Update libaudio from crespo (with FM added)

11 files changed, 757 insertions, 2674 deletions

diff --git a/libaudio/Android.mk b/libaudio/Android.mk
index 4af67bb..183ede3 100644
--- a/libaudio/Android.mk
+++ b/libaudio/Android.mk
@@ -1,65 +1,45 @@
-ifneq ($(TARGET_PROVIDES_LIBAUDIO),true)
-
 LOCAL_PATH:= $(call my-dir)
 
 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)
-
-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)
-
-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)
+LOCAL_SRC_FILES:= \
+	AudioHardware.cpp
 
-include $(CLEAR_VARS)
-LOCAL_SRC_FILES:= AudioHardware.cpp alsa_mixer.c alsa_pcm.c
+LOCAL_MODULE := audio.primary.aries
 LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/hw
-LOCAL_MODULE:= audio.primary.aries
-LOCAL_MODULE_TAGS := optional
-LOCAL_STATIC_LIBRARIES := libmedia_helper
-LOCAL_WHOLE_STATIC_LIBRARIES := libaudiohw_legacy
-LOCAL_SHARED_LIBRARIES:= libc libcutils libutils libmedia libhardware_legacy
-# TODO: Fix A2DP
-#ifeq ($(BOARD_HAVE_BLUETOOTH),true)
-#  LOCAL_SHARED_LIBRARIES += liba2dp
-#endif
+LOCAL_STATIC_LIBRARIES:= libmedia_helper
+LOCAL_SHARED_LIBRARIES:= \
+	libutils \
+	libhardware_legacy \
+	libtinyalsa \
+	libaudioutils
 
-# TODO: Fix FM
-#ifeq ($(BOARD_HAVE_FM_RADIO),true)
-#  LOCAL_CFLAGS += -DHAVE_FM_RADIO
-#endif
+LOCAL_WHOLE_STATIC_LIBRARIES := libaudiohw_legacy
+LOCAL_MODULE_TAGS := optional
 
-ifeq ($(TARGET_SIMULATOR),true)
- LOCAL_LDLIBS += -ldl
-else
- LOCAL_SHARED_LIBRARIES += libdl
-endif
+LOCAL_SHARED_LIBRARIES += libdl
+LOCAL_C_INCLUDES += \
+	external/tinyalsa/include \
+	system/media/audio_effects/include \
+	system/media/audio_utils/include
 
 include $(BUILD_SHARED_LIBRARY)
 
 include $(CLEAR_VARS)
-LOCAL_SRC_FILES:= AudioPolicyManager.cpp
+
+LOCAL_SRC_FILES := AudioPolicyManager.cpp
+LOCAL_SHARED_LIBRARIES := libcutils libutils libmedia
+LOCAL_STATIC_LIBRARIES := libmedia_helper
+LOCAL_WHOLE_STATIC_LIBRARIES := libaudiopolicy_legacy
+LOCAL_MODULE := audio_policy.aries
 LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/hw
-LOCAL_MODULE:= audio_policy.aries
 LOCAL_MODULE_TAGS := optional
-LOCAL_STATIC_LIBRARIES := libmedia_helper
-LOCAL_WHOLE_STATIC_LIBRARIES:= libaudiopolicy_legacy
-LOCAL_SHARED_LIBRARIES:= libc libcutils libutils libmedia
+
 ifeq ($(BOARD_HAVE_BLUETOOTH),true)
   LOCAL_CFLAGS += -DWITH_A2DP
 endif
-include $(BUILD_SHARED_LIBRARY)
 
+ifeq ($(BOARD_HAVE_FM_RADIO),true)
+  LOCAL_CFLAGS += -DHAVE_FM_RADIO
 endif
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/libaudio/AudioHardware.cpp b/libaudio/AudioHardware.cpp
index 5d3f43a..b0f23c3 100644
--- a/libaudio/AudioHardware.cpp
+++ b/libaudio/AudioHardware.cpp
@@ -17,7 +17,6 @@
 #include <math.h>
 
 #define LOG_NDEBUG 0
-
 #define LOG_TAG "AudioHardware"
 
 #include <utils/Log.h>
@@ -34,10 +33,10 @@
 
 #include "AudioHardware.h"
 #include <media/AudioRecord.h>
-#include <hardware_legacy/power.h>
+#include <audio_effects/effect_aec.h>
 
 extern "C" {
-#include "alsa_audio.h"
+#include <tinyalsa/asoundlib.h>
 }
 
 #ifdef HAVE_FM_RADIO
@@ -47,8 +46,13 @@ extern "C" {
 
 namespace android_audio_legacy {
 
-const uint32_t AudioHardware::inputSamplingRates[] = {
-        8000, 11025, 16000, 22050, 44100
+const uint32_t AudioHardware::inputConfigTable[][AudioHardware::INPUT_CONFIG_CNT] = {
+        {8000, 4},
+        {11025, 4},
+        {16000, 2},
+        {22050, 2},
+        {32000, 1},
+        {44100, 1}
 };
 
 //  trace driver operations for dump
@@ -80,7 +84,6 @@ enum {
 const char *AudioHardware::inputPathNameDefault = "Default";
 const char *AudioHardware::inputPathNameCamcorder = "Camcorder";
 const char *AudioHardware::inputPathNameVoiceRecognition = "Voice Recognition";
-const char *AudioHardware::inputPathNameVoiceCommunication = "Voice Communication";
 
 AudioHardware::AudioHardware() :
     mInit(false),
@@ -90,12 +93,14 @@ AudioHardware::AudioHardware() :
     mPcmOpenCnt(0),
     mMixerOpenCnt(0),
     mInCallAudioMode(false),
+    mVoiceVol(1.0f),
     mInputSource(AUDIO_SOURCE_DEFAULT),
     mBluetoothNrec(true),
     mTTYMode(TTY_MODE_OFF),
     mSecRilLibHandle(NULL),
     mRilClient(0),
     mActivatedCP(false),
+    mEchoReference(NULL),
 #ifdef HAVE_FM_RADIO
     mFmFd(-1),
     mFmVolume(1),
@@ -251,6 +256,7 @@ AudioStreamOut* AudioHardware::openOutputStream(
 
 void AudioHardware::closeOutputStream(AudioStreamOut* out) {
     sp <AudioStreamOutALSA> spOut;
+    sp<AudioStreamInALSA> spIn;
     {
         Mutex::Autolock lock(mLock);
         if (mOutput == 0 || mOutput.get() != out) {
@@ -259,7 +265,16 @@ void AudioHardware::closeOutputStream(AudioStreamOut* out) {
         }
         spOut = mOutput;
         mOutput.clear();
+        if (mEchoReference != NULL) {
+            spIn = getActiveInput_l();
+        }
+    }
+    if (spIn != 0) {
+        // this will safely release the echo reference by calling releaseEchoReference()
+        // after placing the active input in standby
+        spIn->standby();
     }
+
     spOut.clear();
 }
 
