mutex held
+status_t AudioHardware::setMicMute_nosync(bool state)
+{
+    if (mMicMute != state) {
+        mMicMute = state;
+        return NO_ERROR; //doAudioRouteOrMute(SND_DEVICE_CURRENT);
+    }
+    return NO_ERROR;
+}
+
+status_t AudioHardware::getMicMute(bool* state)
+{
+    *state = mMicMute;
+    return NO_ERROR;
+}
+
+status_t AudioHardware::setParameters(const String8& keyValuePairs)
+{
+    AudioParameter param = AudioParameter(keyValuePairs);
+    String8 value;
+    String8 key;
+    const char BT_NREC_KEY[] = "bt_headset_nrec";
+    const char BT_NAME_KEY[] = "bt_headset_name";
+    const char BT_NREC_VALUE_ON[] = "on";
+
+
+    LOGV("setParameters() %s", keyValuePairs.string());
+
+    if (keyValuePairs.length() == 0) return BAD_VALUE;
+
+    key = String8(BT_NREC_KEY);
+    if (param.get(key, value) == NO_ERROR) {
+        if (value == BT_NREC_VALUE_ON) {
+            mBluetoothNrec = true;
+        } else {
+            mBluetoothNrec = false;
+            LOGI("Turning noise reduction and echo cancellation off for BT "
+                 "headset");
+        }
+    }
+    key = String8(BT_NAME_KEY);
+    if (param.get(key, value) == NO_ERROR) {
+        mBluetoothId = 0;
+#if 0
+        for (int i = 0; i < mNumSndEndpoints; i++) {
+            if (!strcasecmp(value.string(), mSndEndpoints[i].name)) {
+                mBluetoothId = mSndEndpoints[i].id;
+                LOGI("Using custom acoustic parameters for %s", value.string());
+                break;
+            }
+        }
+#endif
+        if (mBluetoothId == 0) {
+            LOGI("Using default acoustic parameters "
+                 "(%s not in acoustic database)", value.string());
+            doRouting();
+        }
+    }
+    return NO_ERROR;
+}
+
+String8 AudioHardware::getParameters(const String8& keys)
+{
+    AudioParameter param = AudioParameter(keys);
+    return param.toString();
+}
+
+size_t AudioHardware::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
+{
+    if (format != AudioSystem::PCM_16_BIT) {
+        LOGW("getInputBufferSize bad format: %d", format);
+        return 0;
+    }
+    if (channelCount < 1 || channelCount > 2) {
+        LOGW("getInputBufferSize bad channel count: %d", channelCount);
+        return 0;
+    }
+
+    return 2048*channelCount;
+}
+
+status_t AudioHardware::setVoiceVolume(float v)
+{
+    if (v < 0.0) {
+        LOGW("setVoiceVolume(%f) under 0.0, assuming 0.0\n", v);
+        v = 0.0;
+    } else if (v > 1.0) {
+        LOGW("setVoiceVolume(%f) over 1.0, assuming 1.0\n", v);
+        v = 1.0;
+    }
+
+    int vol = lrint(v * 15.0);
+    LOGD("setVoiceVolume(%f)\n", v);
+    LOGI("Setting in-call volume to %d (available range is 0 to 15)\n", vol);
+
+    Mutex::Autolock lock(mLock);
+    {
+        int fd = open("/dev/audio_ctl", O_RDWR);
+        if (fd < 0) {
+            LOGE("could not open audio_ctl: %s\n", strerror(errno));
+            return NO_ERROR;
+        }
+        if (ioctl(fd, CPCAP_AUDIO_OUT_SET_VOLUME, vol) < 0)
+            LOGE("could not set volume: %s\n", strerror(errno));
+        close(fd);
+    }
+    return NO_ERROR;
+}
+
+status_t AudioHardware::setMasterVolume(float v)
+{
+    Mutex::Autolock lock(mLock);
+    int vol = ceil(v * 5.0);
+    LOGI("Set master volume to %d.\n", vol);
+    /*
+    set_volume_rpc(SND_DEVICE_HANDSET, SND_METHOD_VOICE, vol);
+    set_volume_rpc(SND_DEVICE_SPEAKER, SND_METHOD_VOICE, vol);
+    set_volume_rpc(SND_DEVICE_BT,      SND_METHOD_VOICE, vol);
+    set_volume_rpc(SND_DEVICE_HEADSET, SND_METHOD_VOICE, vol);
+    */
+    // We return an error code here to let the audioflinger do in-software
+    // volume on top of the maximum volume that we set through the SND API.
