am 2739a509: Removed ALSA user space library and utilities

* commit '2739a5099838025963269b6ff0e448c9c8a3ba52':
  Removed ALSA user space library and utilities

16 files changed, 4441 insertions, 3327 deletions

diff --git a/libaudio/Android.mk b/libaudio/Android.mk
index 04f24e0..5ff0a21 100755..100644
--- a/libaudio/Android.mk
+++ b/libaudio/Android.mk
@@ -1,68 +1,53 @@
-# hardware/libaudio-alsa/Android.mk
-#
-# Copyright 2008 Wind River Systems
-#
+LOCAL_PATH:= $(call my-dir)
 
 ifeq ($(TARGET_DEVICE),crespo)
-ifeq ($(BOARD_USES_ALSA_AUDIO),true)
-ifeq ($(filter-out s5pc110 s5pc100 s5p6440,$(TARGET_BOARD_PLATFORM)),)
-ifeq ($(BOARD_USES_GENERIC_AUDIO),false)
 
-  LOCAL_PATH := $(call my-dir)
-
-  include $(CLEAR_VARS)
-
-  LOCAL_ARM_MODE := arm
-  LOCAL_CFLAGS := -D_POSIX_SOURCE
-  LOCAL_WHOLE_STATIC_LIBRARIES := libasound
-
-  ifneq ($(ALSA_DEFAULT_SAMPLE_RATE),)
-    LOCAL_CFLAGS += -DALSA_DEFAULT_SAMPLE_RATE=$(ALSA_DEFAULT_SAMPLE_RATE)
-  endif
-
-  LOCAL_C_INCLUDES += device/samsung/crespo/alsa-lib/include
-  LOCAL_SRC_FILES := AudioHardwareALSA.cpp
-
-  LOCAL_MODULE := libaudio
+include $(CLEAR_VARS)
+LOCAL_SRC_FILES:= aplay.c alsa_pcm.c alsa_mixer.c
+LOCAL_MODULE:= aplay
+LOCAL_SHARED_LIBRARIES:= libc libcutils
+LOCAL_MODULE_TAGS:= debug
+include $(BUILD_EXECUTABLE)
 
-  LOCAL_STATIC_LIBRARIES += libaudiointerface
+include $(CLEAR_VARS)
+LOCAL_SRC_FILES:= arec.c alsa_pcm.c
+LOCAL_MODULE:= arec
+LOCAL_SHARED_LIBRARIES:= libc libcutils
+LOCAL_MODULE_TAGS:= debug
+include $(BUILD_EXECUTABLE)
 
-  LOCAL_SHARED_LIBRARIES := \
-    libcutils \
-    libutils \
-    libmedia \
-    libhardware_legacy \
-    libdl \
-    libc
+include $(CLEAR_VARS)
+LOCAL_SRC_FILES:= amix.c alsa_mixer.c
+LOCAL_MODULE:= amix
+LOCAL_SHARED_LIBRARIES := libc libcutils
+LOCAL_MODULE_TAGS:= debug
+include $(BUILD_EXECUTABLE)
 
+include $(CLEAR_VARS)
+LOCAL_SRC_FILES:= AudioHardware.cpp alsa_mixer.c alsa_pcm.c
+LOCAL_MODULE:= libaudio
+LOCAL_STATIC_LIBRARIES:= libaudiointerface
+LOCAL_SHARED_LIBRARIES:= libc libcutils libutils libmedia libhardware_legacy
 ifeq ($(BOARD_HAVE_BLUETOOTH),true)
   LOCAL_SHARED_LIBRARIES += liba2dp
 endif
 
-  include $(BUILD_SHARED_LIBRARY)
-
-# To build audiopolicy library 
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES:=               \
-    AudioPolicyManager.cpp
-
-LOCAL_SHARED_LIBRARIES := \
-    libcutils \
-    libutils \
-    libmedia
+ifeq ($(TARGET_SIMULATOR),true)
+ LOCAL_LDLIBS += -ldl
+else
+ LOCAL_SHARED_LIBRARIES += libdl
+endif
 
-LOCAL_STATIC_LIBRARIES := libaudiopolicybase
+include $(BUILD_SHARED_LIBRARY)
 
+include $(CLEAR_VARS)
+LOCAL_SRC_FILES:= AudioPolicyManager.cpp
 LOCAL_MODULE:= libaudiopolicy
-
+LOCAL_STATIC_LIBRARIES:= libaudiopolicybase
+LOCAL_SHARED_LIBRARIES:= libc libcutils libutils libmedia
 ifeq ($(BOARD_HAVE_BLUETOOTH),true)
   LOCAL_CFLAGS += -DWITH_A2DP
 endif
-
 include $(BUILD_SHARED_LIBRARY)
 