@@ -370,8 +385,10 @@ status_t AudioHardware::setMode(int mode)
     status = AudioHardwareBase::setMode(mode);
     LOGV("setMode() : new %d, old %d", mMode, prevMode);
     if (status == NO_ERROR) {
+        bool modeNeedsCPActive = mMode == AudioSystem::MODE_IN_CALL ||
+                                    mMode == AudioSystem::MODE_RINGTONE;
         // activate call clock in radio when entering in call or ringtone mode
-        if (prevMode == AudioSystem::MODE_NORMAL)
+        if (modeNeedsCPActive)
         {
             if ((!mActivatedCP) && (mSecRilLibHandle) && (connectRILDIfRequired() == OK)) {
                 setCallClockSync(mRilClient, SOUND_CLOCK_START);
@@ -393,18 +410,19 @@ status_t AudioHardware::setMode(int mode)
             openPcmOut_l();
             openMixer_l();
             setInputSource_l(AUDIO_SOURCE_DEFAULT);
+            setVoiceVolume_l(mVoiceVol);
             mInCallAudioMode = true;
         }
-        if (mMode == AudioSystem::MODE_NORMAL && mInCallAudioMode) {
+        if (mMode != AudioSystem::MODE_IN_CALL && mInCallAudioMode) {
             setInputSource_l(mInputSource);
             if (mMixer != NULL) {
                 TRACE_DRIVER_IN(DRV_MIXER_GET)
-                struct mixer_ctl *ctl= mixer_get_control(mMixer, "Playback Path", 0);
+                struct mixer_ctl *ctl= mixer_get_ctl_by_name(mMixer, "Playback Path");
                 TRACE_DRIVER_OUT
                 if (ctl != NULL) {
                     LOGV("setMode() reset Playback Path to RCV");
                     TRACE_DRIVER_IN(DRV_MIXER_SEL)
-                    mixer_ctl_select(ctl, "RCV");
+                    mixer_ctl_set_enum_by_string(ctl, "RCV");
                     TRACE_DRIVER_OUT
                 }
             }
@@ -424,7 +442,7 @@ status_t AudioHardware::setMode(int mode)
             mInCallAudioMode = false;
         }
 
-        if (mMode == AudioSystem::MODE_NORMAL) {
+        if (!modeNeedsCPActive) {
             if(mActivatedCP)
                 mActivatedCP = false;
         }
@@ -444,6 +462,7 @@ status_t AudioHardware::setMode(int mode)
         enableFMRadio();
     }
 #endif
+
     return status;
 }
 
@@ -487,6 +506,12 @@ status_t AudioHardware::setParameters(const String8& keyValuePairs)
     const char TTY_MODE_VALUE_VCO[] = "tty_vco";
     const char TTY_MODE_VALUE_HCO[] = "tty_hco";
     const char TTY_MODE_VALUE_FULL[] = "tty_full";
+#ifdef HAVE_FM_RADIO
+    const char FM_RADIO_KEY_ON[] = "fm_on";
+    const char FM_RADIO_KEY_OFF[] = "fm_off";
+    const char FM_RADIO_VALUE_ON[] = "2055";
+    const char FM_RADIO_VALUE_OFF[] = "7";
+#endif
 
     key = String8(BT_NREC_KEY);
     if (param.get(key, value) == NO_ERROR) {
@@ -564,8 +589,8 @@ size_t AudioHardware::getInputBufferSize(uint32_t sampleRate, int format, int ch
         LOGW("getInputBufferSize bad channel count: %d", channelCount);
         return 0;
     }
-    if (sampleRate != 8000 && sampleRate != 11025 && sampleRate != 16000 &&
-            sampleRate != 22050 && sampleRate != 44100) {
+
+    if (sampleRate != getInputSampleRate(sampleRate)) {
         LOGW("getInputBufferSize bad sample rate: %d", sampleRate);
         return 0;
     }
@@ -573,12 +598,21 @@ size_t AudioHardware::getInputBufferSize(uint32_t sampleRate, int format, int ch
     return AudioStreamInALSA::getBufferSize(sampleRate, channelCount);
 }
 
-
 status_t AudioHardware::setVoiceVolume(float volume)
 {
-    LOGD("### setVoiceVolume");
-
     AutoMutex lock(mLock);
+
+    setVoiceVolume_l(volume);
+
+    return NO_ERROR;
+}
+
+void AudioHardware::setVoiceVolume_l(float volume)
+{
+    LOGD("### setVoiceVolume_l");
+
+    mVoiceVol = volume;
+
     if ( (AudioSystem::MODE_IN_CALL == mMode) && (mSecRilLibHandle) &&
          (connectRILDIfRequired() == OK) ) {
 
@@ -622,7 +656,6 @@ status_t AudioHardware::setVoiceVolume(float volume)
         setCallVolume(mRilClient, type, int_volume);
     }
 
-    return NO_ERROR;
 }
 
 status_t AudioHardware::setMasterVolume(float volume)
@@ -775,13 +808,13 @@ status_t AudioHardware::setIncallPath_l(uint32_t device)
 
             if (mMixer != NULL) {
                 TRACE_DRIVER_IN(DRV_MIXER_GET)
-                struct mixer_ctl *ctl= mixer_get_control(mMixer, "Voice Call Path", 0);
+                struct mixer_ctl *ctl= mixer_get_ctl_by_name(mMixer, "Voice Call Path");
                 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));
+                    mixer_ctl_set_enum_by_string(ctl, getVoiceRouteFromDevice(device));
                     TRACE_DRIVER_OUT
                 }
             }
@@ -930,14 +963,22 @@ struct pcm *AudioHardware::openPcmOut_l()
         }
         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;
+        struct pcm_config config = {
+            channels : 2,
+            rate : AUDIO_HW_OUT_SAMPLERATE,
+            period_size : AUDIO_HW_OUT_PERIOD_SZ,
+            period_count : AUDIO_HW_OUT_PERIOD_CNT,
+            format : PCM_FORMAT_S16_LE,
+            start_threshold : 0,
+            stop_threshold : 0,
+            silence_threshold : 0,
+        };
 