+    // return error - software mixer will handle it
+    return -1;
+}
+
+// always call with mutex held
+status_t AudioHardware::doAudioRouteOrMute(uint32_t device)
+{
+#if 0
+    if (device == (uint32_t)SND_DEVICE_BT || device == (uint32_t)SND_DEVICE_CARKIT) {
+        if (mBluetoothId) {
+            device = mBluetoothId;
+        } else if (!mBluetoothNrec) {
+            device = CPCAP_AUDIO_OUT_SPEAKER;
+        }
+    }
+#endif
+
+    {
+        int fd = open("/dev/audio_ctl", O_RDWR);
+        if (fd < 0) {
+            LOGE("could not open audio_ctl: %s\n", strerror(errno));
+            return NO_ERROR;
+        }
+        if (ioctl(fd, CPCAP_AUDIO_OUT_SET_OUTPUT, device) < 0)
+            LOGE("could not set output: %s\n", strerror(errno));
+        close(fd);
+    }
+    return NO_ERROR;
+}
+
+status_t AudioHardware::doRouting()
+{
+    Mutex::Autolock lock(mLock);
+    uint32_t outputDevices = mOutput->devices();
+    status_t ret = NO_ERROR;
+    AudioStreamInTegra *input = getActiveInput_l();
+    uint32_t inputDevice = (input == NULL) ? 0 : input->devices();
+
+    int sndDevice = -1;
+
+    if (inputDevice != 0) {
+        LOGI("do input routing device %x\n", inputDevice);
+        if (inputDevice & AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
+            LOGI("Routing audio to Bluetooth PCM\n");
+            sndDevice = CPCAP_AUDIO_OUT_SPEAKER;
+        } else if (inputDevice & AudioSystem::DEVICE_IN_WIRED_HEADSET) {
+            if ((outputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET) &&
+                (outputDevices & AudioSystem::DEVICE_OUT_SPEAKER)) {
+                LOGI("Routing audio to Wired Headset and Speaker\n");
+                sndDevice = CPCAP_AUDIO_OUT_SPEAKER;
+            } else {
+                LOGI("Routing audio to Wired Headset\n");
+                sndDevice = CPCAP_AUDIO_OUT_HEADSET;
+            }
+        } else {
+            if (outputDevices & AudioSystem::DEVICE_OUT_SPEAKER) {
+                LOGI("Routing audio to Speakerphone\n");
+                sndDevice = CPCAP_AUDIO_OUT_SPEAKER;
+            } else {
+                LOGI("Routing audio to Handset\n");
+                sndDevice = CPCAP_AUDIO_OUT_HEADSET;
+            }
+        }
+    }
+    // if inputDevice == 0, restore output routing
+
+    if (sndDevice == -1) {
+        if (outputDevices & (outputDevices - 1)) {
+            if ((outputDevices & AudioSystem::DEVICE_OUT_SPEAKER) == 0) {
+                LOGW("Hardware does not support requested route combination (%#X),"
+                     " picking closest possible route...", outputDevices);
+            }
+        }
+
+        if (outputDevices &
+            (AudioSystem::DEVICE_OUT_BLUETOOTH_SCO | AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET)) {
+            LOGI("Routing audio to Bluetooth PCM\n");
+            sndDevice = CPCAP_AUDIO_OUT_SPEAKER;
+        } else if (outputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT) {
+            LOGI("Routing audio to Bluetooth PCM\n");
+            sndDevice = CPCAP_AUDIO_OUT_SPEAKER;
+        } else if ((outputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET) &&
+                   (outputDevices & AudioSystem::DEVICE_OUT_SPEAKER)) {
+            LOGI("Routing audio to Wired Headset and Speaker\n");
+            sndDevice = CPCAP_AUDIO_OUT_SPEAKER;
+        } else if (outputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADPHONE) {
+            if (outputDevices & AudioSystem::DEVICE_OUT_SPEAKER) {
+                LOGI("Routing audio to No microphone Wired Headset and Speaker (%d,%x)\n", mMode, outputDevices);
+                sndDevice = CPCAP_AUDIO_OUT_SPEAKER;
+            } else {
+                LOGI("Routing audio to No microphone Wired Headset (%d,%x)\n", mMode, outputDevices);
+                sndDevice = CPCAP_AUDIO_OUT_HEADSET;
+            }
+        } else if (outputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET) {
+            LOGI("Routing audio to Wired Headset\n");
+            sndDevice = CPCAP_AUDIO_OUT_HEADSET;
+        } else if (outputDevices & AudioSystem::DEVICE_OUT_SPEAKER) {
+            LOGI("Routing audio to Speakerphone\n");
+            sndDevice = CPCAP_AUDIO_OUT_SPEAKER;
+        } else {
+            LOGI("Routing audio to Handset\n");
+            sndDevice = CPCAP_AUDIO_OUT_HEADSET;
+        }
+    }
+
+    if (sndDevice != -1 && sndDevice != mCurSndDevice) {
+        ret = doAudioRouteOrMute(sndDevice);
+        mCurSndDevice = sndDevice;
+    }
+
+    return ret;
+}
+
+status_t AudioHardware::checkMicMute()
+{
+    Mutex::Autolock lock(mLock);
+    if (mMode != AudioSystem::MODE_IN_CALL) {
+        setMicMute_nosync(true);
+    }
+
+    return NO_ERROR;
+}
+
+status_t AudioHardware::dumpInternals(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+    result.append("AudioHardware::dumpInternals\n");
+    snprintf(buffer, SIZE, "\tmInit: %s\n", mInit? "true": "false");