 endif
-endif
-endif
-endif
diff --git a/libaudio/AudioHardware.cpp b/libaudio/AudioHardware.cpp
new file mode 100644
index 0000000..10409a1
--- /dev/null
+++ b/libaudio/AudioHardware.cpp
@@ -0,0 +1,2014 @@
+/*
+** Copyright 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.
+*/
+
+#include <math.h>
+
+//#define LOG_NDEBUG 0
+
+#define LOG_TAG "AudioHardware"
+
+#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 <sys/resource.h>
+#include <dlfcn.h>
+#include <fcntl.h>
+
+#include "AudioHardware.h"
+#include <media/AudioRecord.h>
+#include <hardware_legacy/power.h>
+
+extern "C" {
+#include "alsa_audio.h"
+}
+
+
+namespace android {
+
+const uint32_t AudioHardware::inputSamplingRates[] = {
+        8000, 11025, 16000, 22050, 44100
+};
+
+//  trace driver operations for dump
+//
+#define DRIVER_TRACE
+
+enum {
+    DRV_NONE,
+    DRV_PCM_OPEN,
+    DRV_PCM_CLOSE,
+    DRV_PCM_WRITE,
+    DRV_PCM_READ,
+    DRV_MIXER_OPEN,
+    DRV_MIXER_CLOSE,
+    DRV_MIXER_GET,
+    DRV_MIXER_SEL
+};
+
+#ifdef DRIVER_TRACE
+#define TRACE_DRIVER_IN(op) mDriverOp = op;
+#define TRACE_DRIVER_OUT mDriverOp = DRV_NONE;
+#else
+#define TRACE_DRIVER_IN(op)
+#define TRACE_DRIVER_OUT
+#endif
+
+// ----------------------------------------------------------------------------
+
+AudioHardware::AudioHardware() :
+    mInit(false),
+    mMicMute(false),
+    mPcm(NULL),
+    mMixer(NULL),
+    mPcmOpenCnt(0),
+    mMixerOpenCnt(0),
+    mInCallAudioMode(false),
+    mInputSource("Default"),
+    mBluetoothNrec(true),
+    mSecRilLibHandle(NULL),
+    mRilClient(0),
+    mActivatedCP(false),
+    mDriverOp(DRV_NONE)
+{
+    loadRILD();
+    mInit = true;
+}
+
+AudioHardware::~AudioHardware()
+{
+    for (size_t index = 0; index < mInputs.size(); index++) {
+        closeInputStream(mInputs[index].get());
+    }
+    mInputs.clear();
+    closeOutputStream((AudioStreamOut*)mOutput.get());
+
+    if (mMixer) {
+        TRACE_DRIVER_IN(DRV_MIXER_CLOSE)
+        mixer_close(mMixer);
+        TRACE_DRIVER_OUT
+    }
+    if (mPcm) {
+        TRACE_DRIVER_IN(DRV_PCM_CLOSE)
+        pcm_close(mPcm);
+        TRACE_DRIVER_OUT
+    }
+
+    if (mSecRilLibHandle) {
+        if (disconnectRILD(mRilClient) != RIL_CLIENT_ERR_SUCCESS)
+            LOGE("Disconnect_RILD() error");
+
+        if (closeClientRILD(mRilClient) != RIL_CLIENT_ERR_SUCCESS)
+            LOGE("CloseClient_RILD() error");
+
+        mRilClient = 0;
+
+        dlclose(mSecRilLibHandle);
+        mSecRilLibHandle = NULL;
+    }
+
+    mInit = false;
+}
+
+status_t AudioHardware::initCheck()
+{
+    return mInit ? NO_ERROR : NO_INIT;
+}
+
+void AudioHardware::loadRILD(void)
+{
+    mSecRilLibHandle = dlopen("libsecril-client.so", RTLD_NOW);
+
+    if (mSecRilLibHandle) {
+        LOGV("libsecril-client.so is loaded");
+
+        openClientRILD   = (HRilClient (*)(void))
+                              dlsym(mSecRilLibHandle, "OpenClient_RILD");
+        disconnectRILD   = (int (*)(HRilClient))
+                              dlsym(mSecRilLibHandle, "Disconnect_RILD");
+        closeClientRILD  = (int (*)(HRilClient))
+                              dlsym(mSecRilLibHandle, "CloseClient_RILD");
+        isConnectedRILD  = (int (*)(HRilClient))
+                              dlsym(mSecRilLibHandle, "isConnected_RILD");
+        connectRILD      = (int (*)(HRilClient))
+                              dlsym(mSecRilLibHandle, "Connect_RILD");
+        setCallVolume    = (int (*)(HRilClient, SoundType, int))
+                              dlsym(mSecRilLibHandle, "SetCallVolume");
+        setCallAudioPath = (int (*)(HRilClient, AudioPath))
+                              dlsym(mSecRilLibHandle, "SetCallAudioPath");
+        setCallClockSync = (int (*)(HRilClient, SoundClockCondition))
+                              dlsym(mSecRilLibHandle, "SetCallClockSync");
+
+        if (!openClientRILD  || !disconnectRILD   || !closeClientRILD ||
+            !isConnectedRILD || !connectRILD      ||
+            !setCallVolume   || !setCallAudioPath || !setCallClockSync) {
+            LOGE("Can't load all functions from libsecril-client.so");
+
+            dlclose(mSecRilLibHandle);
+            mSecRilLibHandle = NULL;
+        } else {
+            mRilClient = openClientRILD();
+            if (!mRilClient) {
+                LOGE("OpenClient_RILD() error");
+
+                dlclose(mSecRilLibHandle);
+                mSecRilLibHandle = NULL;
+            }
+        }
+    } else {
+        LOGE("Can't load libsecril-client.so");
+    }
+}
+
+status_t AudioHardware::connectRILDIfRequired(void)
+{
+    if (!mSecRilLibHandle) {
+        LOGE("connectIfRequired() lib is not loaded");
+        return INVALID_OPERATION;
+    }
+
+    if (isConnectedRILD(mRilClient)) {
+        return OK;
+    }
+
+    if (connectRILD(mRilClient) != RIL_CLIENT_ERR_SUCCESS) {
+        LOGE("Connect_RILD() error");
+        return INVALID_OPERATION;
+    }
+
+    return OK;
+}
+
+AudioStreamOut* AudioHardware::openOutputStream(
+    uint32_t devices, int *format, uint32_t *channels,
+    uint32_t *sampleRate, status_t *status)
+{
+    sp <AudioStreamOutALSA> out;
+    status_t rc;
+
+    { // scope for the lock
+        Mutex::Autolock lock(mLock);
+
+        // only one output stream allowed
+        if (mOutput != 0) {
+            if (status) {
+                *status = INVALID_OPERATION;
+            }
+            return NULL;
+        }
+
+        out = new AudioStreamOutALSA();
+
+        rc = out->set(this, devices, format, channels, sampleRate);
+        if (rc == NO_ERROR) {
+            mOutput = out;
+        }
+    }
+
+    if (rc != NO_ERROR) {
+        if (out != 0) {
+            out.clear();
+        }
+    }
+    if (status) {
+        *status = rc;
+    }
+
+    return out.get();
+}
+
+void AudioHardware::closeOutputStream(AudioStreamOut* out) {
+    sp <AudioStreamOutALSA> spOut;
+    {
+        Mutex::Autolock lock(mLock);
+        if (mOutput == 0 || mOutput.get() != out) {
+            LOGW("Attempt to close invalid output stream");
+            return;
+        }
+        spOut = mOutput;
+        mOutput.clear();
+    }
+    spOut.clear();
+}
+
+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)) {
+        if (status) {
+            *status = BAD_VALUE;
+        }
+        return NULL;
+    }
+
+    status_t rc = NO_ERROR;
+    sp <AudioStreamInALSA> in;
+
+    { // scope for the lock
+        Mutex::Autolock lock(mLock);
+
+        in = new AudioStreamInALSA();
+        rc = in->set(this, devices, format, channels, sampleRate, acoustic_flags);
+        if (rc == NO_ERROR) {
+            mInputs.add(in);
+        }
+    }
+
+    if (rc != NO_ERROR) {
+        if (in != 0) {
+            in.clear();
+        }
+    }
+    if (status) {
+        *status = rc;
+    }
+
+    LOGV("AudioHardware::openInputStream()%p", in.get());
+    return in.get();
+}
+
+void AudioHardware::closeInputStream(AudioStreamIn* in) {
+
+    sp<AudioStreamInALSA> spIn;
+    {
+        Mutex::Autolock lock(mLock);
+
+        ssize_t index = mInputs.indexOf((AudioStreamInALSA *)in);
+        if (index < 0) {
+            LOGW("Attempt to close invalid input stream");
+            return;
+        }
+        spIn = mInputs[index];
+        mInputs.removeAt(index);
+    }
+    LOGV("AudioHardware::closeInputStream()%p", in);
+    spIn.clear();
+}
+
+
+status_t AudioHardware::setMode(int mode)
+{
+    sp<AudioStreamOutALSA> spOut;
+    sp<AudioStreamInALSA> spIn;
+    status_t status;
+
+    // bump thread priority to speed up mutex acquisition
+    int  priority = getpriority(PRIO_PROCESS, 0);
+    setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_URGENT_AUDIO);
+
+    // Mutex acquisition order is always out -> in -> hw
+    AutoMutex lock(mLock);
+
+    spOut = mOutput;
+    while (spOut != 0) {
+        if (!spOut->checkStandby()) {
+            int cnt = spOut->standbyCnt();
+            mLock.unlock();
+            spOut->lock();
+            mLock.lock();
+            // make sure that another thread did not change output state while the
+            // mutex is released
+            if ((spOut == mOutput) && (cnt == spOut->standbyCnt())) {
+                break;
+            }
+            spOut->unlock();
+            spOut = mOutput;
+        } else {
+            spOut.clear();
+        }
+    }
+    // spOut is not 0 here only if the output is active
+
+    spIn = getActiveInput_l();
+    while (spIn != 0) {
+        int cnt = spIn->standbyCnt();
+        mLock.unlock();
+        spIn->lock();
+        mLock.lock();
+        // make sure that another thread did not change input state while the
+        // mutex is released
+        if ((spIn == getActiveInput_l()) && (cnt == spIn->standbyCnt())) {
+            break;
+        }
+        spIn->unlock();
+        spIn = getActiveInput_l();
+    }
+    // spIn is not 0 here only if the input is active
+
+    setpriority(PRIO_PROCESS, 0, priority);
+
+    int prevMode = mMode;
+    status = AudioHardwareBase::setMode(mode);
+    LOGV("setMode() : new %d, old %d", mMode, prevMode);
+    if (status == NO_ERROR) {
+        // activate call clock in radio when entering in call or ringtone mode
+        if (prevMode == AudioSystem::MODE_NORMAL)
+        {
+            if ((!mActivatedCP) && (mSecRilLibHandle) && (connectRILDIfRequired() == OK)) {
+                setCallClockSync(mRilClient, SOUND_CLOCK_START);
+                mActivatedCP = true;
+            }
+        }
+
+        if (mMode == AudioSystem::MODE_IN_CALL && !mInCallAudioMode) {
+            if (spOut != 0) {
+                LOGV("setMode() in call force output standby");
+                spOut->doStandby_l();
+            }
+            if (spIn != 0) {
+                LOGV("setMode() in call force input standby");
+                spIn->doStandby_l();
+            }
+
+            LOGV("setMode() openPcmOut_l()");
+            openPcmOut_l();
+            openMixer_l();
+            setInputSource_l(String8("Default"));
+            mInCallAudioMode = true;
+        }
+        if (mMode == AudioSystem::MODE_NORMAL && mInCallAudioMode) {
+            setInputSource_l(mInputSource);
+            if (mMixer != NULL) {
+                TRACE_DRIVER_IN(DRV_MIXER_GET)
+                struct mixer_ctl *ctl= mixer_get_control(mMixer, "Playback Path", 0);
+                TRACE_DRIVER_OUT
+                if (ctl != NULL) {
+                    LOGV("setMode() reset Playback Path to RCV");
+                    TRACE_DRIVER_IN(DRV_MIXER_SEL)
+                    mixer_ctl_select(ctl, "RCV");
+                    TRACE_DRIVER_OUT
+                }
+            }
+            LOGV("setMode() closePcmOut_l()");
+            closeMixer_l();
+            closePcmOut_l();
+
+            if (spOut != 0) {
+                LOGV("setMode() off call force output standby");
+                spOut->doStandby_l();
+            }
+            if (spIn != 0) {
+                LOGV("setMode() off call force input standby");
+                spIn->doStandby_l();
+            }
+
+            mInCallAudioMode = false;
+        }
+
+        if (mMode == AudioSystem::MODE_NORMAL) {
+            if(mActivatedCP)
+                mActivatedCP = false;
+        }
+    }
+
+    if (spIn != 0) {
+        spIn->unlock();
+    }
+    if (spOut != 0) {
+        spOut->unlock();
+    }
+
+    return status;
+}
+
+status_t AudioHardware::setMicMute(bool state)
+{
+    LOGV("setMicMute(%d) mMicMute %d", state, mMicMute);
+    sp<AudioStreamInALSA> spIn;
+    {
+        AutoMutex lock(mLock);
+        if (mMicMute != state) {
+            mMicMute = state;
+            // in call mute is handled by RIL
+            if (mMode != AudioSystem::MODE_IN_CALL) {
+                spIn = getActiveInput_l();
+            }
+        }
+    }
+
+    if (spIn != 0) {
+        spIn->standby();
+    }
+
+    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_NREC_VALUE_ON[] = "on";
+
+    key = String8(BT_NREC_KEY);
+    if (param.get(key, value) == NO_ERROR) {
+        if (value == BT_NREC_VALUE_ON) {
+            mBluetoothNrec = true;
+        } else {
+            mBluetoothNrec = false;
+            LOGD("Turning noise reduction and echo cancellation off for BT "
+                 "headset");
+        }
+    }
+
+    return NO_ERROR;
+}
+
+String8 AudioHardware::getParameters(const String8& keys)
+{
+    AudioParameter request = AudioParameter(keys);
+    AudioParameter reply = AudioParameter();
+
+    LOGV("getParameters() %s", keys.string());
+
+    return reply.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;
+    }
+    if (sampleRate != 8000 && sampleRate != 11025 && sampleRate != 16000 &&
+            sampleRate != 22050 && sampleRate != 44100) {
+        LOGW("getInputBufferSize bad sample rate: %d", sampleRate);
+        return 0;
+    }
+
+    return AudioStreamInALSA::getBufferSize(sampleRate, channelCount);
+}
+
+
+status_t AudioHardware::setVoiceVolume(float volume)
+{
+    LOGD("### setVoiceVolume");
+
+    AutoMutex lock(mLock);
+    if ( (AudioSystem::MODE_IN_CALL == mMode) && (mSecRilLibHandle) &&
+         (connectRILDIfRequired() == OK) ) {
+
+        uint32_t device = AudioSystem::DEVICE_OUT_EARPIECE;
+        if (mOutput != 0) {
+            device = mOutput->device();
+        }
+        int int_volume = (int)(volume * 5);
+        SoundType type;
+
+        LOGD("### route(%d) call volume(%f)", device, volume);
+        switch (device) {
+            case AudioSystem::DEVICE_OUT_EARPIECE:
+                LOGD("### earpiece call volume");
+                type = SOUND_TYPE_VOICE;
+                break;
+
+            case AudioSystem::DEVICE_OUT_SPEAKER:
+                LOGD("### speaker call volume");
+                type = SOUND_TYPE_SPEAKER;
+                break;
+
+            case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO:
+            case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
+            case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
+                LOGD("### bluetooth call volume");
+                type = SOUND_TYPE_BTVOICE;
+                break;
+
+            case AudioSystem::DEVICE_OUT_WIRED_HEADSET:
+            case AudioSystem::DEVICE_OUT_WIRED_HEADPHONE: // Use receive path with 3 pole headset.
+                LOGD("### headset call volume");
+                type = SOUND_TYPE_HEADSET;
+                break;
+
+            default:
+                LOGW("### Call volume setting error!!!0x%08x \n", device);
+                type = SOUND_TYPE_VOICE;
+                break;
+        }
+        setCallVolume(mRilClient, type, int_volume);
+    }
+
+    return NO_ERROR;
+}
+
+status_t AudioHardware::setMasterVolume(float volume)
+{
+    LOGV("Set master volume to %f.\n", volume);
+    // 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;
+}
+
+static const int kDumpLockRetries = 50;
+static const int kDumpLockSleep = 20000;
+
+static bool tryLock(Mutex& mutex)
+{
+    bool locked = false;
+    for (int i = 0; i < kDumpLockRetries; ++i) {
+        if (mutex.tryLock() == NO_ERROR) {
+            locked = true;
+            break;
+        }
+        usleep(kDumpLockSleep);
+    }
+    return locked;
+}
+
+status_t AudioHardware::dump(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    bool locked = tryLock(mLock);
+    if (!locked) {
+        snprintf(buffer, SIZE, "\n\tAudioHardware maybe deadlocked\n");
+    } else {
+        mLock.unlock();
+    }
+
+    snprintf(buffer, SIZE, "\tInit %s\n", (mInit) ? "OK" : "Failed");
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tMic Mute %s\n", (mMicMute) ? "ON" : "OFF");
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmPcm: %p\n", mPcm);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmPcmOpenCnt: %d\n", mPcmOpenCnt);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmMixer: %p\n", mMixer);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmMixerOpenCnt: %d\n", mMixerOpenCnt);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tIn Call Audio Mode %s\n",
+             (mInCallAudioMode) ? "ON" : "OFF");
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tInput source %s\n", mInputSource.string());
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmSecRilLibHandle: %p\n", mSecRilLibHandle);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmRilClient: %p\n", mRilClient);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tCP %s\n",
+             (mActivatedCP) ? "Activated" : "Deactivated");
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmDriverOp: %d\n", mDriverOp);
+    result.append(buffer);
+
+    snprintf(buffer, SIZE, "\n\tmOutput %p dump:\n", mOutput.get());
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+    if (mOutput != 0) {
+        mOutput->dump(fd, args);
+    }
+
+    snprintf(buffer, SIZE, "\n\t%d inputs opened:\n", mInputs.size());
+    write(fd, buffer, strlen(buffer));
+    for (size_t i = 0; i < mInputs.size(); i++) {
+        snprintf(buffer, SIZE, "\t- input %d dump:\n", i);
+        write(fd, buffer, strlen(buffer));
+        mInputs[i]->dump(fd, args);
+    }
+
+    return NO_ERROR;
+}
+
+status_t AudioHardware::setIncallPath_l(uint32_t device)
+{
+    LOGV("setIncallPath_l: device %x", device);
+
+    // Setup sound path for CP clocking
+    if ((mSecRilLibHandle) &&
+        (connectRILDIfRequired() == OK)) {
+
+        if (mMode == AudioSystem::MODE_IN_CALL) {
+            LOGD("### incall mode route (%d)", device);
+            AudioPath path;
+            switch(device){
+                case AudioSystem::DEVICE_OUT_EARPIECE:
+                    LOGD("### incall mode earpiece route");
+                    path = SOUND_AUDIO_PATH_HANDSET;
+                    break;
+
+                case AudioSystem::DEVICE_OUT_SPEAKER:
+                    LOGD("### incall mode speaker route");
+                    path = SOUND_AUDIO_PATH_SPEAKER;
+                    break;
+
+                case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO:
+                case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
+                case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
+                    LOGD("### incall mode bluetooth route %s NR", mBluetoothNrec ? "" : "NO");
+                    if (mBluetoothNrec) {
+                        path = SOUND_AUDIO_PATH_BLUETOOTH;
+                    } else {
+                        path = SOUND_AUDIO_PATH_BLUETOOTH_NO_NR;
+                    }
+                    break;
+
+                case AudioSystem::DEVICE_OUT_WIRED_HEADPHONE :
+                    LOGD("### incall mode headphone route");
+                    path = SOUND_AUDIO_PATH_HEADPHONE;
+                    break;
+
+                case AudioSystem::DEVICE_OUT_WIRED_HEADSET :
+                    LOGD("### incall mode headset route");
+                    path = SOUND_AUDIO_PATH_HEADSET;
+                    break;
+
+                default:
+                    LOGW("### incall mode Error!! route = [%d]", device);
+                    path = SOUND_AUDIO_PATH_HANDSET;
+                    break;
+            }
+
+            setCallAudioPath(mRilClient, path);
+
+            if (mMixer != NULL) {
+                TRACE_DRIVER_IN(DRV_MIXER_GET)
+                struct mixer_ctl *ctl= mixer_get_control(mMixer, "Voice Call Path", 0);
+                TRACE_DRIVER_OUT
+                LOGE_IF(ctl == NULL, "setIncallPath_l() could not get mixer ctl");
+                if (ctl != NULL) {
+                    LOGV("setIncallPath_l() Voice Call Path, (%x)", device);
+                    TRACE_DRIVER_IN(DRV_MIXER_SEL)
+                    mixer_ctl_select(ctl, getVoiceRouteFromDevice(device));
+                    TRACE_DRIVER_OUT
+                }
+            }
+        }
+    }
+    return NO_ERROR;
+}
+
+struct pcm *AudioHardware::openPcmOut_l()
+{
+    LOGD("openPcmOut_l() mPcmOpenCnt: %d", mPcmOpenCnt);
+    if (mPcmOpenCnt++ == 0) {
+        if (mPcm != NULL) {
+            LOGE("openPcmOut_l() mPcmOpenCnt == 0 and mPcm == %p\n", mPcm);
+            mPcmOpenCnt--;
+            return NULL;
+        }
+        unsigned flags = PCM_OUT;
+
+        flags |= (AUDIO_HW_OUT_PERIOD_MULT - 1) << PCM_PERIOD_SZ_SHIFT;
+        flags |= (AUDIO_HW_OUT_PERIOD_CNT - PCM_PERIOD_CNT_MIN) << PCM_PERIOD_CNT_SHIFT;
+
+        TRACE_DRIVER_IN(DRV_PCM_OPEN)
+        mPcm = pcm_open(flags);
+        TRACE_DRIVER_OUT
+        if (!pcm_ready(mPcm)) {
+            LOGE("openPcmOut_l() cannot open pcm_out driver: %s\n", pcm_error(mPcm));
+            TRACE_DRIVER_IN(DRV_PCM_CLOSE)
+            pcm_close(mPcm);
+            TRACE_DRIVER_OUT
+            mPcmOpenCnt--;
+            mPcm = NULL;
+        }
+    }
+    return mPcm;
+}
+
+void AudioHardware::closePcmOut_l()
+{
+    LOGD("closePcmOut_l() mPcmOpenCnt: %d", mPcmOpenCnt);
+    if (mPcmOpenCnt == 0) {
+        LOGE("closePcmOut_l() mPcmOpenCnt == 0");
+        return;
+    }
+
+    if (--mPcmOpenCnt == 0) {
+        TRACE_DRIVER_IN(DRV_PCM_CLOSE)
+        pcm_close(mPcm);
+        TRACE_DRIVER_OUT
+        mPcm = NULL;
+    }
+}
+
+struct mixer *AudioHardware::openMixer_l()
+{
+    LOGV("openMixer_l() mMixerOpenCnt: %d", mMixerOpenCnt);
+    if (mMixerOpenCnt++ == 0) {
+        if (mMixer != NULL) {
+            LOGE("openMixer_l() mMixerOpenCnt == 0 and mMixer == %p\n", mMixer);
+            mMixerOpenCnt--;
+            return NULL;
+        }
+        TRACE_DRIVER_IN(DRV_MIXER_OPEN)
+        mMixer = mixer_open();
+        TRACE_DRIVER_OUT
+        if (mMixer == NULL) {
+            LOGE("openMixer_l() cannot open mixer");
+            mMixerOpenCnt--;
+            return NULL;
+        }
+    }
+    return mMixer;
+}
+
+void AudioHardware::closeMixer_l()
+{
+    LOGV("closeMixer_l() mMixerOpenCnt: %d", mMixerOpenCnt);
+    if (mMixerOpenCnt == 0) {
+        LOGE("closeMixer_l() mMixerOpenCnt == 0");
+        return;
+    }
+
+    if (--mMixerOpenCnt == 0) {
+        TRACE_DRIVER_IN(DRV_MIXER_CLOSE)
+        mixer_close(mMixer);
+        TRACE_DRIVER_OUT
+        mMixer = NULL;
+    }
+}
+
+const char *AudioHardware::getOutputRouteFromDevice(uint32_t device)
+{
+    switch (device) {
+    case AudioSystem::DEVICE_OUT_EARPIECE:
+        return "RCV";
+    case AudioSystem::DEVICE_OUT_SPEAKER:
+        if (mMode == AudioSystem::MODE_RINGTONE) return "RING_SPK";
+        else return "SPK";
+    case AudioSystem::DEVICE_OUT_WIRED_HEADPHONE:
+        if (mMode == AudioSystem::MODE_RINGTONE) return "RING_NO_MIC";
+        else return "HP_NO_MIC";
+    case AudioSystem::DEVICE_OUT_WIRED_HEADSET:
+        if (mMode == AudioSystem::MODE_RINGTONE) return "RING_HP";
+        else return "HP";
+    case (AudioSystem::DEVICE_OUT_SPEAKER|AudioSystem::DEVICE_OUT_WIRED_HEADPHONE):
+    case (AudioSystem::DEVICE_OUT_SPEAKER|AudioSystem::DEVICE_OUT_WIRED_HEADSET):
+        if (mMode == AudioSystem::MODE_RINGTONE) return "RING_SPK_HP";
+        else return "SPK_HP";
+    case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO:
+    case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
+    case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
+        return "BT";
+    default:
+        return "OFF";
+    }
+}
+
+const char *AudioHardware::getVoiceRouteFromDevice(uint32_t device)
+{
+    switch (device) {
+    case AudioSystem::DEVICE_OUT_EARPIECE:
+        return "RCV";
+    case AudioSystem::DEVICE_OUT_SPEAKER:
+        return "SPK";
+    case AudioSystem::DEVICE_OUT_WIRED_HEADPHONE:
+        return "HP_NO_MIC";
+    case AudioSystem::DEVICE_OUT_WIRED_HEADSET:
+        return "HP";
+    case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO:
+    case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
+    case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
+        return "BT";
+    default:
+        return "OFF";
+    }
+}
+
+const char *AudioHardware::getInputRouteFromDevice(uint32_t device)
+{
+    if (mMicMute) {
+        return "MIC OFF";
+    }
+
+    switch (device) {
+    case AudioSystem::DEVICE_IN_BUILTIN_MIC:
+        return "Main Mic";
+    case AudioSystem::DEVICE_IN_WIRED_HEADSET:
+        return "Hands Free Mic";
+    case AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET:
+        return "BT Sco Mic";
+    default:
+        return "MIC OFF";
+    }
+}
+
+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
+sp <AudioHardware::AudioStreamInALSA> AudioHardware::getActiveInput_l()
+{
+    sp< AudioHardware::AudioStreamInALSA> spIn;
+
+    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]->checkStandby()) {
+            spIn = mInputs[i];
+            break;
+        }
+    }
+
+    return spIn;
+}
+
+status_t AudioHardware::setInputSource_l(String8 source)
+{
+     LOGV("setInputSource_l(%s)", source.string());
+     if (source != mInputSource) {
+         if ((source == "Default") || (mMode != AudioSystem::MODE_IN_CALL)) {
+             if (mMixer) {
+                 TRACE_DRIVER_IN(DRV_MIXER_GET)
+                 struct mixer_ctl *ctl= mixer_get_control(mMixer, "Input Source", 0);
+                 TRACE_DRIVER_OUT
+                 if (ctl == NULL) {
+                     return NO_INIT;
+                 }
+                 LOGV("mixer_ctl_select, Input Source, (%s)", source.string());
+                 TRACE_DRIVER_IN(DRV_MIXER_SEL)
+                 mixer_ctl_select(ctl, source.string());
+                 TRACE_DRIVER_OUT
+             }
+         }
+         mInputSource = source;
+     }
+
+     return NO_ERROR;
+}
+
+
+//------------------------------------------------------------------------------
+//  AudioStreamOutALSA
+//------------------------------------------------------------------------------
+
+AudioHardware::AudioStreamOutALSA::AudioStreamOutALSA() :
+    mHardware(0), mPcm(0), mMixer(0), mRouteCtl(0),
+    mStandby(true), mDevices(0), mChannels(AUDIO_HW_OUT_CHANNELS),
+    mSampleRate(AUDIO_HW_OUT_SAMPLERATE), mBufferSize(AUDIO_HW_OUT_PERIOD_BYTES),
+    mDriverOp(DRV_NONE), mStandbyCnt(0)
+{
+}
+
+status_t AudioHardware::AudioStreamOutALSA::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;
+    mDevices = devices;
+
+    // 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;
+
+    mChannels = lChannels;
+    mSampleRate = lRate;
+    mBufferSize = AUDIO_HW_OUT_PERIOD_BYTES;
+
+    return NO_ERROR;
+}
+
+AudioHardware::AudioStreamOutALSA::~AudioStreamOutALSA()
+{
+    standby();
+}
+
+ssize_t AudioHardware::AudioStreamOutALSA::write(const void* buffer, size_t bytes)
+{
+    //    LOGV("AudioStreamOutALSA::write(%p, %u)", buffer, bytes);
+    status_t status = NO_INIT;
+    const uint8_t* p = static_cast<const uint8_t*>(buffer);
+    int ret;
+
+    if (mHardware == NULL) return NO_INIT;
+
+    { // scope for the lock
+
+        AutoMutex lock(mLock);
+
+        if (mStandby) {
+            AutoMutex hwLock(mHardware->lock());
+
+            LOGD("AudioHardware pcm playback is exiting standby.");
+            acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioOutLock");
+
+            sp<AudioStreamInALSA> spIn = mHardware->getActiveInput_l();
+            while (spIn != 0) {
+                int cnt = spIn->standbyCnt();
+                mHardware->lock().unlock();
+                // Mutex acquisition order is always out -> in -> hw
+                spIn->lock();
+                mHardware->lock().lock();
+                // make sure that another thread did not change input state
+                // while the mutex is released
+                if ((spIn == mHardware->getActiveInput_l()) &&
+                        (cnt == spIn->standbyCnt())) {
+                    LOGV("AudioStreamOutALSA::write() force input standby");
+                    spIn->close_l();
+                    break;
+                }
+                spIn->unlock();
+                spIn = mHardware->getActiveInput_l();
+            }
+            // spIn is not 0 here only if the input was active and has been
+            // closed above
+
+            // open output before input
+            open_l();
+
+            if (spIn != 0) {
+                if (spIn->open_l() != NO_ERROR) {
+                    spIn->doStandby_l();
+                }
+                spIn->unlock();
+            }
+            if (mPcm == NULL) {
+                release_wake_lock("AudioOutLock");
+                goto Error;
+            }
+            mStandby = false;
+        }
+
+        TRACE_DRIVER_IN(DRV_PCM_WRITE)
+        ret = pcm_write(mPcm,(void*) p, bytes);
+        TRACE_DRIVER_OUT
+
+        if (ret == 0) {
+            return bytes;
+        }
+        LOGW("write error: %d", errno);
+        status = -errno;
+    }
+Error:
+
+    standby();
+
+    // Simulate audio output timing in case of error
+    usleep((((bytes * 1000) / frameSize()) * 1000) / sampleRate());
+
+    return status;
+}
+
+status_t AudioHardware::AudioStreamOutALSA::standby()
+{
+    if (mHardware == NULL) return NO_INIT;
+
+    AutoMutex lock(mLock);
+
+    { // scope for the AudioHardware lock
+        AutoMutex hwLock(mHardware->lock());
+
+        doStandby_l();
+    }
+
+    return NO_ERROR;
+}
+
+void AudioHardware::AudioStreamOutALSA::doStandby_l()
+{
+    mStandbyCnt++;
+
+    if (!mStandby) {
+        LOGD("AudioHardware pcm playback is going to standby.");
+        release_wake_lock("AudioOutLock");
+        mStandby = true;
+    }
+
+    close_l();
+}
+
+void AudioHardware::AudioStreamOutALSA::close_l()
+{
+    if (mMixer) {
+        mHardware->closeMixer_l();
+        mMixer = NULL;
+        mRouteCtl = NULL;
+    }
+    if (mPcm) {
+        mHardware->closePcmOut_l();
+        mPcm = NULL;
+    }
+}
+
+status_t AudioHardware::AudioStreamOutALSA::open_l()
+{
+    LOGV("open pcm_out driver");
+    mPcm = mHardware->openPcmOut_l();
+    if (mPcm == NULL) {
+        return NO_INIT;
+    }
+
+    mMixer = mHardware->openMixer_l();
+    if (mMixer) {
+        LOGV("open playback normal");
+        TRACE_DRIVER_IN(DRV_MIXER_GET)
+        mRouteCtl = mixer_get_control(mMixer, "Playback Path", 0);
+        TRACE_DRIVER_OUT
+    }
+    if (mHardware->mode() != AudioSystem::MODE_IN_CALL) {
+        const char *route = mHardware->getOutputRouteFromDevice(mDevices);
+        LOGV("write() wakeup setting route %s", route);
+        if (mRouteCtl) {
+            TRACE_DRIVER_IN(DRV_MIXER_SEL)
+            mixer_ctl_select(mRouteCtl, route);
+            TRACE_DRIVER_OUT
+        }
+    }
+    return NO_ERROR;
+}
+
+status_t AudioHardware::AudioStreamOutALSA::dump(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    bool locked = tryLock(mLock);
+    if (!locked) {
+        snprintf(buffer, SIZE, "\n\t\tAudioStreamOutALSA maybe deadlocked\n");
+    } else {
+        mLock.unlock();
+    }
+
+    snprintf(buffer, SIZE, "\t\tmHardware: %p\n", mHardware);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmPcm: %p\n", mPcm);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmMixer: %p\n", mMixer);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmRouteCtl: %p\n", mRouteCtl);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tStandby %s\n", (mStandby) ? "ON" : "OFF");
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmDevices: 0x%08x\n", mDevices);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmChannels: 0x%08x\n", mChannels);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmSampleRate: %d\n", mSampleRate);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmBufferSize: %d\n", mBufferSize);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmDriverOp: %d\n", mDriverOp);
+    result.append(buffer);
+
+    ::write(fd, result.string(), result.size());
+
+    return NO_ERROR;
+}
+
+bool AudioHardware::AudioStreamOutALSA::checkStandby()
+{
+    return mStandby;
+}
+
+status_t AudioHardware::AudioStreamOutALSA::setParameters(const String8& keyValuePairs)
+{
+    AudioParameter param = AudioParameter(keyValuePairs);
+    status_t status = NO_ERROR;
+    int device;
+    LOGD("AudioStreamOutALSA::setParameters() %s", keyValuePairs.string());
+
+    if (mHardware == NULL) return NO_INIT;
+
+    {
+        AutoMutex lock(mLock);
+
+        if (param.getInt(String8(AudioParameter::keyRouting), device) == NO_ERROR)
+        {
+            AutoMutex hwLock(mHardware->lock());
+
+            if (mDevices != (uint32_t)device) {
+                mDevices = (uint32_t)device;
+                if (mHardware->mode() != AudioSystem::MODE_IN_CALL) {
+                    doStandby_l();
+                }
+            }
+            if (mHardware->mode() == AudioSystem::MODE_IN_CALL) {
+                mHardware->setIncallPath_l(device);
+            }
+            param.remove(String8(AudioParameter::keyRouting));
+        }
+    }
+
+    if (param.size()) {
+        status = BAD_VALUE;
+    }
+
+
+    return status;
+
+}
+
+String8 AudioHardware::AudioStreamOutALSA::getParameters(const String8& keys)
+{
+    AudioParameter param = AudioParameter(keys);
+    String8 value;
+    String8 key = String8(AudioParameter::keyRouting);
+
+    if (param.get(key, value) == NO_ERROR) {
+        param.addInt(key, (int)mDevices);
+    }
+
+    LOGV("AudioStreamOutALSA::getParameters() %s", param.toString().string());
+    return param.toString();
+}
+
+status_t AudioHardware::AudioStreamOutALSA::getRenderPosition(uint32_t *dspFrames)
+{
+    //TODO
+    return INVALID_OPERATION;
+}
+
+//------------------------------------------------------------------------------
+//  AudioStreamInALSA
+//------------------------------------------------------------------------------
+
+AudioHardware::AudioStreamInALSA::AudioStreamInALSA() :
+    mHardware(0), mPcm(0), mMixer(0), mRouteCtl(0),
+    mStandby(true), mDevices(0), mChannels(AUDIO_HW_IN_CHANNELS), mChannelCount(1),
+    mSampleRate(AUDIO_HW_IN_SAMPLERATE), mBufferSize(AUDIO_HW_IN_PERIOD_BYTES),
+    mDownSampler(NULL), mReadStatus(NO_ERROR), mDriverOp(DRV_NONE),
+    mStandbyCnt(0)
+{
+}
+
+status_t AudioHardware::AudioStreamInALSA::set(
+    AudioHardware* hw, uint32_t devices, int *pFormat,
+    uint32_t *pChannels, uint32_t *pRate, AudioSystem::audio_in_acoustics acoustics)
+{
+    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 = AudioHardware::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("AudioStreamInALSA::set(%d, %d, %u)", *pFormat, *pChannels, *pRate);
+
+    mBufferSize = getBufferSize(*pRate, AudioSystem::popCount(*pChannels));
+    mDevices = devices;
+    mChannels = *pChannels;
+    mChannelCount = AudioSystem::popCount(mChannels);
+    mSampleRate = rate;
+    if (mSampleRate != AUDIO_HW_OUT_SAMPLERATE) {
+        mDownSampler = new AudioHardware::DownSampler(mSampleRate,
+                                                  mChannelCount,
+                                                  AUDIO_HW_IN_PERIOD_SZ,
+                                                  this);
+        status_t status = mDownSampler->initCheck();
+        if (status != NO_ERROR) {
+            delete mDownSampler;
+            LOGW("AudioStreamInALSA::set() downsampler init failed: %d", status);
+            return status;
+        }
+
+        mPcmIn = new int16_t[AUDIO_HW_IN_PERIOD_SZ * mChannelCount];
+    }
+    return NO_ERROR;
+}
+
+AudioHardware::AudioStreamInALSA::~AudioStreamInALSA()
+{
+    standby();
+    if (mDownSampler != NULL) {
+        delete mDownSampler;
+        if (mPcmIn != NULL) {
+            delete[] mPcmIn;
+        }
+    }
+}
+
+ssize_t AudioHardware::AudioStreamInALSA::read(void* buffer, ssize_t bytes)
+{
+    //    LOGV("AudioStreamInALSA::read(%p, %u)", buffer, bytes);
+    status_t status = NO_INIT;
+    int ret;
+
+    if (mHardware == NULL) return NO_INIT;
+
+    { // scope for the lock
+        AutoMutex lock(mLock);
+
+        if (mStandby) {
+            AutoMutex hwLock(mHardware->lock());
+
+            LOGD("AudioHardware pcm capture is exiting standby.");
+            acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioInLock");
+
+            sp<AudioStreamOutALSA> spOut = mHardware->output();
+            while (spOut != 0) {
+                if (!spOut->checkStandby()) {
+                    int cnt = spOut->standbyCnt();
+                    mHardware->lock().unlock();
+                    mLock.unlock();
+                    // Mutex acquisition order is always out -> in -> hw
+                    spOut->lock();
+                    mLock.lock();
+                    mHardware->lock().lock();
+                    // make sure that another thread did not change output state
+                    // while the mutex is released
+                    if ((spOut == mHardware->output()) && (cnt == spOut->standbyCnt())) {
+                        LOGV("AudioStreamInALSA::read() force output standby");
+                        spOut->close_l();
+                        break;
+                    }
+                    spOut->unlock();
+                    spOut = mHardware->output();
+                } else {
+                    spOut.clear();
+                }
+            }
+            // spOut is not 0 here only if the output was active and has been
+            // closed above
+
+            // open output before input
+            if (spOut != 0) {
+                if (spOut->open_l() != NO_ERROR) {
+                    spOut->doStandby_l();
+                }
+                spOut->unlock();
+            }
+
+            open_l();
+
+            if (mPcm == NULL) {
+                release_wake_lock("AudioInLock");
+                goto Error;
+            }
+            mStandby = false;
+        }
+
+
+        if (mDownSampler != NULL) {
+            size_t frames = bytes / frameSize();
+            size_t framesIn = 0;
+            mReadStatus = 0;
+            do {
+                size_t outframes = frames - framesIn;
+                mDownSampler->resample(
+                        (int16_t *)buffer + (framesIn * mChannelCount),
+                        &outframes);
+                framesIn += outframes;
+            } while ((framesIn < frames) && mReadStatus == 0);
+            ret = mReadStatus;
+            bytes = framesIn * frameSize();
+        } else {
+            TRACE_DRIVER_IN(DRV_PCM_READ)
+            ret = pcm_read(mPcm, buffer, bytes);
+            TRACE_DRIVER_OUT
+        }
+
+        if (ret == 0) {
+            return bytes;
+        }
+
+        LOGW("read error: %d", ret);
+        status = ret;
+    }
+
+Error:
+
+    standby();
+
+    // Simulate audio output timing in case of error
+    usleep((((bytes * 1000) / frameSize()) * 1000) / sampleRate());
+
+    return status;
+}
+
+status_t AudioHardware::AudioStreamInALSA::standby()
+{
+    if (mHardware == NULL) return NO_INIT;
+
+    AutoMutex lock(mLock);
+
+    { // scope for AudioHardware lock
+        AutoMutex hwLock(mHardware->lock());
+
+        doStandby_l();
+    }
+    return NO_ERROR;
+}
+
+void AudioHardware::AudioStreamInALSA::doStandby_l()
+{
+    mStandbyCnt++;
+
+    if (!mStandby) {
+        LOGD("AudioHardware pcm capture is going to standby.");
+        release_wake_lock("AudioInLock");
+        mStandby = true;
+    }
+    close_l();
+}
+
+void AudioHardware::AudioStreamInALSA::close_l()
+{
+    if (mMixer) {
+        mHardware->closeMixer_l();
+        mMixer = NULL;
+        mRouteCtl = NULL;
+    }
+
+    if (mPcm) {
+        TRACE_DRIVER_IN(DRV_PCM_CLOSE)
+        pcm_close(mPcm);
+        TRACE_DRIVER_OUT
+        mPcm = NULL;
+    }
+}
+
+status_t AudioHardware::AudioStreamInALSA::open_l()
+{
+    unsigned flags = PCM_IN;
+    if (mChannels == AudioSystem::CHANNEL_IN_MONO) {
+        flags |= PCM_MONO;
+    }
+    flags |= (AUDIO_HW_IN_PERIOD_MULT - 1) << PCM_PERIOD_SZ_SHIFT;
+    flags |= (AUDIO_HW_IN_PERIOD_CNT - PCM_PERIOD_CNT_MIN)
+            << PCM_PERIOD_CNT_SHIFT;
+
+    LOGV("open pcm_in driver");
+    TRACE_DRIVER_IN(DRV_PCM_OPEN)
+    mPcm = pcm_open(flags);
+    TRACE_DRIVER_OUT
+    if (!pcm_ready(mPcm)) {
+        LOGE("cannot open pcm_in driver: %s\n", pcm_error(mPcm));
+        TRACE_DRIVER_IN(DRV_PCM_CLOSE)
+        pcm_close(mPcm);
+        TRACE_DRIVER_OUT
+        mPcm = NULL;
+        return NO_INIT;
+    }
+
+    if (mDownSampler != NULL) {
+        mInPcmInBuf = 0;
+        mDownSampler->reset();
+    }
+
+    mMixer = mHardware->openMixer_l();
+    if (mMixer) {
+        TRACE_DRIVER_IN(DRV_MIXER_GET)
+        mRouteCtl = mixer_get_control(mMixer, "Capture MIC Path", 0);
+        TRACE_DRIVER_OUT
+    }
+
+    if (mHardware->mode() != AudioSystem::MODE_IN_CALL) {
+        const char *route = mHardware->getInputRouteFromDevice(mDevices);
+        LOGV("read() wakeup setting route %s", route);
+        if (mRouteCtl) {
+            TRACE_DRIVER_IN(DRV_MIXER_SEL)
+            mixer_ctl_select(mRouteCtl, route);
+            TRACE_DRIVER_OUT
+        }
+    }
+
+    return NO_ERROR;
+}
+
+status_t AudioHardware::AudioStreamInALSA::dump(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    bool locked = tryLock(mLock);
+    if (!locked) {
+        snprintf(buffer, SIZE, "\n\t\tAudioStreamInALSA maybe deadlocked\n");
+    } else {
+        mLock.unlock();
+    }
+
+    snprintf(buffer, SIZE, "\t\tmHardware: %p\n", mHardware);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmPcm: %p\n", mPcm);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmMixer: %p\n", mMixer);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tStandby %s\n", (mStandby) ? "ON" : "OFF");
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmDevices: 0x%08x\n", mDevices);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmChannels: 0x%08x\n", mChannels);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmSampleRate: %d\n", mSampleRate);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmBufferSize: %d\n", mBufferSize);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmDriverOp: %d\n", mDriverOp);
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+
+    return NO_ERROR;
+}
+
+bool AudioHardware::AudioStreamInALSA::checkStandby()
+{
+    return mStandby;
+}
+
+status_t AudioHardware::AudioStreamInALSA::setParameters(const String8& keyValuePairs)
+{
+    AudioParameter param = AudioParameter(keyValuePairs);
+    status_t status = NO_ERROR;
+    int value;
+    String8 source;
+
+    LOGD("AudioStreamInALSA::setParameters() %s", keyValuePairs.string());
+
+    if (mHardware == NULL) return NO_INIT;
+
+    {
+        AutoMutex lock(mLock);
+