         TRACE_DRIVER_IN(DRV_PCM_OPEN)
-        mPcm = pcm_open(flags);
+        mPcm = pcm_open(0, 0, flags, &config);
         TRACE_DRIVER_OUT
-        if (!pcm_ready(mPcm)) {
-            LOGE("openPcmOut_l() cannot open pcm_out driver: %s\n", pcm_error(mPcm));
+        if (!pcm_is_ready(mPcm)) {
+            LOGE("openPcmOut_l() cannot open pcm_out driver: %s\n", pcm_get_error(mPcm));
             TRACE_DRIVER_IN(DRV_PCM_CLOSE)
             pcm_close(mPcm);
             TRACE_DRIVER_OUT
@@ -974,7 +1015,7 @@ struct mixer *AudioHardware::openMixer_l()
             return NULL;
         }
         TRACE_DRIVER_IN(DRV_MIXER_OPEN)
-        mMixer = mixer_open();
+        mMixer = mixer_open(0);
         TRACE_DRIVER_OUT
         if (mMixer == NULL) {
             LOGE("openMixer_l() cannot open mixer");
@@ -1087,16 +1128,17 @@ const char *AudioHardware::getInputRouteFromDevice(uint32_t device)
 
 uint32_t AudioHardware::getInputSampleRate(uint32_t sampleRate)
 {
-    uint32_t i;
+    size_t i;
     uint32_t prevDelta;
     uint32_t delta;
+    size_t size = sizeof(inputConfigTable)/sizeof(uint32_t)/INPUT_CONFIG_CNT;
 
-    for (i = 0, prevDelta = 0xFFFFFFFF; i < sizeof(inputSamplingRates)/sizeof(uint32_t); i++, prevDelta = delta) {
-        delta = abs(sampleRate - inputSamplingRates[i]);
+    for (i = 0, prevDelta = 0xFFFFFFFF; i < size; i++, prevDelta = delta) {
+        delta = abs(sampleRate - inputConfigTable[i][INPUT_CONFIG_SAMPLE_RATE]);
         if (delta > prevDelta) break;
     }
     // i is always > 0 here
-    return inputSamplingRates[i-1];
+    return inputConfigTable[i-1][INPUT_CONFIG_SAMPLE_RATE];
 }
 
 // getActiveInput_l() must be called with mLock held
@@ -1123,7 +1165,7 @@ status_t AudioHardware::setInputSource_l(audio_source source)
          if ((source == AUDIO_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);
+                 struct mixer_ctl *ctl= mixer_get_ctl_by_name(mMixer, "Input Source");
                  TRACE_DRIVER_OUT
                  if (ctl == NULL) {
                      return NO_INIT;
@@ -1131,11 +1173,9 @@ status_t AudioHardware::setInputSource_l(audio_source source)
                  const char* sourceName;
                  switch (source) {
                      case AUDIO_SOURCE_DEFAULT: // intended fall-through
-                     case AUDIO_SOURCE_MIC:
-                         sourceName = inputPathNameDefault;
-                         break;
+                     case AUDIO_SOURCE_MIC:     // intended fall-through
                      case AUDIO_SOURCE_VOICE_COMMUNICATION:
-                         sourceName = inputPathNameVoiceCommunication;
+                         sourceName = inputPathNameDefault;
                          break;
                      case AUDIO_SOURCE_CAMCORDER:
                          sourceName = inputPathNameCamcorder;
@@ -1149,9 +1189,9 @@ status_t AudioHardware::setInputSource_l(audio_source source)
                      default:
                          return NO_INIT;
                  }
-                 LOGV("mixer_ctl_select, Input Source, (%s)", sourceName);
+                 LOGV("mixer_ctl_set_enum_by_string, Input Source, (%s)", sourceName);
                  TRACE_DRIVER_IN(DRV_MIXER_SEL)
-                 mixer_ctl_select(ctl, sourceName);
+                 mixer_ctl_set_enum_by_string(ctl, sourceName);
                  TRACE_DRIVER_OUT
              }
          }
@@ -1161,6 +1201,42 @@ status_t AudioHardware::setInputSource_l(audio_source source)
      return NO_ERROR;
 }
 
+struct echo_reference_itfe *AudioHardware::getEchoReference(audio_format_t format,
+                                                              uint32_t channelCount,
+                                                              uint32_t samplingRate)
+{
+    LOGV("AudioHardware::getEchoReference %p", mEchoReference);
+    releaseEchoReference(mEchoReference);
+    if (mOutput != NULL) {
+        uint32_t wrChannelCount = popcount(mOutput->channels());
+        uint32_t wrSampleRate = mOutput->sampleRate();
+
+        int status = create_echo_reference(AUDIO_FORMAT_PCM_16_BIT,
+                              channelCount,
+                              samplingRate,
+                              AUDIO_FORMAT_PCM_16_BIT,
+                              wrChannelCount,
+                              wrSampleRate,
+                              &mEchoReference);
+        if (status == 0) {
+            mOutput->addEchoReference(mEchoReference);
+        }
+    }
+    return mEchoReference;
+}
+
+void AudioHardware::releaseEchoReference(struct echo_reference_itfe *reference)
+{
+    LOGV("AudioHardware::releaseEchoReference %p", mEchoReference);
+    if (mEchoReference != NULL && reference == mEchoReference) {
+        if (mOutput != NULL) {
+            mOutput->removeEchoReference(reference);
+        }
+        release_echo_reference(mEchoReference);
+        mEchoReference = NULL;
+    }
+}
+
 
 //------------------------------------------------------------------------------
 //  AudioStreamOutALSA
@@ -1170,7 +1246,7 @@ 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), mSleepReq(false)
+    mDriverOp(DRV_NONE), mStandbyCnt(0), mSleepReq(false), mEchoReference(NULL)
 {
 }
 
@@ -1216,9 +1292,40 @@ AudioHardware::AudioStreamOutALSA::~AudioStreamOutALSA()
     standby();
 }
 