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmMicMute: %s\n", mMicMute? "true": "false");
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmBluetoothNrec: %s\n", mBluetoothNrec? "true": "false");
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmBluetoothId: %d\n", mBluetoothId);
+    result.append(buffer);
+    ::write(fd, result.string(), result.size());
+    return NO_ERROR;
+}
+
+status_t AudioHardware::dump(int fd, const Vector<String16>& args)
+{
+    dumpInternals(fd, args);
+    for (size_t index = 0; index < mInputs.size(); index++) {
+        mInputs[index]->dump(fd, args);
+    }
+
+    if (mOutput) {
+        mOutput->dump(fd, args);
+    }
+    return NO_ERROR;
+}
+
+uint32_t AudioHardware::getInputSampleRate(uint32_t sampleRate)
+{
+    uint32_t i;
+    uint32_t prevDelta;
+    uint32_t delta;
+
+    for (i = 0, prevDelta = 0xFFFFFFFF; i < sizeof(inputSamplingRates)/sizeof(uint32_t); i++, prevDelta = delta) {
+        delta = abs(sampleRate - inputSamplingRates[i]);
+        if (delta > prevDelta) break;
+    }
+    // i is always > 0 here
+    return inputSamplingRates[i-1];
+}
+
+// getActiveInput_l() must be called with mLock held
+AudioHardware::AudioStreamInTegra *AudioHardware::getActiveInput_l()
+{
+    for (size_t i = 0; i < mInputs.size(); i++) {
+        // return first input found not being in standby mode
+        // as only one input can be in this state
+        if (mInputs[i]->state() > AudioStreamInTegra::AUDIO_INPUT_CLOSED) {
+            return mInputs[i];
+        }
+    }
+
+    return NULL;
+}
+// ----------------------------------------------------------------------------
+
+AudioHardware::AudioStreamOutTegra::AudioStreamOutTegra() :
+    mHardware(0), mFd(-1), mStartCount(0), mRetryCount(0), mStandby(true), mDevices(0)
+{
+}
+
+status_t AudioHardware::AudioStreamOutTegra::set(
+        AudioHardware* hw, uint32_t devices, int *pFormat, uint32_t *pChannels, uint32_t *pRate)
+{
+    int lFormat = pFormat ? *pFormat : 0;
+    uint32_t lChannels = pChannels ? *pChannels : 0;
+    uint32_t lRate = pRate ? *pRate : 0;
+
+    mHardware = hw;
+
+    // fix up defaults
+    if (lFormat == 0) lFormat = format();
+    if (lChannels == 0) lChannels = channels();
+    if (lRate == 0) lRate = sampleRate();
+
+    // check values
+    if ((lFormat != format()) ||
+        (lChannels != channels()) ||
+        (lRate != sampleRate())) {
+        if (pFormat) *pFormat = format();
+        if (pChannels) *pChannels = channels();
+        if (pRate) *pRate = sampleRate();
+        return BAD_VALUE;
+    }
+
+    if (pFormat) *pFormat = lFormat;
+    if (pChannels) *pChannels = lChannels;
+    if (pRate) *pRate = lRate;
+
+    mDevices = devices;
+
+    return NO_ERROR;
+}
+
+AudioHardware::AudioStreamOutTegra::~AudioStreamOutTegra()
+{
+    if (mFd >= 0) close(mFd);
+}
+
+ssize_t AudioHardware::AudioStreamOutTegra::write(const void* buffer, size_t bytes)
+{
+    // LOGD("AudioStreamOutTegra::write(%p, %u)", buffer, bytes);
+    status_t status = NO_INIT;
+    size_t count = bytes;
+    const uint8_t* p = static_cast<const uint8_t*>(buffer);
+
+    if (mStandby) {
+        // open driver
+        LOGV("open driver");
+        status = ::open("/dev/audio0_out", O_RDWR);
+        if (status < 0) {
+            LOGE("Cannot open /dev/audio0_out errno: %d", errno);
+            goto Error;
+        }
+        mFd = status;
+
+        mStandby = false;
+    }
+
+    while (count) {
+        ssize_t written = ::write(mFd, p, count);
+        if (written >= 0) {
+            count -= written;
+            p += written;
+        } else {
+            if (errno != EAGAIN) return written;
+            mRetryCount++;
+            LOGW("EAGAIN - retry");
+        }
+    }
+
+    return bytes;
+
+Error:
+    if (mFd >= 0) {
+        ::close(mFd);
+        mFd = -1;
+    }
+    // Simulate audio output timing in case of error
+    usleep(bytes * 1000000 / frameSize() / sampleRate());
+
+    return status;
+}
+
+status_t AudioHardware::AudioStreamOutTegra::standby()
+{
+    status_t status = NO_ERROR;
+    if (!mStandby && mFd >= 0) {
+        ::close(mFd);
+        mFd = -1;
+    }
+    mStandby = true;
+    return status;
+}
+
+status_t AudioHardware::AudioStreamOutTegra::dump(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+    result.append("AudioStreamOutTegra::dump\n");
+    snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tchannels: %d\n", channels());
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tformat: %d\n", format());
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmHardware: %p\n", mHardware);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmFd: %d\n", mFd);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmStartCount: %d\n", mStartCount);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmRetryCount: %d\n", mRetryCount);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmStandby: %s\n", mStandby? "true": "false");
+    result.append(buffer);