+        if (param.get(String8(INPUT_SOURCE_KEY), source) == NO_ERROR) {
+            AutoMutex hwLock(mHardware->lock());
+
+            mHardware->openMixer_l();
+            mHardware->setInputSource_l(source);
+            mHardware->closeMixer_l();
+
+            param.remove(String8(INPUT_SOURCE_KEY));
+        }
+
+        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR)
+        {
+            if (value != 0) {
+                AutoMutex hwLock(mHardware->lock());
+
+                if (mDevices != (uint32_t)value) {
+                    mDevices = (uint32_t)value;
+                    if (mHardware->mode() != AudioSystem::MODE_IN_CALL) {
+                        doStandby_l();
+                    }
+                }
+            }
+            param.remove(String8(AudioParameter::keyRouting));
+        }
+    }
+
+
+    if (param.size()) {
+        status = BAD_VALUE;
+    }
+
+    return status;
+
+}
+
+String8 AudioHardware::AudioStreamInALSA::getParameters(const String8& keys)
+{
+    AudioParameter param = AudioParameter(keys);
+    String8 value;
+    String8 key = String8(AudioParameter::keyRouting);
+
+    if (param.get(key, value) == NO_ERROR) {
+        param.addInt(key, (int)mDevices);
+    }
+
+    LOGV("AudioStreamInALSA::getParameters() %s", param.toString().string());
+    return param.toString();
+}
+
+status_t AudioHardware::AudioStreamInALSA::getNextBuffer(AudioHardware::BufferProvider::Buffer* buffer)
+{
+    if (mPcm == NULL) {
+        buffer->raw = NULL;
+        buffer->frameCount = 0;
+        mReadStatus = NO_INIT;
+        return NO_INIT;
+    }
+
+    if (mInPcmInBuf == 0) {
+        TRACE_DRIVER_IN(DRV_PCM_READ)
+        mReadStatus = pcm_read(mPcm,(void*) mPcmIn, AUDIO_HW_IN_PERIOD_SZ * frameSize());
+        TRACE_DRIVER_OUT
+        if (mReadStatus != 0) {
+            buffer->raw = NULL;
+            buffer->frameCount = 0;
+            return mReadStatus;
+        }
+        mInPcmInBuf = AUDIO_HW_IN_PERIOD_SZ;
+    }
+
+    buffer->frameCount = (buffer->frameCount > mInPcmInBuf) ? mInPcmInBuf : buffer->frameCount;
+    buffer->i16 = mPcmIn + (AUDIO_HW_IN_PERIOD_SZ - mInPcmInBuf) * mChannelCount;
+
+    return mReadStatus;
+}
+
+void AudioHardware::AudioStreamInALSA::releaseBuffer(Buffer* buffer)
+{
+    mInPcmInBuf -= buffer->frameCount;
+}
+
+size_t AudioHardware::AudioStreamInALSA::getBufferSize(uint32_t sampleRate, int channelCount)
+{
+    size_t ratio;
+
+    switch (sampleRate) {
+    case 8000:
+    case 11025:
+        ratio = 4;
+        break;
+    case 16000:
+    case 22050:
+        ratio = 2;
+        break;
+    case 44100:
+    default:
+        ratio = 1;
+        break;
+    }
+
+    return (AUDIO_HW_IN_PERIOD_SZ*channelCount*sizeof(int16_t)) / ratio ;
+}
+
+//------------------------------------------------------------------------------
+//  DownSampler
+//------------------------------------------------------------------------------
+
+/*
+ * 2.30 fixed point FIR filter coefficients for conversion 44100 -> 22050.
+ * (Works equivalently for 22010 -> 11025 or any other halving, of course.)
+ *
+ * Transition band from about 18 kHz, passband ripple < 0.1 dB,
+ * stopband ripple at about -55 dB, linear phase.
+ *
+ * Design and display in MATLAB or Octave using:
+ *
+ * filter = fir1(19, 0.5); filter = round(filter * 2**30); freqz(filter * 2**-30);
+ */
+static const int32_t filter_22khz_coeff[] = {
+    2089257, 2898328, -5820678, -10484531,
+    19038724, 30542725, -50469415, -81505260,
+    152544464, 478517512, 478517512, 152544464,
+    -81505260, -50469415, 30542725, 19038724,
+    -10484531, -5820678, 2898328, 2089257,
+};
+#define NUM_COEFF_22KHZ (sizeof(filter_22khz_coeff) / sizeof(filter_22khz_coeff[0]))
+#define OVERLAP_22KHZ (NUM_COEFF_22KHZ - 2)
+
+/*
+ * Convolution of signals A and reverse(B). (In our case, the filter response
+ * is symmetric, so the reversing doesn't matter.)
+ * A is taken to be in 0.16 fixed-point, and B is taken to be in 2.30 fixed-point.
+ * The answer will be in 16.16 fixed-point, unclipped.
+ *
+ * This function would probably be the prime candidate for SIMD conversion if
+ * you want more speed.
+ */
+int32_t fir_convolve(const int16_t* a, const int32_t* b, int num_samples)
+{
+        int32_t sum = 1 << 13;
+        for (int i = 0; i < num_samples; ++i) {
+                sum += a[i] * (b[i] >> 16);
+        }
+        return sum >> 14;
+}
+
+/* Clip from 16.16 fixed-point to 0.16 fixed-point. */
+int16_t clip(int32_t x)
+{
+    if (x < -32768) {
+        return -32768;
+    } else if (x > 32767) {
+        return 32767;
+    } else {
+        return x;
+    }
+}
+
+/*
+ * Convert a chunk from 44 kHz to 22 kHz. Will update num_samples_in and num_samples_out
+ * accordingly, since it may leave input samples in the buffer due to overlap.
+ *
+ * Input and output are taken to be in 0.16 fixed-point.
+ */
+void resample_2_1(int16_t* input, int16_t* output, int* num_samples_in, int* num_samples_out)
+{
+    if (*num_samples_in < (int)NUM_COEFF_22KHZ) {
+        *num_samples_out = 0;
+        return;
+    }
+
+    int odd_smp = *num_samples_in & 0x1;
+    int num_samples = *num_samples_in - odd_smp - OVERLAP_22KHZ;
+
+    for (int i = 0; i < num_samples; i += 2) {
+            output[i / 2] = clip(fir_convolve(input + i, filter_22khz_coeff, NUM_COEFF_22KHZ));
+    }
+
+    memmove(input, input + num_samples, (OVERLAP_22KHZ + odd_smp) * sizeof(*input));
+    *num_samples_out = num_samples / 2;
+    *num_samples_in = OVERLAP_22KHZ + odd_smp;
+}
+
+/*
+ * 2.30 fixed point FIR filter coefficients for conversion 22050 -> 16000,
+ * or 11025 -> 8000.
+ *
+ * Transition band from about 14 kHz, passband ripple < 0.1 dB,
+ * stopband ripple at about -50 dB, linear phase.
+ *
+ * Design and display in MATLAB or Octave using:
+ *
+ * filter = fir1(23, 16000 / 22050); filter = round(filter * 2**30); freqz(filter * 2**-30);
+ */
+static const int32_t filter_16khz_coeff[] = {
+    2057290, -2973608, 1880478, 4362037,
+    -14639744, 18523609, -1609189, -38502470,
+    78073125, -68353935, -59103896, 617555440,
+    617555440, -59103896, -68353935, 78073125,
+    -38502470, -1609189, 18523609, -14639744,
+    4362037, 1880478, -2973608, 2057290,
+};
+#define NUM_COEFF_16KHZ (sizeof(filter_16khz_coeff) / sizeof(filter_16khz_coeff[0]))
+#define OVERLAP_16KHZ (NUM_COEFF_16KHZ - 1)
+
+/*
+ * Convert a chunk from 22 kHz to 16 kHz. Will update num_samples_in and
+ * num_samples_out accordingly, since it may leave input samples in the buffer
+ * due to overlap.
+ *
+ * This implementation is rather ad-hoc; it first low-pass filters the data
+ * into a temporary buffer, and then converts chunks of 441 input samples at a
+ * time into 320 output samples by simple linear interpolation. A better
+ * implementation would use a polyphase filter bank to do these two operations
+ * in one step.
+ *
+ * Input and output are taken to be in 0.16 fixed-point.
+ */
+
+#define RESAMPLE_16KHZ_SAMPLES_IN 441
+#define RESAMPLE_16KHZ_SAMPLES_OUT 320
+
+void resample_441_320(int16_t* input, int16_t* output, int* num_samples_in, int* num_samples_out)
+{
+    const int num_blocks = (*num_samples_in - OVERLAP_16KHZ) / RESAMPLE_16KHZ_SAMPLES_IN;
+    if (num_blocks < 1) {
+        *num_samples_out = 0;
+        return;
+    }
+
+    for (int i = 0; i < num_blocks; ++i) {
+        uint32_t tmp[RESAMPLE_16KHZ_SAMPLES_IN];
+        for (int j = 0; j < RESAMPLE_16KHZ_SAMPLES_IN; ++j) {
+            tmp[j] = fir_convolve(input + i * RESAMPLE_16KHZ_SAMPLES_IN + j,
+                          filter_16khz_coeff,
+                          NUM_COEFF_16KHZ);
+        }
+
+        const float step_float = (float)RESAMPLE_16KHZ_SAMPLES_IN / (float)RESAMPLE_16KHZ_SAMPLES_OUT;
+
+        uint32_t in_sample_num = 0;   // 16.16 fixed point
+        const uint32_t step = (uint32_t)(step_float * 65536.0f + 0.5f);  // 16.16 fixed point
+        for (int j = 0; j < RESAMPLE_16KHZ_SAMPLES_OUT; ++j, in_sample_num += step) {
+            const uint32_t whole = in_sample_num >> 16;
+            const uint32_t frac = (in_sample_num & 0xffff);  // 0.16 fixed point
+            const int32_t s1 = tmp[whole];
+            const int32_t s2 = tmp[whole + 1];
+            *output++ = clip(s1 + (((s2 - s1) * (int32_t)frac) >> 16));
+        }
+    }
+
+    const int samples_consumed = num_blocks * RESAMPLE_16KHZ_SAMPLES_IN;
+    memmove(input, input + samples_consumed, (*num_samples_in - samples_consumed) * sizeof(*input));
+    *num_samples_in -= samples_consumed;
+    *num_samples_out = RESAMPLE_16KHZ_SAMPLES_OUT * num_blocks;
+}
+
+
+AudioHardware::DownSampler::DownSampler(uint32_t outSampleRate,
+                                    uint32_t channelCount,
+                                    uint32_t frameCount,
+                                    AudioHardware::BufferProvider* provider)
+    :  mStatus(NO_INIT), mProvider(provider), mSampleRate(outSampleRate),
+       mChannelCount(channelCount), mFrameCount(frameCount),
+       mInLeft(NULL), mInRight(NULL), mTmpLeft(NULL), mTmpRight(NULL),
+       mTmp2Left(NULL), mTmp2Right(NULL), mOutLeft(NULL), mOutRight(NULL)
+
+{
+    LOGV("AudioHardware::DownSampler() cstor %p SR %d channels %d frames %d",
+         this, mSampleRate, mChannelCount, mFrameCount);
+
+    if (mSampleRate != 8000 && mSampleRate != 11025 && mSampleRate != 16000 &&
+            mSampleRate != 22050) {
+        LOGW("AudioHardware::DownSampler cstor: bad sampling rate: %d", mSampleRate);
+        return;
+    }
+
+    mInLeft = new int16_t[mFrameCount];
+    mInRight = new int16_t[mFrameCount];
+    mTmpLeft = new int16_t[mFrameCount];
+    mTmpRight = new int16_t[mFrameCount];
+    mTmp2Left = new int16_t[mFrameCount];
+    mTmp2Right = new int16_t[mFrameCount];
+    mOutLeft = new int16_t[mFrameCount];
+    mOutRight = new int16_t[mFrameCount];
+
+    mStatus = NO_ERROR;
+}
+
+AudioHardware::DownSampler::~DownSampler()
+{
+    if (mInLeft) delete[] mInLeft;
+    if (mInRight) delete[] mInRight;
+    if (mTmpLeft) delete[] mTmpLeft;
+    if (mTmpRight) delete[] mTmpRight;
+    if (mTmp2Left) delete[] mTmp2Left;
+    if (mTmp2Right) delete[] mTmp2Right;
+    if (mOutLeft) delete[] mOutLeft;
+    if (mOutRight) delete[] mOutRight;
+}
+
+void AudioHardware::DownSampler::reset()
+{
+    mInInBuf = 0;
+    mInTmpBuf = 0;
+    mInTmp2Buf = 0;
+    mOutBufPos = 0;
+    mInOutBuf = 0;
+}
+
+
+int AudioHardware::DownSampler::resample(int16_t* out, size_t *outFrameCount)
+{
+    if (mStatus != NO_ERROR) {
+        return mStatus;
+    }
+
+    if (out == NULL || outFrameCount == NULL) {
+        return BAD_VALUE;
+    }
+
+    int16_t *outLeft = mTmp2Left;
+    int16_t *outRight = mTmp2Left;
+    if (mSampleRate == 22050) {
+        outLeft = mTmpLeft;
+        outRight = mTmpRight;
+    } else if (mSampleRate == 8000){
+        outLeft = mOutLeft;
+        outRight = mOutRight;
+    }
+
+    int outFrames = 0;
+    int remaingFrames = *outFrameCount;
+
+    if (mInOutBuf) {
+        int frames = (remaingFrames > mInOutBuf) ? mInOutBuf : remaingFrames;
+
+        for (int i = 0; i < frames; ++i) {
+            out[i] = outLeft[mOutBufPos + i];
+        }
+        if (mChannelCount == 2) {
+            for (int i = 0; i < frames; ++i) {
+                out[i * 2] = outLeft[mOutBufPos + i];
+                out[i * 2 + 1] = outRight[mOutBufPos + i];
+            }
+        }
+        remaingFrames -= frames;
+        mInOutBuf -= frames;
+        mOutBufPos += frames;
+        outFrames += frames;
+    }
+
+    while (remaingFrames) {
+        LOGW_IF((mInOutBuf != 0), "mInOutBuf should be 0 here");
+
+        AudioHardware::BufferProvider::Buffer buf;
+        buf.frameCount =  mFrameCount - mInInBuf;
+        int ret = mProvider->getNextBuffer(&buf);
+        if (buf.raw == NULL) {
+            *outFrameCount = outFrames;
+            return ret;
+        }
+
+        for (size_t i = 0; i < buf.frameCount; ++i) {
+            mInLeft[i + mInInBuf] = buf.i16[i];
+        }
+        if (mChannelCount == 2) {
+            for (size_t i = 0; i < buf.frameCount; ++i) {
+                mInLeft[i + mInInBuf] = buf.i16[i * 2];
+                mInRight[i + mInInBuf] = buf.i16[i * 2 + 1];
+            }
+        }
+        mInInBuf += buf.frameCount;
+        mProvider->releaseBuffer(&buf);
+
+        /* 44010 -> 22050 */
+        {
+            int samples_in_left = mInInBuf;
+            int samples_out_left;
+            resample_2_1(mInLeft, mTmpLeft + mInTmpBuf, &samples_in_left, &samples_out_left);
+
+            if (mChannelCount == 2) {
+                int samples_in_right = mInInBuf;
+                int samples_out_right;
+                resample_2_1(mInRight, mTmpRight + mInTmpBuf, &samples_in_right, &samples_out_right);
+            }
+
+            mInInBuf = samples_in_left;
+            mInTmpBuf += samples_out_left;
+            mInOutBuf = samples_out_left;
+        }
+
+        if (mSampleRate == 11025 || mSampleRate == 8000) {
+            /* 22050 - > 11025 */
+            int samples_in_left = mInTmpBuf;
+            int samples_out_left;
+            resample_2_1(mTmpLeft, mTmp2Left + mInTmp2Buf, &samples_in_left, &samples_out_left);
+
+            if (mChannelCount == 2) {
+                int samples_in_right = mInTmpBuf;
+                int samples_out_right;
+                resample_2_1(mTmpRight, mTmp2Right + mInTmp2Buf, &samples_in_right, &samples_out_right);
+            }
+
+
+            mInTmpBuf = samples_in_left;
+            mInTmp2Buf += samples_out_left;
+            mInOutBuf = samples_out_left;
+
+            if (mSampleRate == 8000) {
+                /* 11025 -> 8000*/
+                int samples_in_left = mInTmp2Buf;
+                int samples_out_left;
+                resample_441_320(mTmp2Left, mOutLeft, &samples_in_left, &samples_out_left);
+
+                if (mChannelCount == 2) {
+                    int samples_in_right = mInTmp2Buf;
+                    int samples_out_right;
+                    resample_441_320(mTmp2Right, mOutRight, &samples_in_right, &samples_out_right);
+                }
+
+                mInTmp2Buf = samples_in_left;
+                mInOutBuf = samples_out_left;
+            } else {
+                mInTmp2Buf = 0;
+            }
+
+        } else if (mSampleRate == 16000) {
+            /* 22050 -> 16000*/
+            int samples_in_left = mInTmpBuf;
+            int samples_out_left;
+            resample_441_320(mTmpLeft, mTmp2Left, &samples_in_left, &samples_out_left);
+
+            if (mChannelCount == 2) {
+                int samples_in_right = mInTmpBuf;
+                int samples_out_right;
+                resample_441_320(mTmpRight, mTmp2Right, &samples_in_right, &samples_out_right);
+            }
+
+            mInTmpBuf = samples_in_left;
+            mInOutBuf = samples_out_left;
+        } else {
+            mInTmpBuf = 0;
+        }
+
+        int frames = (remaingFrames > mInOutBuf) ? mInOutBuf : remaingFrames;
+
+        for (int i = 0; i < frames; ++i) {
+            out[outFrames + i] = outLeft[i];
+        }
+        if (mChannelCount == 2) {
+            for (int i = 0; i < frames; ++i) {
+                out[(outFrames + i) * 2] = outLeft[i];
+                out[(outFrames + i) * 2 + 1] = outRight[i];
+            }
+        }
+        remaingFrames -= frames;
+        outFrames += frames;
+        mOutBufPos = frames;
+        mInOutBuf -= frames;
+    }
+
+    return 0;
+}
+
+
+
+
+
+
+
+//------------------------------------------------------------------------------
+//  Factory
+//------------------------------------------------------------------------------
+
+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..1379495
--- /dev/null
+++ b/libaudio/AudioHardware.h
@@ -0,0 +1,348 @@
+/*
+** 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>
+
+#include "secril-client.h"
+
+extern "C" {
+    struct pcm;
+    struct mixer;
+    struct mixer_ctl;
+};
+
+namespace android {
+
+// TODO: determine actual audio DSP and hardware latency
+// Additionnal latency introduced by audio DSP and hardware in ms
+#define AUDIO_HW_OUT_LATENCY_MS 0
+// Default audio output sample rate
+#define AUDIO_HW_OUT_SAMPLERATE 44100
+// Default audio output channel mask
+#define AUDIO_HW_OUT_CHANNELS (AudioSystem::CHANNEL_OUT_STEREO)
+// Default audio output sample format
+#define AUDIO_HW_OUT_FORMAT (AudioSystem::PCM_16_BIT)
+// Kernel pcm out buffer size in frames at 44.1kHz
+#define AUDIO_HW_OUT_PERIOD_MULT 8 // (8 * 128 = 1024 frames)
+#define AUDIO_HW_OUT_PERIOD_SZ (PCM_PERIOD_SZ_MIN * AUDIO_HW_OUT_PERIOD_MULT)
+#define AUDIO_HW_OUT_PERIOD_CNT 4
+// Default audio output buffer size in bytes
+#define AUDIO_HW_OUT_PERIOD_BYTES (AUDIO_HW_OUT_PERIOD_SZ * 2 * sizeof(int16_t))
+
+// Default audio input sample rate
+#define AUDIO_HW_IN_SAMPLERATE 8000
+// Default audio input channel mask
+#define AUDIO_HW_IN_CHANNELS (AudioSystem::CHANNEL_IN_MONO)
+// Default audio input sample format
+#define AUDIO_HW_IN_FORMAT (AudioSystem::PCM_16_BIT)
+// Number of buffers in audio driver for input
+#define AUDIO_HW_NUM_IN_BUF 2
+// Kernel pcm in buffer size in frames at 44.1kHz (before resampling)
+#define AUDIO_HW_IN_PERIOD_MULT 16  // (16 * 128 = 2048 frames)
+#define AUDIO_HW_IN_PERIOD_SZ (PCM_PERIOD_SZ_MIN * AUDIO_HW_IN_PERIOD_MULT)
+#define AUDIO_HW_IN_PERIOD_CNT 2
+// Default audio input buffer size in bytes (8kHz mono)
+#define AUDIO_HW_IN_PERIOD_BYTES ((AUDIO_HW_IN_PERIOD_SZ*sizeof(int16_t))/8)
+
+#define INPUT_SOURCE_KEY "Input Source"
+
+class AudioHardware : public AudioHardwareBase
+{
+    class AudioStreamOutALSA;
+    class AudioStreamInALSA;
+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);
+
+    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);
+
+    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);
+
+            int  mode() { return mMode; }
+            const char *getOutputRouteFromDevice(uint32_t device);
+            const char *getInputRouteFromDevice(uint32_t device);
+            const char *getVoiceRouteFromDevice(uint32_t device);
+
+            status_t setIncallPath_l(uint32_t device);
+
+            status_t setInputSource_l(String8 source);
+
+    static uint32_t    getInputSampleRate(uint32_t sampleRate);
+           sp <AudioStreamInALSA> getActiveInput_l();
+
+           Mutex& lock() { return mLock; }
+
+           struct pcm *openPcmOut_l();
+           void closePcmOut_l();
+
+           struct mixer *openMixer_l();
+           void closeMixer_l();
+
+           sp <AudioStreamOutALSA>  output() { return mOutput; }
+
+protected:
+    virtual status_t dump(int fd, const Vector<String16>& args);
+
+private:
+
+    bool            mInit;
+    bool            mMicMute;
+    sp <AudioStreamOutALSA>                 mOutput;
+    SortedVector < sp<AudioStreamInALSA> >   mInputs;
+    Mutex           mLock;
+    struct pcm*     mPcm;
+    struct mixer*   mMixer;
+    uint32_t        mPcmOpenCnt;
+    uint32_t        mMixerOpenCnt;
+    bool            mInCallAudioMode;
+
+    String8         mInputSource;
+    bool            mBluetoothNrec;
+    void*           mSecRilLibHandle;
+    HRilClient      mRilClient;
+    bool            mActivatedCP;
+    HRilClient      (*openClientRILD)  (void);
+    int             (*disconnectRILD)  (HRilClient);
+    int             (*closeClientRILD) (HRilClient);
+    int             (*isConnectedRILD) (HRilClient);
+    int             (*connectRILD)     (HRilClient);
+    int             (*setCallVolume)   (HRilClient, SoundType, int);
+    int             (*setCallAudioPath)(HRilClient, AudioPath);
+    int             (*setCallClockSync)(HRilClient, SoundClockCondition);
+    void            loadRILD(void);
+    status_t        connectRILDIfRequired(void);
+
+    //  trace driver operations for dump
+    int             mDriverOp;
+
+    static uint32_t         checkInputSampleRate(uint32_t sampleRate);
+    static const uint32_t   inputSamplingRates[];
+
+    class AudioStreamOutALSA : public AudioStreamOut, public RefBase
+    {
+    public:
+        AudioStreamOutALSA();
+        virtual ~AudioStreamOutALSA();
+        status_t set(AudioHardware* mHardware,
+                     uint32_t devices,
+                     int *pFormat,
+                     uint32_t *pChannels,
+                     uint32_t *pRate);
+        virtual uint32_t sampleRate()
+            const { return mSampleRate; }
+        virtual size_t bufferSize()
+            const { return mBufferSize; }
+        virtual uint32_t channels()
+            const { return mChannels; }
+        virtual int format()
+            const { return AUDIO_HW_OUT_FORMAT; }
+        virtual uint32_t latency()
+            const { return (1000 * AUDIO_HW_OUT_PERIOD_CNT *
+                            (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();
+                bool checkStandby();
+
+        virtual status_t dump(int fd, const Vector<String16>& args);
+        virtual status_t setParameters(const String8& keyValuePairs);
+        virtual String8 getParameters(const String8& keys);
+        uint32_t device() { return mDevices; }
+        virtual status_t getRenderPosition(uint32_t *dspFrames);
+
+                void doStandby_l();
+                void close_l();
+                status_t open_l();
+                int standbyCnt() { return mStandbyCnt; }
+
+                void lock() { mLock.lock(); }
+                void unlock() { mLock.unlock(); }
+
+    private:
+
+        Mutex mLock;
+        AudioHardware* mHardware;
+        struct pcm *mPcm;
+        struct mixer *mMixer;
+        struct mixer_ctl *mRouteCtl;
+        const char *next_route;
+        bool mStandby;
+        uint32_t mDevices;
+        uint32_t mChannels;
+        uint32_t mSampleRate;
+        size_t mBufferSize;
+        //  trace driver operations for dump
+        int mDriverOp;
+        int mStandbyCnt;
+    };
+
+    class DownSampler;
+
+    class BufferProvider
+    {
+    public:
+
+        struct Buffer {
+            union {
+                void*       raw;
+                short*      i16;
+                int8_t*     i8;
+            };
+            size_t frameCount;
+        };
+
+        virtual ~BufferProvider() {}
+
+        virtual status_t getNextBuffer(Buffer* buffer) = 0;
+        virtual void releaseBuffer(Buffer* buffer) = 0;
+    };
+
+    class DownSampler {
+    public:
+        DownSampler(uint32_t outSampleRate,
+                  uint32_t channelCount,
+                  uint32_t frameCount,
+                  BufferProvider* provider);
+
+        virtual ~DownSampler();
+
+                void reset();
+                status_t initCheck() { return mStatus; }
+                int resample(int16_t* out, size_t *outFrameCount);
+
+    private:
+        status_t    mStatus;
+        BufferProvider* mProvider;
+        uint32_t mSampleRate;
+        uint32_t mChannelCount;
+        uint32_t mFrameCount;
+        int16_t *mInLeft;
+        int16_t *mInRight;
+        int16_t *mTmpLeft;
+        int16_t *mTmpRight;
+        int16_t *mTmp2Left;
+        int16_t *mTmp2Right;
+        int16_t *mOutLeft;
+        int16_t *mOutRight;
+        int mInInBuf;
+        int mInTmpBuf;
+        int mInTmp2Buf;
+        int mOutBufPos;
+        int mInOutBuf;
+    };
+
+
+    class AudioStreamInALSA : public AudioStreamIn, public BufferProvider, public RefBase
+    {
+
+     public:
+                    AudioStreamInALSA();
+        virtual     ~AudioStreamInALSA();
+        status_t    set(AudioHardware* hw,
+                    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 AUDIO_HW_IN_FORMAT; }
+        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();
+                bool checkStandby();
+        virtual status_t setParameters(const String8& keyValuePairs);
+        virtual String8 getParameters(const String8& keys);
+        virtual unsigned int getInputFramesLost() const { return 0; }
+                uint32_t device() { return mDevices; }
+                void doStandby_l();
+                void close_l();
+                status_t open_l();
+                int standbyCnt() { return mStandbyCnt; }
+
+        static size_t getBufferSize(uint32_t sampleRate, int channelCount);
+
+        // BufferProvider
+        virtual status_t getNextBuffer(BufferProvider::Buffer* buffer);
+        virtual void releaseBuffer(BufferProvider::Buffer* buffer);
+
+        void lock() { mLock.lock(); }
+        void unlock() { mLock.unlock(); }
+
+    private:
+        Mutex mLock;
+        AudioHardware* mHardware;
+        struct pcm *mPcm;
+        struct mixer *mMixer;
+        struct mixer_ctl *mRouteCtl;
+        const char *next_route;
+        bool mStandby;
+        uint32_t mDevices;
+        uint32_t mChannels;
+        uint32_t mChannelCount;
+        uint32_t mSampleRate;
+        size_t mBufferSize;
+        DownSampler *mDownSampler;
+        status_t mReadStatus;
+        size_t mInPcmInBuf;
+        int16_t *mPcmIn;
+        //  trace driver operations for dump
+        int mDriverOp;
+        int mStandbyCnt;
+    };
+
+};
+
+}; // namespace android
+
+#endif
diff --git a/libaudio/AudioHardwareALSA.cpp b/libaudio/AudioHardwareALSA.cpp
deleted file mode 100755
index 45223f3..0000000
--- a/libaudio/AudioHardwareALSA.cpp
+++ /dev/null
@@ -1,2654 +0,0 @@
-/* AudioHardwareALSA.cpp
- **
- ** Copyright 2008-2009 Wind River Systems
- **
- ** 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 <errno.h>
-#include <stdarg.h>
-#include <stdint.h>
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <fcntl.h>
-#include <stdlib.h>
-#include <unistd.h>
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "AudioHardwareALSA"
-#include <utils/Log.h>
-#include <utils/String8.h>
-
-#include <cutils/properties.h>
-#include <media/AudioRecord.h>
-#include <hardware_legacy/power.h>
-
-#include <alsa/asoundlib.h>
-#include "AudioHardwareALSA.h"
-// #define READ_FRAME_SIZE 2080
-// #define READ_FRAME_SIZE_STANDARD 4160
-
-#include <dlfcn.h>
-
-#define SND_MIXER_VOL_RANGE_MIN  (0)
-#define SND_MIXER_VOL_RANGE_MAX  (100)
-
-#define ALSA_NAME_MAX 128
-
-#define ALSA_STRCAT(x,y) \
-    if (strlen(x) + strlen(y) < ALSA_NAME_MAX) \
-        strcat(x, y);
-
-extern "C"
-{
-    extern int ffs(int i);
-
-    //
-    // Make sure this prototype is consistent with what's in
-    // external/libasound/alsa-lib-1.0.16/src/pcm/pcm_null.c!
-    //
-    extern int snd_pcm_null_open(snd_pcm_t **pcmp,
-                                 const char *name,
-                                 snd_pcm_stream_t stream,
-                                 int mode);
-
-    //
-    // Function for dlsym() to look up for creating a new AudioHardwareInterface.
-    //
-    android::AudioHardwareInterface *createAudioHardware(void) {
-        return new android::AudioHardwareALSA();
-    }
-}         // extern "C"
-
-namespace android
-{
-
-typedef AudioSystem::audio_devices audio_routes;
-#define ROUTE_ALL            AudioSystem::DEVICE_OUT_ALL
-#define ROUTE_EARPIECE       AudioSystem::DEVICE_OUT_EARPIECE
-#define ROUTE_HEADSET        AudioSystem::DEVICE_OUT_WIRED_HEADSET
-#define ROUTE_HEADPHONE      AudioSystem::DEVICE_OUT_WIRED_HEADPHONE
-#define ROUTE_SPEAKER        AudioSystem::DEVICE_OUT_SPEAKER
-#define ROUTE_BLUETOOTH_SCO  AudioSystem::DEVICE_OUT_BLUETOOTH_SCO
-#define ROUTE_BLUETOOTH_SCO_HEADSET  AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET
-#define ROUTE_BLUETOOTH_SCO_CARKIT  AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT
-#define ROUTE_BLUETOOTH_A2DP AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP
-#define ROUTE_BLUETOOTH_A2DP_HEADPHONES AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES
-#define ROUTE_BLUETOOTH_A2DP_SPEAKER AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
-
-// ----------------------------------------------------------------------------
-
-
-static const char _nullALSADeviceName[] = "NULL_Device";
-
-static void ALSAErrorHandler(const char *file,
-                             int line,
-                             const char *function,
-                             int err,
-                             const char *fmt,
-                             ...)
-{
-    char buf[BUFSIZ];
-    va_list arg;
-    int l;
-
-    va_start(arg, fmt);
-    l = snprintf(buf, BUFSIZ, "%s:%i:(%s) ", file, line, function);
-    vsnprintf(buf + l, BUFSIZ - l, fmt, arg);
-    buf[BUFSIZ-1] = '\0';
-    LOGE("ALSALib %s.", buf);
-    va_end(arg);
-}
-
-// ----------------------------------------------------------------------------
-
-/* The following table(s) need to match in order of the route bits
- */
-static const char *deviceSuffix[] = {
-    // output devices
-    /* ROUTE_EARPIECE       */ "_Earpiece",
-    /* ROUTE_SPEAKER        */ "_Speaker",
-    /* ROUTE_HEADSET        */ "_Headset",
-    /* ROUTE_HEADPHONE      */ "_Headphone",
-    /* ROUTE_BLUETOOTH_SCO  */ "_Bluetooth",
-    /* ROUTE_BLUETOOTH_SCO_HEADSET */ "_Bluetooth",
-    /* ROUTE_BLUETOOTH_SCO_CARKIT  */ "_Bluetooth", //"_Bluetooth_Carkit"
-    /* ROUTE_BLUETOOTH_A2DP        */ "_Bluetooth", //"_Bluetooth-A2DP"
-    /* ROUTE_BLUETOOTH_A2DP_HEADPHONES */ "_Bluetooth", //"_Bluetooth-A2DP_HeadPhone"
-    /* ROUTE_BLUETOOTH_A2DP_SPEAKER    */ "_Bluetooth",     // "_Bluetooth-A2DP_Speaker"
-    /* ROUTE_AUX_DIGITAL */ "_AuxDigital",
-    /* ROUTE_TV_OUT */ "_TvOut",
-    /* ROUTE_AUX_DIGITAL */ "_ExtraDockSpeaker",
-    /* ROUTE_NULL */ "_Null",
-    /* ROUTE_NULL */ "_Null",
-    /* ROUTE_DEFAULT */ "_OutDefault",
-
-    // input devices
-    /* ROUTE_COMMUNICATION         */ "_Communication",
-    /* ROUTE_AMBIENT               */ "_Ambient",
-    /* ROUTE_BUILTIN_MIC           */ "_Speaker",
-    /* ROUTE_BLUETOOTH_SCO_HEADSET */ "_Bluetooth",
-    /* ROUTE_WIRED_HEADSET         */ "_Headset",
-    /* ROUTE_AUX_DIGITAL           */ "_AuxDigital",
-    /* ROUTE_VOICE_CALL            */ "_VoiceCall",
-    /* ROUTE_BACK_MIC              */ "_BackMic",
-    /* ROUTE_IN_DEFAULT            */ "_InDefault",
-};
-
-static const int deviceSuffixLen = (sizeof(deviceSuffix) / sizeof(char *));
-
-struct mixer_info_t;
-
-struct alsa_properties_t
-{
-    const audio_routes  routes;
-    const char         *propName;
-    const char         *propDefault;
-    mixer_info_t       *mInfo;
-};
-
-static alsa_properties_t masterPlaybackProp = {
-    ROUTE_ALL, "alsa.mixer.playback.master", "PCM", NULL
-};
-
-static alsa_properties_t masterCaptureProp = {
-    ROUTE_ALL, "alsa.mixer.capture.master", "Capture", NULL
-};
-
-static alsa_properties_t
-mixerMasterProp[SND_PCM_STREAM_LAST+1] = {
-    { ROUTE_ALL, "alsa.mixer.playback.master",  "PCM",     NULL},
-    { ROUTE_ALL, "alsa.mixer.capture.master",   "Capture", NULL}
-};
-
-static alsa_properties_t
-mixerProp[][SND_PCM_STREAM_LAST+1] = {
-    {
-        {ROUTE_EARPIECE,       "alsa.mixer.playback.earpiece",       "Earpiece", NULL},
-        {ROUTE_EARPIECE,       "alsa.mixer.capture.earpiece",        "Capture",  NULL}
-    },
-    {
-        {ROUTE_SPEAKER,        "alsa.mixer.playback.speaker",        "Speaker", NULL},
-        {ROUTE_SPEAKER,        "alsa.mixer.capture.speaker",         "",        NULL}
-    },
-    {
-        {ROUTE_BLUETOOTH_SCO,  "alsa.mixer.playback.bluetooth.sco",  "Bluetooth",         NULL},
-        {ROUTE_BLUETOOTH_SCO,  "alsa.mixer.capture.bluetooth.sco",   "Bluetooth Capture", NULL}
-    },
-    {
-        {ROUTE_HEADSET,        "alsa.mixer.playback.headset",        "Headphone", NULL},
-        {ROUTE_HEADSET,        "alsa.mixer.capture.headset",         "Capture",   NULL}
-    },
-    {
-        {ROUTE_BLUETOOTH_A2DP, "alsa.mixer.playback.bluetooth.a2dp", "Bluetooth A2DP",         NULL},
-        {ROUTE_BLUETOOTH_A2DP, "alsa.mixer.capture.bluetooth.a2dp",  "Bluetooth A2DP Capture", NULL}
-    },
-    {
-        {static_cast<audio_routes>(0), NULL, NULL, NULL},
-        {static_cast<audio_routes>(0), NULL, NULL, NULL}
-    }
-};
-
-const uint32_t AudioHardwareALSA::inputSamplingRates[] = {
-        8000, 11025, 16000, 22050, 44100
-};
-
-// ----------------------------------------------------------------------------
-
-AudioHardwareALSA::AudioHardwareALSA() :
-    mOutput(0),
-    mInput(0),
-    mSecRilLibHandle(NULL),
-    mRilClient(0),
-    mVrModeEnabled(false),
-    mActivatedCP(false),
-    mBluetoothECOff(false)
-{
-    snd_lib_error_set_handler(&ALSAErrorHandler);
-    mMixer = new ALSAMixer;
-
-    loadRILD();
-}
-
-AudioHardwareALSA::~AudioHardwareALSA()
-{
-    if (mOutput) delete mOutput;
-    if (mInput) delete mInput;
-    if (mMixer) delete mMixer;
-
-    if (mSecRilLibHandle) {
-        if (disconnectRILD(mRilClient) != RIL_CLIENT_ERR_SUCCESS)
-            LOGE("Disconnect_RILD() error");
-
-        if (closeClientRILD(mRilClient) != RIL_CLIENT_ERR_SUCCESS)
-            LOGE("CloseClient_RILD() error");
-
-        mRilClient = 0;
-
-        dlclose(mSecRilLibHandle);
-        mSecRilLibHandle = NULL;
-    }
-}
-
-
-void AudioHardwareALSA::loadRILD(void)
-{
-    mSecRilLibHandle = dlopen("libsecril-client.so", RTLD_NOW);
-
-    if (mSecRilLibHandle) {
-        LOGV("libsecril-client.so is loaded");
-
-        openClientRILD   = (HRilClient (*)(void))
-                              dlsym(mSecRilLibHandle, "OpenClient_RILD");
-        disconnectRILD   = (int (*)(HRilClient))
-                              dlsym(mSecRilLibHandle, "Disconnect_RILD");
-        closeClientRILD  = (int (*)(HRilClient))
-                              dlsym(mSecRilLibHandle, "CloseClient_RILD");
-        isConnectedRILD  = (int (*)(HRilClient))
-                              dlsym(mSecRilLibHandle, "isConnected_RILD");
-        connectRILD      = (int (*)(HRilClient))
-                              dlsym(mSecRilLibHandle, "Connect_RILD");
-        setCallVolume    = (int (*)(HRilClient, SoundType, int))
-                              dlsym(mSecRilLibHandle, "SetCallVolume");
-        setCallAudioPath = (int (*)(HRilClient, AudioPath))
-                              dlsym(mSecRilLibHandle, "SetCallAudioPath");
-        setCallClockSync = (int (*)(HRilClient, SoundClockCondition))
-                              dlsym(mSecRilLibHandle, "SetCallClockSync");
-
-        if (!openClientRILD  || !disconnectRILD   || !closeClientRILD ||
-            !isConnectedRILD || !connectRILD      ||
-            !setCallVolume   || !setCallAudioPath || !setCallClockSync) {
-            LOGE("Can't load all functions from libsecril-client.so");
-
-            dlclose(mSecRilLibHandle);
-            mSecRilLibHandle = NULL;
-        } else {
-            mRilClient = openClientRILD();
-            if (!mRilClient) {
-                LOGE("OpenClient_RILD() error");
-
-                dlclose(mSecRilLibHandle);
-                mSecRilLibHandle = NULL;
-            }
-        }
-    } else {
-        LOGE("Can't load libsecril-client.so");
-    }
-}
-
-
-status_t AudioHardwareALSA::initCheck()
-{
-    if (mMixer && mMixer->isValid())
-        return NO_ERROR;
-    else
-        return NO_INIT;
-}
-
-
-status_t AudioHardwareALSA::connectRILDIfRequired(void)
-{
-    if (!mSecRilLibHandle) {
-        LOGE("connectIfRequired() lib is not loaded");
-        return INVALID_OPERATION;
-    }
-
-    if (isConnectedRILD(mRilClient)) {
-        return OK;
-    }
-
-    if (connectRILD(mRilClient) != RIL_CLIENT_ERR_SUCCESS) {
-        LOGE("Connect_RILD() error");
-        return INVALID_OPERATION;
-    }
-
-    return OK;
-}
-
-
-status_t AudioHardwareALSA::setVoiceVolume(float volume)
-{
-    LOGI("### setVoiceVolume");
-
-    AutoMutex lock(mLock);
-    // sangsu fix : transmic volume level IPC to modem
-    if ( (AudioSystem::MODE_IN_CALL == mMode) && (mSecRilLibHandle) &&
-         (connectRILDIfRequired() == OK) ) {
-
-        uint32_t routes = AudioSystem::ROUTE_EARPIECE;
-        if (mOutput != NULL) {
-            routes = mOutput->device();
-        }
-        int int_volume = (int)(volume * 5);
-
-        LOGI("### route(%d) call volume(%f)", routes, volume);
-        switch (routes) {
-            case AudioSystem::ROUTE_EARPIECE:
-                LOGI("### earpiece call volume");
-                setCallVolume(mRilClient, SOUND_TYPE_VOICE, int_volume);
-                break;
-
-            case AudioSystem::ROUTE_SPEAKER:
-                LOGI("### speaker call volume");
-                setCallVolume(mRilClient, SOUND_TYPE_SPEAKER, int_volume);
-                break;
-
-            case AudioSystem::ROUTE_BLUETOOTH_SCO:
-            case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
-            case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
-            case AudioSystem::ROUTE_BLUETOOTH_A2DP:
-                LOGI("### bluetooth call volume");
-                setCallVolume(mRilClient, SOUND_TYPE_BTVOICE, int_volume);
-                break;
-
-            case AudioSystem::ROUTE_HEADSET:
-            case AudioSystem::ROUTE_HEADPHONE: // Use receive path with 3 pole headset.
-                LOGI("### headset call volume");
-                setCallVolume(mRilClient, SOUND_TYPE_HEADSET, int_volume);
-                break;
-
-            default:
-                LOGE("### Call volume setting error!!!0x%08x \n", routes);
-                break;
-        }
-    }
-    // sangsu fix end
-
-    // The voice volume is used by the VOICE_CALL audio stream.
-    if (mMixer)
-        return mMixer->setVolume(ROUTE_EARPIECE, volume);
-    else
-        return INVALID_OPERATION;
-}
-
-status_t AudioHardwareALSA::setMasterVolume(float volume)
-{
-    if (mMixer)
-        return mMixer->setMasterVolume(volume);
-    else
-        return INVALID_OPERATION;
-}
-
-AudioStreamOut *
-AudioHardwareALSA::openOutputStream(
-                        uint32_t devices,
-                        int *format,
-                        uint32_t *channels,
-                        uint32_t *sampleRate,
-                        status_t *status)
-{
-    AudioStreamOutALSA *out = NULL;
-    status_t  ret = NO_ERROR;
-    {
-        AutoMutex lock(mLock);
-
-        // only one output stream allowed
-        if (mOutput) {
-            ret = ALREADY_EXISTS;
-            goto exit;
-        }
-
-        LOGV("[[[[[[[[\n%s - format = %d, channels = %d, sampleRate = %d, devices = %d]]]]]]]]\n", __func__, *format, *channels, *sampleRate,devices);
-
-        out = new AudioStreamOutALSA(this);
-
-        ret = out->set(format, channels, sampleRate);
-
-        if (ret == NO_ERROR) {
-            mOutput = out;
-        }
-    }
-exit:
-    if (ret == NO_ERROR) {
-        // Some information is expected to be available immediately after
-        // the device is open.
-        /* Tushar - Sets the current device output here - we may set device here */
-        LOGI("%s] Setting ALSA device.", __func__);
-        mOutput->setDevice(mMode, devices, PLAYBACK); /* tushar - Enable all devices as of now */
-    } else if (out) {
-        delete out;
-    }
-    if (status) {
-        *status = ret;
-    }
-    return mOutput;
-}
-
-void
-AudioHardwareALSA::closeOutputStream(AudioStreamOut* out)
-{
-    /* TODO:Tushar: May lead to segmentation fault - check*/
-    //delete out;
-    {
-        AutoMutex lock(mLock);
-
-        if (mOutput == 0 || mOutput != out) {
-            LOGW("Attempt to close invalid output stream");
-            return;
-        }
-        mOutput = 0;
-    }
-    delete out;
-}
-
-
-AudioStreamIn*
-AudioHardwareALSA::openInputStream(
-                                uint32_t devices,
-                                int *format,
-                                uint32_t *channels,
-                                uint32_t *sampleRate,
-                                status_t *status,
-                                AudioSystem::audio_in_acoustics acoustics)
-{
-    AudioStreamInALSA *in = NULL;
-    status_t ret = NO_ERROR;
-    {
-        AutoMutex lock(mLock);
-
-        // only one input stream allowed
-        if (mInput) {
-            ret = ALREADY_EXISTS;
-            goto exit;
-        }
-
-        in = new AudioStreamInALSA(this);
-
-        ret = in->set(format, channels, sampleRate);
-        if (ret == NO_ERROR) {
-            mInput = in;
-        }
-    }
-exit:
-    if (ret == NO_ERROR) {
-        // Some information is expected to be available immediately after
-        // the device is open.
-        mInput->setDevice(mMode, devices, CAPTURE);  /* Tushar - as per modified arch */
-    } else if (in != NULL) {
-        delete in;
-    }
-    if (status) {
-        *status = ret;
-    }
-    return mInput;
-}
-
-void
-AudioHardwareALSA::closeInputStream(AudioStreamIn* in)
-{
-    /* TODO:Tushar: May lead to segmentation fault - check*/
-    //delete in;
-    {
-        AutoMutex lock(mLock);
-
-        if (mInput == 0 || mInput != in) {
-            LOGW("Attempt to close invalid input stream");
-            return;
-        } else {
-            mInput = 0;
-        }
-    }
-    delete in;
-}
-
-
-status_t AudioHardwareALSA::doRouting(uint32_t device, bool force)
-{
-    AutoMutex lock(mLock);
-    return doRouting_l(device, force);
-}
-
-status_t AudioHardwareALSA::doRouting_l(uint32_t device, bool force)
-{
-    status_t ret;
-    int mode = mMode;        // Prevent to changing mode on setup sequence.
-
-    LOGV("doRouting: device %x, force %d", device, force);
-
-    if (mOutput) {
-        //device = 0; /* Tushar - temp implementation */
-        if (device == AudioSystem::DEVICE_OUT_DEFAULT) {
-            device = mOutput->device();
-        }
-
-        // Setup sound path for CP clocking
-        if ( (AudioSystem::MODE_IN_CALL == mode) && (mSecRilLibHandle) &&
-             (connectRILDIfRequired() == OK) ) {
-
-            LOGI("### incall mode route (%d)", device);
-
-            switch(device){
-                case AudioSystem::ROUTE_EARPIECE:
-                    LOGI("### incall mode earpiece route");
-                    setCallAudioPath(mRilClient, SOUND_AUDIO_PATH_HANDSET);
-                    break;
-
-                case AudioSystem::ROUTE_SPEAKER:
-                    LOGI("### incall mode speaker route");
-                    setCallAudioPath(mRilClient, SOUND_AUDIO_PATH_SPEAKER);
-                    break;
-
-                case AudioSystem::ROUTE_BLUETOOTH_SCO:
-                case AudioSystem::ROUTE_BLUETOOTH_SCO_HEADSET:
-                case AudioSystem::ROUTE_BLUETOOTH_SCO_CARKIT:
-                    LOGI("### incall mode bluetooth route %s NR", mBluetoothECOff ? "NO" : "");
-                    if (mBluetoothECOff)
-                        setCallAudioPath(mRilClient, SOUND_AUDIO_PATH_BLUETOOTH_NO_NR);
-                    else
-                        setCallAudioPath(mRilClient, SOUND_AUDIO_PATH_BLUETOOTH);
-                    break;
-
-                case AudioSystem::ROUTE_HEADSET :
-                case AudioSystem::ROUTE_HEADPHONE :
-                    LOGI("### incall mode headset route");
-                    setCallAudioPath(mRilClient, SOUND_AUDIO_PATH_HEADSET);
-                    break;
-
-                case AudioSystem::ROUTE_BLUETOOTH_A2DP:
-                    LOGI("### incall mode bluetooth route");
-                    setCallAudioPath(mRilClient, SOUND_AUDIO_PATH_BLUETOOTH);
-                    break;
-
-                default:
-                    LOGE("### incall mode Error!! route = [%d]", device);
-                    break;
-            }
-        }
-
-        ret = mOutput->setDevice(mode, device, PLAYBACK, force);
-
-        return ret;
-    }
-
-    return NO_INIT;
-}
-
-
-status_t AudioHardwareALSA::setMicMute(bool state)
-{
-    if (mMixer)
-        return mMixer->setCaptureMuteState(ROUTE_EARPIECE, state);
-
-    return NO_INIT;
-}
-
-status_t AudioHardwareALSA::getMicMute(bool *state)
-{
-    if (mMixer)
-        return mMixer->getCaptureMuteState(ROUTE_EARPIECE, state);
-
-    return NO_ERROR;
-}
-
-status_t AudioHardwareALSA::dump(int fd, const Vector<String16>& args)
-{
-    return NO_ERROR;
-}
-
-
-uint32_t AudioHardwareALSA::bufferRatio(uint32_t samplingRate) {
-    switch (samplingRate) {
-    case 8000:
-    case 11025:
-        return 4;
-    case 16000:
-    case 22050:
-        return 2;
-    case 44100:
-    default:
-        break;
-    }
-    return 1;
-}
-
-
-size_t AudioHardwareALSA::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
-{
-    if (sampleRate != 8000 && sampleRate != 11025 && sampleRate != 16000 &&
-            sampleRate != 22050 && sampleRate != 44100) {
-        LOGW("getInputBufferSize bad sampling rate: %d", sampleRate);
-        return 0;
-    }
-    if (format != AudioSystem::PCM_16_BIT) {
-        LOGW("getInputBufferSize bad format: %d", format);
-        return 0;
-    }
-    if (channelCount != 1) {
-        LOGW("getInputBufferSize bad channel count: %d", channelCount);
-        return 0;
-    }
-
-    size_t size = (PERIOD_SZ_CAPTURE / bufferRatio(sampleRate)) * sizeof(int16_t);
-    LOGV("getInputBufferSize() rate %d, ratio %d", sampleRate, size);
-    return size;
-
-}
-
-uint32_t AudioHardwareALSA::checkInputSampleRate(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];
-}
-
-status_t AudioHardwareALSA::setMode(int mode)
-{
-    AutoMutex lock(mLock);
-    int prevMode = mMode;
-    status_t status = AudioHardwareBase::setMode(mode);
-    LOGV("setMode() : new %d, old %d", mMode, prevMode);
-    if (status == NO_ERROR) {
-        if ( (mMode == AudioSystem::MODE_RINGTONE) || (mMode == AudioSystem::MODE_IN_CALL) )
-        {
-            if ( (!mActivatedCP) && (mSecRilLibHandle) && (connectRILDIfRequired() == OK) ) {
-                setCallClockSync(mRilClient, SOUND_CLOCK_START);
-                mActivatedCP = true;
-            }
-        }
-
-        // make sure that doAudioRouteOrMute() is called by doRouting()
-        // when entering or exiting in call mode even if the new device
-        // selected is the same as current one.
-        if ((prevMode != AudioSystem::MODE_IN_CALL) && (mMode == AudioSystem::MODE_IN_CALL)) {
-            LOGV("setMode() entering call");
-            doRouting_l(AudioSystem::DEVICE_OUT_DEFAULT, true);
-            setVoiceRecordGain_l(false);
-        }
-        if ((prevMode == AudioSystem::MODE_IN_CALL) && (mMode != AudioSystem::MODE_IN_CALL)) {
-            LOGV("setMode() exiting call");
-            doRouting_l(AudioSystem::DEVICE_OUT_DEFAULT, true);
-            if (mOutput != NULL && !mOutput->isActive()) {
-                mOutput->close();
-            }
-        }
-
-        if (mMode == AudioSystem::MODE_NORMAL) {
-            if(mActivatedCP)
-                mActivatedCP = false;
-        }
-    }
-
-    return status;
-}
-
-int AudioHardwareALSA::setVoiceRecordGain(bool enable)
-{
-    AutoMutex lock(mLock);
-    return setVoiceRecordGain_l(enable);
-}
-
-int AudioHardwareALSA::setVoiceRecordGain_l(bool enable)
-{
-     LOGI("[%s], enable=%d", __func__, enable);
-     if (enable != mVrModeEnabled &&
-         !(enable && (mMode == AudioSystem::MODE_IN_CALL))) {
-         ALSAControl *alsaControl = new ALSAControl();
-         status_t ret = alsaControl->set("Recognition Control", enable ? RECOGNITION_ON : RECOGNITION_OFF);
-         delete alsaControl;
-         mVrModeEnabled = enable;
-     }
-
-     return NO_ERROR;
-}
-
-status_t AudioHardwareALSA::setParameters(const String8& keyValuePairs)
-{
-    AudioParameter param = AudioParameter(keyValuePairs);
-    String8 bt_nrec_key = String8("bt_headset_nrec");
-    String8 value;
-
-    LOGV("setParameters(%s)", keyValuePairs.string());
-
-    if (param.get(bt_nrec_key, value) == NO_ERROR) {
-        setBluetoothNrEcOnOff((value == "on") ? false : true);
-    }
-
-    return NO_ERROR;
-}
-
-void AudioHardwareALSA::setBluetoothNrEcOnOff(bool disable)
-{
-    LOGV("setBluetoothNrEcOnOff(%s)", disable ? "true" : "false");
-
-    if (disable != mBluetoothECOff)
-    {
-        mBluetoothECOff = disable;
-
-        if ( (mOutput)          && (AudioSystem::MODE_IN_CALL == mMode) &&
-             (mSecRilLibHandle) && (connectRILDIfRequired() == OK)) {
-
-            uint32_t device = mOutput->device();
-
-            switch (device) {
-                case AudioSystem::ROUTE_BLUETOOTH_SCO:
-                case AudioSystem::ROUTE_BLUETOOTH_SCO_HEADSET:
-                case AudioSystem::ROUTE_BLUETOOTH_SCO_CARKIT:
-                    LOGV("### incall mode bluetooth EC %s route", mBluetoothECOff ? "OFF" : "ON");
-                    if (mBluetoothECOff)
-                        setCallAudioPath(mRilClient, SOUND_AUDIO_PATH_BLUETOOTH_NO_NR);
-                    else
-                        setCallAudioPath(mRilClient, SOUND_AUDIO_PATH_BLUETOOTH);
-                    break;
-
-                default :
-                    LOGE("Bluetooth path is not activated!!");
-                    break;
-            }
-        }
-    }
-}
-
-
-// ----------------------------------------------------------------------------
-
-ALSAStreamOps::ALSAStreamOps() :
-    mHandle(0),
-    mHardwareParams(0),
-    mSoftwareParams(0),
-    mDevice(0)
-{
-    if (snd_pcm_hw_params_malloc(&mHardwareParams) < 0) {
-        LOG_ALWAYS_FATAL("Failed to allocate ALSA hardware parameters!");
-    }
-
-    if (snd_pcm_sw_params_malloc(&mSoftwareParams) < 0) {
-        LOG_ALWAYS_FATAL("Failed to allocate ALSA software parameters!");
-    }
-}
-
-ALSAStreamOps::~ALSAStreamOps()
-{
-    AutoMutex lock(mLock);
-
-    close();
-
-    if (mHardwareParams)
-        snd_pcm_hw_params_free(mHardwareParams);
-
-    if (mSoftwareParams)
-        snd_pcm_sw_params_free(mSoftwareParams);
-}
-
-
-status_t ALSAStreamOps::set(int      *pformat,
-                            uint32_t *pchannels,
-                            uint32_t *prate)
-{
-    int lformat = pformat ? *pformat : 0;
-    unsigned int lchannels = pchannels ? *pchannels : 0;
-    unsigned int lrate = prate ? *prate : 0;
-
-
-    LOGD("ALSAStreamOps - input   - format = %d, channels = %d, rate = %d\n", lformat, lchannels, lrate);
-    LOGD("ALSAStreamOps - default - format = %d, channelCount = %d, rate = %d\n", mDefaults->format, mDefaults->channelCount, mDefaults->sampleRate);
-
-    if (lformat == 0) lformat = getAndroidFormat(mDefaults->format);//format();
-    if (lchannels == 0) lchannels = getAndroidChannels(mDefaults->channelCount);// channelCount();
-    if (lrate == 0) lrate = mDefaults->sampleRate;
-
-    if ( (lformat != getAndroidFormat(mDefaults->format)) ||
-          (lchannels != getAndroidChannels(mDefaults->channelCount)) ) {
-        if (pformat)   *pformat = getAndroidFormat(mDefaults->format);
-        if (pchannels) *pchannels = getAndroidChannels(mDefaults->channelCount);
-        return BAD_VALUE;
-    }
-    if (mDefaults->direction == SND_PCM_STREAM_PLAYBACK) {
-        if (lrate != mDefaults->sampleRate) {
-            if (prate) *prate = mDefaults->sampleRate;
-            return BAD_VALUE;
-        }
-    } else {
-        uint32_t rate = AudioHardwareALSA::checkInputSampleRate(lrate);
-        if (rate != lrate) {
-            if (prate) *prate = rate;
-            return BAD_VALUE;
-        }
-        lrate = rate;
-    }
-    mDefaults->bufferRatio = AudioHardwareALSA::bufferRatio(lrate);
-    mDefaults->sampleRate = lrate;
-
-    if(pformat)     *pformat = getAndroidFormat(mDefaults->format);
-    if(pchannels)   *pchannels = getAndroidChannels(mDefaults->channelCount);
-    if(prate)       *prate = mDefaults->sampleRate;
-
-    return NO_ERROR;
-}
-
-
-uint32_t ALSAStreamOps::sampleRate() const
-{
-    return mDefaults->sampleRate;
-}
-
-status_t ALSAStreamOps::sampleRate(uint32_t rate)
-{
-    const char *stream;
-    unsigned int requestedRate;
-    int err;
-
-    if (!mHandle)
-        return NO_INIT;
-
-    stream = streamName();
-    requestedRate = rate;
-    err = snd_pcm_hw_params_set_rate_near(mHandle,
-                                          mHardwareParams,
-                                          &requestedRate,
-                                          0);
-
-    if (err < 0) {
-        LOGE("Unable to set %s sample rate to %u: %s",
-            stream, rate, snd_strerror(err));
-        return BAD_VALUE;
-    }
-    if (requestedRate != rate) {
-        // Some devices have a fixed sample rate, and can not be changed.
-        // This may cause resampling problems; i.e. PCM playback will be too
-        // slow or fast.
-        LOGW("Requested rate (%u HZ) does not match actual rate (%u HZ)",
-            rate, requestedRate);
-    }
-    else {
-        LOGD("Set %s sample rate to %u HZ", stream, requestedRate);
-    }
-    return NO_ERROR;
-}
-
-//
-// Return the number of bytes (not frames)
-//
-size_t ALSAStreamOps::bufferSize() const
-{
-    int err;
-
-    size_t size = ((mDefaults->periodSize / mDefaults->bufferRatio) * mDefaults->channelCount *
-            snd_pcm_format_physical_width(mDefaults->format)) / 8;
-    LOGV("bufferSize() channelCount %d, bufferRatio %d, size %d",
-         mDefaults->channelCount, mDefaults->bufferRatio, size);
-    return size;
-
-}
-
-int ALSAStreamOps::getAndroidFormat(snd_pcm_format_t format)
-{
-    int pcmFormatBitWidth;
-    int audioSystemFormat;
-
-    pcmFormatBitWidth = snd_pcm_format_physical_width(format);
-    audioSystemFormat = AudioSystem::DEFAULT;
-    switch(pcmFormatBitWidth) {
-        case 8:
-            audioSystemFormat = AudioSystem::PCM_8_BIT;
-            break;
-
-        case 16:
-            audioSystemFormat = AudioSystem::PCM_16_BIT;
-            break;
-
-        default:
-            LOG_FATAL("Unknown AudioSystem bit width %i!", pcmFormatBitWidth);
-    }
-
-    return audioSystemFormat;
-
-}
-
-int ALSAStreamOps::format() const
-{
-    snd_pcm_format_t ALSAFormat;
-    int pcmFormatBitWidth;
-    int audioSystemFormat;
-
-    if (snd_pcm_hw_params_get_format(mHardwareParams, &ALSAFormat) < 0) {
-        return -1;
-    }
-
-    pcmFormatBitWidth = snd_pcm_format_physical_width(ALSAFormat);
-    audioSystemFormat = AudioSystem::DEFAULT;
-    switch(pcmFormatBitWidth) {
-        case 8:
-            audioSystemFormat = AudioSystem::PCM_8_BIT;
-            break;
-
-        case 16:
-            audioSystemFormat = AudioSystem::PCM_16_BIT;
-            break;
-
-        default:
-            LOG_FATAL("Unknown AudioSystem bit width %i!", pcmFormatBitWidth);
-    }
-
-    return audioSystemFormat;
-}
-
-uint32_t ALSAStreamOps::getAndroidChannels(int channelCount) const
-{
-    int AudioSystemChannels = AudioSystem::DEFAULT;
-
-    if (mDefaults->direction == SND_PCM_STREAM_PLAYBACK) {
-        switch(channelCount){
-        case 1:
-            AudioSystemChannels = AudioSystem::CHANNEL_OUT_MONO;
-            break;
-        case 2:
-            AudioSystemChannels = AudioSystem::CHANNEL_OUT_STEREO;
-            break;
-        case 4:
-            AudioSystemChannels = AudioSystem::CHANNEL_OUT_QUAD;
-            break;
-        case 6:
-            AudioSystemChannels = AudioSystem::CHANNEL_OUT_5POINT1;
-            break;
-        default:
-            LOGE("FATAL: AudioSystem does not support %d output channels.", channelCount);
-        }
-    } else {
-        switch(channelCount){
-        case 1:
-            AudioSystemChannels = AudioSystem::CHANNEL_IN_MONO;
-            break;
-        case 2:
-            AudioSystemChannels = AudioSystem::CHANNEL_IN_STEREO;
-            break;
-        default:
-            LOGE("FATAL: AudioSystem does not support %d input channels.", channelCount);
-        }
-
-    }
-    return AudioSystemChannels;
-}
-
-uint32_t ALSAStreamOps::channels() const
-{
-    return getAndroidChannels(mDefaults->channelCount);
-}
-
-int ALSAStreamOps::channelCount() const
-{
-    return mDefaults->channelCount;
-}
-
-status_t ALSAStreamOps::channelCount(int channelCount) {
-    int err;
-
-    if (!mHandle)
-        return NO_INIT;
-
-    err = snd_pcm_hw_params_set_channels(mHandle, mHardwareParams, channelCount);
-    if (err < 0) {
-        LOGE("Unable to set channel count to %i: %s",
-            channelCount, snd_strerror(err));
-        return BAD_VALUE;
-    }
-
-    LOGD("Using %i %s for %s.",
-         channelCount, channelCount == 1 ? "channel" : "channels", streamName());
-
-    return NO_ERROR;
-}
-
-status_t ALSAStreamOps::open(int mode, uint32_t device)
-{
-    const char *stream = streamName();
-    const char *devName = deviceName(mode, device);
-
-    int         err;
-
-    LOGI("Try to open ALSA %s device %s", stream, devName);
-
-    for(;;) {
-        // The PCM stream is opened in blocking mode, per ALSA defaults.  The
-        // AudioFlinger seems to assume blocking mode too, so asynchronous mode
-        // should not be used.
-        err = snd_pcm_open(&mHandle, devName, mDefaults->direction, 0);
-        if (err == 0) break;
-
-        // See if there is a less specific name we can try.
-        // Note: We are changing the contents of a const char * here.
-        char *tail = strrchr(devName, '_');
-        if (! tail) break;
-        *tail = 0;
-    }
-
-    if (err < 0) {
-        // None of the Android defined audio devices exist. Open a generic one.
-        devName = "hw:00,1";    // 090507 SMDKC110 Froyo
-
-        err = snd_pcm_open(&mHandle, devName, mDefaults->direction, 0);
-        if (err < 0) {
-            // Last resort is the NULL device (i.e. the bit bucket).
-            devName = _nullALSADeviceName;
-            err = snd_pcm_open(&mHandle, devName, mDefaults->direction, 0);
-        }
-    }
-
-    mDevice = device;
-
-    LOGI("Initialized ALSA %s device %s", stream, devName);
-    return err;
-}
-
-void ALSAStreamOps::close()
-{
-    snd_pcm_t *handle = mHandle;
-    mHandle = NULL;
-
-    if (handle) {
-        LOGV("ALSAStreamOps::close()");
-        snd_pcm_drain(handle);
-        snd_pcm_close(handle);
-    }
-}
-
-status_t ALSAStreamOps::setSoftwareParams()
-{
-    if (!mHandle)
-        return NO_INIT;
-
-    int err;
-
-    // Get the current software parameters
-    err = snd_pcm_sw_params_current(mHandle, mSoftwareParams);
-    if (err < 0) {
-        LOGE("Unable to get software parameters: %s", snd_strerror(err));
-        return NO_INIT;
-    }
-
-    snd_pcm_uframes_t bufferSize = 0;
-    snd_pcm_uframes_t periodSize = 0;
-    snd_pcm_uframes_t startThreshold;
-
-    // Configure ALSA to start the transfer when the buffer is almost full.
-    snd_pcm_get_params(mHandle, &bufferSize, &periodSize);
-    LOGE("bufferSize %d, periodSize %d\n", (int)bufferSize, (int)periodSize);
-
-    if (mDefaults->direction == SND_PCM_STREAM_PLAYBACK) {
-        // For playback, configure ALSA to start the transfer when the
-        // buffer is almost full.
-        startThreshold = (bufferSize / periodSize) * periodSize;
-        //startThreshold = 1;
-    }
-    else {
-        // For recording, configure ALSA to start the transfer on the
-        // first frame.
-        startThreshold = 1;
-    }
-
-    err = snd_pcm_sw_params_set_start_threshold(mHandle,
-        mSoftwareParams,
-        startThreshold);
-    if (err < 0) {
-        LOGE("Unable to set start threshold to %lu frames: %s",
-            startThreshold, snd_strerror(err));
-        return NO_INIT;
-    }
-
-    // Stop the transfer when the buffer is full.
-    err = snd_pcm_sw_params_set_stop_threshold(mHandle,
-                                               mSoftwareParams,
-                                               bufferSize);
-    if (err < 0) {
-        LOGE("Unable to set stop threshold to %lu frames: %s",
-            bufferSize, snd_strerror(err));
-        return NO_INIT;
-    }
-
-    // Allow the transfer to start when at least periodSize samples can be
-    // processed.
-    err = snd_pcm_sw_params_set_avail_min(mHandle,
-                                          mSoftwareParams,
-                                          periodSize);
-    if (err < 0) {
-        LOGE("Unable to configure available minimum to %lu: %s",
-            periodSize, snd_strerror(err));
-        return NO_INIT;
-    }
-
-    // Commit the software parameters back to the device.
-    err = snd_pcm_sw_params(mHandle, mSoftwareParams);
-    if (err < 0) {
-        LOGE("Unable to configure software parameters: %s",
-            snd_strerror(err));
-        return NO_INIT;
-    }
-
-    return NO_ERROR;
-}
-
-status_t ALSAStreamOps::setPCMFormat(snd_pcm_format_t format)
-{
-    const char *formatDesc;
-    const char *formatName;
-    bool validFormat;
-    int err;
-
-    // snd_pcm_format_description() and snd_pcm_format_name() do not perform
-    // proper bounds checking.
-    validFormat = (static_cast<int>(format) > SND_PCM_FORMAT_UNKNOWN) &&
-        (static_cast<int>(format) <= SND_PCM_FORMAT_LAST);
-    formatDesc = validFormat ?
-        snd_pcm_format_description(format) : "Invalid Format";
-    formatName = validFormat ?
-        snd_pcm_format_name(format) : "UNKNOWN";
-
-    err = snd_pcm_hw_params_set_format(mHandle, mHardwareParams, format);
-    if (err < 0) {
-        LOGE("Unable to configure PCM format %s (%s): %s",
-            formatName, formatDesc, snd_strerror(err));
-        return NO_INIT;
-    }
-
-    LOGD("Set %s PCM format to %s (%s)", streamName(), formatName, formatDesc);
-    return NO_ERROR;
-}
-
-status_t ALSAStreamOps::setHardwareResample(bool resample)
-{
-    int err;
-
-    err = snd_pcm_hw_params_set_rate_resample(mHandle,
-                                              mHardwareParams,
-                                              static_cast<int>(resample));
-    if (err < 0) {
-        LOGE("Unable to %s hardware resampling: %s",
-            resample ? "enable" : "disable",
-            snd_strerror(err));
-        return NO_INIT;
-    }
-    return NO_ERROR;
-}
-
-const char *ALSAStreamOps::streamName()
-{
-    // Don't use snd_pcm_stream(mHandle), as the PCM stream may not be
-    // opened yet.  In such case, snd_pcm_stream() will abort().
-    return snd_pcm_stream_name(mDefaults->direction);
-}
-
-//
-// Set playback or capture PCM device.  It's possible to support audio output
-// or input from multiple devices by using the ALSA plugins, but this is
-// not supported for simplicity.
-//
-// The AudioHardwareALSA API does not allow one to set the input routing.
-//
-// If the "routes" value does not map to a valid device, the default playback
-// device is used.
-//
-status_t ALSAStreamOps::setDevice(int mode, uint32_t device, uint audio_mode)
-{
-    // Close off previously opened device.
-    // It would be nice to determine if the underlying device actually
-    // changes, but we might be manipulating mixer settings (see asound.conf).
-    //
-    close();
-
-    const char *stream = streamName();
-
-
-    LOGD("\n------------------------>>>>>> ALSA OPEN mode %d,device %d \n",mode,device);
-
-    status_t    status = open (mode, device);
-    int     err;
-    unsigned int period_val;
-
-    if (status != NO_ERROR)
-        return status;
-
-    err = snd_pcm_hw_params_any(mHandle, mHardwareParams);
-    if (err < 0) {
-        LOGE("Unable to configure hardware: %s", snd_strerror(err));
-        return NO_INIT;
-    }
-
-    status = setPCMFormat(mDefaults->format);
-
-    // Set the interleaved read and write format.
-    err = snd_pcm_hw_params_set_access(mHandle, mHardwareParams,
-                                       SND_PCM_ACCESS_RW_INTERLEAVED);
-    if (err < 0) {
-        LOGE("Unable to configure PCM read/write format: %s",
-            snd_strerror(err));
-        return NO_INIT;
-    }
-
-
-    //
-    // Some devices do not have the default two channels.  Force an error to
-    // prevent AudioMixer from crashing and taking the whole system down.
-    //
-    // Note that some devices will return an -EINVAL if the channel count
-    // is queried before it has been set.  i.e. calling channelCount()
-    // before channelCount(channels) may return -EINVAL.
-    //
-    status = channelCount(mDefaults->channelCount);
-    if (status != NO_ERROR)
-        return status;
-
-    // Don't check for failure; some devices do not support the default
-    // sample rate.
-    // FIXME:: always use default sampling rate
-    sampleRate(DEFAULT_SAMPLE_RATE);
-
-    snd_pcm_uframes_t bufferSize = mDefaults->bufferSize;
-    snd_pcm_uframes_t periodSize = mDefaults->periodSize;
-    period_val = bufferSize/periodSize;
-
-    unsigned int latency = mDefaults->latency;
-
-    // Make sure we have at least the size we originally wanted
-    err = snd_pcm_hw_params_set_buffer_size(mHandle, mHardwareParams, bufferSize);
-    if (err < 0) {
-        LOGE("Unable to set buffer size to %d:  %s",
-             (int)bufferSize, snd_strerror(err));
-        return NO_INIT;
-    }
-
-    err = snd_pcm_hw_params_set_period_size (mHandle, mHardwareParams, periodSize, NULL);
-    if (err < 0) {
-        LOGE("Unable to set the period size for latency: %s", snd_strerror(err));
-        return NO_INIT;
-    }
-
-    err = snd_pcm_hw_params_get_period_size (mHardwareParams, &periodSize, NULL);
-    if (err < 0) {
-        LOGE("Unable to get the period size for latency: %s", snd_strerror(err));
-        return NO_INIT;
-    }
-
-//    err = snd_pcm_hw_params_get_period_size (mHardwareParams, &periodSize, NULL);
-//    if (err < 0) {
-//        LOGE("Unable to get the period size for latency: %s", snd_strerror(err));
-//        return NO_INIT;
-//    }
-
-//    // Setup buffers for latency
-//    err = snd_pcm_hw_params_set_buffer_time_near (mHandle, mHardwareParams,
-//                                                  &latency, NULL);
-//    if(audio_mode == PLAYBACK) {
-//        period_val = PERIODS_PLAYBACK;
-//        if(snd_pcm_hw_params_set_periods(mHandle, mHardwareParams, period_val, 0) < 0)
-//            LOGE("Fail to set period size %d for playback", period_val);
-//    }
-//    else
-//        period_val = PERIODS_CAPTURE;
-//
-//    if (err < 0) {
-//        LOGD("snd_pcm_hw_params_set_buffer_time_near() failed: %s", snd_strerror(err));
-//        /* That didn't work, set the period instead */
-//        unsigned int periodTime = latency / period_val;
-//        err = snd_pcm_hw_params_set_period_time_near (mHandle, mHardwareParams,
-//                                                      &periodTime, NULL);
-//        if (err < 0) {
-//            LOGE("Unable to set the period time for latency: %s", snd_strerror(err));
-//            return NO_INIT;
-//        }
-//        err = snd_pcm_hw_params_get_period_size (mHardwareParams, &periodSize, NULL);
-//        if (err < 0) {
-//            LOGE("Unable to get the period size for latency: %s", snd_strerror(err));
-//            return NO_INIT;
-//        }
-//        bufferSize = periodSize * period_val;
-//        if (bufferSize < mDefaults->bufferSize)
-//            bufferSize = mDefaults->bufferSize;
-//        err = snd_pcm_hw_params_set_buffer_size_near (mHandle, mHardwareParams, &bufferSize);
-//        if (err < 0) {
-//            LOGE("Unable to set the buffer size for latency: %s", snd_strerror(err));
-//            return NO_INIT;
-//        }
-//    } else {
-//        LOGD("snd_pcm_hw_params_set_buffer_time_near() OK");
-//        // OK, we got buffer time near what we expect. See what that did for bufferSize.
-//        err = snd_pcm_hw_params_get_buffer_size (mHardwareParams, &bufferSize);
-//        if (err < 0) {
-//            LOGE("Unable to get the buffer size for latency: %s", snd_strerror(err));
-//            return NO_INIT;
-//        }
-//        // Does set_buffer_time_near change the passed value? It should.
-//        err = snd_pcm_hw_params_get_buffer_time (mHardwareParams, &latency, NULL);
-//        if (err < 0) {
-//            LOGE("Unable to get the buffer time for latency: %s", snd_strerror(err));
-//            return NO_INIT;
-//        }
-//        LOGD("got latency %d for bufferSize %d", latency, bufferSize);
-//        unsigned int periodTime = latency / period_val;
-//        LOGD("got latency %d for bufferSize %d => periodTime %d", latency, bufferSize, periodTime);
-//        err = snd_pcm_hw_params_set_period_time_near (mHandle, mHardwareParams,
-//                                                      &periodTime, NULL);
-//        if (err < 0) {
-//            LOGE("Unable to set the period time for latency: %s", snd_strerror(err));
-//            return NO_INIT;
-//        }
-//        err = snd_pcm_hw_params_get_period_size (mHardwareParams, &periodSize, NULL);
-//        if (err < 0) {
-//            LOGE("Unable to get the period size for latency: %s", snd_strerror(err));
-//            return NO_INIT;
-//        }
-//    }
-
-    LOGD("Buffer size: %d", (int)bufferSize);
-    LOGD("Period size: %d", (int)periodSize);
-    LOGD("Latency: %d", (int)latency);
-
-    mDefaults->bufferSize = bufferSize;
-    mDefaults->latency = latency;
-    mDefaults->periodSize = periodSize;
-
-    // Commit the hardware parameters back to the device.
-    err = snd_pcm_hw_params(mHandle, mHardwareParams);
-    if (err < 0) {
-        LOGE("Unable to set hardware parameters: %s", snd_strerror(err));
-        return NO_INIT;
-    }
-
-    status = setSoftwareParams();
-
-    return status;
-}
-
-const char *ALSAStreamOps::deviceName(int mode, uint32_t device)
-{
-    static char devString[ALSA_NAME_MAX];
-    int dev;
-    int hasDevExt = 0;
-
-    strcpy (devString, mDefaults->devicePrefix);
-
-    for (dev=0; device; dev++)
-        if (device & (1 << dev)) {
-            /* Don't go past the end of our list */
-            if (dev >= deviceSuffixLen)
-                break;
-            ALSA_STRCAT (devString, deviceSuffix[dev]);
-            device &= ~(1 << dev);
-            hasDevExt = 1;
-        }
-
-    if (hasDevExt)
-        switch (mode) {
-            case AudioSystem::MODE_NORMAL:
-                ALSA_STRCAT (devString, "_normal");
-                break;
-            case AudioSystem::MODE_RINGTONE:
-                ALSA_STRCAT (devString, "_ringtone");
-                break;
-            case AudioSystem::MODE_IN_CALL:
-                ALSA_STRCAT (devString, "_incall");
-                break;
-        };
-
-    return devString;
-}
-
-// ----------------------------------------------------------------------------
-
-AudioStreamOutALSA::AudioStreamOutALSA(AudioHardwareALSA *parent) :
-    mParent(parent),
-    mPowerLock(false)
-{
-    static StreamDefaults _defaults = {
-        devicePrefix   : "AndroidPlayback",
-        direction      : SND_PCM_STREAM_PLAYBACK,
-        format         : SND_PCM_FORMAT_S16_LE,   // AudioSystem::PCM_16_BIT
-        channelCount       : 2,
-        sampleRate     : DEFAULT_SAMPLE_RATE,
-        bufferRatio   : 1,
-        latency        : LATENCY_PLAYBACK_MS,               // Desired Delay in usec
-        bufferSize     : BUFFER_SZ_PLAYBACK,                // Desired Number of samples
-        periodSize     : PERIOD_SZ_PLAYBACK
-};
-
-    setStreamDefaults(&_defaults);
-}
-
-AudioStreamOutALSA::~AudioStreamOutALSA()
-{
-    standby();
-}
-
-
-/* New arch */
-status_t AudioStreamOutALSA::setVolume(float left, float right)
-{
-    if (! mParent->mMixer || ! mDevice)
-        return NO_INIT;
-
-    /** Tushar - Need to decide on the volume value
-     * that we pass onto the mixer. */
-    return mParent->mMixer->setVolume (mDevice, (left + right)/2);
-}
-
-status_t AudioStreamOutALSA::setVolume(float volume)
-{
-    if (! mParent->mMixer || ! mDevice)
-        return NO_INIT;
-
-    return mParent->mMixer->setVolume (mDevice, volume);
-}
-
-/* New Arch */
-status_t    AudioStreamOutALSA::setParameters(const String8& keyValuePairs)
-{
-    AudioParameter param = AudioParameter(keyValuePairs);
-    status_t status = NO_ERROR;
-    int device;
-    int value;
-    LOGD("AudioStreamOutALSA::setParameters() %s", keyValuePairs.string());
-
-    if (param.getInt(String8(AudioParameter::keyRouting), device) == NO_ERROR)
-    {
-        mParent->doRouting(device);
-
-        param.remove(String8(AudioParameter::keyRouting));
-    }
-
-    if (param.size()) {
-        status = BAD_VALUE;
-    }
-    return status;
-}
-
-
-String8  AudioStreamOutALSA::getParameters(const String8& keys)
-{
-    AudioParameter param = AudioParameter(keys);
-    String8 value;
-    String8 key = String8(AudioParameter::keyRouting);
-
-    if (param.get(key, value) == NO_ERROR) {
-        param.addInt(key, (int)mDevice);
-    }
-
-    LOGD("AudioStreamOutALSA::getParameters() %s", param.toString().string());
-    return param.toString();
-}
-
-
-status_t  AudioStreamOutALSA::getRenderPosition(uint32_t *dspFrames)
-{
-
-    //TODO: enable when supported by driver
-    return INVALID_OPERATION;
-}
-
-
-ssize_t AudioStreamOutALSA::write(const void *buffer, size_t bytes)
-{
-    snd_pcm_sframes_t n;
-    size_t            sent = 0;
-    status_t          err;
-
-    mParent->lock().lock();
-    AutoMutex lock(mLock);
-
-    if (!mPowerLock) {
-        LOGD("Calling setDevice from write @..%d.\n",__LINE__);
-        ALSAStreamOps::setDevice(mParent->mode(), mDevice, PLAYBACK);
-        acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioOutLock");
-        mPowerLock = true;
-    }
-    mParent->lock().unlock();
-
-    do {
-        // write correct number of bytes per attempt
-        n = snd_pcm_writei(mHandle, (char *) buffer + sent, snd_pcm_bytes_to_frames(mHandle, bytes
-                - sent));
-        if (n == -EBADFD) {
-            LOGD("Calling setDevice.. pcm_write returned error @..%d.\n",__LINE__);
-            // Somehow the stream is in a bad state. The driver probably
-            // has a bug and snd_pcm_recover() doesn't seem to handle this.
-            ALSAStreamOps::setDevice(mParent->mode(), mDevice, PLAYBACK);
-        } else if (n < 0) {
-            if (mHandle) {
-                // snd_pcm_recover() will return 0 if successful in recovering from
-                //             // an error, or -errno if the error was unrecoverable.
-                // We can make silent bit on as we are now handling the under-run and there will not be any data loss due to under-run
-                n = snd_pcm_recover(mHandle, n, 1);
-                if (n)
-                    return static_cast<ssize_t> (n);
-            }
-        } else
-            sent += static_cast<ssize_t> (snd_pcm_frames_to_bytes(mHandle, n));
-    } while (mHandle && sent < bytes);
-    //LOGI("Request Bytes=%d, Actual Written=%d",bytes,sent);
-    return snd_pcm_frames_to_bytes(mHandle, sent);
-}
-
-
-status_t AudioStreamOutALSA::dump(int fd, const Vector<String16>& args)
-{
-    return NO_ERROR;
-}
-
-status_t AudioStreamOutALSA::setDevice(int mode,
-                                       uint32_t newDevice,
-                                       uint32_t audio_mode,
-                                       bool force)
-{
-    AutoMutex lock(mLock);
-
-    LOGV("AudioStreamOutALSA::setDevice(mode %d, newDevice %x, audio_mode %d), mDevice %x",
-         mode, newDevice, audio_mode, mDevice);
-    if (newDevice != mDevice || force) {
-        return ALSAStreamOps::setDevice(mode, newDevice, audio_mode);
-    }
-    return NO_ERROR;
-}
-
-status_t AudioStreamOutALSA::standby() {
-    AutoMutex _l(mParent->lock());
-    AutoMutex lock(mLock);
-    LOGD("Inside AudioStreamOutALSA::standby\n");
-
-    if (mParent->mode() != AudioSystem::MODE_IN_CALL) {
-        ALSAStreamOps::close();
-    }
-
-    if (mPowerLock) {
-        release_wake_lock("AudioOutLock");
-        mPowerLock = false;
-    }
-    return NO_ERROR;
-}
-
-
-#define USEC_TO_MSEC(x) ((x + 999) / 1000)
-
-uint32_t AudioStreamOutALSA::latency() const
-{
-    // Android wants latency in milliseconds.
-    return USEC_TO_MSEC (mDefaults->latency);
-}
-
-// ----------------------------------------------------------------------------
-
-AudioStreamInALSA::AudioStreamInALSA(AudioHardwareALSA *parent) :
-    mParent(parent), mPowerLock(false),
-    mDownSampler(NULL), mPcmIn(NULL)
-{
-    static StreamDefaults _defaults = {
-        devicePrefix   : "AndroidRecord",
-        direction      : SND_PCM_STREAM_CAPTURE,
-        format         : SND_PCM_FORMAT_S16_LE,   // AudioSystem::PCM_16_BIT
-        channelCount       : 1,
-        sampleRate     : DEFAULT_SAMPLE_RATE,
-        bufferRatio   : 1,
-        latency        : LATENCY_CAPTURE_MS,// Desired Delay in usec
-        bufferSize     : BUFFER_SZ_CAPTURE,      // Desired Number of samples
-        periodSize     : PERIOD_SZ_CAPTURE
-        };
-
-    setStreamDefaults(&_defaults);
-}
-
-status_t AudioStreamInALSA::set(int *pformat,
-                            uint32_t *pchannels,
-                            uint32_t *prate)
-{
-    status_t status = ALSAStreamOps::set(pformat, pchannels, prate);
-    if (status == NO_ERROR && prate && *prate != DEFAULT_SAMPLE_RATE) {
-        mDownSampler = new ALSADownsampler(*prate,
-                                           mDefaults->channelCount,
-                                           PERIOD_SZ_CAPTURE,
-                                           this);
-        status = mDownSampler->initCheck();
-        if (status != NO_ERROR) {
-            return status;
-        }
-        mPcmIn = new int16_t[PERIOD_SZ_CAPTURE * mDefaults->channelCount];
-    }
-    return status;
-}
-
-AudioStreamInALSA::~AudioStreamInALSA()
-{
-    standby();
-    if (mDownSampler != NULL) {
-        delete mDownSampler;
-    }
-    if (mPcmIn != NULL) {
-        delete[] mPcmIn;
-    }
-}
-
-status_t AudioStreamInALSA::setGain(float gain)
-{
-    if (mParent->mMixer)
-        return mParent->mMixer->setMasterGain (gain);
-    else
-        return NO_INIT;
-}
-
-ssize_t AudioStreamInALSA::read(void *buffer, ssize_t bytes)
-{
-    snd_pcm_sframes_t n;
-
-    mParent->lock().lock();
-    AutoMutex lock(mLock);
-    if (!mPowerLock) {
-        acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioInLock");
-
-        LOGD("Calling setDevice from read@..%d.\n",__LINE__);
-        ALSAStreamOps::setDevice(mParent->mode(), mDevice, CAPTURE);
-
-        if (mDownSampler != NULL) {
-            mDownSampler->reset();
-            mReadStatus = 0;
-            mInPcmInBuf = 0;
-        }
-        mPowerLock = true;
-    }
-    mParent->lock().unlock();
-
-    if (!mHandle) {
-        return -1;
-    }
-
-    // FIXME: only support reads of exactly bufferSize() for now
-    if (bytes != (ssize_t)bufferSize()) {
-        LOGW("AudioStreamInALSA::read bad read size %d expected %d", (int)bytes, bufferSize());
-        return -1;
-    }
-
-    size_t frames = snd_pcm_bytes_to_frames(mHandle, bytes);
-    do {
-        if (mDownSampler) {
-            status_t status = mDownSampler->resample((int16_t *)buffer, &frames);
-            if (status != NO_ERROR) {
-                if (mReadStatus != 0) {
-                    n = mReadStatus;
-                } else {
-                    n = status;
-                }
-                frames = snd_pcm_bytes_to_frames(mHandle, bytes);
-            } else {
-                n = frames;
-            }
-        } else {
-            n = snd_pcm_readi(mHandle,
-                              (uint8_t *)buffer,
-                              frames);
-        }
-        if (n < 0) {
-            LOGD("AudioStreamInALSA::read error %d", (int)n);
-            n = snd_pcm_recover(mHandle, n, 0);
-            LOGD("AudioStreamInALSA::snd_pcm_recover error %d", (int)n);
-            if (n)
-                return static_cast<ssize_t> (n);
-        }
-    } while (n == 0);
-
-    return snd_pcm_frames_to_bytes(mHandle, n);
-}
-
-status_t AudioStreamInALSA::dump(int fd, const Vector<String16>& args)
-{
-    return NO_ERROR;
-}
-
-status_t AudioStreamInALSA::setDevice(int mode,
-                                      uint32_t newDevice,
-                                      uint32_t audio_mode,
-                                      bool force)
-{
-    AutoMutex lock(mLock);
-
-    return ALSAStreamOps::setDevice(mode, newDevice, audio_mode);
-}
-
-status_t AudioStreamInALSA::standby()
-{
-    AutoMutex _l(mParent->lock());
-    AutoMutex lock(mLock);
-
-    LOGD("Entering AudioStreamInALSA::standby\n");
-
-    ALSAStreamOps::close();
-
-    if (mPowerLock) {
-        release_wake_lock ("AudioInLock");
-        mPowerLock = false;
-    }
-
-    return NO_ERROR;
-}
-
-/* New Arch */
-status_t    AudioStreamInALSA::setParameters(const String8& keyValuePairs)
-{
-    AudioParameter param = AudioParameter(keyValuePairs);
-    String8 key = String8("vr_mode");
-    status_t status = NO_ERROR;
-    int value;
-    LOGD("AudioStreamInALSA::setParameters() %s", keyValuePairs.string());
-
-
-    if (param.getInt(key, value) == NO_ERROR) {
-        mParent->setVoiceRecordGain((value != 0));
-        param.remove(key);
-    }
-
-    key = String8(AudioParameter::keyRouting);
-    if (param.getInt(key, value) == NO_ERROR) {
-        if(mHandle != NULL && value != 0)
-            setDevice(mParent->mode(), value, CAPTURE);
-        param.remove(key);
-    }
-
-    if (param.size()) {
-        status = BAD_VALUE;
-    }
-    return status;
-}
-
-
-String8  AudioStreamInALSA::getParameters(const String8& keys)
-{
-    AudioParameter param = AudioParameter(keys);
-    String8 value;
-    String8 key = String8(AudioParameter::keyRouting);
-
-    if (param.get(key, value) == NO_ERROR) {
-        param.addInt(key, (int)mDevice);
-    }
-
-    LOGD("AudioStreamInALSA::getParameters() %s", param.toString().string());
-    return param.toString();
-}
-
-status_t AudioStreamInALSA::getNextBuffer(ALSABufferProvider::Buffer* buffer)
-{
-    if (mHandle == NULL) {
-        buffer->raw = NULL;
-        buffer->frameCount = 0;
-        return NO_INIT;
-    }
-
-    if (mInPcmInBuf == 0) {
-        while (mInPcmInBuf < PERIOD_SZ_CAPTURE) {
-            mReadStatus = snd_pcm_readi(mHandle,
-                              (uint8_t *)mPcmIn +
-                              (mInPcmInBuf * mDefaults->channelCount * sizeof(int16_t)),
-                              PERIOD_SZ_CAPTURE - mInPcmInBuf);
-            if (mReadStatus < 0) {
-                buffer->raw = NULL;
-                buffer->frameCount = 0;
-                LOGV("resampler read error %d", mReadStatus);
-                return mReadStatus;
-            }
-            mInPcmInBuf += mReadStatus;
-        }
-    }
-
-    buffer->frameCount = (buffer->frameCount > mInPcmInBuf) ? mInPcmInBuf : buffer->frameCount;
-    buffer->i16 = mPcmIn + (PERIOD_SZ_CAPTURE - mInPcmInBuf) * mDefaults->channelCount;
-
-    return NO_ERROR;
-}
-
-void AudioStreamInALSA::releaseBuffer(ALSABufferProvider::Buffer* buffer)
-{
-    mInPcmInBuf -= buffer->frameCount;
-}
-
-
-// ----------------------------------------------------------------------------
-
-struct mixer_info_t
-{
-    mixer_info_t() :
-        elem(0),
-        min(SND_MIXER_VOL_RANGE_MIN),
-        max(SND_MIXER_VOL_RANGE_MAX),
-        mute(false)
-    {
-    }
-
-    snd_mixer_elem_t *elem;
-    long              min;
-    long              max;
-    long              volume;
-    bool              mute;
-    char              name[ALSA_NAME_MAX];
-};
-
-static int initMixer (snd_mixer_t **mixer, const char *name)
-{
-    int err;
-
-    if ((err = snd_mixer_open(mixer, 0)) < 0) {
-        LOGE("Unable to open mixer: %s", snd_strerror(err));
-        return err;
-    }
-
-    if ((err = snd_mixer_attach(*mixer, name)) < 0) {
-        LOGE("Unable to attach mixer to device %s: %s",
-            name, snd_strerror(err));
-
-        if ((err = snd_mixer_attach(*mixer, "hw:00")) < 0) {
-            LOGE("Unable to attach mixer to device default: %s",
-                snd_strerror(err));
-
-            snd_mixer_close (*mixer);
-            *mixer = NULL;
-            return err;
-        }
-    }
-
-    if ((err = snd_mixer_selem_register(*mixer, NULL, NULL)) < 0) {
-        LOGE("Unable to register mixer elements: %s", snd_strerror(err));
-        snd_mixer_close (*mixer);
-        *mixer = NULL;
-        return err;
-    }
-
-    // Get the mixer controls from the kernel
-    if ((err = snd_mixer_load(*mixer)) < 0) {
-        LOGE("Unable to load mixer elements: %s", snd_strerror(err));
-        snd_mixer_close (*mixer);
-        *mixer = NULL;
-        return err;
-    }
-
-    return 0;
-}
-
-typedef int (*hasVolume_t)(snd_mixer_elem_t*);
-
-static const hasVolume_t hasVolume[] = {
-    snd_mixer_selem_has_playback_volume,
-    snd_mixer_selem_has_capture_volume
-};
-
-typedef int (*getVolumeRange_t)(snd_mixer_elem_t*, long int*, long int*);
-
-static const getVolumeRange_t getVolumeRange[] = {
-    snd_mixer_selem_get_playback_volume_range,
-    snd_mixer_selem_get_capture_volume_range
-};
-
-typedef int (*setVolume_t)(snd_mixer_elem_t*, long int);
-
-static const setVolume_t setVol[] = {
-    snd_mixer_selem_set_playback_volume_all,
-    snd_mixer_selem_set_capture_volume_all