+int AudioHardware::AudioStreamOutALSA::getPlaybackDelay(size_t frames,
+                                                        struct echo_reference_buffer *buffer)
+{
+    size_t kernelFr;
+
+    int rc = pcm_get_htimestamp(mPcm, &kernelFr, &buffer->time_stamp);
+    if (rc < 0) {
+        buffer->time_stamp.tv_sec  = 0;
+        buffer->time_stamp.tv_nsec = 0;
+        buffer->delay_ns           = 0;
+        LOGV("getPlaybackDelay(): pcm_get_htimestamp error, setting playbackTimestamp to 0");
+        return rc;
+    }
+
+    kernelFr = pcm_get_buffer_size(mPcm) - kernelFr;
+
+    // adjust render time stamp with delay added by current driver buffer.
+    // Add the duration of current frame as we want the render time of the last
+    // sample being written.
+    long delayNs = (long)(((int64_t)(kernelFr + frames)* 1000000000) /AUDIO_HW_OUT_SAMPLERATE);
+
+    LOGV("AudioStreamOutALSA::getPlaybackDelay delayNs: [%ld], "\
+         "kernelFr:[%d], frames:[%d], buffSize:[%d], time_stamp:[%ld].[%ld]",
+         delayNs, (int)kernelFr, (int)frames, pcm_get_buffer_size(mPcm),
+         (long)buffer->time_stamp.tv_sec, buffer->time_stamp.tv_nsec);
+
+    buffer->delay_ns = delayNs;
+
+    return 0;
+}
+
 ssize_t AudioHardware::AudioStreamOutALSA::write(const void* buffer, size_t bytes)
 {
-    //    LOGV("AudioStreamOutALSA::write(%p, %u)", buffer, bytes);
+    LOGV("-----AudioStreamInALSA::write(%p, %d) START", buffer, (int)bytes);
     status_t status = NO_INIT;
     const uint8_t* p = static_cast<const uint8_t*>(buffer);
     int ret;
@@ -1239,8 +1346,6 @@ ssize_t AudioHardware::AudioStreamOutALSA::write(const void* buffer, size_t byte
             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->prepareLock();
@@ -1272,29 +1377,37 @@ ssize_t AudioHardware::AudioStreamOutALSA::write(const void* buffer, size_t byte
                 spIn->unlock();
             }
             if (mPcm == NULL) {
-                release_wake_lock("AudioOutLock");
                 goto Error;
             }
             mStandby = false;
         }
 
+        if (mEchoReference != NULL) {
+            struct echo_reference_buffer b;
+            b.raw = (void *)buffer;
+            b.frame_count = bytes / frameSize();
+
+            getPlaybackDelay(bytes / frameSize(), &b);
+            mEchoReference->write(mEchoReference, &b);
+        }
+
         TRACE_DRIVER_IN(DRV_PCM_WRITE)
         ret = pcm_write(mPcm,(void*) p, bytes);
         TRACE_DRIVER_OUT
 
         if (ret == 0) {
+            LOGV("-----AudioStreamInALSA::write(%p, %d) END", buffer, (int)bytes);
             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());
-
+    LOGE("AudioStreamOutALSA::write END WITH ERROR !!!!!!!!!(%p, %u)", buffer, bytes);
     return status;
 }
 
@@ -1302,12 +1415,16 @@ status_t AudioHardware::AudioStreamOutALSA::standby()
 {
     if (mHardware == NULL) return NO_INIT;
 
-    AutoMutex lock(mLock);
+    mSleepReq = true;
+    {
+        AutoMutex lock(mLock);
+        mSleepReq = false;
 
-    { // scope for the AudioHardware lock
-        AutoMutex hwLock(mHardware->lock());
+        { // scope for the AudioHardware lock
+            AutoMutex hwLock(mHardware->lock());
 
-        doStandby_l();
+            doStandby_l();
+        }
     }
 
     return NO_ERROR;
@@ -1319,7 +1436,10 @@ void AudioHardware::AudioStreamOutALSA::doStandby_l()
 
     if (!mStandby) {
         LOGD("AudioHardware pcm playback is going to standby.");
-        release_wake_lock("AudioOutLock");
+        // stop echo reference capture
+        if (mEchoReference != NULL) {
+            mEchoReference->write(mEchoReference, NULL);
+        }
         mStandby = true;
     }
 
@@ -1351,7 +1471,7 @@ status_t AudioHardware::AudioStreamOutALSA::open_l()
     if (mMixer) {
         LOGV("open playback normal");
         TRACE_DRIVER_IN(DRV_MIXER_GET)
-        mRouteCtl = mixer_get_control(mMixer, "Playback Path", 0);
+        mRouteCtl = mixer_get_ctl_by_name(mMixer, "Playback Path");
         TRACE_DRIVER_OUT
     }
     if (mHardware->mode() != AudioSystem::MODE_IN_CALL) {
@@ -1359,7 +1479,7 @@ status_t AudioHardware::AudioStreamOutALSA::open_l()
         LOGV("write() wakeup setting route %s", route);
         if (mRouteCtl) {
             TRACE_DRIVER_IN(DRV_MIXER_SEL)
-            mixer_ctl_select(mRouteCtl, route);
+            mixer_ctl_set_enum_by_string(mRouteCtl, route);
             TRACE_DRIVER_OUT
         }
     }
@@ -1413,16 +1533,16 @@ bool AudioHardware::AudioStreamOutALSA::checkStandby()
 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;
 
+    mSleepReq = true;
     {
         AutoMutex lock(mLock);
-
+        mSleepReq = false;
         if (param.getInt(String8(AudioParameter::keyRouting), device) == NO_ERROR)
         {
             if (device != 0) {
@@ -1430,7 +1550,6 @@ status_t AudioHardware::AudioStreamOutALSA::setParameters(const String8& keyValu
 
                 if (mDevices != (uint32_t)device) {
                     mDevices = (uint32_t)device;
-
                     if (mHardware->mode() != AudioSystem::MODE_IN_CALL) {
                         doStandby_l();
                     }
@@ -1447,7 +1566,9 @@ status_t AudioHardware::AudioStreamOutALSA::setParameters(const String8& keyValu
         status = BAD_VALUE;
     }
 
+
     return status;
+
 }
 
 String8 AudioHardware::AudioStreamOutALSA::getParameters(const String8& keys)
@@ -1488,6 +1609,24 @@ void AudioHardware::AudioStreamOutALSA::unlock() {
     mLock.unlock();
 }
 