+    ::write(fd, result.string(), result.size());
+    return NO_ERROR;
+}
+
+bool AudioHardware::AudioStreamOutTegra::checkStandby()
+{
+    return mStandby;
+}
+
+status_t AudioHardware::AudioStreamOutTegra::setParameters(const String8& keyValuePairs)
+{
+    AudioParameter param = AudioParameter(keyValuePairs);
+    String8 key = String8(AudioParameter::keyRouting);
+    status_t status = NO_ERROR;
+    int device;
+    LOGV("AudioStreamOutTegra::setParameters() %s", keyValuePairs.string());
+
+    if (param.getInt(key, device) == NO_ERROR) {
+        mDevices = device;
+        LOGV("set output routing %x", mDevices);
+        status = mHardware->doRouting();
+        param.remove(key);
+    }
+
+    if (param.size()) {
+        status = BAD_VALUE;
+    }
+    return status;
+}
+
+String8 AudioHardware::AudioStreamOutTegra::getParameters(const String8& keys)
+{
+    AudioParameter param = AudioParameter(keys);
+    String8 value;
+    String8 key = String8(AudioParameter::keyRouting);
+
+    if (param.get(key, value) == NO_ERROR) {
+        LOGV("get routing %x", mDevices);
+        param.addInt(key, (int)mDevices);
+    }
+
+    LOGV("AudioStreamOutTegra::getParameters() %s", param.toString().string());
+    return param.toString();
+}
+
+status_t AudioHardware::AudioStreamOutTegra::getRenderPosition(uint32_t *dspFrames)
+{
+    //TODO: enable when supported by driver
+    return INVALID_OPERATION;
+}
+
+// ----------------------------------------------------------------------------
+
+AudioHardware::AudioStreamInTegra::AudioStreamInTegra() :
+    mHardware(0), mFd(-1), mState(AUDIO_INPUT_CLOSED), mRetryCount(0),
+#if 0
+    mFormat(AUDIO_HW_IN_FORMAT), mChannels(AUDIO_HW_IN_CHANNELS),
+    mSampleRate(AUDIO_HW_IN_SAMPLERATE), mBufferSize(AUDIO_HW_IN_BUFFERSIZE),
+#endif
+    mAcoustics((AudioSystem::audio_in_acoustics)0), mDevices(0)
+{
+}
+
+status_t AudioHardware::AudioStreamInTegra::set(
+        AudioHardware* hw, uint32_t devices, int *pFormat, uint32_t *pChannels, uint32_t *pRate,
+        AudioSystem::audio_in_acoustics acoustic_flags)
+{
+#if 1
+    return NO_ERROR;
+#else
+    if (pFormat == 0 || *pFormat != AUDIO_HW_IN_FORMAT) {
+        *pFormat = AUDIO_HW_IN_FORMAT;
+        return BAD_VALUE;
+    }
+    if (pRate == 0) {
+        return BAD_VALUE;
+    }
+    uint32_t rate = hw->getInputSampleRate(*pRate);
+    if (rate != *pRate) {
+        *pRate = rate;
+        return BAD_VALUE;
+    }
+
+    if (pChannels == 0 || (*pChannels != AudioSystem::CHANNEL_IN_MONO &&
+        *pChannels != AudioSystem::CHANNEL_IN_STEREO)) {
+        *pChannels = AUDIO_HW_IN_CHANNELS;
+        return BAD_VALUE;
+    }
+
+    mHardware = hw;
+
+    LOGV("AudioStreamInTegra::set(%d, %d, %u)", *pFormat, *pChannels, *pRate);
+    if (mFd >= 0) {
+        LOGE("Audio record already open");
+        return -EPERM;
+    }
+
+    // open audio input device
+    status_t status = ::open("/dev/msm_pcm_in", O_RDWR);
+    if (status < 0) {
+        LOGE("Cannot open /dev/msm_pcm_in errno: %d", errno);
+        goto Error;
+    }
+    mFd = status;
+
+    // configuration
+    LOGV("get config");
+    struct msm_audio_config config;
+    status = ioctl(mFd, AUDIO_GET_CONFIG, &config);
+    if (status < 0) {
+        LOGE("Cannot read config");
+        goto Error;
+    }
+
+    LOGV("set config");
+    config.channel_count = AudioSystem::popCount(*pChannels);
+    config.sample_rate = *pRate;
+    config.buffer_size = bufferSize();
+    config.buffer_count = 2;
+    config.codec_type = CODEC_TYPE_PCM;
+    status = ioctl(mFd, AUDIO_SET_CONFIG, &config);
+    if (status < 0) {
+        LOGE("Cannot set config");
+        if (ioctl(mFd, AUDIO_GET_CONFIG, &config) == 0) {
+            if (config.channel_count == 1) {
+                *pChannels = AudioSystem::CHANNEL_IN_MONO;
+            } else {
+                *pChannels = AudioSystem::CHANNEL_IN_STEREO;
+            }
+            *pRate = config.sample_rate;
+        }
+        goto Error;
+    }
+
+    LOGV("confirm config");
+    status = ioctl(mFd, AUDIO_GET_CONFIG, &config);
+    if (status < 0) {
+        LOGE("Cannot read config");
+        goto Error;
+    }
+    LOGV("buffer_size: %u", config.buffer_size);
+    LOGV("buffer_count: %u", config.buffer_count);
+    LOGV("channel_count: %u", config.channel_count);
+    LOGV("sample_rate: %u", config.sample_rate);
+
+    mDevices = devices;
+    mFormat = AUDIO_HW_IN_FORMAT;
+    mChannels = *pChannels;
+    mSampleRate = config.sample_rate;
+    mBufferSize = config.buffer_size;
+
+    //mHardware->setMicMute_nosync(false);
+    mState = AUDIO_INPUT_OPENED;
+
+    if (!acoustic)
+        return NO_ERROR;
+
+    audpre_index = calculate_audpre_table_index(mSampleRate);
+    tx_iir_index = (audpre_index * 2) + (hw->checkOutputStandby() ? 0 : 1);
+    LOGD("audpre_index = %d, tx_iir_index = %d\n", audpre_index, tx_iir_index);
+
+    /**
+     * If audio-preprocessing failed, we should not block record.