-};
-
-ALSAMixer::ALSAMixer()
-{
-    int err;
-
-    initMixer (&mMixer[SND_PCM_STREAM_PLAYBACK], "AndroidPlayback");
-    initMixer (&mMixer[SND_PCM_STREAM_CAPTURE], "AndroidRecord");
-
-    snd_mixer_selem_id_t *sid;
-    snd_mixer_selem_id_alloca(&sid);
-
-    for (int i = 0; i <= SND_PCM_STREAM_LAST; i++) {
-
-        mixer_info_t *info = mixerMasterProp[i].mInfo = new mixer_info_t;
-
-        property_get (mixerMasterProp[i].propName,
-                      info->name,
-                      mixerMasterProp[i].propDefault);
-
-        for (snd_mixer_elem_t *elem = snd_mixer_first_elem(mMixer[i]);
-             elem;
-             elem = snd_mixer_elem_next(elem)) {
-
-            if (!snd_mixer_selem_is_active(elem))
-                continue;
-
-            snd_mixer_selem_get_id(elem, sid);
-
-            // Find PCM playback volume control element.
-            const char *elementName = snd_mixer_selem_id_get_name(sid);
-
-            if (hasVolume[i] (elem))
-                LOGD ("Mixer: element name: '%s'", elementName);
-
-            if (info->elem == NULL &&
-                strcmp(elementName, info->name) == 0 &&
-                hasVolume[i] (elem)) {
-
-                info->elem = elem;
-                getVolumeRange[i] (elem, &info->min, &info->max);
-                info->volume = info->max;
-                setVol[i] (elem, info->volume);
-                if (i == SND_PCM_STREAM_PLAYBACK &&
-                    snd_mixer_selem_has_playback_switch (elem))
-                    snd_mixer_selem_set_playback_switch_all (elem, 1);
-                break;
-            }
-        }
-
-        LOGD ("Mixer: master '%s' %s.", info->name, info->elem ? "found" : "not found");
-
-        for (int j = 0; mixerProp[j][i].routes; j++) {
-
-            mixer_info_t *info = mixerProp[j][i].mInfo = new mixer_info_t;
-
-            property_get (mixerProp[j][i].propName,
-                          info->name,
-                          mixerProp[j][i].propDefault);
-
-            for (snd_mixer_elem_t *elem = snd_mixer_first_elem(mMixer[i]);
-                 elem;
-                 elem = snd_mixer_elem_next(elem)) {
-
-                if (!snd_mixer_selem_is_active(elem))
-                    continue;
-
-                snd_mixer_selem_get_id(elem, sid);
-
-                // Find PCM playback volume control element.
-                const char *elementName = snd_mixer_selem_id_get_name(sid);
-
-               if (info->elem == NULL &&
-                    strcmp(elementName, info->name) == 0 &&
-                    hasVolume[i] (elem)) {
-
-                    info->elem = elem;
-                    getVolumeRange[i] (elem, &info->min, &info->max);
-                    info->volume = info->max;
-                    setVol[i] (elem, info->volume);
-                    if (i == SND_PCM_STREAM_PLAYBACK &&
-                        snd_mixer_selem_has_playback_switch (elem))
-                        snd_mixer_selem_set_playback_switch_all (elem, 1);
-                    break;
-                }
-            }
-            LOGD ("Mixer: route '%s' %s.", info->name, info->elem ? "found" : "not found");
-        }
-    }
-    LOGD("mixer initialized.");
-}
-
-ALSAMixer::~ALSAMixer()
-{
-    for (int i = 0; i <= SND_PCM_STREAM_LAST; i++) {
-        if (mMixer[i]) snd_mixer_close (mMixer[i]);
-        if (mixerMasterProp[i].mInfo) {
-            delete mixerMasterProp[i].mInfo;
-            mixerMasterProp[i].mInfo = NULL;
-        }
-        for (int j = 0; mixerProp[j][i].routes; j++) {
-            if (mixerProp[j][i].mInfo) {
-                delete mixerProp[j][i].mInfo;
-                mixerProp[j][i].mInfo = NULL;
-            }
-        }
-    }
-    LOGD("mixer destroyed.");
-}
-
-status_t ALSAMixer::setMasterVolume(float volume)
-{
-    mixer_info_t *info = mixerMasterProp[SND_PCM_STREAM_PLAYBACK].mInfo;
-    if (!info || !info->elem) return INVALID_OPERATION;
-
-    long minVol = info->min;
-    long maxVol = info->max;
-
-    // Make sure volume is between bounds.
-    long vol = minVol + volume * (maxVol - minVol);
-    if (vol > maxVol) vol = maxVol;
-    if (vol < minVol) vol = minVol;
-
-    info->volume = vol;
-    snd_mixer_selem_set_playback_volume_all (info->elem, vol);
-
-    return NO_ERROR;
-}
-
-status_t ALSAMixer::setMasterGain(float gain)
-{
-    mixer_info_t *info = mixerMasterProp[SND_PCM_STREAM_CAPTURE].mInfo;
-    if (!info || !info->elem) return INVALID_OPERATION;
-
-    long minVol = info->min;
-    long maxVol = info->max;
-
-    // Make sure volume is between bounds.
-    long vol = minVol + gain * (maxVol - minVol);
-    if (vol > maxVol) vol = maxVol;
-    if (vol < minVol) vol = minVol;
-
-    info->volume = vol;
-    snd_mixer_selem_set_capture_volume_all (info->elem, vol);
-
-    return NO_ERROR;
-}
-
-status_t ALSAMixer::setVolume(uint32_t device, float volume)
-{
-    for (int j = 0; mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes; j++)
-        if (mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes & device) {
-
-            mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_PLAYBACK].mInfo;
-            if (!info || !info->elem) return INVALID_OPERATION;
-
-            long minVol = info->min;
-            long maxVol = info->max;
-
-            // Make sure volume is between bounds.
-            long vol = minVol + volume * (maxVol - minVol);
-            if (vol > maxVol) vol = maxVol;
-            if (vol < minVol) vol = minVol;
-
-            info->volume = vol;
-            snd_mixer_selem_set_playback_volume_all (info->elem, vol);
-        }
-
-    return NO_ERROR;
-}
-
-status_t ALSAMixer::setGain(uint32_t device, float gain)
-{
-    for (int j = 0; mixerProp[j][SND_PCM_STREAM_CAPTURE].routes; j++)
-        if (mixerProp[j][SND_PCM_STREAM_CAPTURE].routes & device) {
-
-            mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_CAPTURE].mInfo;
-            if (!info || !info->elem) return INVALID_OPERATION;
-
-            long minVol = info->min;
-            long maxVol = info->max;
-
-            // Make sure volume is between bounds.
-            long vol = minVol + gain * (maxVol - minVol);
-            if (vol > maxVol) vol = maxVol;
-            if (vol < minVol) vol = minVol;
-
-            info->volume = vol;
-            snd_mixer_selem_set_capture_volume_all (info->elem, vol);
-        }
-
-    return NO_ERROR;
-}
-
-status_t ALSAMixer::setCaptureMuteState(uint32_t device, bool state)
-{
-    for (int j = 0; mixerProp[j][SND_PCM_STREAM_CAPTURE].routes; j++)
-        if (mixerProp[j][SND_PCM_STREAM_CAPTURE].routes & device) {
-
-            mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_CAPTURE].mInfo;
-            if (!info || !info->elem) return INVALID_OPERATION;
-
-            if (snd_mixer_selem_has_capture_switch (info->elem)) {
-
-                int err = snd_mixer_selem_set_capture_switch_all (info->elem, static_cast<int>(!state));
-                if (err < 0) {
-                    LOGE("Unable to %s capture mixer switch %s",
-                        state ? "enable" : "disable", info->name);
-                    return INVALID_OPERATION;
-                }
-            }
-
-            info->mute = state;
-        }
-
-    return NO_ERROR;
-}
-
-status_t ALSAMixer::getCaptureMuteState(uint32_t device, bool *state)
-{
-    if (! state) return BAD_VALUE;
-
-    for (int j = 0; mixerProp[j][SND_PCM_STREAM_CAPTURE].routes; j++)
-        if (mixerProp[j][SND_PCM_STREAM_CAPTURE].routes & device) {
-
-            mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_CAPTURE].mInfo;
-            if (!info || !info->elem) return INVALID_OPERATION;
-
-            *state = info->mute;
-            return NO_ERROR;
-        }
-
-    return BAD_VALUE;
-}
-
-status_t ALSAMixer::setPlaybackMuteState(uint32_t device, bool state)
-{
-
-    LOGE("\n set playback mute device %d, state %d \n", device,state);
-
-    for (int j = 0; mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes; j++)
-        if (mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes & device) {
-
-            mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_PLAYBACK].mInfo;
-            if (!info || !info->elem) return INVALID_OPERATION;
-
-            if (snd_mixer_selem_has_playback_switch (info->elem)) {
-
-                int err = snd_mixer_selem_set_playback_switch_all (info->elem, static_cast<int>(!state));
-                if (err < 0) {
-                    LOGE("Unable to %s playback mixer switch %s",
-                        state ? "enable" : "disable", info->name);
-                    return INVALID_OPERATION;
-                }
-            }
-
-            info->mute = state;
-        }
-
-    return NO_ERROR;
-}
-
-status_t ALSAMixer::getPlaybackMuteState(uint32_t device, bool *state)
-{
-    if (! state) return BAD_VALUE;
-
-    for (int j = 0; mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes; j++)
-        if (mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes & device) {
-
-            mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_PLAYBACK].mInfo;
-            if (!info || !info->elem) return INVALID_OPERATION;
-
-            *state = info->mute;
-            return NO_ERROR;
-        }
-
-    return BAD_VALUE;
-}
-
-// ----------------------------------------------------------------------------
-
-ALSAControl::ALSAControl(const char *device)
-{
-    snd_ctl_open(&mHandle, device, 0);
-}
-
-ALSAControl::~ALSAControl()
-{
-    if (mHandle) snd_ctl_close(mHandle);
-}
-
-status_t ALSAControl::get(const char *name, unsigned int &value, int index)
-{
-    if (!mHandle) return NO_INIT;
-
-    snd_ctl_elem_id_t *id;
-    snd_ctl_elem_info_t *info;
-    snd_ctl_elem_value_t *control;
-
-    snd_ctl_elem_id_alloca(&id);
-    snd_ctl_elem_info_alloca(&info);
-    snd_ctl_elem_value_alloca(&control);
-
-    snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER);
-    snd_ctl_elem_id_set_name(id, name);
-    snd_ctl_elem_info_set_id(info, id);
-
-    int ret = snd_ctl_elem_info(mHandle, info);
-    if (ret < 0) return BAD_VALUE;
-
-    snd_ctl_elem_info_get_id(info, id);
-    snd_ctl_elem_type_t type = snd_ctl_elem_info_get_type(info);
-    unsigned int count = snd_ctl_elem_info_get_count(info);
-    if ((unsigned int)index >= count) return BAD_VALUE;
-
-    snd_ctl_elem_value_set_id(control, id);
-
-    ret = snd_ctl_elem_read(mHandle, control);
-    if (ret < 0) return BAD_VALUE;
-
-    switch (type) {
-        case SND_CTL_ELEM_TYPE_BOOLEAN:
-            value = snd_ctl_elem_value_get_boolean(control, index);
-            break;
-        case SND_CTL_ELEM_TYPE_INTEGER:
-            value = snd_ctl_elem_value_get_integer(control, index);
-            break;
-        case SND_CTL_ELEM_TYPE_INTEGER64:
-            value = snd_ctl_elem_value_get_integer64(control, index);
-            break;
-        case SND_CTL_ELEM_TYPE_ENUMERATED:
-            value = snd_ctl_elem_value_get_enumerated(control, index);
-            break;
-        case SND_CTL_ELEM_TYPE_BYTES:
-            value = snd_ctl_elem_value_get_byte(control, index);
-            break;
-        default:
-            return BAD_VALUE;
-    }
-
-    return NO_ERROR;
-}
-
-status_t ALSAControl::set(const char *name, unsigned int value, int index)
-{
-    if (!mHandle) return NO_INIT;
-
-    snd_ctl_elem_id_t *id;
-    snd_ctl_elem_info_t *info;
-    snd_ctl_elem_value_t *control;
-
-    snd_ctl_elem_id_alloca(&id);
-    snd_ctl_elem_info_alloca(&info);
-    snd_ctl_elem_value_alloca(&control);
-
-    snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER);
-    snd_ctl_elem_id_set_name(id, name);
-    snd_ctl_elem_info_set_id(info, id);
-
-    int ret = snd_ctl_elem_info(mHandle, info);
-    if (ret < 0) return BAD_VALUE;
-
-    snd_ctl_elem_info_get_id(info, id);
-    snd_ctl_elem_type_t type = snd_ctl_elem_info_get_type(info);
-    unsigned int count = snd_ctl_elem_info_get_count(info);
-
-    if (index >= (int)count) return BAD_VALUE;
-
-    if (index == -1)
-        index = 0; // Range over all of them
-    else
-        count = index + 1; // Just do the one specified
-
-    snd_ctl_elem_value_set_id(control, id);
-
-    for (unsigned int i = index; i < count; i++)
-        switch (type) {
-            case SND_CTL_ELEM_TYPE_BOOLEAN:
-                snd_ctl_elem_value_set_boolean(control, i, value);
-                break;
-            case SND_CTL_ELEM_TYPE_INTEGER:
-                snd_ctl_elem_value_set_integer(control, i, value);
-                break;
-            case SND_CTL_ELEM_TYPE_INTEGER64:
-                snd_ctl_elem_value_set_integer64(control, i, value);
-                break;
-            case SND_CTL_ELEM_TYPE_ENUMERATED:
-                snd_ctl_elem_value_set_enumerated(control, i, value);
-                break;
-            case SND_CTL_ELEM_TYPE_BYTES:
-                snd_ctl_elem_value_set_byte(control, i, value);
-                break;
-            default:
-                break;
-        }
-    ret = snd_ctl_elem_write(mHandle, control);
-    return (ret < 0) ? BAD_VALUE : NO_ERROR;
-}
-
-//------------------------------------------------------------------------------
-// Downsampler
-//------------------------------------------------------------------------------
-
-/*
- * 2.30 fixed point FIR filter coefficients for conversion 44100 -> 22050.
- * (Works equivalently for 22010 -> 11025 or any other halving, of course.)
- *
- * Transition band from about 18 kHz, passband ripple < 0.1 dB,
- * stopband ripple at about -55 dB, linear phase.
- *
- * Design and display in MATLAB or Octave using:
- *
- * filter = fir1(19, 0.5); filter = round(filter * 2**30); freqz(filter * 2**-30);
- */
-static const int32_t filter_22khz_coeff[] = {
-    2089257, 2898328, -5820678, -10484531,
-    19038724, 30542725, -50469415, -81505260,
-    152544464, 478517512, 478517512, 152544464,
-    -81505260, -50469415, 30542725, 19038724,
-    -10484531, -5820678, 2898328, 2089257,
-};
-#define NUM_COEFF_22KHZ (sizeof(filter_22khz_coeff) / sizeof(filter_22khz_coeff[0]))
-#define OVERLAP_22KHZ (NUM_COEFF_22KHZ - 2)
-
-/*
- * Convolution of signals A and reverse(B). (In our case, the filter response
- * is symmetric, so the reversing doesn't matter.)
- * A is taken to be in 0.16 fixed-point, and B is taken to be in 2.30 fixed-point.
- * The answer will be in 16.16 fixed-point, unclipped.
- *
- * This function would probably be the prime candidate for SIMD conversion if
- * you want more speed.
- */
-int32_t fir_convolve(const int16_t* a, const int32_t* b, int num_samples)
-{
-        int32_t sum = 1 << 13;
-        for (int i = 0; i < num_samples; ++i) {
-                sum += a[i] * (b[i] >> 16);
-        }
-        return sum >> 14;
-}
-
-/* Clip from 16.16 fixed-point to 0.16 fixed-point. */
-int16_t clip(int32_t x)
-{
-    if (x < -32768) {
-        return -32768;
-    } else if (x > 32767) {
-        return 32767;
-    } else {
-        return x;
-    }
-}
-
-/*
- * Convert a chunk from 44 kHz to 22 kHz. Will update num_samples_in and num_samples_out
- * accordingly, since it may leave input samples in the buffer due to overlap.
- *
- * Input and output are taken to be in 0.16 fixed-point.
- */
-void resample_2_1(int16_t* input, int16_t* output, int* num_samples_in, int* num_samples_out)
-{
-    if (*num_samples_in < (int)NUM_COEFF_22KHZ) {
-        *num_samples_out = 0;
-        return;
-    }
-
-    int odd_smp = *num_samples_in & 0x1;
-    int num_samples = *num_samples_in - odd_smp - OVERLAP_22KHZ;
-
-    for (int i = 0; i < num_samples; i += 2) {
-            output[i / 2] = clip(fir_convolve(input + i, filter_22khz_coeff, NUM_COEFF_22KHZ));
-    }
-
-    memmove(input, input + num_samples, (OVERLAP_22KHZ + odd_smp) * sizeof(*input));
-    *num_samples_out = num_samples / 2;
-    *num_samples_in = OVERLAP_22KHZ + odd_smp;
-}
-
-/*
- * 2.30 fixed point FIR filter coefficients for conversion 22050 -> 16000,
- * or 11025 -> 8000.
- *
- * Transition band from about 14 kHz, passband ripple < 0.1 dB,
- * stopband ripple at about -50 dB, linear phase.
- *
- * Design and display in MATLAB or Octave using:
- *
- * filter = fir1(23, 16000 / 22050); filter = round(filter * 2**30); freqz(filter * 2**-30);
- */
-static const int32_t filter_16khz_coeff[] = {
-    2057290, -2973608, 1880478, 4362037,
-    -14639744, 18523609, -1609189, -38502470,
-    78073125, -68353935, -59103896, 617555440,
-    617555440, -59103896, -68353935, 78073125,
-    -38502470, -1609189, 18523609, -14639744,
-    4362037, 1880478, -2973608, 2057290,
-};
-#define NUM_COEFF_16KHZ (sizeof(filter_16khz_coeff) / sizeof(filter_16khz_coeff[0]))
-#define OVERLAP_16KHZ (NUM_COEFF_16KHZ - 1)
-
-/*
- * Convert a chunk from 22 kHz to 16 kHz. Will update num_samples_in and
- * num_samples_out accordingly, since it may leave input samples in the buffer
- * due to overlap.
- *
- * This implementation is rather ad-hoc; it first low-pass filters the data
- * into a temporary buffer, and then converts chunks of 441 input samples at a
- * time into 320 output samples by simple linear interpolation. A better
- * implementation would use a polyphase filter bank to do these two operations
- * in one step.
- *
- * Input and output are taken to be in 0.16 fixed-point.
- */
-
-#define RESAMPLE_16KHZ_SAMPLES_IN 441
-#define RESAMPLE_16KHZ_SAMPLES_OUT 320
-
-void resample_441_320(int16_t* input, int16_t* output, int* num_samples_in, int* num_samples_out)
-{
-    const int num_blocks = (*num_samples_in - OVERLAP_16KHZ) / RESAMPLE_16KHZ_SAMPLES_IN;
-    if (num_blocks < 1) {
-        *num_samples_out = 0;
-        return;
-    }
-
-    for (int i = 0; i < num_blocks; ++i) {
-        uint32_t tmp[RESAMPLE_16KHZ_SAMPLES_IN];
-        for (int j = 0; j < RESAMPLE_16KHZ_SAMPLES_IN; ++j) {
-            tmp[j] = fir_convolve(input + i * RESAMPLE_16KHZ_SAMPLES_IN + j,
-                          filter_16khz_coeff,
-                          NUM_COEFF_16KHZ);
-        }
-
-        const float step_float = (float)RESAMPLE_16KHZ_SAMPLES_IN / (float)RESAMPLE_16KHZ_SAMPLES_OUT;
-
-        uint32_t in_sample_num = 0;   // 16.16 fixed point
-        const uint32_t step = (uint32_t)(step_float * 65536.0f + 0.5f);  // 16.16 fixed point
-        for (int j = 0; j < RESAMPLE_16KHZ_SAMPLES_OUT; ++j, in_sample_num += step) {
-            const uint32_t whole = in_sample_num >> 16;
-            const uint32_t frac = (in_sample_num & 0xffff);  // 0.16 fixed point
-            const int32_t s1 = tmp[whole];
-            const int32_t s2 = tmp[whole + 1];
-            *output++ = clip(s1 + (((s2 - s1) * (int32_t)frac) >> 16));
-        }
-    }
-
-    const int samples_consumed = num_blocks * RESAMPLE_16KHZ_SAMPLES_IN;
-    memmove(input, input + samples_consumed, (*num_samples_in - samples_consumed) * sizeof(*input));
-    *num_samples_in -= samples_consumed;
-    *num_samples_out = RESAMPLE_16KHZ_SAMPLES_OUT * num_blocks;
-}
-
-
-ALSADownsampler::ALSADownsampler(uint32_t outSampleRate,
-                                    uint32_t channelCount,
-                                    uint32_t frameCount,
-                                    ALSABufferProvider* provider)
-    :  mStatus(NO_INIT), mProvider(provider), mSampleRate(outSampleRate),
-       mChannelCount(channelCount), mFrameCount(frameCount),
-       mInLeft(NULL), mInRight(NULL), mTmpLeft(NULL), mTmpRight(NULL),
-       mTmp2Left(NULL), mTmp2Right(NULL), mOutLeft(NULL), mOutRight(NULL)
-
-{
-    LOGV("ALSADownsampler() cstor SR %d channels %d frames %d",
-         mSampleRate, mChannelCount, mFrameCount);
-
-    if (mSampleRate != 8000 && mSampleRate != 11025 && mSampleRate != 16000 &&
-            mSampleRate != 22050) {
-        LOGW("ALSADownsampler cstor: bad sampling rate: %d", mSampleRate);
-        return;
-    }
-
-    mInLeft = new int16_t[mFrameCount];
-    mInRight = new int16_t[mFrameCount];
-    mTmpLeft = new int16_t[mFrameCount];
-    mTmpRight = new int16_t[mFrameCount];
-    mTmp2Left = new int16_t[mFrameCount];
-    mTmp2Right = new int16_t[mFrameCount];
-    mOutLeft = new int16_t[mFrameCount];
-    mOutRight = new int16_t[mFrameCount];
-
-    mStatus = NO_ERROR;
-}
-
-ALSADownsampler::~ALSADownsampler()
-{
-    if (mInLeft) delete[] mInLeft;
-    if (mInRight) delete[] mInRight;
-    if (mTmpLeft) delete[] mTmpLeft;
-    if (mTmpRight) delete[] mTmpRight;
-    if (mTmp2Left) delete[] mTmp2Left;
-    if (mTmp2Right) delete[] mTmp2Right;
-    if (mOutLeft) delete[] mOutLeft;
-    if (mOutRight) delete[] mOutRight;
-}
-
-void ALSADownsampler::reset()
-{
-    mInInBuf = 0;
-    mInTmpBuf = 0;
-    mInTmp2Buf = 0;
-    mOutBufPos = 0;
-    mInOutBuf = 0;
-}
-
-
-int ALSADownsampler::resample(int16_t* out, size_t *outFrameCount)
-{
-    if (mStatus != NO_ERROR) {
-        return mStatus;
-    }
-
-    if (out == NULL || outFrameCount == NULL) {
-        return BAD_VALUE;
-    }
-
-    int16_t *outLeft = mTmp2Left;
-    int16_t *outRight = mTmp2Left;
-    if (mSampleRate == 22050) {
-        outLeft = mTmpLeft;
-        outRight = mTmpRight;
-    } else if (mSampleRate == 8000){
-        outLeft = mOutLeft;
-        outRight = mOutRight;
-    }
-
-    int outFrames = 0;
-    int remaingFrames = *outFrameCount;
-
-    if (mInOutBuf) {
-        int frames = (remaingFrames > mInOutBuf) ? mInOutBuf : remaingFrames;
-
-        for (int i = 0; i < frames; ++i) {
-            out[i] = outLeft[mOutBufPos + i];
-        }
-        if (mChannelCount == 2) {
-            for (int i = 0; i < frames; ++i) {
-                out[i * 2] = outLeft[mOutBufPos + i];
-                out[i * 2 + 1] = outRight[mOutBufPos + i];
-            }
-        }
-        remaingFrames -= frames;
-        mInOutBuf -= frames;
-        mOutBufPos += frames;
-        outFrames += frames;
-    }
-
-    while (remaingFrames) {
-        LOGW_IF((mInOutBuf != 0), "mInOutBuf should be 0 here");
-
-        ALSABufferProvider::Buffer buf;
-        buf.frameCount =  mFrameCount - mInInBuf;
-        int ret = mProvider->getNextBuffer(&buf);
-        if (buf.raw == NULL) {
-            *outFrameCount = outFrames;
-            return ret;
-        }
-
-        for (size_t i = 0; i < buf.frameCount; ++i) {
-            mInLeft[i + mInInBuf] = buf.i16[i];
-        }
-        if (mChannelCount == 2) {
-            for (size_t i = 0; i < buf.frameCount; ++i) {
-                mInLeft[i + mInInBuf] = buf.i16[i * 2];
-                mInRight[i + mInInBuf] = buf.i16[i * 2 + 1];
-            }
-        }
-        mInInBuf += buf.frameCount;
-        mProvider->releaseBuffer(&buf);
-
-        /* 44010 -> 22050 */
-        {
-            int samples_in_left = mInInBuf;
-            int samples_out_left;
-            resample_2_1(mInLeft, mTmpLeft + mInTmpBuf, &samples_in_left, &samples_out_left);
-
-            if (mChannelCount == 2) {
-                int samples_in_right = mInInBuf;
-                int samples_out_right;
-                resample_2_1(mInRight, mTmpRight + mInTmpBuf, &samples_in_right, &samples_out_right);
-            }
-
-            mInInBuf = samples_in_left;
-            mInTmpBuf += samples_out_left;
-            mInOutBuf = samples_out_left;
-        }
-
-        if (mSampleRate == 11025 || mSampleRate == 8000) {
-            /* 22050 - > 11025 */
-            int samples_in_left = mInTmpBuf;
-            int samples_out_left;
-            resample_2_1(mTmpLeft, mTmp2Left + mInTmp2Buf, &samples_in_left, &samples_out_left);
-
-            if (mChannelCount == 2) {
-                int samples_in_right = mInTmpBuf;
-                int samples_out_right;
-                resample_2_1(mTmpRight, mTmp2Right + mInTmp2Buf, &samples_in_right, &samples_out_right);
-            }
-
-
-            mInTmpBuf = samples_in_left;
-            mInTmp2Buf += samples_out_left;
-            mInOutBuf = samples_out_left;
-
-            if (mSampleRate == 8000) {
-                /* 11025 -> 8000*/
-                int samples_in_left = mInTmp2Buf;
-                int samples_out_left;
-                resample_441_320(mTmp2Left, mOutLeft, &samples_in_left, &samples_out_left);
-
-                if (mChannelCount == 2) {
-                    int samples_in_right = mInTmp2Buf;
-                    int samples_out_right;
-                    resample_441_320(mTmp2Right, mOutRight, &samples_in_right, &samples_out_right);
-                }
-
-                mInTmp2Buf = samples_in_left;
-                mInOutBuf = samples_out_left;
-            } else {
-                mInTmp2Buf = 0;
-            }
-
-        } else if (mSampleRate == 16000) {
-            /* 22050 -> 16000*/
-            int samples_in_left = mInTmpBuf;
-            int samples_out_left;
-            resample_441_320(mTmpLeft, mTmp2Left, &samples_in_left, &samples_out_left);
-
-            if (mChannelCount == 2) {
-                int samples_in_right = mInTmpBuf;
-                int samples_out_right;
-                resample_441_320(mTmpRight, mTmp2Right, &samples_in_right, &samples_out_right);
-            }
-
-            mInTmpBuf = samples_in_left;
-            mInOutBuf = samples_out_left;
-        } else {
-            mInTmpBuf = 0;
-        }
-
-        int frames = (remaingFrames > mInOutBuf) ? mInOutBuf : remaingFrames;
-
-        for (int i = 0; i < frames; ++i) {
-            out[outFrames + i] = outLeft[i];
-        }
-        if (mChannelCount == 2) {
-            for (int i = 0; i < frames; ++i) {
-                out[(outFrames + i) * 2] = outLeft[i];
-                out[(outFrames + i) * 2 + 1] = outRight[i];
-            }
-        }
-        remaingFrames -= frames;
-        outFrames += frames;
-        mOutBufPos = frames;
-        mInOutBuf -= frames;
-    }
-
-    return 0;
-}
-
-};        // namespace android
-
diff --git a/libaudio/AudioHardwareALSA.h b/libaudio/AudioHardwareALSA.h
deleted file mode 100755
index c3b6caa..0000000
--- a/libaudio/AudioHardwareALSA.h
+++ /dev/null
@@ -1,432 +0,0 @@
-/* AudioHardwareALSA.h
- **
- ** Copyright 2008, Wind River Systems
- **
- ** 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_ALSA_H
-#define ANDROID_AUDIO_HARDWARE_ALSA_H
-
-#include <stdint.h>
-#include <sys/types.h>
-#include <alsa/asoundlib.h>
-
-#include <hardware_legacy/AudioHardwareBase.h>
-
-// sangsu fix : headers for IPC
-#include "secril-client.h"
-
-#ifndef ALSA_DEFAULT_SAMPLE_RATE
-#define ALSA_DEFAULT_SAMPLE_RATE 44100 // in Hz
-#endif
-
-#define DEFAULT_SAMPLE_RATE ALSA_DEFAULT_SAMPLE_RATE
-
-#define PLAYBACK    0
-#define PERIOD_SZ_PLAYBACK   1024
-#define PERIODS_PLAYBACK     4
-#define BUFFER_SZ_PLAYBACK   (PERIODS_PLAYBACK * PERIOD_SZ_PLAYBACK)
-#define LATENCY_PLAYBACK_MS  ((BUFFER_SZ_PLAYBACK * 1000 / DEFAULT_SAMPLE_RATE) * 1000)
-
-#define CAPTURE     1
-#define PERIOD_SZ_CAPTURE   1024
-#define PERIODS_CAPTURE     4
-#define BUFFER_SZ_CAPTURE   (PERIODS_CAPTURE * PERIOD_SZ_CAPTURE)
-#define LATENCY_CAPTURE_MS  ((BUFFER_SZ_CAPTURE * 1000 / DEFAULT_SAMPLE_RATE) * 1000)
-
-//Recognition param
-#define RECOGNITION_OFF	0
-#define RECOGNITION_ON	1
-
-namespace android
-{
-
-    class AudioHardwareALSA;
-
-    // ----------------------------------------------------------------------------
-
-    class ALSAMixer
-    {
-        public:
-                                    ALSAMixer();
-            virtual                ~ALSAMixer();
-
-            bool                    isValid() { return !!mMixer[SND_PCM_STREAM_PLAYBACK]; }
-            status_t                setMasterVolume(float volume);
-            status_t                setMasterGain(float gain);
-
-            status_t                setVolume(uint32_t device, float volume);
-            status_t                setGain(uint32_t device, float gain);
-
-            status_t                setCaptureMuteState(uint32_t device, bool state);
-            status_t                getCaptureMuteState(uint32_t device, bool *state);
-            status_t                setPlaybackMuteState(uint32_t device, bool state);
-            status_t                getPlaybackMuteState(uint32_t device, bool *state);
-
-        private:
-            snd_mixer_t            *mMixer[SND_PCM_STREAM_LAST+1];
-    };
-
-    class ALSAControl
-    {
-        public:
-                                    ALSAControl(const char *device = "default");
-            virtual                ~ALSAControl();
-
-            status_t                get(const char *name, unsigned int &value, int index = 0);
-            status_t                set(const char *name, unsigned int value, int index = -1);
-
-        private:
-            snd_ctl_t              *mHandle;
-    };
-
-    class ALSAStreamOps
-    {
-     public:
-                uint32_t            device() { return mDevice; }
-                void                close();
-
-        protected:
-            friend class AudioStreamOutALSA;
-            friend class AudioStreamInALSA;
-
-            struct StreamDefaults
-            {
-                const char *        devicePrefix;
-                snd_pcm_stream_t    direction;       // playback or capture
-                snd_pcm_format_t    format;
-                int                 channelCount;
-                uint32_t            sampleRate;
-                uint32_t            bufferRatio;
-                unsigned int        latency;         // Delay in usec
-                unsigned int        bufferSize;      // Size of sample buffer
-                unsigned int        periodSize;      // Size of sample buffer
-            };
-
-                                    ALSAStreamOps();
-            virtual                ~ALSAStreamOps();
-
-            status_t                set(int *format,
-                                        uint32_t *channels,
-                                        uint32_t *rate);
-            virtual uint32_t        sampleRate() const;
-            status_t                sampleRate(uint32_t rate);
-            virtual size_t          bufferSize() const;
-            virtual int             format() const;
-                    int             getAndroidFormat(snd_pcm_format_t format);
-
-            virtual uint32_t        channels() const;
-                    int             channelCount() const;
-                    status_t        channelCount(int channelCount);
-                    uint32_t        getAndroidChannels(int channelCount) const;
-
-            status_t                open(int mode, uint32_t device);
-            status_t                setSoftwareParams();
-            status_t                setPCMFormat(snd_pcm_format_t format);
-            status_t                setHardwareResample(bool resample);
-
-            const char             *streamName();
-            status_t                setDevice(int mode, uint32_t device, uint32_t audio_mode);
-
-            const char             *deviceName(int mode, uint32_t device);
-
-            void                    setStreamDefaults(StreamDefaults *dev) {
-                mDefaults = dev;
-            }
-
-            Mutex                   mLock;
-
-        private:
-            snd_pcm_t              *mHandle;
-            snd_pcm_hw_params_t    *mHardwareParams;
-            snd_pcm_sw_params_t    *mSoftwareParams;
-            uint32_t                mDevice;
-
-            StreamDefaults         *mDefaults;
-    };
-
-    // ----------------------------------------------------------------------------
-
-    class AudioStreamOutALSA : public AudioStreamOut, public ALSAStreamOps
-    {
-        public:
-                                    AudioStreamOutALSA(AudioHardwareALSA *parent);
-            virtual                ~AudioStreamOutALSA();
-
-
-            status_t                set(int *format,
-                                        uint32_t *channelCount,
-                                        uint32_t *sampleRate){
-                return ALSAStreamOps::set(format, channelCount, sampleRate);
-            }
-
-            virtual uint32_t        sampleRate() const {
-                return ALSAStreamOps::sampleRate();
-            }
-
-            virtual size_t          bufferSize() const
-            {
-                return ALSAStreamOps::bufferSize();
-            }
-
-            virtual uint32_t        channels() const
-            {
-                return ALSAStreamOps::channels();
-            }
-
-            virtual int             format() const
-            {
-                return ALSAStreamOps::format();
-            }
-
-            virtual uint32_t        latency() const;
-
-            virtual ssize_t         write(const void *buffer, size_t bytes);
-            virtual status_t        dump(int fd, const Vector<String16>& args);
-                    status_t        setDevice(int mode, uint32_t newDevice, uint32_t audio_mode,
-                                              bool force = false);
-            virtual status_t        setVolume(float left, float right); //Tushar: New arch
-
-            status_t                setVolume(float volume);
-
-            status_t                standby();
-
-            virtual status_t        setParameters(const String8& keyValuePairs);
-            virtual String8         getParameters(const String8& keys);
-
-            virtual status_t        getRenderPosition(uint32_t *dspFrames);
-                    bool            isActive() { return mPowerLock; }
-
-        private:
-            AudioHardwareALSA      *mParent;
-            bool                    mPowerLock;
-    };
-
-    class ALSADownsampler;
-
-    class ALSABufferProvider
-    {
-    public:
-
-        struct Buffer {
-            union {
-                void*       raw;
-                short*      i16;
-                int8_t*     i8;
-            };
-            size_t frameCount;
-        };
-
-        virtual ~ALSABufferProvider() {}
-
-        virtual status_t getNextBuffer(Buffer* buffer) = 0;
-        virtual void releaseBuffer(Buffer* buffer) = 0;
-    };
-
-    class AudioStreamInALSA : public AudioStreamIn, public ALSAStreamOps, public ALSABufferProvider
-    {
-        public:
-                                    AudioStreamInALSA(AudioHardwareALSA *parent);
-            virtual                ~AudioStreamInALSA();
-
-            status_t                set(int *format,
-                                        uint32_t *channelCount,
-                                        uint32_t *sampleRate);
-
-            virtual uint32_t        sampleRate() const {
-                return ALSAStreamOps::sampleRate();
-            }
-
-            virtual size_t          bufferSize() const
-            {
-                return ALSAStreamOps::bufferSize();
-            }
-
-            virtual uint32_t        channels() const
-            {
-                return ALSAStreamOps::channels();
-            }
-
-            virtual int             format() const
-            {
-                return ALSAStreamOps::format();
-            }
-
-            virtual ssize_t         read(void* buffer, ssize_t bytes);
-            virtual status_t        dump(int fd, const Vector<String16>& args);
-                    status_t        setDevice(int mode, uint32_t newDevice, uint32_t audio_mode,
-                                              bool force = false);
-
-            virtual status_t        setGain(float gain);
-
-            virtual status_t        standby();
-
-            virtual status_t        setParameters(const String8& keyValuePairs);
-            virtual String8         getParameters(const String8& keys);
-
-            virtual unsigned int    getInputFramesLost() const { return 0; }
-
-                    bool            isActive() { return mPowerLock; }
-
-            // ALSABufferProvider
-            virtual status_t getNextBuffer(ALSABufferProvider::Buffer* buffer);
-            virtual void releaseBuffer(ALSABufferProvider::Buffer* buffer);
-
-        private:
-            AudioHardwareALSA      *mParent;
-            bool                    mPowerLock;
-            ALSADownsampler *mDownSampler;
-            status_t mReadStatus;
-            size_t mInPcmInBuf;
-            int16_t *mPcmIn;
-    };
-
-    class AudioHardwareALSA : public AudioHardwareBase
-    {
-        public:
-                                    AudioHardwareALSA();
-            virtual                ~AudioHardwareALSA();
-
-            /**
-             * check to see if the audio hardware interface has been initialized.
-             * return status based on values defined in include/utils/Errors.h
-             */
-            virtual status_t        initCheck();
-
-            /** set the audio volume of a voice call. Range is between 0.0 and 1.0 */
-            virtual status_t        setVoiceVolume(float volume);
-
-            virtual status_t        setMode(int mode);
-
-            /**
-             * set the audio volume for all audio activities other than voice call.
-             * Range between 0.0 and 1.0. If any value other than NO_ERROR is returned,
-             * the software mixer will emulate this capability.
-             */
-            virtual status_t        setMasterVolume(float volume);
-
-            // mic mute
-            virtual status_t        setMicMute(bool state);
-            virtual status_t        getMicMute(bool* state);
-            virtual size_t          getInputBufferSize(uint32_t sampleRate,
-                                                       int format,
-                                                       int channelCount);
-
-            /** This method creates and opens the audio hardware output stream */
-            virtual AudioStreamOut* openOutputStream(uint32_t devices,
-                                                     int *format = 0,
-                                                     uint32_t *channels = 0,
-                                                     uint32_t *sampleRate = 0,
-                                                     status_t *status = 0);
-            virtual void            closeOutputStream(AudioStreamOut* out);
-
-            /** This method creates and opens the audio hardware input stream */
-            virtual AudioStreamIn*  openInputStream(uint32_t devices,
-                                                   int *format,
-                                                   uint32_t *channels,
-                                                   uint32_t *sampleRate,
-                                                   status_t *status,
-                                                   AudioSystem::audio_in_acoustics acoustics);
-            virtual void            closeInputStream(AudioStreamIn* in);
-
-            static uint32_t         checkInputSampleRate(uint32_t sampleRate);
-            static const uint32_t   inputSamplingRates[];
-            static uint32_t         bufferRatio(uint32_t samplingRate);
-
-                   int              mode() { return mMode; }
-                   Mutex&           lock() { return mLock; }
-
-                   int              setVoiceRecordGain(bool enable);
-                   int              setVoiceRecordGain_l(bool enable);
-
-            virtual status_t        setParameters(const String8& keyValuePairs);
-
-        protected:
-            /**
-             * doRouting actually initiates the routing. A call to setRouting
-             * or setMode may result in a routing change. The generic logic calls
-             * doRouting when required. If the device has any special requirements these
-             * methods can be overriden.
-             */
-                    status_t        doRouting(uint32_t device, bool force = false);
-                    status_t        doRouting_l(uint32_t device, bool force = false);
-
-            virtual status_t        dump(int fd, const Vector<String16>& args);
-
-            friend class AudioStreamOutALSA;
-            friend class AudioStreamInALSA;
-
-            ALSAMixer              *mMixer;
-            AudioStreamOutALSA     *mOutput;
-            AudioStreamInALSA      *mInput;
-
-        private:
-            Mutex                   mLock;
-            void                   *mSecRilLibHandle;
-            HRilClient              mRilClient;
-            bool                    mVrModeEnabled;
-            bool                    mActivatedCP;
-            bool                    mBluetoothECOff;
-
-            HRilClient (*openClientRILD)  (void);
-            int        (*disconnectRILD)  (HRilClient);
-            int        (*closeClientRILD) (HRilClient);
-            int        (*isConnectedRILD) (HRilClient);
-            int        (*connectRILD)     (HRilClient);
-            int        (*setCallVolume)   (HRilClient, SoundType, int);
-            int        (*setCallAudioPath)(HRilClient, AudioPath);
-            int        (*setCallClockSync)(HRilClient, SoundClockCondition);
-
-            void                    loadRILD(void);
-            status_t                connectRILDIfRequired(void);
-            void                    setBluetoothNrEcOnOff(bool disable);
-
-    };
-
-    class ALSADownsampler {
-    public:
-        ALSADownsampler(uint32_t outSampleRate,
-                  uint32_t channelCount,
-                  uint32_t frameCount,
-                  ALSABufferProvider* provider);
-
-        virtual ~ALSADownsampler();
-
-                void reset();
-                status_t initCheck() { return mStatus; }
-                int resample(int16_t* out, size_t *outFrameCount);
-
-    private:
-        status_t    mStatus;
-        ALSABufferProvider* mProvider;
-        uint32_t mSampleRate;
-        uint32_t mChannelCount;
-        uint32_t mFrameCount;
-        int16_t *mInLeft;
-        int16_t *mInRight;
-        int16_t *mTmpLeft;
-        int16_t *mTmpRight;
-        int16_t *mTmp2Left;
-        int16_t *mTmp2Right;
-        int16_t *mOutLeft;
-        int16_t *mOutRight;
-        int mInInBuf;
-        int mInTmpBuf;
-        int mInTmp2Buf;
-        int mOutBufPos;
-        int mInOutBuf;
-    };
-
-};        // namespace android
-#endif    // ANDROID_AUDIO_HARDWARE_ALSA_H
diff --git a/libaudio/AudioPolicyManager.cpp b/libaudio/AudioPolicyManager.cpp
index 41c0fd0..93d70d8 100644
--- a/libaudio/AudioPolicyManager.cpp
+++ b/libaudio/AudioPolicyManager.cpp
@@ -42,4 +42,33 @@ extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
     delete interface;
 }
 