+void AudioHardware::AudioStreamOutALSA::addEchoReference(struct echo_reference_itfe *reference)
+{
+    LOGV("AudioStreamOutALSA::addEchoReference %p", mEchoReference);
+    if (mEchoReference == NULL) {
+        mEchoReference = reference;
+    }
+}
+
+void AudioHardware::AudioStreamOutALSA::removeEchoReference(struct echo_reference_itfe *reference)
+{
+    LOGV("AudioStreamOutALSA::removeEchoReference %p", mEchoReference);
+    if (mEchoReference == reference) {
+        mEchoReference->write(mEchoReference, NULL);
+        mEchoReference = NULL;
+    }
+}
+
+
 //------------------------------------------------------------------------------
 //  AudioStreamInALSA
 //------------------------------------------------------------------------------
@@ -1496,8 +1635,10 @@ 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), mSleepReq(false)
+    mDownSampler(NULL), mReadStatus(NO_ERROR), mInputBuf(NULL),
+    mDriverOp(DRV_NONE), mStandbyCnt(0), mSleepReq(false),
+    mProcBuf(NULL), mProcBufSize(0), mRefBuf(NULL), mRefBufSize(0),
+    mEchoReference(NULL), mNeedEchoReference(false)
 {
 }
 
@@ -1534,38 +1675,283 @@ status_t AudioHardware::AudioStreamInALSA::set(
     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;
+        mBufferProvider.mProvider.get_next_buffer = getNextBufferStatic;
+        mBufferProvider.mProvider.release_buffer = releaseBufferStatic;
+        mBufferProvider.mInputStream = this;
+        int status = create_resampler(AUDIO_HW_OUT_SAMPLERATE,
+                                                    mSampleRate,
+                                                    mChannelCount,
+                                                    RESAMPLER_QUALITY_VOIP,
+                                                    &mBufferProvider.mProvider,
+                                                    &mDownSampler);
+        if (status != 0) {
             LOGW("AudioStreamInALSA::set() downsampler init failed: %d", status);
+            mDownSampler = NULL;
             return status;
         }
-
-        mPcmIn = new int16_t[AUDIO_HW_IN_PERIOD_SZ * mChannelCount];
     }
+    mInputBuf = 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;
+        release_resampler(mDownSampler);
+    }
+    delete[] mInputBuf;
+    delete[] mProcBuf;
+}
+
+// readFrames() reads frames from kernel driver, down samples to capture rate if necessary
+// and output the number of frames requested to the buffer specified
+ssize_t AudioHardware::AudioStreamInALSA::readFrames(void* buffer, ssize_t frames)
+{
+    ssize_t framesWr = 0;
+    while (framesWr < frames) {
+        size_t framesRd = frames - framesWr;
+        if (mDownSampler != NULL) {
+            mDownSampler->resample_from_provider(mDownSampler,
+                    (int16_t *)((char *)buffer + framesWr * frameSize()),
+                    &framesRd);
+        } else {
+            struct resampler_buffer buf = {
+                    { raw : NULL, },
+                    frame_count : framesRd,
+            };
+            getNextBuffer(&buf);
+            if (buf.raw != NULL) {
+                memcpy((char *)buffer + framesWr * frameSize(),
+                        buf.raw,
+                        buf.frame_count * frameSize());
+                framesRd = buf.frame_count;
+            }
+            releaseBuffer(&buf);
         }
+        // mReadStatus is updated by getNextBuffer() also called by
+        // mDownSampler->resample_from_provider()
+        if (mReadStatus != 0) {
+            return mReadStatus;
+        }
+        framesWr += framesRd;
+    }
+    return framesWr;
+}
+
+// processFrames() reads frames from kernel driver (via readFrames()), calls the active
+// audio pre processings and output the number of frames requested to the buffer specified
+ssize_t AudioHardware::AudioStreamInALSA::processFrames(void* buffer, ssize_t frames)
+{
+    ssize_t framesWr = 0;
+    while (framesWr < frames) {
+        // first reload enough frames at the end of process input buffer
+        if (mProcFramesIn < (size_t)frames) {
+            // expand process input buffer if necessary
+            if (mProcBufSize < (size_t)frames) {
+                mProcBufSize = (size_t)frames;
+                mProcBuf = (int16_t *)realloc(mProcBuf,
+                                              mProcBufSize * mChannelCount * sizeof(int16_t));
+                LOGV("processFrames(): mProcBuf %p size extended to %d frames",
+                     mProcBuf, mProcBufSize);
+            }
+            ssize_t framesRd = readFrames(mProcBuf + mProcFramesIn * mChannelCount,
+                                          frames - mProcFramesIn);
+            if (framesRd < 0) {
+                framesWr = framesRd;
+                break;
+            }
+            mProcFramesIn += framesRd;
+        }
+
+        if (mEchoReference != NULL) {
+            pushEchoReference(mProcFramesIn);
+        }
+
+        //inBuf.frameCount and outBuf.frameCount indicate respectively the maximum number of frames
+        //to be consumed and produced by process()
+        audio_buffer_t inBuf = {
+                mProcFramesIn,
+                {mProcBuf}
+        };
+        audio_buffer_t outBuf = {
+                frames - framesWr,
+                {(int16_t *)buffer + framesWr * mChannelCount}
+        };
+
+        for (size_t i = 0; i < mPreprocessors.size(); i++) {
+            (*mPreprocessors[i])->process(mPreprocessors[i],
+                                                   &inBuf,
+                                                   &outBuf);
+        }
+
+        // process() has updated the number of frames consumed and produced in
+        // inBuf.frameCount and outBuf.frameCount respectively
+        // move remaining frames to the beginning of mProcBuf
+        mProcFramesIn -= inBuf.frameCount;
+        if (mProcFramesIn) {
+            memcpy(mProcBuf,
+                   mProcBuf + inBuf.frameCount * mChannelCount,