+     */
+    int (*msm72xx_set_audpre_params)(int, int);
+    msm72xx_set_audpre_params = (int (*)(int, int))::dlsym(acoustic, "msm72xx_set_audpre_params");
+    status = msm72xx_set_audpre_params(audpre_index, tx_iir_index);
+    if (status < 0)
+        LOGE("Cannot set audpre parameters");
+
+    int (*msm72xx_enable_audpre)(int, int, int);
+    msm72xx_enable_audpre = (int (*)(int, int, int))::dlsym(acoustic, "msm72xx_enable_audpre");
+    mAcoustics = acoustic_flags;
+    status = msm72xx_enable_audpre((int)acoustic_flags, audpre_index, tx_iir_index);
+    if (status < 0)
+        LOGE("Cannot enable audpre");
+
+    return NO_ERROR;
+
+Error:
+    if (mFd >= 0) {
+        ::close(mFd);
+        mFd = -1;
+    }
+    return status;
+#endif
+}
+
+AudioHardware::AudioStreamInTegra::~AudioStreamInTegra()
+{
+    LOGV("AudioStreamInTegra destructor");
+    standby();
+}
+
+ssize_t AudioHardware::AudioStreamInTegra::read( void* buffer, ssize_t bytes)
+{
+#if 1
+    return 0;
+#else
+    LOGV("AudioStreamInTegra::read(%p, %ld)", buffer, bytes);
+    if (!mHardware) return -1;
+
+    size_t count = bytes;
+    uint8_t* p = static_cast<uint8_t*>(buffer);
+
+    if (mState < AUDIO_INPUT_OPENED) {
+        Mutex::Autolock lock(mHardware->mLock);
+        if (set(mHardware, mDevices, &mFormat, &mChannels, &mSampleRate, mAcoustics) != NO_ERROR) {
+            return -1;
+        }
+    }
+
+    if (mState < AUDIO_INPUT_STARTED) {
+        mState = AUDIO_INPUT_STARTED;
+        // force routing to input device
+        mHardware->clearCurDevice();
+        mHardware->doRouting();
+        if (ioctl(mFd, AUDIO_START, 0)) {
+            LOGE("Error starting record");
+            standby();
+            return -1;
+        }
+    }
+
+    while (count) {
+        ssize_t bytesRead = ::read(mFd, buffer, count);
+        if (bytesRead >= 0) {
+            count -= bytesRead;
+            p += bytesRead;
+        } else {
+            if (errno != EAGAIN) return bytesRead;
+            mRetryCount++;
+            LOGW("EAGAIN - retrying");
+        }
+    }
+    return bytes;
+#endif
+}
+
+status_t AudioHardware::AudioStreamInTegra::standby()
+{
+#if 0
+    if (mState > AUDIO_INPUT_CLOSED) {
+        if (mFd >= 0) {
+            ::close(mFd);
+            mFd = -1;
+        }
+        mState = AUDIO_INPUT_CLOSED;
+    }
+    if (!mHardware) return -1;
+    // restore output routing if necessary
+    mHardware->clearCurDevice();
+    mHardware->doRouting();
+#endif
+    return NO_ERROR;
+}
+
+status_t AudioHardware::AudioStreamInTegra::dump(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+    result.append("AudioStreamInTegra::dump\n");
+    snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tchannels: %d\n", channels());
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tformat: %d\n", format());
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmHardware: %p\n", mHardware);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmFd count: %d\n", mFd);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmState: %d\n", mState);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmRetryCount: %d\n", mRetryCount);
+    result.append(buffer);
+    ::write(fd, result.string(), result.size());
+    return NO_ERROR;
+}
+
+status_t AudioHardware::AudioStreamInTegra::setParameters(const String8& keyValuePairs)
+{
+#if 1
+    return NO_ERROR;
+#else
+    AudioParameter param = AudioParameter(keyValuePairs);
+    String8 key = String8(AudioParameter::keyRouting);
+    status_t status = NO_ERROR;
+    int device;
+    LOGV("AudioStreamInTegra::setParameters() %s", keyValuePairs.string());
+
+    if (param.getInt(key, device) == NO_ERROR) {
+        LOGV("set input routing %x", device);
+        if (device & (device - 1)) {
+            status = BAD_VALUE;
+        } else {
+            mDevices = device;
+            status = mHardware->doRouting();
+        }
+        param.remove(key);
+    }
+
+    if (param.size()) {
+        status = BAD_VALUE;
+    }
+    return status;
+#endif
+}
+
+String8 AudioHardware::AudioStreamInTegra::getParameters(const String8& keys)
+{
+    AudioParameter param = AudioParameter(keys);
+    String8 value;
+    String8 key = String8(AudioParameter::keyRouting);
+
+    if (param.get(key, value) == NO_ERROR) {
+        LOGV("get routing %x", mDevices);
+        param.addInt(key, (int)mDevices);
+    }
+