+
+status_t AudioPolicyManager::startInput(audio_io_handle_t input)
+{
+    status_t status = AudioPolicyManagerBase::startInput(input);
+
+    if (status == NO_ERROR) {
+        AudioInputDescriptor *inputDesc = mInputs.valueFor(input);
+        String8 key = String8("Input Source");
+        String8 value;
+        switch(inputDesc->mInputSource) {
+        case AUDIO_SOURCE_VOICE_RECOGNITION:
+            value = String8("Voice Recognition");
+            break;
+        case AUDIO_SOURCE_CAMCORDER:
+            value = String8("Camcorder");
+            break;
+        case AUDIO_SOURCE_DEFAULT:
+        case AUDIO_SOURCE_MIC:
+            value = String8("Default");
+        default:
+            break;
+        }
+        AudioParameter param = AudioParameter();
+        param.add(key, value);
+        mpClientInterface->setParameters(input, param.toString());
+    }
+    return status;
+}
+
 }; // namespace android
diff --git a/libaudio/AudioPolicyManager.h b/libaudio/AudioPolicyManager.h
index 03141e5..ae283db 100644
--- a/libaudio/AudioPolicyManager.h
+++ b/libaudio/AudioPolicyManager.h
@@ -34,6 +34,7 @@ public:
 