+                   mProcFramesIn * mChannelCount * sizeof(int16_t));
+        }
+
+        // if not enough frames were passed to process(), read more and retry.
+        if (outBuf.frameCount == 0) {
+            continue;
+        }
+        framesWr += outBuf.frameCount;
+    }
+    return framesWr;
+}
+
+int32_t AudioHardware::AudioStreamInALSA::updateEchoReference(size_t frames)
+{
+    struct echo_reference_buffer b;
+    b.delay_ns = 0;
+
+    LOGV("updateEchoReference1 START, frames = [%d], mRefFramesIn = [%d],  b.frame_count = [%d]",
+         frames, mRefFramesIn, frames - mRefFramesIn);
+    if (mRefFramesIn < frames) {
+        if (mRefBufSize < frames) {
+            mRefBufSize = frames;
+            mRefBuf = (int16_t *)realloc(mRefBuf,
+                                         mRefBufSize * mChannelCount * sizeof(int16_t));
+        }
+
+        b.frame_count = frames - mRefFramesIn;
+        b.raw = (void *)(mRefBuf + mRefFramesIn * mChannelCount);
+
+        getCaptureDelay(frames, &b);
+
+        if (mEchoReference->read(mEchoReference, &b) == NO_ERROR)
+        {
+            mRefFramesIn += b.frame_count;
+            LOGV("updateEchoReference2: mRefFramesIn:[%d], mRefBufSize:[%d], "\
+                 "frames:[%d], b.frame_count:[%d]", mRefFramesIn, mRefBufSize,frames,b.frame_count);
+        }
+
+    }else{
+        LOGV("updateEchoReference3: NOT enough frames to read ref buffer");
+    }
+    return b.delay_ns;
+}
+
+void AudioHardware::AudioStreamInALSA::pushEchoReference(size_t frames)
+{
+    // read frames from echo reference buffer and update echo delay
+    // mRefFramesIn is updated with frames available in mRefBuf
+    int32_t delayUs = (int32_t)(updateEchoReference(frames)/1000);
+
+    if (mRefFramesIn < frames) {
+        frames = mRefFramesIn;
+    }
+
+    audio_buffer_t refBuf = {
+            frames,
+            {mRefBuf}
+    };
+
+    for (size_t i = 0; i < mPreprocessors.size(); i++) {
+        if ((*mPreprocessors[i])->process_reverse == NULL) {
+            continue;
+        }
+        (*mPreprocessors[i])->process_reverse(mPreprocessors[i],
+                                               &refBuf,
+                                               NULL);
+        setPreProcessorEchoDelay(mPreprocessors[i], delayUs);
+    }
+
+    mRefFramesIn -= refBuf.frameCount;
+    if (mRefFramesIn) {
+    LOGV("pushEchoReference5: shifting mRefBuf down by = %d frames", mRefFramesIn);
+        memcpy(mRefBuf,
+               mRefBuf + refBuf.frameCount * mChannelCount,
+               mRefFramesIn * mChannelCount * sizeof(int16_t));
+    }
+}
+
+status_t AudioHardware::AudioStreamInALSA::setPreProcessorEchoDelay(effect_handle_t handle,
+                                                                    int32_t delayUs)
+{
+    uint32_t buf[sizeof(effect_param_t) / sizeof(uint32_t) + 2];
+    effect_param_t *param = (effect_param_t *)buf;
+
+    param->psize = sizeof(uint32_t);
+    param->vsize = sizeof(uint32_t);
+    *(uint32_t *)param->data = AEC_PARAM_ECHO_DELAY;
+    *((int32_t *)param->data + 1) = delayUs;
+
+    LOGV("setPreProcessorEchoDelay: %d us", delayUs);
+
+    return setPreprocessorParam(handle, param);
+}
+
+status_t AudioHardware::AudioStreamInALSA::setPreprocessorParam(effect_handle_t handle,
+                                                                effect_param_t *param)
+{
+    uint32_t size = sizeof(int);
+    uint32_t psize = ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + param->vsize;
+
+    status_t status = (*handle)->command(handle,
+                                           EFFECT_CMD_SET_PARAM,
+                                           sizeof (effect_param_t) + psize,
+                                           param,
+                                           &size,
+                                           &param->status);
+    if (status == NO_ERROR) {
+        status = param->status;
     }
+    return status;
+}
+
+void AudioHardware::AudioStreamInALSA::getCaptureDelay(size_t frames,
+                                                       struct echo_reference_buffer *buffer)
+{
+
+    // read frames available in kernel driver buffer
+    size_t kernelFr;
+    struct timespec tstamp;
+
+    if (pcm_get_htimestamp(mPcm, &kernelFr, &tstamp) < 0) {
+        buffer->time_stamp.tv_sec  = 0;
+        buffer->time_stamp.tv_nsec = 0;
+        buffer->delay_ns           = 0;
+        LOGW("read getCaptureDelay(): pcm_htimestamp error");
+        return;
+    }
+
+    // read frames available in audio HAL input buffer
+    // add number of frames being read as we want the capture time of first sample in current
+    // buffer
+    long bufDelay = (long)(((int64_t)(mInputFramesIn + mProcFramesIn) * 1000000000)
+                                    / AUDIO_HW_IN_SAMPLERATE);
+    // add delay introduced by resampler
+    long rsmpDelay = 0;
+    if (mDownSampler) {
+        rsmpDelay = mDownSampler->delay_ns(mDownSampler);
+    }
+
+    long kernelDelay = (long)(((int64_t)kernelFr * 1000000000) / AUDIO_HW_IN_SAMPLERATE);
+
+    // correct capture time stamp
+    long delayNs = kernelDelay + bufDelay + rsmpDelay;
+
+    buffer->time_stamp = tstamp;
+    buffer->delay_ns   = delayNs;
+    LOGV("AudioStreamInALSA::getCaptureDelay TimeStamp = [%ld].[%ld], delayCaptureNs: [%d],"\
+         " kernelDelay:[%ld], bufDelay:[%ld], rsmpDelay:[%ld], kernelFr:[%d], "\
+         "mInputFramesIn:[%d], mProcFramesIn:[%d], frames:[%d]",
+         buffer->time_stamp.tv_sec , buffer->time_stamp.tv_nsec, buffer->delay_ns,
+         kernelDelay, bufDelay, rsmpDelay, kernelFr, mInputFramesIn, mProcFramesIn, frames);
+
 }
 
 ssize_t AudioHardware::AudioStreamInALSA::read(void* buffer, ssize_t bytes)
 {
-    //    LOGV("AudioStreamInALSA::read(%p, %u)", buffer, bytes);
+    LOGV("-----AudioStreamInALSA::read(%p, %d) START", buffer, (int)bytes);
     status_t status = NO_INIT;
-    int ret;
 
     if (mHardware == NULL) return NO_INIT;
 