+    LOGV("AudioStreamInTegra::getParameters() %s", param.toString().string());
+    return param.toString();
+}
+
+// ----------------------------------------------------------------------------
+
+extern "C" AudioHardwareInterface* createAudioHardware(void) {
+    return new AudioHardware();
+}
+
+}; // namespace android
diff --git a/libaudio/AudioHardware.h b/libaudio/AudioHardware.h
new file mode 100644
index 0000000..d87d560
--- /dev/null
+++ b/libaudio/AudioHardware.h
@@ -0,0 +1,195 @@
+/*
+** Copyright 2008, 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.
+*/
+
+#ifndef ANDROID_AUDIO_HARDWARE_H
+#define ANDROID_AUDIO_HARDWARE_H
+
+#include <stdint.h>
+#include <sys/types.h>
+
+#include <utils/threads.h>
+#include <utils/SortedVector.h>
+
+#include <hardware_legacy/AudioHardwareBase.h>
+
+extern "C" {
+#include <linux/msm_audio.h>
+}
+
+namespace android {
+
+#include "cpcap_audio.h"
+
+#define AUDIO_HW_NUM_OUT_BUF 4
+#define AUDIO_HW_OUT_LATENCY_MS 0
+
+class AudioHardware : public  AudioHardwareBase
+{
+    class AudioStreamOutTegra;
+    class AudioStreamInTegra;
+
+public:
+                        AudioHardware();
+    virtual             ~AudioHardware();
+    virtual status_t    initCheck();
+
+    virtual status_t    setVoiceVolume(float volume);
+    virtual status_t    setMasterVolume(float volume);
+
+    virtual status_t    setMode(int mode);
+
+    // mic mute
+    virtual status_t    setMicMute(bool state);
+    virtual status_t    getMicMute(bool* state);
+
+    virtual status_t    setParameters(const String8& keyValuePairs);
+    virtual String8     getParameters(const String8& keys);
+
+    // create I/O streams
+    virtual AudioStreamOut* openOutputStream(
+                                uint32_t devices,
+                                int *format=0,
+                                uint32_t *channels=0,
+                                uint32_t *sampleRate=0,
+                                status_t *status=0);
+
+    virtual AudioStreamIn* openInputStream(
+
+                                uint32_t devices,
+                                int *format,
+                                uint32_t *channels,
+                                uint32_t *sampleRate,
+                                status_t *status,
+                                AudioSystem::audio_in_acoustics acoustics);
+
+    virtual    void        closeOutputStream(AudioStreamOut* out);
+    virtual    void        closeInputStream(AudioStreamIn* in);
+
+    virtual    size_t      getInputBufferSize(uint32_t sampleRate, int format, int channelCount);
+               void        clearCurDevice() { mCurSndDevice = -1; }
+
+protected:
+    virtual status_t    dump(int fd, const Vector<String16>& args);
+
+private:
+
+    status_t    doAudioRouteOrMute(uint32_t device);
+    status_t    setMicMute_nosync(bool state);
+    status_t    checkMicMute();
+    status_t    dumpInternals(int fd, const Vector<String16>& args);
+    uint32_t    getInputSampleRate(uint32_t sampleRate);
+    bool        checkOutputStandby();
+    status_t    doRouting();
+    AudioStreamInTegra*   getActiveInput_l();
+
+    class AudioStreamOutTegra : public AudioStreamOut {
+    public:
+                            AudioStreamOutTegra();
+        virtual             ~AudioStreamOutTegra();
+                status_t    set(AudioHardware* mHardware,
+                                uint32_t devices,
+                                int *pFormat,
+                                uint32_t *pChannels,
+                                uint32_t *pRate);
+        virtual uint32_t    sampleRate() const { return 44100; }
+        // must be 32-bit aligned - driver only seems to like 4800
+        virtual size_t      bufferSize() const { return 4800; }
+        virtual uint32_t    channels() const { return AudioSystem::CHANNEL_OUT_STEREO; }
+        virtual int         format() const { return AudioSystem::PCM_16_BIT; }
+        virtual uint32_t    latency() const { return (1000*AUDIO_HW_NUM_OUT_BUF*(bufferSize()/frameSize()))/sampleRate()+AUDIO_HW_OUT_LATENCY_MS; }
+        virtual status_t    setVolume(float left, float right) { return INVALID_OPERATION; }
+        virtual ssize_t     write(const void* buffer, size_t bytes);
+        virtual status_t    standby();
+        virtual status_t    dump(int fd, const Vector<String16>& args);
+                bool        checkStandby();