         virtual ~AudioPolicyManager() {}
 
+        virtual status_t startInput(audio_io_handle_t input);
 protected:
         // true is current platform implements a back microphone
         virtual bool hasBackMicrophone() const { return false; }
diff --git a/libaudio/NOTICE b/libaudio/NOTICE
deleted file mode 100644
index ada44e1..0000000
--- a/libaudio/NOTICE
+++ /dev/null
@@ -1,191 +0,0 @@
-
-   Copyright (c) 2005-2008, The Android Open Source Project
-   Copyright 2008 Wind River Systems
-
-   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,
-      and conversions to other media types.
-
-      "Work" shall mean the work of authorship, whether in Source or
-      Object form, made available under the License, as indicated by a
-      copyright notice that is included in or attached to the work
-      (an example is provided in the Appendix below).
-
-      "Derivative Works" shall mean any work, whether in Source or Object
-      form, that is based on (or derived from) the Work and for which the
-      editorial revisions, annotations, elaborations, or other modifications
-      represent, as a whole, an original work of authorship. For the purposes
-      of this License, Derivative Works shall not include works that remain
-      separable from, or merely link (or bind by name) to the interfaces of,
-      the Work and Derivative Works thereof.
-
-      "Contribution" shall mean any work of authorship, including
-      the original version of the Work and any modifications or additions
-      to that Work or Derivative Works thereof, that is intentionally
-      submitted to Licensor for inclusion in the Work by the copyright owner
-      or by an individual or Legal Entity authorized to submit on behalf of
-      the copyright owner. For the purposes of this definition, "submitted"
-      means any form of electronic, verbal, or written communication sent
-      to the Licensor or its representatives, including but not limited to
-      communication on electronic mailing lists, source code control systems,
-      and issue tracking systems that are managed by, or on behalf of, the
-      Licensor for the purpose of discussing and improving the Work, but
-      excluding communication that is conspicuously marked or otherwise
-      designated in writing by the copyright owner as "Not a Contribution."
-
-      "Contributor" shall mean Licensor and any individual or Legal Entity
-      on behalf of whom a Contribution has been received by Licensor and
-      subsequently incorporated within the Work.
-
-   2. Grant of Copyright License. Subject to the terms and conditions of
-      this License, each Contributor hereby grants to You a perpetual,
-      worldwide, non-exclusive, no-charge, royalty-free, irrevocable
-      copyright license to reproduce, prepare Derivative Works of,
-      publicly display, publicly perform, sublicense, and distribute the
-      Work and such Derivative Works in Source or Object form.
-
-   3. Grant of Patent License. Subject to the terms and conditions of
-      this License, each Contributor hereby grants to You a perpetual,
-      worldwide, non-exclusive, no-charge, royalty-free, irrevocable
-      (except as stated in this section) patent license to make, have made,
-      use, offer to sell, sell, import, and otherwise transfer the Work,
-      where such license applies only to those patent claims licensable
-      by such Contributor that are necessarily infringed by their
-      Contribution(s) alone or by combination of their Contribution(s)
-      with the Work to which such Contribution(s) was submitted. If You
-      institute patent litigation against any entity (including a
-      cross-claim or counterclaim in a lawsuit) alleging that the Work
-      or a Contribution incorporated within the Work constitutes direct
-      or contributory patent infringement, then any patent licenses
-      granted to You under this License for that Work shall terminate
-      as of the date such litigation is filed.
-
-   4. Redistribution. You may reproduce and distribute copies of the
-      Work or Derivative Works thereof in any medium, with or without
-      modifications, and in Source or Object form, provided that You
-      meet the following conditions:
-
-      (a) You must give any other recipients of the Work or
-          Derivative Works a copy of this License; and
-
-      (b) You must cause any modified files to carry prominent notices
-          stating that You changed the files; and
-
-      (c) You must retain, in the Source form of any Derivative Works
-          that You distribute, all copyright, patent, trademark, and
-          attribution notices from the Source form of the Work,
-          excluding those notices that do not pertain to any part of
-          the Derivative Works; and
-
-      (d) If the Work includes a "NOTICE" text file as part of its
-          distribution, then any Derivative Works that You distribute must
-          include a readable copy of the attribution notices contained
-          within such NOTICE file, excluding those notices that do not
-          pertain to any part of the Derivative Works, in at least one
-          of the following places: within a NOTICE text file distributed
-          as part of the Derivative Works; within the Source form or
-          documentation, if provided along with the Derivative Works; or,
-          within a display generated by the Derivative Works, if and
-          wherever such third-party notices normally appear. The contents
-          of the NOTICE file are for informational purposes only and
-          do not modify the License. You may add Your own attribution
-          notices within Derivative Works that You distribute, alongside
-          or as an addendum to the NOTICE text from the Work, provided
-          that such additional attribution notices cannot be construed
-          as modifying the License.
-
-      You may add Your own copyright statement to Your modifications and
-      may provide additional or different license terms and conditions
-      for use, reproduction, or distribution of Your modifications, or
-      for any such Derivative Works as a whole, provided Your use,
-      reproduction, and distribution of the Work otherwise complies with
-      the conditions stated in this License.
-
-   5. Submission of Contributions. Unless You explicitly state otherwise,
-      any Contribution intentionally submitted for inclusion in the Work
-      by You to the Licensor shall be under the terms and conditions of
-      this License, without any additional terms or conditions.
-      Notwithstanding the above, nothing herein shall supersede or modify
-      the terms of any separate license agreement you may have executed
-      with Licensor regarding such Contributions.
-
-   6. Trademarks. This License does not grant permission to use the trade
-      names, trademarks, service marks, or product names of the Licensor,
-      except as required for reasonable and customary use in describing the
-      origin of the Work and reproducing the content of the NOTICE file.
-
-   7. Disclaimer of Warranty. Unless required by applicable law or
-      agreed to in writing, Licensor provides the Work (and each
-      Contributor provides its Contributions) on an "AS IS" BASIS,
-      WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or
-      implied, including, without limitation, any warranties or conditions
-      of TITLE, NON-INFRINGEMENT, MERCHANTABILITY, or FITNESS FOR A
-      PARTICULAR PURPOSE. You are solely responsible for determining the
-      appropriateness of using or redistributing the Work and assume any
-      risks associated with Your exercise of permissions under this License.
-
-   8. Limitation of Liability. In no event and under no legal theory,
-      whether in tort (including negligence), contract, or otherwise,
-      unless required by applicable law (such as deliberate and grossly
-      negligent acts) or agreed to in writing, shall any Contributor be
-      liable to You for damages, including any direct, indirect, special,
-      incidental, or consequential damages of any character arising as a
-      result of this License or out of the use or inability to use the
-      Work (including but not limited to damages for loss of goodwill,
-      work stoppage, computer failure or malfunction, or any and all
-      other commercial damages or losses), even if such Contributor
-      has been advised of the possibility of such damages.
-
-   9. Accepting Warranty or Additional Liability. While redistributing
-      the Work or Derivative Works thereof, You may choose to offer,
-      and charge a fee for, acceptance of support, warranty, indemnity,
-      or other liability obligations and/or rights consistent with this
-      License. However, in accepting such obligations, You may act only
-      on Your own behalf and on Your sole responsibility, not on behalf
-      of any other Contributor, and only if You agree to indemnify,
-      defend, and hold each Contributor harmless for any liability
-      incurred by, or claims asserted against, such Contributor by reason
-      of your accepting any such warranty or additional liability.
-
-   END OF TERMS AND CONDITIONS
-
diff --git a/libaudio/alsa_audio.h b/libaudio/alsa_audio.h
new file mode 100644
index 0000000..3cb86d9
--- /dev/null
+++ b/libaudio/alsa_audio.h
@@ -0,0 +1,77 @@
+/*
+** Copyright 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.
+*/
+
+#ifndef _AUDIO_H_
+#define _AUDIO_H_
+
+struct pcm;
+
+#define PCM_OUT        0x00000000
+#define PCM_IN         0x10000000
+
+#define PCM_STEREO     0x00000000
+#define PCM_MONO       0x01000000
+
+#define PCM_44100HZ    0x00000000
+#define PCM_48000HZ    0x00100000
+#define PCM_8000HZ     0x00200000
+#define PCM_RATE_MASK  0x00F00000
+
+#define PCM_PERIOD_CNT_MIN 2
+#define PCM_PERIOD_CNT_SHIFT 16
+#define PCM_PERIOD_CNT_MASK (0xF << PCM_PERIOD_CNT_SHIFT)
+#define PCM_PERIOD_SZ_MIN 128
+#define PCM_PERIOD_SZ_SHIFT 12
+#define PCM_PERIOD_SZ_MASK (0xF << PCM_PERIOD_SZ_SHIFT)
+
+/* Acquire/release a pcm channel.
+ * Returns non-zero on error
+ */
+struct pcm *pcm_open(unsigned flags);
+int pcm_close(struct pcm *pcm);
+int pcm_ready(struct pcm *pcm);
+
+/* Returns a human readable reason for the last error. */
+const char *pcm_error(struct pcm *pcm);
+
+/* Returns the buffer size (int bytes) that should be used for pcm_write.
+ * This will be 1/2 of the actual fifo size.
+ */
+unsigned pcm_buffer_size(struct pcm *pcm);
+
+/* Write data to the fifo.
+ * Will start playback on the first write or on a write that
+ * occurs after a fifo underrun.
+ */
+int pcm_write(struct pcm *pcm, void *data, unsigned count);
+int pcm_read(struct pcm *pcm, void *data, unsigned count);
+
+struct mixer;
+struct mixer_ctl;
+
+struct mixer *mixer_open(void);
+void mixer_close(struct mixer *mixer);
+void mixer_dump(struct mixer *mixer);
+
+struct mixer_ctl *mixer_get_control(struct mixer *mixer,
+                                    const char *name, unsigned index);
+struct mixer_ctl *mixer_get_nth_control(struct mixer *mixer, unsigned n);
+
+int mixer_ctl_set(struct mixer_ctl *ctl, unsigned percent);
+int mixer_ctl_select(struct mixer_ctl *ctl, const char *value);
+void mixer_ctl_print(struct mixer_ctl *ctl);
+
+#endif
diff --git a/libaudio/alsa_mixer.c b/libaudio/alsa_mixer.c
new file mode 100644
index 0000000..3036ef8
--- /dev/null
+++ b/libaudio/alsa_mixer.c
@@ -0,0 +1,371 @@
+/*
+** Copyright 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.
+*/
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <errno.h>
+#include <ctype.h>
+
+#include <linux/ioctl.h>
+#define __force
+#define __bitwise
+#define __user
+#include "asound.h"
+
+#include "alsa_audio.h"
+
+static const char *elem_iface_name(snd_ctl_elem_iface_t n)
+{
+    switch (n) {
+    case SNDRV_CTL_ELEM_IFACE_CARD: return "CARD";
+    case SNDRV_CTL_ELEM_IFACE_HWDEP: return "HWDEP";
+    case SNDRV_CTL_ELEM_IFACE_MIXER: return "MIXER";
+    case SNDRV_CTL_ELEM_IFACE_PCM: return "PCM";
+    case SNDRV_CTL_ELEM_IFACE_RAWMIDI: return "MIDI";
+    case SNDRV_CTL_ELEM_IFACE_TIMER: return "TIMER";
+    case SNDRV_CTL_ELEM_IFACE_SEQUENCER: return "SEQ";
+    default: return "???";
+    }
+}
+
+static const char *elem_type_name(snd_ctl_elem_type_t n)
+{
+    switch (n) {
+    case SNDRV_CTL_ELEM_TYPE_NONE: return "NONE";
+    case SNDRV_CTL_ELEM_TYPE_BOOLEAN: return "BOOL";
+    case SNDRV_CTL_ELEM_TYPE_INTEGER: return "INT32";
+    case SNDRV_CTL_ELEM_TYPE_ENUMERATED: return "ENUM";
+    case SNDRV_CTL_ELEM_TYPE_BYTES: return "BYTES";
+    case SNDRV_CTL_ELEM_TYPE_IEC958: return "IEC958";
+    case SNDRV_CTL_ELEM_TYPE_INTEGER64: return "INT64";
+    default: return "???";
+    }
+}
+
+
+struct mixer_ctl {
+    struct mixer *mixer;
+    struct snd_ctl_elem_info *info;
+    char **ename;
+};
+
+struct mixer {
+    int fd;
+    struct snd_ctl_elem_info *info;
+    struct mixer_ctl *ctl;
+    unsigned count;
+};
+
+void mixer_close(struct mixer *mixer)
+{
+    unsigned n,m;
+
+    if (mixer->fd >= 0)
+        close(mixer->fd);
+
+    if (mixer->ctl) {
+        for (n = 0; n < mixer->count; n++) {
+            if (mixer->ctl[n].ename) {
+                unsigned max = mixer->ctl[n].info->value.enumerated.items;
+                for (m = 0; m < max; m++)
+                    free(mixer->ctl[n].ename[m]);
+                free(mixer->ctl[n].ename);
+            }
+        }
+        free(mixer->ctl);
+    }
+
+    if (mixer->info)
+        free(mixer->info);
+
+    free(mixer);
+}
+
+struct mixer *mixer_open(void)
+{
+    struct snd_ctl_elem_list elist;
+    struct snd_ctl_elem_info tmp;
+    struct snd_ctl_elem_id *eid = NULL;
+    struct mixer *mixer = NULL;
+    unsigned n, m;
+    int fd;
+
+    fd = open("/dev/snd/controlC0", O_RDWR);
+    if (fd < 0)
+        return 0;
+
+    memset(&elist, 0, sizeof(elist));
+    if (ioctl(fd, SNDRV_CTL_IOCTL_ELEM_LIST, &elist) < 0)
+        goto fail;
+
+    mixer = calloc(1, sizeof(*mixer));
+    if (!mixer)
+        goto fail;
+
+    mixer->ctl = calloc(elist.count, sizeof(struct mixer_ctl));
+    mixer->info = calloc(elist.count, sizeof(struct snd_ctl_elem_info));
+    if (!mixer->ctl || !mixer->info)
+        goto fail;
+
+    eid = calloc(elist.count, sizeof(struct snd_ctl_elem_id));
+    if (!eid)
+        goto fail;
+
+    mixer->count = elist.count;
+    mixer->fd = fd;
+    elist.space = mixer->count;
+    elist.pids = eid;
+    if (ioctl(fd, SNDRV_CTL_IOCTL_ELEM_LIST, &elist) < 0)
+        goto fail;
+
+    for (n = 0; n < mixer->count; n++) {
+        struct snd_ctl_elem_info *ei = mixer->info + n;
+        ei->id.numid = eid[n].numid;
+        if (ioctl(fd, SNDRV_CTL_IOCTL_ELEM_INFO, ei) < 0)
+            goto fail;
+        mixer->ctl[n].info = ei;
+        mixer->ctl[n].mixer = mixer;
+        if (ei->type == SNDRV_CTL_ELEM_TYPE_ENUMERATED) {
+            char **enames = calloc(ei->value.enumerated.items, sizeof(char*));
+            if (!enames)
+                goto fail;
+            mixer->ctl[n].ename = enames;
+            for (m = 0; m < ei->value.enumerated.items; m++) {
+                memset(&tmp, 0, sizeof(tmp));
+                tmp.id.numid = ei->id.numid;
+                tmp.value.enumerated.item = m;
+                if (ioctl(fd, SNDRV_CTL_IOCTL_ELEM_INFO, &tmp) < 0)
+                    goto fail;
+                enames[m] = strdup(tmp.value.enumerated.name);
+                if (!enames[m])
+                    goto fail;
+            }
+        }
+    }
+
+    free(eid);
+    return mixer;
+
+fail:
+    if (eid)
+        free(eid);
+    if (mixer)
+        mixer_close(mixer);
+    else if (fd >= 0)
+        close(fd);
+    return 0;
+}
+
+void mixer_dump(struct mixer *mixer)
+{
+    unsigned n, m;
+
+    printf("  id iface dev sub idx num perms     type   name\n");
+    for (n = 0; n < mixer->count; n++) {
+        struct snd_ctl_elem_info *ei = mixer->info + n;
+
+        printf("%4d %5s %3d %3d %3d %3d %c%c%c%c%c%c%c%c%c %-6s %s",
+               ei->id.numid, elem_iface_name(ei->id.iface),
+               ei->id.device, ei->id.subdevice, ei->id.index,
+               ei->count,
+               (ei->access & SNDRV_CTL_ELEM_ACCESS_READ) ? 'r' : ' ',
+               (ei->access & SNDRV_CTL_ELEM_ACCESS_WRITE) ? 'w' : ' ',
+               (ei->access & SNDRV_CTL_ELEM_ACCESS_VOLATILE) ? 'V' : ' ',
+               (ei->access & SNDRV_CTL_ELEM_ACCESS_TIMESTAMP) ? 'T' : ' ',
+               (ei->access & SNDRV_CTL_ELEM_ACCESS_TLV_READ) ? 'R' : ' ',
+               (ei->access & SNDRV_CTL_ELEM_ACCESS_TLV_WRITE) ? 'W' : ' ',
+               (ei->access & SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND) ? 'C' : ' ',
+               (ei->access & SNDRV_CTL_ELEM_ACCESS_INACTIVE) ? 'I' : ' ',
+               (ei->access & SNDRV_CTL_ELEM_ACCESS_LOCK) ? 'L' : ' ',
+               elem_type_name(ei->type),
+               ei->id.name);
+        switch (ei->type) {
+        case SNDRV_CTL_ELEM_TYPE_INTEGER:
+            printf(ei->value.integer.step ?
+                   " { %ld-%ld, %ld }\n" : " { %ld-%ld }",
+                   ei->value.integer.min,
+                   ei->value.integer.max,
+                   ei->value.integer.step);
+            break;
+        case SNDRV_CTL_ELEM_TYPE_INTEGER64:
+            printf(ei->value.integer64.step ?
+                   " { %lld-%lld, %lld }\n" : " { %lld-%lld }",
+                   ei->value.integer64.min,
+                   ei->value.integer64.max,
+                   ei->value.integer64.step);
+            break;
+        case SNDRV_CTL_ELEM_TYPE_ENUMERATED: {
+            unsigned m;
+            printf(" { %s=0", mixer->ctl[n].ename[0]);
+            for (m = 1; m < ei->value.enumerated.items; m++)
+                printf(", %s=%d", mixer->ctl[n].ename[m],m);
+            printf(" }");
+            break;
+        }
+        }
+        printf("\n");
+    }
+}
+
+struct mixer_ctl *mixer_get_control(struct mixer *mixer,
+                                    const char *name, unsigned index)
+{
+    unsigned n;
+    for (n = 0; n < mixer->count; n++) {
+        if (mixer->info[n].id.index == index) {
+            if (!strcmp(name, (char*) mixer->info[n].id.name)) {
+                return mixer->ctl + n;
+            }
+        }
+    }
+    return 0;
+}
+
+struct mixer_ctl *mixer_get_nth_control(struct mixer *mixer, unsigned n)
+{
+    if (n < mixer->count)
+        return mixer->ctl + n;
+    return 0;
+}
+
+void mixer_ctl_print(struct mixer_ctl *ctl)
+{
+    struct snd_ctl_elem_value ev;
+    unsigned n;
+
+    memset(&ev, 0, sizeof(ev));
+    ev.id.numid = ctl->info->id.numid;
+    if (ioctl(ctl->mixer->fd, SNDRV_CTL_IOCTL_ELEM_READ, &ev))
+        return;
+    printf("%s:", ctl->info->id.name);
+
+    switch (ctl->info->type) {
+    case SNDRV_CTL_ELEM_TYPE_BOOLEAN:
+        for (n = 0; n < ctl->info->count; n++)
+            printf(" %s", ev.value.integer.value[n] ? "ON" : "OFF");
+        break;
+    case SNDRV_CTL_ELEM_TYPE_INTEGER: {
+        for (n = 0; n < ctl->info->count; n++)
+            printf(" %ld", ev.value.integer.value[n]);
+        break;
+    }
+    case SNDRV_CTL_ELEM_TYPE_INTEGER64:
+        for (n = 0; n < ctl->info->count; n++)
+            printf(" %lld", ev.value.integer64.value[n]);
+        break;
+    case SNDRV_CTL_ELEM_TYPE_ENUMERATED:
+        for (n = 0; n < ctl->info->count; n++) {
+            unsigned v = ev.value.enumerated.item[n];
+            printf(" %d (%s)", v,
+                   (v < ctl->info->value.enumerated.items) ? ctl->ename[v] : "???");
+        }
+        break;
+    default:
+        printf(" ???");
+    }
+    printf("\n");
+}
+
+static long scale_int(struct snd_ctl_elem_info *ei, unsigned _percent)
+{
+    long percent;
+    long range;
+
+    if (_percent > 100)
+        percent = 100;
+    else
+        percent = (long) _percent;
+
+    range = (ei->value.integer.max - ei->value.integer.min);
+
+    return ei->value.integer.min + (range * percent) / 100LL;
+}
+
+static long long scale_int64(struct snd_ctl_elem_info *ei, unsigned _percent)
+{
+    long long percent;
+    long long range;
+
+    if (_percent > 100)
+        percent = 100;
+    else
+        percent = (long) _percent;
+
+    range = (ei->value.integer.max - ei->value.integer.min) * 100LL;
+
+    return ei->value.integer.min + (range / percent);
+}
+
+int mixer_ctl_set(struct mixer_ctl *ctl, unsigned percent)
+{
+    struct snd_ctl_elem_value ev;
+    unsigned n;
+
+    memset(&ev, 0, sizeof(ev));
+    ev.id.numid = ctl->info->id.numid;
+    switch (ctl->info->type) {
+    case SNDRV_CTL_ELEM_TYPE_BOOLEAN:
+        for (n = 0; n < ctl->info->count; n++)
+            ev.value.integer.value[n] = !!percent;
+        break;
+    case SNDRV_CTL_ELEM_TYPE_INTEGER: {
+        long value = scale_int(ctl->info, percent);
+        for (n = 0; n < ctl->info->count; n++)
+            ev.value.integer.value[n] = value;
+        break;
+    }
+    case SNDRV_CTL_ELEM_TYPE_INTEGER64: {
+        long long value = scale_int64(ctl->info, percent);
+        for (n = 0; n < ctl->info->count; n++)
+            ev.value.integer64.value[n] = value;
+        break;
+    }
+    default:
+        errno = EINVAL;
+        return -1;
+    }
+
+    return ioctl(ctl->mixer->fd, SNDRV_CTL_IOCTL_ELEM_WRITE, &ev);
+}
+
+int mixer_ctl_select(struct mixer_ctl *ctl, const char *value)
+{
+    unsigned n, max;
+    struct snd_ctl_elem_value ev;
+
+    if (ctl->info->type != SNDRV_CTL_ELEM_TYPE_ENUMERATED) {
+        errno = EINVAL;
+        return -1;
+    }
+
+    max = ctl->info->value.enumerated.items;
+    for (n = 0; n < max; n++) {
+        if (!strcmp(value, ctl->ename[n])) {
+            memset(&ev, 0, sizeof(ev));
+            ev.value.enumerated.item[0] = n;
+            ev.id.numid = ctl->info->id.numid;
+            if (ioctl(ctl->mixer->fd, SNDRV_CTL_IOCTL_ELEM_WRITE, &ev) < 0)
+                return -1;
+            return 0;
+        }
+    }
+
+    errno = EINVAL;
+    return -1;
+}
diff --git a/libaudio/alsa_pcm.c b/libaudio/alsa_pcm.c
new file mode 100644
index 0000000..5673391
--- /dev/null
+++ b/libaudio/alsa_pcm.c
@@ -0,0 +1,405 @@
+/*
+** Copyright 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.
+*/
+
+#define LOG_TAG "alsa_pcm"
+//#define LOG_NDEBUG 0
+#include <cutils/log.h>
+#include <cutils/config_utils.h>
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <fcntl.h>
+#include <stdarg.h>
+#include <string.h>
+#include <errno.h>
+#include <unistd.h>
+
+#include <sys/ioctl.h>
+#include <sys/mman.h>
+#include <sys/time.h>
+
+#include <linux/ioctl.h>
+
+#include "alsa_audio.h"
+
+#define __force
+#define __bitwise
+#define __user
+#include "asound.h"
+
+#define DEBUG 0
+
+/* alsa parameter manipulation cruft */
+
+#define PARAM_MAX SNDRV_PCM_HW_PARAM_LAST_INTERVAL
+
+static inline int param_is_mask(int p)
+{
+    return (p >= SNDRV_PCM_HW_PARAM_FIRST_MASK) &&
+        (p <= SNDRV_PCM_HW_PARAM_LAST_MASK);
+}
+
+static inline int param_is_interval(int p)
+{
+    return (p >= SNDRV_PCM_HW_PARAM_FIRST_INTERVAL) &&
+        (p <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL);
+}
+
+static inline struct snd_interval *param_to_interval(struct snd_pcm_hw_params *p, int n)
+{
+    return &(p->intervals[n - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL]);
+}
+
+static inline struct snd_mask *param_to_mask(struct snd_pcm_hw_params *p, int n)
+{
+    return &(p->masks[n - SNDRV_PCM_HW_PARAM_FIRST_MASK]);
+}
+
+static void param_set_mask(struct snd_pcm_hw_params *p, int n, unsigned bit)
+{
+    if (bit >= SNDRV_MASK_MAX)
+        return;
+    if (param_is_mask(n)) {
+        struct snd_mask *m = param_to_mask(p, n);
+        m->bits[0] = 0;
+        m->bits[1] = 0;
+        m->bits[bit >> 5] |= (1 << (bit & 31));
+    }
+}
+
+static void param_set_min(struct snd_pcm_hw_params *p, int n, unsigned val)
+{
+    if (param_is_interval(n)) {
+        struct snd_interval *i = param_to_interval(p, n);
+        i->min = val;
+    }
+}
+
+static void param_set_max(struct snd_pcm_hw_params *p, int n, unsigned val)
+{
+    if (param_is_interval(n)) {
+        struct snd_interval *i = param_to_interval(p, n);
+        i->max = val;
+    }
+}
+
+static void param_set_int(struct snd_pcm_hw_params *p, int n, unsigned val)
+{
+    if (param_is_interval(n)) {
+        struct snd_interval *i = param_to_interval(p, n);
+        i->min = val;
+        i->max = val;
+        i->integer = 1;
+    }
+}
+
+static void param_init(struct snd_pcm_hw_params *p)
+{
+    int n;
+    memset(p, 0, sizeof(*p));
+    for (n = SNDRV_PCM_HW_PARAM_FIRST_MASK;
+         n <= SNDRV_PCM_HW_PARAM_LAST_MASK; n++) {
+            struct snd_mask *m = param_to_mask(p, n);
+            m->bits[0] = ~0;
+            m->bits[1] = ~0;
+    }
+    for (n = SNDRV_PCM_HW_PARAM_FIRST_INTERVAL;
+         n <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; n++) {
+            struct snd_interval *i = param_to_interval(p, n);
+            i->min = 0;
+            i->max = ~0;
+    }
+}
+
+/* debugging gunk */
+
+#if DEBUG
+static const char *param_name[PARAM_MAX+1] = {
+    [SNDRV_PCM_HW_PARAM_ACCESS] = "access",
+    [SNDRV_PCM_HW_PARAM_FORMAT] = "format",
+    [SNDRV_PCM_HW_PARAM_SUBFORMAT] = "subformat",
+
+    [SNDRV_PCM_HW_PARAM_SAMPLE_BITS] = "sample_bits",
+    [SNDRV_PCM_HW_PARAM_FRAME_BITS] = "frame_bits",
+    [SNDRV_PCM_HW_PARAM_CHANNELS] = "channels",
+    [SNDRV_PCM_HW_PARAM_RATE] = "rate",
+    [SNDRV_PCM_HW_PARAM_PERIOD_TIME] = "period_time",
+    [SNDRV_PCM_HW_PARAM_PERIOD_SIZE] = "period_size",
+    [SNDRV_PCM_HW_PARAM_PERIOD_BYTES] = "period_bytes",
+    [SNDRV_PCM_HW_PARAM_PERIODS] = "periods",
+    [SNDRV_PCM_HW_PARAM_BUFFER_TIME] = "buffer_time",
+    [SNDRV_PCM_HW_PARAM_BUFFER_SIZE] = "buffer_size",
+    [SNDRV_PCM_HW_PARAM_BUFFER_BYTES] = "buffer_bytes",
+    [SNDRV_PCM_HW_PARAM_TICK_TIME] = "tick_time",
+};
+
+static void param_dump(struct snd_pcm_hw_params *p)
+{
+    int n;
+
+    for (n = SNDRV_PCM_HW_PARAM_FIRST_MASK;
+         n <= SNDRV_PCM_HW_PARAM_LAST_MASK; n++) {
+            struct snd_mask *m = param_to_mask(p, n);
+            LOGV("%s = %08x%08x\n", param_name[n],
+                   m->bits[1], m->bits[0]);
+    }
+    for (n = SNDRV_PCM_HW_PARAM_FIRST_INTERVAL;
+         n <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; n++) {
+            struct snd_interval *i = param_to_interval(p, n);
+            LOGV("%s = (%d,%d) omin=%d omax=%d int=%d empty=%d\n",
+                   param_name[n], i->min, i->max, i->openmin,
+                   i->openmax, i->integer, i->empty);
+    }
+    LOGV("info = %08x\n", p->info);
+    LOGV("msbits = %d\n", p->msbits);
+    LOGV("rate = %d/%d\n", p->rate_num, p->rate_den);
+    LOGV("fifo = %d\n", (int) p->fifo_size);
+}
+
+static void info_dump(struct snd_pcm_info *info)
+{
+    LOGV("device = %d\n", info->device);
+    LOGV("subdevice = %d\n", info->subdevice);
+    LOGV("stream = %d\n", info->stream);
+    LOGV("card = %d\n", info->card);
+    LOGV("id = '%s'\n", info->id);
+    LOGV("name = '%s'\n", info->name);
+    LOGV("subname = '%s'\n", info->subname);
+    LOGV("dev_class = %d\n", info->dev_class);
+    LOGV("dev_subclass = %d\n", info->dev_subclass);
+    LOGV("subdevices_count = %d\n", info->subdevices_count);
+    LOGV("subdevices_avail = %d\n", info->subdevices_avail);
+}
+#else
+static void param_dump(struct snd_pcm_hw_params *p) {}
+static void info_dump(struct snd_pcm_info *info) {}
+#endif
+
+#define PCM_ERROR_MAX 128
+
+struct pcm {
+    int fd;
+    unsigned flags;
+    int running:1;
+    int underruns;
+    unsigned buffer_size;
+    char error[PCM_ERROR_MAX];
+};
+
+unsigned pcm_buffer_size(struct pcm *pcm)
+{
+    return pcm->buffer_size;
+}
+
+const char* pcm_error(struct pcm *pcm)
+{
+    return pcm->error;
+}
+
+static int oops(struct pcm *pcm, int e, const char *fmt, ...)
+{
+    va_list ap;
+    int sz;
+
+    va_start(ap, fmt);
+    vsnprintf(pcm->error, PCM_ERROR_MAX, fmt, ap);
+    va_end(ap);
+    sz = strlen(pcm->error);
+
+    if (errno)
+        snprintf(pcm->error + sz, PCM_ERROR_MAX - sz,
+                 ": %s", strerror(e));
+    return -1;
+}
+
+int pcm_write(struct pcm *pcm, void *data, unsigned count)
+{
+    struct snd_xferi x;
+
+    if (pcm->flags & PCM_IN)
+        return -EINVAL;
+
+    x.buf = data;
+    x.frames = (pcm->flags & PCM_MONO) ? (count / 2) : (count / 4);
+
+    for (;;) {
+        if (!pcm->running) {
+            if (ioctl(pcm->fd, SNDRV_PCM_IOCTL_PREPARE))
+                return oops(pcm, errno, "cannot prepare channel");
+            if (ioctl(pcm->fd, SNDRV_PCM_IOCTL_WRITEI_FRAMES, &x))
+                return oops(pcm, errno, "cannot write initial data");
+            pcm->running = 1;
+            return 0;
+        }
+        if (ioctl(pcm->fd, SNDRV_PCM_IOCTL_WRITEI_FRAMES, &x)) {
+            pcm->running = 0;
+            if (errno == EPIPE) {
+                    /* we failed to make our window -- try to restart */
+                pcm->underruns++;
+                continue;
+            }
+            return oops(pcm, errno, "cannot write stream data");
+        }
+        return 0;
+    }
+}
+
+int pcm_read(struct pcm *pcm, void *data, unsigned count)
+{
+    struct snd_xferi x;
+
+    if (!(pcm->flags & PCM_IN))
+        return -EINVAL;
+
+    x.buf = data;
+    x.frames = (pcm->flags & PCM_MONO) ? (count / 2) : (count / 4);
+
+//    LOGV("read() %d frames", x.frames);
+    for (;;) {
+        if (!pcm->running) {
+            if (ioctl(pcm->fd, SNDRV_PCM_IOCTL_PREPARE))
+                return oops(pcm, errno, "cannot prepare channel");
+            if (ioctl(pcm->fd, SNDRV_PCM_IOCTL_START))
+                return oops(pcm, errno, "cannot start channel");
+            pcm->running = 1;
+        }
+        if (ioctl(pcm->fd, SNDRV_PCM_IOCTL_READI_FRAMES, &x)) {
+            pcm->running = 0;
+            if (errno == EPIPE) {
+                    /* we failed to make our window -- try to restart */
+                pcm->underruns++;
+                continue;
+            }
+            return oops(pcm, errno, "cannot read stream data");
+        }
+//        LOGV("read() got %d frames", x.frames);
+        return 0;
+    }
+}
+
+static struct pcm bad_pcm = {
+    .fd = -1,
+};
+
+int pcm_close(struct pcm *pcm)
+{
+    if (pcm == &bad_pcm)
+        return 0;
+
+    if (pcm->fd >= 0)
+        close(pcm->fd);
+    pcm->running = 0;
+    pcm->buffer_size = 0;
+    pcm->fd = -1;
+    return 0;
+}
+
+struct pcm *pcm_open(unsigned flags)
+{
+    const char *dname;
+    struct pcm *pcm;
+    struct snd_pcm_info info;
+    struct snd_pcm_hw_params params;
+    struct snd_pcm_sw_params sparams;
+    unsigned period_sz;
+    unsigned period_cnt;
+
+    LOGV("pcm_open(0x%08x)",flags);
+
+    pcm = calloc(1, sizeof(struct pcm));
+    if (!pcm)
+        return &bad_pcm;
+
+    if (flags & PCM_IN) {
+        dname = "/dev/snd/pcmC0D0c";
+    } else {
+        dname = "/dev/snd/pcmC0D0p";
+    }
+
+    LOGV("pcm_open() period sz multiplier %d",
+         ((flags & PCM_PERIOD_SZ_MASK) >> PCM_PERIOD_SZ_SHIFT) + 1);
+    period_sz = 128 * (((flags & PCM_PERIOD_SZ_MASK) >> PCM_PERIOD_SZ_SHIFT) + 1);
+    LOGV("pcm_open() period cnt %d",
+         ((flags & PCM_PERIOD_CNT_MASK) >> PCM_PERIOD_CNT_SHIFT) + PCM_PERIOD_CNT_MIN);
+    period_cnt = ((flags & PCM_PERIOD_CNT_MASK) >> PCM_PERIOD_CNT_SHIFT) + PCM_PERIOD_CNT_MIN;
+
+    pcm->flags = flags;
+    pcm->fd = open(dname, O_RDWR);
+    if (pcm->fd < 0) {
+        oops(pcm, errno, "cannot open device '%s'");
+        return pcm;
+    }
+
+    if (ioctl(pcm->fd, SNDRV_PCM_IOCTL_INFO, &info)) {
+        oops(pcm, errno, "cannot get info - %s");
+        goto fail;
+    }
+    info_dump(&info);
+
+    LOGV("pcm_open() period_cnt %d period_sz %d channels %d",
+         period_cnt, period_sz, (flags & PCM_MONO) ? 1 : 2);
+
+    param_init(&params);
+    param_set_mask(&params, SNDRV_PCM_HW_PARAM_ACCESS,
+                   SNDRV_PCM_ACCESS_RW_INTERLEAVED);
+    param_set_mask(&params, SNDRV_PCM_HW_PARAM_FORMAT,
+                   SNDRV_PCM_FORMAT_S16_LE);
+    param_set_mask(&params, SNDRV_PCM_HW_PARAM_SUBFORMAT,
+                   SNDRV_PCM_SUBFORMAT_STD);
+    param_set_min(&params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, period_sz);
+    param_set_int(&params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, 16);
+    param_set_int(&params, SNDRV_PCM_HW_PARAM_FRAME_BITS,
+                  (flags & PCM_MONO) ? 16 : 32);
+    param_set_int(&params, SNDRV_PCM_HW_PARAM_CHANNELS,
+                  (flags & PCM_MONO) ? 1 : 2);
+    param_set_int(&params, SNDRV_PCM_HW_PARAM_PERIODS, period_cnt);
+    param_set_int(&params, SNDRV_PCM_HW_PARAM_RATE, 44100);
+
+    if (ioctl(pcm->fd, SNDRV_PCM_IOCTL_HW_PARAMS, &params)) {
+        oops(pcm, errno, "cannot set hw params");
+        goto fail;
+    }
+    param_dump(&params);
+
+    memset(&sparams, 0, sizeof(sparams));
+    sparams.tstamp_mode = SNDRV_PCM_TSTAMP_NONE;
+    sparams.period_step = 1;
+    sparams.avail_min = 1;
+    sparams.start_threshold = period_cnt * period_sz;
+    sparams.stop_threshold = period_cnt * period_sz;
+    sparams.xfer_align = period_sz / 2; /* needed for old kernels */
+    sparams.silence_size = 0;
+    sparams.silence_threshold = 0;
+
+    if (ioctl(pcm->fd, SNDRV_PCM_IOCTL_SW_PARAMS, &sparams)) {
+        oops(pcm, errno, "cannot set sw params");
+        goto fail;
+    }
+
+    pcm->buffer_size = period_cnt * period_sz;
+    pcm->underruns = 0;
+    return pcm;
+
+fail:
+    close(pcm->fd);
+    pcm->fd = -1;
+    return pcm;
+}
+
+int pcm_ready(struct pcm *pcm)
+{
+    return pcm->fd >= 0;
+}
diff --git a/libaudio/amix.c b/libaudio/amix.c
new file mode 100644
index 0000000..d978caa
--- /dev/null
+++ b/libaudio/amix.c
@@ -0,0 +1,78 @@
+/*
+** Copyright 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.
+*/
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <errno.h>
+#include <ctype.h>
+
+#include "alsa_audio.h"
+
+
+struct mixer_ctl *get_ctl(struct mixer *mixer, char *name)
+{
+    char *p;
+    unsigned idx = 0;
+
+    if (isdigit(name[0]))
+        return mixer_get_nth_control(mixer, atoi(name) - 1);
+
+    p = strrchr(name, '#');
+    if (p) {
+        *p++ = 0;
+        idx = atoi(p);
+    }
+
+    return mixer_get_control(mixer, name, idx);
+}
+
+int main(int argc, char **argv)
+{
+    struct mixer *mixer;
+    struct mixer_ctl *ctl;
+    int r;
+
+    mixer = mixer_open();
+    if (!mixer)
+        return -1;
+
+    if (argc == 1) {
+        mixer_dump(mixer);
+        return 0;
+    }
+
+    ctl = get_ctl(mixer, argv[1]);
+    argc -= 2;
+    argv += 2;
+
+    if (!ctl) {
+        fprintf(stderr,"can't find control\n");
+        return -1;
+    }
+
+    if (argc) {
+        if (isdigit(argv[0][0]))