@@ -1582,38 +1968,38 @@ ssize_t AudioHardware::AudioStreamInALSA::read(void* buffer, ssize_t bytes)
             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->prepareLock();
-                    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->prepareLock();
+                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()) {
+                    break;
                 }
+                spOut->unlock();
+                spOut = mHardware->output();
             }
-            // 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();
+                if (!spOut->checkStandby()) {
+                    LOGV("AudioStreamInALSA::read() force output standby");
+                    spOut->close_l();
+                    if (spOut->open_l() != NO_ERROR) {
+                        spOut->doStandby_l();
+                    }
+                }
+                LOGV("AudioStreamInALSA exit standby mNeedEchoReference %d mEchoReference %p",
+                     mNeedEchoReference, mEchoReference);
+                if (mNeedEchoReference && mEchoReference == NULL) {
+                    mEchoReference = mHardware->getEchoReference(AUDIO_FORMAT_PCM_16_BIT,
+                                                                 mChannelCount,
+                                                                 mSampleRate);
                 }
                 spOut->unlock();
             }
@@ -1621,38 +2007,27 @@ ssize_t AudioHardware::AudioStreamInALSA::read(void* buffer, ssize_t bytes)
             open_l();
 
             if (mPcm == NULL) {
-                release_wake_lock("AudioInLock");
                 goto Error;
             }
             mStandby = false;
         }
 
+        size_t framesRq = bytes / mChannelCount/sizeof(int16_t);
+        ssize_t framesRd;
 
-        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();
+        if (mPreprocessors.size() == 0) {
+            framesRd = readFrames(buffer, framesRq);
         } else {
-            TRACE_DRIVER_IN(DRV_PCM_READ)
-            ret = pcm_read(mPcm, buffer, bytes);
-            TRACE_DRIVER_OUT
+            framesRd = processFrames(buffer, framesRq);
         }
 
-        if (ret == 0) {
-            return bytes;
+        if (framesRd >= 0) {
+            LOGV("-----AudioStreamInALSA::read(%p, %d) END", buffer, (int)bytes);
+            return framesRd * mChannelCount * sizeof(int16_t);
         }
 
-        LOGW("read error: %d", ret);
-        status = ret;
+        LOGW("read error: %d", (int)framesRd);
+        status = framesRd;
     }
 
 Error:
@@ -1661,7 +2036,7 @@ Error:
 
     // Simulate audio output timing in case of error
     usleep((((bytes * 1000) / frameSize()) * 1000) / sampleRate());
-
+    LOGE("-----AudioStreamInALSA::read(%p, %d) END ERROR", buffer, (int)bytes);
     return status;
 }
 
@@ -1669,12 +2044,16 @@ status_t AudioHardware::AudioStreamInALSA::standby()
 {
     if (mHardware == NULL) return NO_INIT;
 
-    AutoMutex lock(mLock);
+    mSleepReq = true;
+    {
+        AutoMutex lock(mLock);
+        mSleepReq = false;
 
-    { // scope for AudioHardware lock
-        AutoMutex hwLock(mHardware->lock());
+        { // scope for AudioHardware lock
+            AutoMutex hwLock(mHardware->lock());
 
-        doStandby_l();
+            doStandby_l();
+        }
     }
     return NO_ERROR;
 }
@@ -1685,7 +2064,24 @@ void AudioHardware::AudioStreamInALSA::doStandby_l()
 
     if (!mStandby) {
         LOGD("AudioHardware pcm capture is going to standby.");
-        release_wake_lock("AudioInLock");
+        if (mEchoReference != NULL) {
+            // stop reading from echo reference
+            mEchoReference->read(mEchoReference, NULL);
+            // Mutex acquisition order is always out -> in -> hw
+            sp<AudioStreamOutALSA> spOut = mHardware->output();
+            if (spOut != 0) {
+                spOut->prepareLock();
+                mHardware->lock().unlock();
+                mLock.unlock();
+                spOut->lock();
+                mLock.lock();
+                mHardware->lock().lock();
+                mHardware->releaseEchoReference(mEchoReference);
+                spOut->unlock();
+            }
+            mEchoReference = NULL;
+        }
+
         mStandby = true;
     }
     close_l();
@@ -1705,24 +2101,36 @@ void AudioHardware::AudioStreamInALSA::close_l()
         TRACE_DRIVER_OUT
         mPcm = NULL;
     }
+
+    delete[] mProcBuf;
+    mProcBuf = NULL;
+    mProcBufSize = 0;
+    delete[] mRefBuf;
+    mRefBuf = NULL;
+    mRefBufSize = 0;
 }
 
 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;
+
+    struct pcm_config config = {
+        channels : mChannelCount,
+        rate : AUDIO_HW_IN_SAMPLERATE,
+        period_size : AUDIO_HW_IN_PERIOD_SZ,
+        period_count : AUDIO_HW_IN_PERIOD_CNT,
+        format : PCM_FORMAT_S16_LE,
+        start_threshold : 0,
+        stop_threshold : 0,
+        silence_threshold : 0,
+    };
 
     LOGV("open pcm_in driver");
     TRACE_DRIVER_IN(DRV_PCM_OPEN)
-    mPcm = pcm_open(flags);
+    mPcm = pcm_open(0, 0, flags, &config);
     TRACE_DRIVER_OUT
-    if (!pcm_ready(mPcm)) {
-        LOGE("cannot open pcm_in driver: %s\n", pcm_error(mPcm));
+    if (!pcm_is_ready(mPcm)) {
+        LOGE("cannot open pcm_in driver: %s\n", pcm_get_error(mPcm));
         TRACE_DRIVER_IN(DRV_PCM_CLOSE)
         pcm_close(mPcm);
         TRACE_DRIVER_OUT
@@ -1731,14 +2139,19 @@ status_t AudioHardware::AudioStreamInALSA::open_l()
     }
 
     if (mDownSampler != NULL) {
-        mInPcmInBuf = 0;
-        mDownSampler->reset();
+        mDownSampler->reset(mDownSampler);
     }
+    mInputFramesIn = 0;
+
+    mProcBufSize = 0;
+    mProcFramesIn = 0;
+    mRefBufSize = 0;
+    mRefFramesIn = 0;
 
     mMixer = mHardware->openMixer_l();
     if (mMixer) {
         TRACE_DRIVER_IN(DRV_MIXER_GET)
-        mRouteCtl = mixer_get_control(mMixer, "Capture MIC Path", 0);
+        mRouteCtl = mixer_get_ctl_by_name(mMixer, "Capture MIC Path");
         TRACE_DRIVER_OUT
     }
 