+        virtual status_t    setParameters(const String8& keyValuePairs);
+        virtual String8     getParameters(const String8& keys);
+                uint32_t    devices() { return mDevices; }
+        virtual status_t    getRenderPosition(uint32_t *dspFrames);
+
+    private:
+                AudioHardware* mHardware;
+                int         mFd;
+                int         mStartCount;
+                int         mRetryCount;
+                bool        mStandby;
+                uint32_t    mDevices;
+    };
+
+    class AudioStreamInTegra : public AudioStreamIn {
+    public:
+        enum input_state {
+            AUDIO_INPUT_CLOSED,
+            AUDIO_INPUT_OPENED,
+            AUDIO_INPUT_STARTED
+        };
+
+                            AudioStreamInTegra();
+        virtual             ~AudioStreamInTegra();
+                status_t    set(AudioHardware* mHardware,
+                                uint32_t devices,
+                                int *pFormat,
+                                uint32_t *pChannels,
+                                uint32_t *pRate,
+                                AudioSystem::audio_in_acoustics acoustics);
+        virtual size_t      bufferSize() const { return mBufferSize; }
+        virtual uint32_t    channels() const { return mChannels; }
+        virtual int         format() const { return mFormat; }
+        virtual uint32_t    sampleRate() const { return mSampleRate; }
+        virtual status_t    setGain(float gain) { return INVALID_OPERATION; }
+        virtual ssize_t     read(void* buffer, ssize_t bytes);
+        virtual status_t    dump(int fd, const Vector<String16>& args);
+        virtual status_t    standby();
+        virtual status_t    setParameters(const String8& keyValuePairs);
+        virtual String8     getParameters(const String8& keys);
+        virtual unsigned int  getInputFramesLost() const { return 0; }
+                uint32_t    devices() { return mDevices; }
+                int         state() const { return mState; }
+
+    private:
+                AudioHardware* mHardware;
+                int         mFd;
+                int         mState;
+                int         mRetryCount;
+                int         mFormat;
+                uint32_t    mChannels;
+                uint32_t    mSampleRate;
+                size_t      mBufferSize;
+                AudioSystem::audio_in_acoustics mAcoustics;
+                uint32_t    mDevices;
+    };
+
+            static const uint32_t inputSamplingRates[];
+            bool        mInit;
+            bool        mMicMute;
+            bool        mBluetoothNrec;
+            uint32_t    mBluetoothId;
+            AudioStreamOutTegra*  mOutput;
+            SortedVector <AudioStreamInTegra*>   mInputs;
+
+            int mCurSndDevice;
+
+     friend class AudioStreamInTegra;
+            Mutex       mLock;
+
+            int sndDevice;
+};
+
+// ----------------------------------------------------------------------------
+
+}; // namespace android
+
+#endif // ANDROID_AUDIO_HARDWARE_MSM72XX_H
diff --git a/libaudio/AudioPolicyManager.cpp b/libaudio/AudioPolicyManager.cpp
new file mode 100644
index 0000000..ca8f3bf
--- /dev/null
+++ b/libaudio/AudioPolicyManager.cpp
@@ -0,0 +1,45 @@
+/*
+ * Copyright (C) 2009 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.
+ */
+
+#define LOG_TAG "AudioPolicyManager"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+#include "AudioPolicyManager.h"
+#include <media/mediarecorder.h>
+
+namespace android {
+
+
+
+// ----------------------------------------------------------------------------
+// AudioPolicyManager for msm7k platform
+// Common audio policy manager code is implemented in AudioPolicyManagerBase class
+// ----------------------------------------------------------------------------
+
+// ---  class factory
+
+
+extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface)
+{
+    return new AudioPolicyManager(clientInterface);
+}
+
+extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
+{
+    delete interface;
+}
+
+}; // namespace android
diff --git a/libaudio/AudioPolicyManager.h b/libaudio/AudioPolicyManager.h
new file mode 100644
index 0000000..b591c1f
--- /dev/null
+++ b/libaudio/AudioPolicyManager.h
@@ -0,0 +1,46 @@
+/*
+ * Copyright (C) 2009 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.