+            r = mixer_ctl_set(ctl, atoi(argv[0]));
+        else
+            r = mixer_ctl_select(ctl, argv[0]);
+        if (r)
+            fprintf(stderr,"oops: %s\n", strerror(errno));
+    } else {
+        mixer_ctl_print(ctl);
+    }
+    return 0;
+}
diff --git a/libaudio/aplay.c b/libaudio/aplay.c
new file mode 100644
index 0000000..0ac0ac0
--- /dev/null
+++ b/libaudio/aplay.c
@@ -0,0 +1,140 @@
+/*
+** Copyright 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.
+*/
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <stdint.h>
+#include <string.h>
+
+#include "alsa_audio.h"
+
+#define ID_RIFF 0x46464952
+#define ID_WAVE 0x45564157
+#define ID_FMT  0x20746d66
+#define ID_DATA 0x61746164
+
+#define FORMAT_PCM 1
+
+struct wav_header {
+    uint32_t riff_id;
+    uint32_t riff_sz;
+    uint32_t riff_fmt;
+    uint32_t fmt_id;
+    uint32_t fmt_sz;
+    uint16_t audio_format;
+    uint16_t num_channels;
+    uint32_t sample_rate;
+    uint32_t byte_rate;       /* sample_rate * num_channels * bps / 8 */
+    uint16_t block_align;     /* num_channels * bps / 8 */
+    uint16_t bits_per_sample;
+    uint32_t data_id;
+    uint32_t data_sz;
+};
+
+int play_file(unsigned rate, unsigned channels, int fd, unsigned count)
+{
+    struct pcm *pcm;
+    struct mixer *mixer;
+    struct pcm_ctl *ctl = NULL;
+    unsigned bufsize;
+    char *data;
+    unsigned flags = PCM_OUT;
+
+    if (channels == 1)
+        flags |= PCM_MONO;
+    else
+        flags |= PCM_STEREO;
+
+    pcm = pcm_open(flags);
+    if (!pcm_ready(pcm)) {
+        pcm_close(pcm);
+        return -1;
+    }
+
+    mixer = mixer_open();
+    if (mixer)
+        ctl = mixer_get_control(mixer,"Playback Path", 0);
+    
+    bufsize = pcm_buffer_size(pcm);
+    data = malloc(bufsize);
+    if (!data) {
+        fprintf(stderr,"could not allocate %d bytes\n", count);
+        return -1;
+    }
+
+    while (read(fd, data, bufsize) == bufsize) {
+        if (pcm_write(pcm, data, bufsize))
+            break;
+        
+            /* HACK: remove */
+        if (ctl) {
+            //mixer_ctl_select(ctl, "SPK");
+            ctl = 0;
+        }
+    }
+    pcm_close(pcm);
+    return 0;
+}
+
+int play_wav(const char *fn)
+{
+    struct wav_header hdr;
+    unsigned rate, channels;
+    int fd;
+    fd = open(fn, O_RDONLY);
+    if (fd < 0) {
+        fprintf(stderr, "aplay: cannot open '%s'\n", fn);
+        return -1;
+    }
+    if (read(fd, &hdr, sizeof(hdr)) != sizeof(hdr)) {
+        fprintf(stderr, "aplay: cannot read header\n");
+        return -1;
+    }
+    fprintf(stderr,"aplay: %d ch, %d hz, %d bit, %s\n",
+            hdr.num_channels, hdr.sample_rate, hdr.bits_per_sample,
+            hdr.audio_format == FORMAT_PCM ? "PCM" : "unknown");
+    
+    if ((hdr.riff_id != ID_RIFF) ||
+        (hdr.riff_fmt != ID_WAVE) ||
+        (hdr.fmt_id != ID_FMT)) {
+        fprintf(stderr, "aplay: '%s' is not a riff/wave file\n", fn);
+        return -1;
+    }
+    if ((hdr.audio_format != FORMAT_PCM) ||
+        (hdr.fmt_sz != 16)) {
+        fprintf(stderr, "aplay: '%s' is not pcm format\n", fn);
+        return -1;
+    }
+    if (hdr.bits_per_sample != 16) {
+        fprintf(stderr, "aplay: '%s' is not 16bit per sample\n", fn);
+        return -1;
+    }
+
+    return play_file(hdr.sample_rate, hdr.num_channels, fd, hdr.data_sz);
+}
+
+int main(int argc, char **argv)
+{
+    if (argc != 2) {
+        fprintf(stderr,"usage: aplay <file>\n");
+        return -1;
+    }
+
+    return play_wav(argv[1]);
+}
+
diff --git a/libaudio/arec.c b/libaudio/arec.c
new file mode 100644
index 0000000..b1e9eda
--- /dev/null
+++ b/libaudio/arec.c
@@ -0,0 +1,128 @@
+/*
+** Copyright 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.
+*/
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <stdint.h>
+#include <string.h>
+
+#include "alsa_audio.h"
+
+#define ID_RIFF 0x46464952
+#define ID_WAVE 0x45564157
+#define ID_FMT  0x20746d66
+#define ID_DATA 0x61746164
+
+#define FORMAT_PCM 1
+
+struct wav_header {
+    uint32_t riff_id;
+    uint32_t riff_sz;
+    uint32_t riff_fmt;
+    uint32_t fmt_id;
+    uint32_t fmt_sz;
+    uint16_t audio_format;
+    uint16_t num_channels;
+    uint32_t sample_rate;
+    uint32_t byte_rate;       /* sample_rate * num_channels * bps / 8 */
+    uint16_t block_align;     /* num_channels * bps / 8 */
+    uint16_t bits_per_sample;
+    uint32_t data_id;
+    uint32_t data_sz;
+};
+
+int record_file(unsigned rate, unsigned channels, int fd, unsigned count)
+{
+    struct pcm *pcm;
+    unsigned avail, xfer, bufsize;
+    char *data, *next;
+    int r;
+
+    pcm = pcm_open(PCM_IN|PCM_MONO);
+    if (!pcm_ready(pcm)) {
+        pcm_close(pcm);
+        goto fail;
+    }
+
+    bufsize = pcm_buffer_size(pcm);
+    
+    data = malloc(bufsize);
+    if (!data) {
+        fprintf(stderr,"could not allocate %d bytes\n", count);
+        return -1;
+    }
+
+    while (!pcm_read(pcm, data, bufsize)) {
+        if (write(fd, data, bufsize) != bufsize) {
+            fprintf(stderr,"could not write %d bytes\n", bufsize);
+            return -1;
+        }
+    }
+
+    close(fd);
+    pcm_close(pcm);
+    return 0;
+    
+fail:
+    fprintf(stderr,"pcm error: %s\n", pcm_error(pcm));
+    return -1;
+}
+
+int rec_wav(const char *fn)
+{
+    struct wav_header hdr;
+    unsigned rate, channels;
+    int fd;
+    fd = open(fn, O_WRONLY | O_CREAT | O_TRUNC, 0664);
+    if (fd < 0) {
+        fprintf(stderr, "arec: cannot open '%s'\n", fn);
+        return -1;
+    }
+
+    hdr.riff_id = ID_RIFF;
+    hdr.riff_fmt = ID_WAVE;
+    hdr.fmt_id = ID_FMT;
+    hdr.audio_format = FORMAT_PCM;
+    hdr.fmt_sz = 16;
+    hdr.bits_per_sample = 16;
+    hdr.num_channels = 1;
+    hdr.data_sz = 0;
+    hdr.sample_rate = 44100;
+
+    if (write(fd, &hdr, sizeof(hdr)) != sizeof(hdr)) {
+        fprintf(stderr, "arec: cannot write header\n");
+        return -1;
+    }
+    fprintf(stderr,"arec: %d ch, %d hz, %d bit, %s\n",
+            hdr.num_channels, hdr.sample_rate, hdr.bits_per_sample,
+            hdr.audio_format == FORMAT_PCM ? "PCM" : "unknown");
+    
+
+    return record_file(hdr.sample_rate, hdr.num_channels, fd, hdr.data_sz);
+}
+
+int main(int argc, char **argv)
+{
+    if (argc != 2) {
+        fprintf(stderr,"usage: arec <file>\n");
+        return -1;
+    }
+
+    return rec_wav(argv[1]);
+}
+
diff --git a/libaudio/asound.h b/libaudio/asound.h
new file mode 100644
index 0000000..6a17f29
--- /dev/null
+++ b/libaudio/asound.h
@@ -0,0 +1,814 @@
+/****************************************************************************
+ ****************************************************************************
+ ***
+ ***   This header was automatically generated from a Linux kernel header
+ ***   of the same name, to make information necessary for userspace to
+ ***   call into the kernel available to libc.  It contains only constants,
+ ***   structures, and macros generated from the original header, and thus,
+ ***   contains no copyrightable information.
+ ***
+ ****************************************************************************
+ ****************************************************************************/
+#ifndef __SOUND_ASOUND_H
+#define __SOUND_ASOUND_H
+
+#include <linux/types.h>
+
+#define SNDRV_PROTOCOL_VERSION(major, minor, subminor) (((major)<<16)|((minor)<<8)|(subminor))
+#define SNDRV_PROTOCOL_MAJOR(version) (((version)>>16)&0xffff)
+#define SNDRV_PROTOCOL_MINOR(version) (((version)>>8)&0xff)
+#define SNDRV_PROTOCOL_MICRO(version) ((version)&0xff)
+#define SNDRV_PROTOCOL_INCOMPATIBLE(kversion, uversion)   (SNDRV_PROTOCOL_MAJOR(kversion) != SNDRV_PROTOCOL_MAJOR(uversion) ||   (SNDRV_PROTOCOL_MAJOR(kversion) == SNDRV_PROTOCOL_MAJOR(uversion) &&   SNDRV_PROTOCOL_MINOR(kversion) != SNDRV_PROTOCOL_MINOR(uversion)))
+
+struct snd_aes_iec958 {
+ unsigned char status[24];
+ unsigned char subcode[147];
+ unsigned char pad;
+ unsigned char dig_subframe[4];
+};
+
+#define SNDRV_HWDEP_VERSION SNDRV_PROTOCOL_VERSION(1, 0, 1)
+
+enum {
+ SNDRV_HWDEP_IFACE_OPL2 = 0,
+ SNDRV_HWDEP_IFACE_OPL3,
+ SNDRV_HWDEP_IFACE_OPL4,
+ SNDRV_HWDEP_IFACE_SB16CSP,
+ SNDRV_HWDEP_IFACE_EMU10K1,
+ SNDRV_HWDEP_IFACE_YSS225,
+ SNDRV_HWDEP_IFACE_ICS2115,
+ SNDRV_HWDEP_IFACE_SSCAPE,
+ SNDRV_HWDEP_IFACE_VX,
+ SNDRV_HWDEP_IFACE_MIXART,
+ SNDRV_HWDEP_IFACE_USX2Y,
+ SNDRV_HWDEP_IFACE_EMUX_WAVETABLE,
+ SNDRV_HWDEP_IFACE_BLUETOOTH,
+ SNDRV_HWDEP_IFACE_USX2Y_PCM,
+ SNDRV_HWDEP_IFACE_PCXHR,
+ SNDRV_HWDEP_IFACE_SB_RC,
+ SNDRV_HWDEP_IFACE_HDA,
+ SNDRV_HWDEP_IFACE_USB_STREAM,
+
+ SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_USB_STREAM
+};
+
+struct snd_hwdep_info {
+ unsigned int device;
+ int card;
+ unsigned char id[64];
+ unsigned char name[80];
+ int iface;
+ unsigned char reserved[64];
+};
+
+struct snd_hwdep_dsp_status {
+ unsigned int version;
+ unsigned char id[32];
+ unsigned int num_dsps;
+ unsigned int dsp_loaded;
+ unsigned int chip_ready;
+ unsigned char reserved[16];
+};
+
+struct snd_hwdep_dsp_image {
+ unsigned int index;
+ unsigned char name[64];
+ unsigned char __user *image;
+ size_t length;
+ unsigned long driver_data;
+};
+
+#define SNDRV_HWDEP_IOCTL_PVERSION _IOR ('H', 0x00, int)
+#define SNDRV_HWDEP_IOCTL_INFO _IOR ('H', 0x01, struct snd_hwdep_info)
+#define SNDRV_HWDEP_IOCTL_DSP_STATUS _IOR('H', 0x02, struct snd_hwdep_dsp_status)
+#define SNDRV_HWDEP_IOCTL_DSP_LOAD _IOW('H', 0x03, struct snd_hwdep_dsp_image)
+
+#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 10)
+
+typedef unsigned long snd_pcm_uframes_t;
+typedef signed long snd_pcm_sframes_t;
+
+enum {
+ SNDRV_PCM_CLASS_GENERIC = 0,
+ SNDRV_PCM_CLASS_MULTI,
+ SNDRV_PCM_CLASS_MODEM,
+ SNDRV_PCM_CLASS_DIGITIZER,
+
+ SNDRV_PCM_CLASS_LAST = SNDRV_PCM_CLASS_DIGITIZER,
+};
+
+enum {
+ SNDRV_PCM_SUBCLASS_GENERIC_MIX = 0,
+ SNDRV_PCM_SUBCLASS_MULTI_MIX,
+
+ SNDRV_PCM_SUBCLASS_LAST = SNDRV_PCM_SUBCLASS_MULTI_MIX,
+};
+
+enum {
+ SNDRV_PCM_STREAM_PLAYBACK = 0,
+ SNDRV_PCM_STREAM_CAPTURE,
+ SNDRV_PCM_STREAM_LAST = SNDRV_PCM_STREAM_CAPTURE,
+};
+
+typedef int __bitwise snd_pcm_access_t;
+#define SNDRV_PCM_ACCESS_MMAP_INTERLEAVED ((__force snd_pcm_access_t) 0)  
+#define SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED ((__force snd_pcm_access_t) 1)  
+#define SNDRV_PCM_ACCESS_MMAP_COMPLEX ((__force snd_pcm_access_t) 2)  
+#define SNDRV_PCM_ACCESS_RW_INTERLEAVED ((__force snd_pcm_access_t) 3)  
+#define SNDRV_PCM_ACCESS_RW_NONINTERLEAVED ((__force snd_pcm_access_t) 4)  
+#define SNDRV_PCM_ACCESS_LAST SNDRV_PCM_ACCESS_RW_NONINTERLEAVED
+
+typedef int __bitwise snd_pcm_format_t;
+#define SNDRV_PCM_FORMAT_S8 ((__force snd_pcm_format_t) 0)
+#define SNDRV_PCM_FORMAT_U8 ((__force snd_pcm_format_t) 1)
+#define SNDRV_PCM_FORMAT_S16_LE ((__force snd_pcm_format_t) 2)
+#define SNDRV_PCM_FORMAT_S16_BE ((__force snd_pcm_format_t) 3)
+#define SNDRV_PCM_FORMAT_U16_LE ((__force snd_pcm_format_t) 4)
+#define SNDRV_PCM_FORMAT_U16_BE ((__force snd_pcm_format_t) 5)
+#define SNDRV_PCM_FORMAT_S24_LE ((__force snd_pcm_format_t) 6)  
+#define SNDRV_PCM_FORMAT_S24_BE ((__force snd_pcm_format_t) 7)  
+#define SNDRV_PCM_FORMAT_U24_LE ((__force snd_pcm_format_t) 8)  
+#define SNDRV_PCM_FORMAT_U24_BE ((__force snd_pcm_format_t) 9)  
+#define SNDRV_PCM_FORMAT_S32_LE ((__force snd_pcm_format_t) 10)
+#define SNDRV_PCM_FORMAT_S32_BE ((__force snd_pcm_format_t) 11)
+#define SNDRV_PCM_FORMAT_U32_LE ((__force snd_pcm_format_t) 12)
+#define SNDRV_PCM_FORMAT_U32_BE ((__force snd_pcm_format_t) 13)
+#define SNDRV_PCM_FORMAT_FLOAT_LE ((__force snd_pcm_format_t) 14)  
+#define SNDRV_PCM_FORMAT_FLOAT_BE ((__force snd_pcm_format_t) 15)  
+#define SNDRV_PCM_FORMAT_FLOAT64_LE ((__force snd_pcm_format_t) 16)  
+#define SNDRV_PCM_FORMAT_FLOAT64_BE ((__force snd_pcm_format_t) 17)  
+#define SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE ((__force snd_pcm_format_t) 18)  
+#define SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE ((__force snd_pcm_format_t) 19)  
+#define SNDRV_PCM_FORMAT_MU_LAW ((__force snd_pcm_format_t) 20)
+#define SNDRV_PCM_FORMAT_A_LAW ((__force snd_pcm_format_t) 21)
+#define SNDRV_PCM_FORMAT_IMA_ADPCM ((__force snd_pcm_format_t) 22)
+#define SNDRV_PCM_FORMAT_MPEG ((__force snd_pcm_format_t) 23)
+#define SNDRV_PCM_FORMAT_GSM ((__force snd_pcm_format_t) 24)
+#define SNDRV_PCM_FORMAT_SPECIAL ((__force snd_pcm_format_t) 31)
+#define SNDRV_PCM_FORMAT_S24_3LE ((__force snd_pcm_format_t) 32)  
+#define SNDRV_PCM_FORMAT_S24_3BE ((__force snd_pcm_format_t) 33)  
+#define SNDRV_PCM_FORMAT_U24_3LE ((__force snd_pcm_format_t) 34)  
+#define SNDRV_PCM_FORMAT_U24_3BE ((__force snd_pcm_format_t) 35)  
+#define SNDRV_PCM_FORMAT_S20_3LE ((__force snd_pcm_format_t) 36)  
+#define SNDRV_PCM_FORMAT_S20_3BE ((__force snd_pcm_format_t) 37)  
+#define SNDRV_PCM_FORMAT_U20_3LE ((__force snd_pcm_format_t) 38)  
+#define SNDRV_PCM_FORMAT_U20_3BE ((__force snd_pcm_format_t) 39)  
+#define SNDRV_PCM_FORMAT_S18_3LE ((__force snd_pcm_format_t) 40)  
+#define SNDRV_PCM_FORMAT_S18_3BE ((__force snd_pcm_format_t) 41)  
+#define SNDRV_PCM_FORMAT_U18_3LE ((__force snd_pcm_format_t) 42)  
+#define SNDRV_PCM_FORMAT_U18_3BE ((__force snd_pcm_format_t) 43)  
+#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_U18_3BE
+
+#ifdef SNDRV_LITTLE_ENDIAN
+#define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_LE
+#define SNDRV_PCM_FORMAT_U16 SNDRV_PCM_FORMAT_U16_LE
+#define SNDRV_PCM_FORMAT_S24 SNDRV_PCM_FORMAT_S24_LE
+#define SNDRV_PCM_FORMAT_U24 SNDRV_PCM_FORMAT_U24_LE
+#define SNDRV_PCM_FORMAT_S32 SNDRV_PCM_FORMAT_S32_LE
+#define SNDRV_PCM_FORMAT_U32 SNDRV_PCM_FORMAT_U32_LE
+#define SNDRV_PCM_FORMAT_FLOAT SNDRV_PCM_FORMAT_FLOAT_LE
+#define SNDRV_PCM_FORMAT_FLOAT64 SNDRV_PCM_FORMAT_FLOAT64_LE
+#define SNDRV_PCM_FORMAT_IEC958_SUBFRAME SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE
+#endif
+#ifdef SNDRV_BIG_ENDIAN
+#define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_BE
+#define SNDRV_PCM_FORMAT_U16 SNDRV_PCM_FORMAT_U16_BE
+#define SNDRV_PCM_FORMAT_S24 SNDRV_PCM_FORMAT_S24_BE
+#define SNDRV_PCM_FORMAT_U24 SNDRV_PCM_FORMAT_U24_BE
+#define SNDRV_PCM_FORMAT_S32 SNDRV_PCM_FORMAT_S32_BE
+#define SNDRV_PCM_FORMAT_U32 SNDRV_PCM_FORMAT_U32_BE
+#define SNDRV_PCM_FORMAT_FLOAT SNDRV_PCM_FORMAT_FLOAT_BE
+#define SNDRV_PCM_FORMAT_FLOAT64 SNDRV_PCM_FORMAT_FLOAT64_BE
+#define SNDRV_PCM_FORMAT_IEC958_SUBFRAME SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE
+#endif
+
+typedef int __bitwise snd_pcm_subformat_t;
+#define SNDRV_PCM_SUBFORMAT_STD ((__force snd_pcm_subformat_t) 0)
+#define SNDRV_PCM_SUBFORMAT_LAST SNDRV_PCM_SUBFORMAT_STD
+
+#define SNDRV_PCM_INFO_MMAP 0x00000001  
+#define SNDRV_PCM_INFO_MMAP_VALID 0x00000002  
+#define SNDRV_PCM_INFO_DOUBLE 0x00000004  
+#define SNDRV_PCM_INFO_BATCH 0x00000010  
+#define SNDRV_PCM_INFO_INTERLEAVED 0x00000100  
+#define SNDRV_PCM_INFO_NONINTERLEAVED 0x00000200  
+#define SNDRV_PCM_INFO_COMPLEX 0x00000400  
+#define SNDRV_PCM_INFO_BLOCK_TRANSFER 0x00010000  
+#define SNDRV_PCM_INFO_OVERRANGE 0x00020000  
+#define SNDRV_PCM_INFO_RESUME 0x00040000  
+#define SNDRV_PCM_INFO_PAUSE 0x00080000  
+#define SNDRV_PCM_INFO_HALF_DUPLEX 0x00100000  
+#define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000  
+#define SNDRV_PCM_INFO_SYNC_START 0x00400000  
+#define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000  
+
+typedef int __bitwise snd_pcm_state_t;
+#define SNDRV_PCM_STATE_OPEN ((__force snd_pcm_state_t) 0)  
+#define SNDRV_PCM_STATE_SETUP ((__force snd_pcm_state_t) 1)  
+#define SNDRV_PCM_STATE_PREPARED ((__force snd_pcm_state_t) 2)  
+#define SNDRV_PCM_STATE_RUNNING ((__force snd_pcm_state_t) 3)  
+#define SNDRV_PCM_STATE_XRUN ((__force snd_pcm_state_t) 4)  
+#define SNDRV_PCM_STATE_DRAINING ((__force snd_pcm_state_t) 5)  
+#define SNDRV_PCM_STATE_PAUSED ((__force snd_pcm_state_t) 6)  
+#define SNDRV_PCM_STATE_SUSPENDED ((__force snd_pcm_state_t) 7)  
+#define SNDRV_PCM_STATE_DISCONNECTED ((__force snd_pcm_state_t) 8)  
+#define SNDRV_PCM_STATE_LAST SNDRV_PCM_STATE_DISCONNECTED
+
+enum {
+ SNDRV_PCM_MMAP_OFFSET_DATA = 0x00000000,
+ SNDRV_PCM_MMAP_OFFSET_STATUS = 0x80000000,
+ SNDRV_PCM_MMAP_OFFSET_CONTROL = 0x81000000,
+};
+
+union snd_pcm_sync_id {
+ unsigned char id[16];
+ unsigned short id16[8];
+ unsigned int id32[4];
+};
+
+struct snd_pcm_info {
+ unsigned int device;
+ unsigned int subdevice;
+ int stream;
+ int card;
+ unsigned char id[64];
+ unsigned char name[80];
+ unsigned char subname[32];
+ int dev_class;
+ int dev_subclass;
+ unsigned int subdevices_count;
+ unsigned int subdevices_avail;
+ union snd_pcm_sync_id sync;
+ unsigned char reserved[64];
+};
+
+typedef int snd_pcm_hw_param_t;
+#define SNDRV_PCM_HW_PARAM_ACCESS 0  
+#define SNDRV_PCM_HW_PARAM_FORMAT 1  
+#define SNDRV_PCM_HW_PARAM_SUBFORMAT 2  
+#define SNDRV_PCM_HW_PARAM_FIRST_MASK SNDRV_PCM_HW_PARAM_ACCESS
+#define SNDRV_PCM_HW_PARAM_LAST_MASK SNDRV_PCM_HW_PARAM_SUBFORMAT
+
+#define SNDRV_PCM_HW_PARAM_SAMPLE_BITS 8  
+#define SNDRV_PCM_HW_PARAM_FRAME_BITS 9  
+#define SNDRV_PCM_HW_PARAM_CHANNELS 10  
+#define SNDRV_PCM_HW_PARAM_RATE 11  
+#define SNDRV_PCM_HW_PARAM_PERIOD_TIME 12  
+#define SNDRV_PCM_HW_PARAM_PERIOD_SIZE 13  
+#define SNDRV_PCM_HW_PARAM_PERIOD_BYTES 14  
+#define SNDRV_PCM_HW_PARAM_PERIODS 15  
+#define SNDRV_PCM_HW_PARAM_BUFFER_TIME 16  
+#define SNDRV_PCM_HW_PARAM_BUFFER_SIZE 17  
+#define SNDRV_PCM_HW_PARAM_BUFFER_BYTES 18  
+#define SNDRV_PCM_HW_PARAM_TICK_TIME 19  
+#define SNDRV_PCM_HW_PARAM_FIRST_INTERVAL SNDRV_PCM_HW_PARAM_SAMPLE_BITS
+#define SNDRV_PCM_HW_PARAM_LAST_INTERVAL SNDRV_PCM_HW_PARAM_TICK_TIME
+
+#define SNDRV_PCM_HW_PARAMS_NORESAMPLE (1<<0)  
+
+struct snd_interval {
+ unsigned int min, max;
+ unsigned int openmin:1,
+ openmax:1,
+ integer:1,
+ empty:1;
+};
+
+#define SNDRV_MASK_MAX 256
+
+struct snd_mask {
+ __u32 bits[(SNDRV_MASK_MAX+31)/32];
+};
+
+struct snd_pcm_hw_params {
+ unsigned int flags;
+ struct snd_mask masks[SNDRV_PCM_HW_PARAM_LAST_MASK -
+ SNDRV_PCM_HW_PARAM_FIRST_MASK + 1];
+ struct snd_mask mres[5];
+ struct snd_interval intervals[SNDRV_PCM_HW_PARAM_LAST_INTERVAL -
+ SNDRV_PCM_HW_PARAM_FIRST_INTERVAL + 1];
+ struct snd_interval ires[9];
+ unsigned int rmask;
+ unsigned int cmask;
+ unsigned int info;
+ unsigned int msbits;
+ unsigned int rate_num;
+ unsigned int rate_den;
+ snd_pcm_uframes_t fifo_size;
+ unsigned char reserved[64];
+};
+
+enum {
+ SNDRV_PCM_TSTAMP_NONE = 0,
+ SNDRV_PCM_TSTAMP_ENABLE,
+ SNDRV_PCM_TSTAMP_LAST = SNDRV_PCM_TSTAMP_ENABLE,
+};
+
+struct snd_pcm_sw_params {
+ int tstamp_mode;
+ unsigned int period_step;
+ unsigned int sleep_min;
+ snd_pcm_uframes_t avail_min;
+ snd_pcm_uframes_t xfer_align;
+ snd_pcm_uframes_t start_threshold;
+ snd_pcm_uframes_t stop_threshold;
+ snd_pcm_uframes_t silence_threshold;
+ snd_pcm_uframes_t silence_size;
+ snd_pcm_uframes_t boundary;
+ unsigned char reserved[64];
+};
+
+struct snd_pcm_channel_info {
+ unsigned int channel;
+ __kernel_off_t offset;
+ unsigned int first;
+ unsigned int step;
+};
+
+struct snd_pcm_status {
+ snd_pcm_state_t state;
+ struct timespec trigger_tstamp;
+ struct timespec tstamp;
+ snd_pcm_uframes_t appl_ptr;
+ snd_pcm_uframes_t hw_ptr;
+ snd_pcm_sframes_t delay;
+ snd_pcm_uframes_t avail;
+ snd_pcm_uframes_t avail_max;
+ snd_pcm_uframes_t overrange;
+ snd_pcm_state_t suspended_state;
+ unsigned char reserved[60];
+};
+
+struct snd_pcm_mmap_status {
+ snd_pcm_state_t state;
+ int pad1;
+ snd_pcm_uframes_t hw_ptr;
+ struct timespec tstamp;
+ snd_pcm_state_t suspended_state;
+};
+
+struct snd_pcm_mmap_control {
+ snd_pcm_uframes_t appl_ptr;
+ snd_pcm_uframes_t avail_min;
+};
+
+#define SNDRV_PCM_SYNC_PTR_HWSYNC (1<<0)  
+#define SNDRV_PCM_SYNC_PTR_APPL (1<<1)  
+#define SNDRV_PCM_SYNC_PTR_AVAIL_MIN (1<<2)  
+
+struct snd_pcm_sync_ptr {
+ unsigned int flags;
+ union {
+ struct snd_pcm_mmap_status status;
+ unsigned char reserved[64];
+ } s;
+ union {
+ struct snd_pcm_mmap_control control;
+ unsigned char reserved[64];
+ } c;
+};
+
+struct snd_xferi {
+ snd_pcm_sframes_t result;
+ void __user *buf;
+ snd_pcm_uframes_t frames;
+};
+
+struct snd_xfern {
+ snd_pcm_sframes_t result;
+ void __user * __user *bufs;
+ snd_pcm_uframes_t frames;
+};
+
+enum {
+ SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0,
+ SNDRV_PCM_TSTAMP_TYPE_MONOTONIC,
+ SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC,
+};
+
+#define SNDRV_PCM_IOCTL_PVERSION _IOR('A', 0x00, int)
+#define SNDRV_PCM_IOCTL_INFO _IOR('A', 0x01, struct snd_pcm_info)
+#define SNDRV_PCM_IOCTL_TSTAMP _IOW('A', 0x02, int)
+#define SNDRV_PCM_IOCTL_TTSTAMP _IOW('A', 0x03, int)
+#define SNDRV_PCM_IOCTL_HW_REFINE _IOWR('A', 0x10, struct snd_pcm_hw_params)
+#define SNDRV_PCM_IOCTL_HW_PARAMS _IOWR('A', 0x11, struct snd_pcm_hw_params)
+#define SNDRV_PCM_IOCTL_HW_FREE _IO('A', 0x12)
+#define SNDRV_PCM_IOCTL_SW_PARAMS _IOWR('A', 0x13, struct snd_pcm_sw_params)
+#define SNDRV_PCM_IOCTL_STATUS _IOR('A', 0x20, struct snd_pcm_status)
+#define SNDRV_PCM_IOCTL_DELAY _IOR('A', 0x21, snd_pcm_sframes_t)
+#define SNDRV_PCM_IOCTL_HWSYNC _IO('A', 0x22)
+#define SNDRV_PCM_IOCTL_SYNC_PTR _IOWR('A', 0x23, struct snd_pcm_sync_ptr)
+#define SNDRV_PCM_IOCTL_CHANNEL_INFO _IOR('A', 0x32, struct snd_pcm_channel_info)
+#define SNDRV_PCM_IOCTL_PREPARE _IO('A', 0x40)
+#define SNDRV_PCM_IOCTL_RESET _IO('A', 0x41)
+#define SNDRV_PCM_IOCTL_START _IO('A', 0x42)
+#define SNDRV_PCM_IOCTL_DROP _IO('A', 0x43)
+#define SNDRV_PCM_IOCTL_DRAIN _IO('A', 0x44)
+#define SNDRV_PCM_IOCTL_PAUSE _IOW('A', 0x45, int)
+#define SNDRV_PCM_IOCTL_REWIND _IOW('A', 0x46, snd_pcm_uframes_t)
+#define SNDRV_PCM_IOCTL_RESUME _IO('A', 0x47)
+#define SNDRV_PCM_IOCTL_XRUN _IO('A', 0x48)
+#define SNDRV_PCM_IOCTL_FORWARD _IOW('A', 0x49, snd_pcm_uframes_t)
+#define SNDRV_PCM_IOCTL_WRITEI_FRAMES _IOW('A', 0x50, struct snd_xferi)
+#define SNDRV_PCM_IOCTL_READI_FRAMES _IOR('A', 0x51, struct snd_xferi)
+#define SNDRV_PCM_IOCTL_WRITEN_FRAMES _IOW('A', 0x52, struct snd_xfern)
+#define SNDRV_PCM_IOCTL_READN_FRAMES _IOR('A', 0x53, struct snd_xfern)
+#define SNDRV_PCM_IOCTL_LINK _IOW('A', 0x60, int)
+#define SNDRV_PCM_IOCTL_UNLINK _IO('A', 0x61)
+
+#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 0)
+
+enum {
+ SNDRV_RAWMIDI_STREAM_OUTPUT = 0,
+ SNDRV_RAWMIDI_STREAM_INPUT,
+ SNDRV_RAWMIDI_STREAM_LAST = SNDRV_RAWMIDI_STREAM_INPUT,
+};
+
+#define SNDRV_RAWMIDI_INFO_OUTPUT 0x00000001
+#define SNDRV_RAWMIDI_INFO_INPUT 0x00000002
+#define SNDRV_RAWMIDI_INFO_DUPLEX 0x00000004
+
+struct snd_rawmidi_info {
+ unsigned int device;
+ unsigned int subdevice;
+ int stream;
+ int card;
+ unsigned int flags;
+ unsigned char id[64];
+ unsigned char name[80];
+ unsigned char subname[32];
+ unsigned int subdevices_count;
+ unsigned int subdevices_avail;
+ unsigned char reserved[64];
+};
+
+struct snd_rawmidi_params {
+ int stream;
+ size_t buffer_size;
+ size_t avail_min;
+ unsigned int no_active_sensing: 1;
+ unsigned char reserved[16];
+};
+
+struct snd_rawmidi_status {
+ int stream;
+ struct timespec tstamp;
+ size_t avail;
+ size_t xruns;
+ unsigned char reserved[16];
+};
+
+#define SNDRV_RAWMIDI_IOCTL_PVERSION _IOR('W', 0x00, int)
+#define SNDRV_RAWMIDI_IOCTL_INFO _IOR('W', 0x01, struct snd_rawmidi_info)
+#define SNDRV_RAWMIDI_IOCTL_PARAMS _IOWR('W', 0x10, struct snd_rawmidi_params)
+#define SNDRV_RAWMIDI_IOCTL_STATUS _IOWR('W', 0x20, struct snd_rawmidi_status)
+#define SNDRV_RAWMIDI_IOCTL_DROP _IOW('W', 0x30, int)
+#define SNDRV_RAWMIDI_IOCTL_DRAIN _IOW('W', 0x31, int)
+
+#define SNDRV_TIMER_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 6)
+
+enum {
+ SNDRV_TIMER_CLASS_NONE = -1,
+ SNDRV_TIMER_CLASS_SLAVE = 0,
+ SNDRV_TIMER_CLASS_GLOBAL,
+ SNDRV_TIMER_CLASS_CARD,
+ SNDRV_TIMER_CLASS_PCM,
+ SNDRV_TIMER_CLASS_LAST = SNDRV_TIMER_CLASS_PCM,
+};
+
+enum {
+ SNDRV_TIMER_SCLASS_NONE = 0,
+ SNDRV_TIMER_SCLASS_APPLICATION,
+ SNDRV_TIMER_SCLASS_SEQUENCER,
+ SNDRV_TIMER_SCLASS_OSS_SEQUENCER,
+ SNDRV_TIMER_SCLASS_LAST = SNDRV_TIMER_SCLASS_OSS_SEQUENCER,
+};
+
+#define SNDRV_TIMER_GLOBAL_SYSTEM 0
+#define SNDRV_TIMER_GLOBAL_RTC 1
+#define SNDRV_TIMER_GLOBAL_HPET 2
+#define SNDRV_TIMER_GLOBAL_HRTIMER 3
+
+#define SNDRV_TIMER_FLG_SLAVE (1<<0)  
+
+struct snd_timer_id {
+ int dev_class;
+ int dev_sclass;
+ int card;
+ int device;
+ int subdevice;
+};
+
+struct snd_timer_ginfo {
+ struct snd_timer_id tid;
+ unsigned int flags;
+ int card;
+ unsigned char id[64];
+ unsigned char name[80];
+ unsigned long reserved0;
+ unsigned long resolution;
+ unsigned long resolution_min;
+ unsigned long resolution_max;
+ unsigned int clients;
+ unsigned char reserved[32];
+};
+
+struct snd_timer_gparams {
+ struct snd_timer_id tid;
+ unsigned long period_num;
+ unsigned long period_den;
+ unsigned char reserved[32];
+};
+
+struct snd_timer_gstatus {
+ struct snd_timer_id tid;
+ unsigned long resolution;
+ unsigned long resolution_num;
+ unsigned long resolution_den;
+ unsigned char reserved[32];
+};
+
+struct snd_timer_select {
+ struct snd_timer_id id;
+ unsigned char reserved[32];
+};
+
+struct snd_timer_info {
+ unsigned int flags;
+ int card;
+ unsigned char id[64];
+ unsigned char name[80];
+ unsigned long reserved0;
+ unsigned long resolution;
+ unsigned char reserved[64];
+};
+
+#define SNDRV_TIMER_PSFLG_AUTO (1<<0)  
+#define SNDRV_TIMER_PSFLG_EXCLUSIVE (1<<1)  
+#define SNDRV_TIMER_PSFLG_EARLY_EVENT (1<<2)  
+
+struct snd_timer_params {
+ unsigned int flags;
+ unsigned int ticks;
+ unsigned int queue_size;
+ unsigned int reserved0;
+ unsigned int filter;
+ unsigned char reserved[60];
+};
+
+struct snd_timer_status {
+ struct timespec tstamp;
+ unsigned int resolution;
+ unsigned int lost;
+ unsigned int overrun;
+ unsigned int queue;
+ unsigned char reserved[64];
+};
+
+#define SNDRV_TIMER_IOCTL_PVERSION _IOR('T', 0x00, int)
+#define SNDRV_TIMER_IOCTL_NEXT_DEVICE _IOWR('T', 0x01, struct snd_timer_id)
+#define SNDRV_TIMER_IOCTL_TREAD _IOW('T', 0x02, int)
+#define SNDRV_TIMER_IOCTL_GINFO _IOWR('T', 0x03, struct snd_timer_ginfo)
+#define SNDRV_TIMER_IOCTL_GPARAMS _IOW('T', 0x04, struct snd_timer_gparams)
+#define SNDRV_TIMER_IOCTL_GSTATUS _IOWR('T', 0x05, struct snd_timer_gstatus)
+#define SNDRV_TIMER_IOCTL_SELECT _IOW('T', 0x10, struct snd_timer_select)
+#define SNDRV_TIMER_IOCTL_INFO _IOR('T', 0x11, struct snd_timer_info)
+#define SNDRV_TIMER_IOCTL_PARAMS _IOW('T', 0x12, struct snd_timer_params)
+#define SNDRV_TIMER_IOCTL_STATUS _IOR('T', 0x14, struct snd_timer_status)
+
+#define SNDRV_TIMER_IOCTL_START _IO('T', 0xa0)
+#define SNDRV_TIMER_IOCTL_STOP _IO('T', 0xa1)
+#define SNDRV_TIMER_IOCTL_CONTINUE _IO('T', 0xa2)
+#define SNDRV_TIMER_IOCTL_PAUSE _IO('T', 0xa3)
+
+struct snd_timer_read {
+ unsigned int resolution;
+ unsigned int ticks;
+};
+
+enum {
+ SNDRV_TIMER_EVENT_RESOLUTION = 0,
+ SNDRV_TIMER_EVENT_TICK,
+ SNDRV_TIMER_EVENT_START,
+ SNDRV_TIMER_EVENT_STOP,
+ SNDRV_TIMER_EVENT_CONTINUE,
+ SNDRV_TIMER_EVENT_PAUSE,
+ SNDRV_TIMER_EVENT_EARLY,
+ SNDRV_TIMER_EVENT_SUSPEND,
+ SNDRV_TIMER_EVENT_RESUME,
+
+ SNDRV_TIMER_EVENT_MSTART = SNDRV_TIMER_EVENT_START + 10,
+ SNDRV_TIMER_EVENT_MSTOP = SNDRV_TIMER_EVENT_STOP + 10,
+ SNDRV_TIMER_EVENT_MCONTINUE = SNDRV_TIMER_EVENT_CONTINUE + 10,
+ SNDRV_TIMER_EVENT_MPAUSE = SNDRV_TIMER_EVENT_PAUSE + 10,
+ SNDRV_TIMER_EVENT_MSUSPEND = SNDRV_TIMER_EVENT_SUSPEND + 10,
+ SNDRV_TIMER_EVENT_MRESUME = SNDRV_TIMER_EVENT_RESUME + 10,
+};
+
+struct snd_timer_tread {
+ int event;
+ struct timespec tstamp;
+ unsigned int val;
+};
+
+#define SNDRV_CTL_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 6)
+
+struct snd_ctl_card_info {
+ int card;
+ int pad;
+ unsigned char id[16];
+ unsigned char driver[16];
+ unsigned char name[32];
+ unsigned char longname[80];
+ unsigned char reserved_[16];
+ unsigned char mixername[80];
+ unsigned char components[128];
+};
+
+typedef int __bitwise snd_ctl_elem_type_t;
+#define SNDRV_CTL_ELEM_TYPE_NONE ((__force snd_ctl_elem_type_t) 0)  
+#define SNDRV_CTL_ELEM_TYPE_BOOLEAN ((__force snd_ctl_elem_type_t) 1)  
+#define SNDRV_CTL_ELEM_TYPE_INTEGER ((__force snd_ctl_elem_type_t) 2)  
+#define SNDRV_CTL_ELEM_TYPE_ENUMERATED ((__force snd_ctl_elem_type_t) 3)  
+#define SNDRV_CTL_ELEM_TYPE_BYTES ((__force snd_ctl_elem_type_t) 4)  
+#define SNDRV_CTL_ELEM_TYPE_IEC958 ((__force snd_ctl_elem_type_t) 5)  
+#define SNDRV_CTL_ELEM_TYPE_INTEGER64 ((__force snd_ctl_elem_type_t) 6)  
+#define SNDRV_CTL_ELEM_TYPE_LAST SNDRV_CTL_ELEM_TYPE_INTEGER64
+
+typedef int __bitwise snd_ctl_elem_iface_t;
+#define SNDRV_CTL_ELEM_IFACE_CARD ((__force snd_ctl_elem_iface_t) 0)  
+#define SNDRV_CTL_ELEM_IFACE_HWDEP ((__force snd_ctl_elem_iface_t) 1)  
+#define SNDRV_CTL_ELEM_IFACE_MIXER ((__force snd_ctl_elem_iface_t) 2)  
+#define SNDRV_CTL_ELEM_IFACE_PCM ((__force snd_ctl_elem_iface_t) 3)  
+#define SNDRV_CTL_ELEM_IFACE_RAWMIDI ((__force snd_ctl_elem_iface_t) 4)  
+#define SNDRV_CTL_ELEM_IFACE_TIMER ((__force snd_ctl_elem_iface_t) 5)  
+#define SNDRV_CTL_ELEM_IFACE_SEQUENCER ((__force snd_ctl_elem_iface_t) 6)  
+#define SNDRV_CTL_ELEM_IFACE_LAST SNDRV_CTL_ELEM_IFACE_SEQUENCER
+
+#define SNDRV_CTL_ELEM_ACCESS_READ (1<<0)
+#define SNDRV_CTL_ELEM_ACCESS_WRITE (1<<1)
+#define SNDRV_CTL_ELEM_ACCESS_READWRITE (SNDRV_CTL_ELEM_ACCESS_READ|SNDRV_CTL_ELEM_ACCESS_WRITE)
+#define SNDRV_CTL_ELEM_ACCESS_VOLATILE (1<<2)  
+#define SNDRV_CTL_ELEM_ACCESS_TIMESTAMP (1<<3)  
+#define SNDRV_CTL_ELEM_ACCESS_TLV_READ (1<<4)  
+#define SNDRV_CTL_ELEM_ACCESS_TLV_WRITE (1<<5)  
+#define SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE (SNDRV_CTL_ELEM_ACCESS_TLV_READ|SNDRV_CTL_ELEM_ACCESS_TLV_WRITE)
+#define SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND (1<<6)  
+#define SNDRV_CTL_ELEM_ACCESS_INACTIVE (1<<8)  
+#define SNDRV_CTL_ELEM_ACCESS_LOCK (1<<9)  
+#define SNDRV_CTL_ELEM_ACCESS_OWNER (1<<10)  
+#define SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK (1<<28)   
+#define SNDRV_CTL_ELEM_ACCESS_USER (1<<29)  
+
+#define SNDRV_CTL_POWER_D0 0x0000  
+#define SNDRV_CTL_POWER_D1 0x0100  
+#define SNDRV_CTL_POWER_D2 0x0200  
+#define SNDRV_CTL_POWER_D3 0x0300  
+#define SNDRV_CTL_POWER_D3hot (SNDRV_CTL_POWER_D3|0x0000)  
+#define SNDRV_CTL_POWER_D3cold (SNDRV_CTL_POWER_D3|0x0001)  
+
+struct snd_ctl_elem_id {
+ unsigned int numid;
+ snd_ctl_elem_iface_t iface;
+ unsigned int device;
+ unsigned int subdevice;
+ unsigned char name[44];
+ unsigned int index;
+};
+
+struct snd_ctl_elem_list {
+ unsigned int offset;
+ unsigned int space;
+ unsigned int used;
+ unsigned int count;
+ struct snd_ctl_elem_id __user *pids;
+ unsigned char reserved[50];
+};
+
+struct snd_ctl_elem_info {
+ struct snd_ctl_elem_id id;
+ snd_ctl_elem_type_t type;
+ unsigned int access;
+ unsigned int count;
+ __kernel_pid_t owner;
+ union {
+ struct {
+ long min;
+ long max;
+ long step;
+ } integer;
+ struct {
+ long long min;
+ long long max;
+ long long step;
+ } integer64;
+ struct {
+ unsigned int items;
+ unsigned int item;
+ char name[64];
+ } enumerated;
+ unsigned char reserved[128];
+ } value;
+ union {
+ unsigned short d[4];
+ unsigned short *d_ptr;
+ } dimen;
+ unsigned char reserved[64-4*sizeof(unsigned short)];
+};
+
+struct snd_ctl_elem_value {
+ struct snd_ctl_elem_id id;
+ unsigned int indirect: 1;
+ union {
+ union {
+ long value[128];
+ long *value_ptr;
+ } integer;
+ union {
+ long long value[64];
+ long long *value_ptr;
+ } integer64;
+ union {
+ unsigned int item[128];
+ unsigned int *item_ptr;
+ } enumerated;
+ union {
+ unsigned char data[512];
+ unsigned char *data_ptr;
+ } bytes;
+ struct snd_aes_iec958 iec958;
+ } value;
+ struct timespec tstamp;
+ unsigned char reserved[128-sizeof(struct timespec)];
+};
+
+struct snd_ctl_tlv {
+ unsigned int numid;
+ unsigned int length;
+ unsigned int tlv[0];
+};
+
+#define SNDRV_CTL_IOCTL_PVERSION _IOR('U', 0x00, int)
+#define SNDRV_CTL_IOCTL_CARD_INFO _IOR('U', 0x01, struct snd_ctl_card_info)
+#define SNDRV_CTL_IOCTL_ELEM_LIST _IOWR('U', 0x10, struct snd_ctl_elem_list)
+#define SNDRV_CTL_IOCTL_ELEM_INFO _IOWR('U', 0x11, struct snd_ctl_elem_info)
+#define SNDRV_CTL_IOCTL_ELEM_READ _IOWR('U', 0x12, struct snd_ctl_elem_value)
+#define SNDRV_CTL_IOCTL_ELEM_WRITE _IOWR('U', 0x13, struct snd_ctl_elem_value)
+#define SNDRV_CTL_IOCTL_ELEM_LOCK _IOW('U', 0x14, struct snd_ctl_elem_id)
+#define SNDRV_CTL_IOCTL_ELEM_UNLOCK _IOW('U', 0x15, struct snd_ctl_elem_id)
+#define SNDRV_CTL_IOCTL_SUBSCRIBE_EVENTS _IOWR('U', 0x16, int)
+#define SNDRV_CTL_IOCTL_ELEM_ADD _IOWR('U', 0x17, struct snd_ctl_elem_info)
+#define SNDRV_CTL_IOCTL_ELEM_REPLACE _IOWR('U', 0x18, struct snd_ctl_elem_info)
+#define SNDRV_CTL_IOCTL_ELEM_REMOVE _IOWR('U', 0x19, struct snd_ctl_elem_id)
+#define SNDRV_CTL_IOCTL_TLV_READ _IOWR('U', 0x1a, struct snd_ctl_tlv)
+#define SNDRV_CTL_IOCTL_TLV_WRITE _IOWR('U', 0x1b, struct snd_ctl_tlv)
+#define SNDRV_CTL_IOCTL_TLV_COMMAND _IOWR('U', 0x1c, struct snd_ctl_tlv)
+#define SNDRV_CTL_IOCTL_HWDEP_NEXT_DEVICE _IOWR('U', 0x20, int)
+#define SNDRV_CTL_IOCTL_HWDEP_INFO _IOR('U', 0x21, struct snd_hwdep_info)
+#define SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE _IOR('U', 0x30, int)
+#define SNDRV_CTL_IOCTL_PCM_INFO _IOWR('U', 0x31, struct snd_pcm_info)
+#define SNDRV_CTL_IOCTL_PCM_PREFER_SUBDEVICE _IOW('U', 0x32, int)
+#define SNDRV_CTL_IOCTL_RAWMIDI_NEXT_DEVICE _IOWR('U', 0x40, int)
+#define SNDRV_CTL_IOCTL_RAWMIDI_INFO _IOWR('U', 0x41, struct snd_rawmidi_info)
+#define SNDRV_CTL_IOCTL_RAWMIDI_PREFER_SUBDEVICE _IOW('U', 0x42, int)
+#define SNDRV_CTL_IOCTL_POWER _IOWR('U', 0xd0, int)
+#define SNDRV_CTL_IOCTL_POWER_STATE _IOR('U', 0xd1, int)
+
+enum sndrv_ctl_event_type {
+ SNDRV_CTL_EVENT_ELEM = 0,
+ SNDRV_CTL_EVENT_LAST = SNDRV_CTL_EVENT_ELEM,
+};
+
+#define SNDRV_CTL_EVENT_MASK_VALUE (1<<0)  
+#define SNDRV_CTL_EVENT_MASK_INFO (1<<1)  
+#define SNDRV_CTL_EVENT_MASK_ADD (1<<2)  
+#define SNDRV_CTL_EVENT_MASK_TLV (1<<3)  
+#define SNDRV_CTL_EVENT_MASK_REMOVE (~0U)  
+
+struct snd_ctl_event {
+ int type;
+ union {
+ struct {
+ unsigned int mask;
+ struct snd_ctl_elem_id id;
+ } elem;
+ unsigned char data8[60];
+ } data;
+};
+
+#define SNDRV_CTL_NAME_NONE ""
+#define SNDRV_CTL_NAME_PLAYBACK "Playback "
+#define SNDRV_CTL_NAME_CAPTURE "Capture "
+
+#define SNDRV_CTL_NAME_IEC958_NONE ""
+#define SNDRV_CTL_NAME_IEC958_SWITCH "Switch"
+#define SNDRV_CTL_NAME_IEC958_VOLUME "Volume"
+#define SNDRV_CTL_NAME_IEC958_DEFAULT "Default"
+#define SNDRV_CTL_NAME_IEC958_MASK "Mask"
+#define SNDRV_CTL_NAME_IEC958_CON_MASK "Con Mask"
+#define SNDRV_CTL_NAME_IEC958_PRO_MASK "Pro Mask"
+#define SNDRV_CTL_NAME_IEC958_PCM_STREAM "PCM Stream"
+#define SNDRV_CTL_NAME_IEC958(expl,direction,what) "IEC958 " expl SNDRV_CTL_NAME_##direction SNDRV_CTL_NAME_IEC958_##what
+
+#endif
+
diff --git a/libaudio/secril-client.h b/libaudio/secril-client.h
index 2473ae7..7bbaa03 100755..100644
--- a/libaudio/secril-client.h
+++ b/libaudio/secril-client.h
@@ -138,7 +138,8 @@ typedef enum _AudioPath {
     SOUND_AUDIO_PATH_HEADSET,
     SOUND_AUDIO_PATH_SPEAKER,
     SOUND_AUDIO_PATH_BLUETOOTH,
-    SOUND_AUDIO_PATH_BLUETOOTH_NO_NR
+    SOUND_AUDIO_PATH_BLUETOOTH_NO_NR,
+    SOUND_AUDIO_PATH_HEADPHONE
 } AudioPath;
 
 /**