+ */
+
+
+#include <stdint.h>
+#include <sys/types.h>
+#include <utils/Timers.h>
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
+#include <hardware_legacy/AudioPolicyManagerBase.h>
+
+
+namespace android {
+
+class AudioPolicyManager: public AudioPolicyManagerBase
+{
+
+public:
+                AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
+                : AudioPolicyManagerBase(clientInterface) {}
+
+        virtual ~AudioPolicyManager() {}
+
+protected:
+        // true is current platform implements a back microphone
+        virtual bool hasBackMicrophone() const { return false; }
+#ifdef WITH_A2DP
+        // true is current platform supports suplication of notifications and ringtones over A2DP output
+        virtual bool a2dpUsedForSonification() const { return true; }
+#endif
+
+};
+};
diff --git a/libaudio/MODULE_LICENSE_APACHE2 b/libaudio/MODULE_LICENSE_APACHE2
new file mode 100644
index 0000000..e69de29
--- /dev/null
+++ b/libaudio/MODULE_LICENSE_APACHE2
diff --git a/libaudio/NOTICE b/libaudio/NOTICE
new file mode 100644
index 0000000..3237da6
--- /dev/null
+++ b/libaudio/NOTICE
@@ -0,0 +1,190 @@
+
+   Copyright (c) 2008-2009, 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.
+
+   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.
+
+
+                                 Apache License
+                           Version 2.0, January 2004
+                        http://www.apache.org/licenses/
+
+   TERMS AND CONDITIONS FOR USE, REPRODUCTION, AND DISTRIBUTION
+
+   1. Definitions.
+
+      "License" shall mean the terms and conditions for use, reproduction,
+      and distribution as defined by Sections 1 through 9 of this document.
+
+      "Licensor" shall mean the copyright owner or entity authorized by
+      the copyright owner that is granting the License.
+
+      "Legal Entity" shall mean the union of the acting entity and all
+      other entities that control, are controlled by, or are under common
+      control with that entity. For the purposes of this definition,
+      "control" means (i) the power, direct or indirect, to cause the
+      direction or management of such entity, whether by contract or
+      otherwise, or (ii) ownership of fifty percent (50%) or more of the
+      outstanding shares, or (iii) beneficial ownership of such entity.
+
+      "You" (or "Your") shall mean an individual or Legal Entity
+      exercising permissions granted by this License.
+
+      "Source" form shall mean the preferred form for making modifications,
+      including but not limited to software source code, documentation
+      source, and configuration files.
+
+      "Object" form shall mean any form resulting from mechanical
+      transformation or translation of a Source form, including but
+      not limited to compiled object code, generated documentation,
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+      any Contribution intentionally submitted for inclusion in the Work
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+      the terms of any separate license agreement you may have executed
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+
diff --git a/libaudio/cpcap_audio.h b/libaudio/cpcap_audio.h
new file mode 100644
index 0000000..3bcda24
--- /dev/null
+++ b/libaudio/cpcap_audio.h
@@ -0,0 +1,39 @@
+/* include/linux/cpcap_audio.h
+ *
+ * Copyright (C) 2010 Google, Inc.
+ *
+ * This software is licensed under the terms of the GNU General Public
+ * License version 2, as published by the Free Software Foundation, and
+ * may be copied, distributed, and modified under those terms.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of 
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ */
+
+#ifndef _CPCAP_AUDIO_H
+#define _CPCAP_AUDIO_H
+
+#include <asm/ioctl.h>
+
+#define CPCAP_AUDIO_MAGIC 'c'
+
+#define CPCAP_AUDIO_OUT_SPEAKER		0
+#define CPCAP_AUDIO_OUT_HEADSET		1
+#define CPCAP_AUDIO_OUT_MAX		1
+
+#define CPCAP_AUDIO_OUT_SET_OUTPUT _IOW(CPCAP_AUDIO_MAGIC, 0, unsigned int)
+
+#define CPCAP_AUDIO_OUT_VOL_MIN 0
+#define CPCAP_AUDIO_OUT_VOL_MAX 15
+
+#define CPCAP_AUDIO_OUT_SET_VOLUME _IOW(CPCAP_AUDIO_MAGIC, 1, unsigned int)
+
+#define CPCAP_AUDIO_OUT_GET_OUTPUT \
+			_IOR(CPCAP_AUDIO_MAGIC, 2, unsigned int *)
+#define CPCAP_AUDIO_OUT_GET_VOLUME \
+			_IOR(CPCAP_AUDIO_MAGIC, 3, unsigned int *)
+
+#endif/*_CPCAP_AUDIO_H*/
diff --git a/overlay/packages/apps/Phone/res/values/config.xml b/overlay/packages/apps/Phone/res/values/config.xml
new file mode 100644
index 0000000..23e2bce
--- /dev/null
+++ b/overlay/packages/apps/Phone/res/values/config.xml
@@ -0,0 +1,23 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- 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.
+-->
+
+<!-- Stingray-specific phone app resources. -->
+<resources>
+
+    <!-- This device is not "voice capable"; it's data-only. -->
+    <bool name="voice_capable">false</bool>
+
+</resources>
diff --git a/ueventd.stingray.rc b/ueventd.stingray.rc
index 95c0b0c..80f8c8c 100644
--- a/ueventd.stingray.rc
+++ b/ueventd.stingray.rc
@@ -3,3 +3,4 @@
 /dev/knvmap	0666	system	system
 /dev/nvmap	0666	system	system
 /dev/ov5650	0666	media	camera
+/dev/audio*	0660	system	audio