@@ -1747,7 +2160,7 @@ status_t AudioHardware::AudioStreamInALSA::open_l()
         LOGV("read() wakeup setting route %s", route);
         if (mRouteCtl) {
             TRACE_DRIVER_IN(DRV_MIXER_SEL)
-            mixer_ctl_select(mRouteCtl, route);
+            mixer_ctl_set_enum_by_string(mRouteCtl, route);
             TRACE_DRIVER_OUT
         }
     }
@@ -1806,8 +2219,10 @@ status_t AudioHardware::AudioStreamInALSA::setParameters(const String8& keyValue
 
     if (mHardware == NULL) return NO_INIT;
 
+    mSleepReq = true;
     {
         AutoMutex lock(mLock);
+        mSleepReq = false;
 
         if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR) {
             AutoMutex hwLock(mHardware->lock());
@@ -1858,58 +2273,123 @@ String8 AudioHardware::AudioStreamInALSA::getParameters(const String8& keys)
     return param.toString();
 }
 
-status_t AudioHardware::AudioStreamInALSA::getNextBuffer(AudioHardware::BufferProvider::Buffer* buffer)
+status_t AudioHardware::AudioStreamInALSA::addAudioEffect(effect_handle_t effect)
+{
+    LOGV("AudioStreamInALSA::addAudioEffect() %p", effect);
+
+    effect_descriptor_t desc;
+    status_t status = (*effect)->get_descriptor(effect, &desc);
+    if (status == 0) {
+        if (memcmp(&desc.type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0) {
+            LOGV("AudioStreamInALSA::addAudioEffect() mNeedEchoReference true");
+            mNeedEchoReference = true;
+            standby();
+        }
+        LOGV("AudioStreamInALSA::addAudioEffect() name %s", desc.name);
+    } else {
+        LOGV("AudioStreamInALSA::addAudioEffect() get_descriptor() error");
+    }
+
+    AutoMutex lock(mLock);
+    mPreprocessors.add(effect);
+    return NO_ERROR;
+}
+
+status_t AudioHardware::AudioStreamInALSA::removeAudioEffect(effect_handle_t effect)
+{
+    status_t status = INVALID_OPERATION;
+    LOGV("AudioStreamInALSA::removeAudioEffect() %p", effect);
+    {
+        AutoMutex lock(mLock);
+        for (size_t i = 0; i < mPreprocessors.size(); i++) {
+            if (mPreprocessors[i] == effect) {
+                mPreprocessors.removeAt(i);
+                status = NO_ERROR;
+                break;
+            }
+        }
+    }
+
+    if (status == NO_ERROR) {
+        effect_descriptor_t desc;
+        if ((*effect)->get_descriptor(effect, &desc) == 0) {
+            if (memcmp(&desc.type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0) {
+                LOGV("AudioStreamInALSA::removeAudioEffect() mNeedEchoReference false");
+                mNeedEchoReference = false;
+                standby();
+            }
+        }
+    }
+
+    return status;
+}
+
+extern "C" {
+int AudioHardware::AudioStreamInALSA::getNextBufferStatic(
+                                                    struct resampler_buffer_provider *provider,
+                                                    struct resampler_buffer* buffer)
+{
+    ResamplerBufferProvider *bufferProvider = (ResamplerBufferProvider *)provider;
+    return bufferProvider->mInputStream->getNextBuffer(buffer);
+}
+
+void AudioHardware::AudioStreamInALSA::releaseBufferStatic(
+                                                    struct resampler_buffer_provider *provider,
+                                                    struct resampler_buffer* buffer)
+{
+    ResamplerBufferProvider *bufferProvider = (ResamplerBufferProvider *)provider;
+    return bufferProvider->mInputStream->releaseBuffer(buffer);
+}
+
+}; // extern "C"
+
+status_t AudioHardware::AudioStreamInALSA::getNextBuffer(struct resampler_buffer *buffer)
 {
     if (mPcm == NULL) {
         buffer->raw = NULL;
-        buffer->frameCount = 0;
+        buffer->frame_count = 0;
         mReadStatus = NO_INIT;
         return NO_INIT;
     }
 
-    if (mInPcmInBuf == 0) {
+    if (mInputFramesIn == 0) {
         TRACE_DRIVER_IN(DRV_PCM_READ)
-        mReadStatus = pcm_read(mPcm,(void*) mPcmIn, AUDIO_HW_IN_PERIOD_SZ * frameSize());
+        mReadStatus = pcm_read(mPcm,(void*) mInputBuf, AUDIO_HW_IN_PERIOD_SZ * frameSize());
         TRACE_DRIVER_OUT
         if (mReadStatus != 0) {
             buffer->raw = NULL;
-            buffer->frameCount = 0;
+            buffer->frame_count = 0;
             return mReadStatus;
         }
-        mInPcmInBuf = AUDIO_HW_IN_PERIOD_SZ;
+        mInputFramesIn = AUDIO_HW_IN_PERIOD_SZ;
     }
 
-    buffer->frameCount = (buffer->frameCount > mInPcmInBuf) ? mInPcmInBuf : buffer->frameCount;
-    buffer->i16 = mPcmIn + (AUDIO_HW_IN_PERIOD_SZ - mInPcmInBuf) * mChannelCount;
+    buffer->frame_count = (buffer->frame_count > mInputFramesIn) ? mInputFramesIn:buffer->frame_count;
+    buffer->i16 = mInputBuf + (AUDIO_HW_IN_PERIOD_SZ - mInputFramesIn) * mChannelCount;
 
     return mReadStatus;
 }
 
-void AudioHardware::AudioStreamInALSA::releaseBuffer(Buffer* buffer)
+void AudioHardware::AudioStreamInALSA::releaseBuffer(struct resampler_buffer *buffer)
 {
-    mInPcmInBuf -= buffer->frameCount;
+    mInputFramesIn -= buffer->frame_count;
 }
 
 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;
-    }
+    size_t i;
+    size_t size = sizeof(inputConfigTable)/sizeof(uint32_t)/INPUT_CONFIG_CNT;
 
-    return (AUDIO_HW_IN_PERIOD_SZ*channelCount*sizeof(int16_t)) / ratio ;
+    for (i = 0; i < size; i++) {
+        if (sampleRate == inputConfigTable[i][INPUT_CONFIG_SAMPLE_RATE]) {
+            return (AUDIO_HW_IN_PERIOD_SZ*channelCount*sizeof(int16_t)) /
+                    inputConfigTable[i][INPUT_CONFIG_BUFFER_RATIO];
+        }
+    }
+    // this should never happen as getBufferSize() is always called after getInputSampleRate()
+    // that checks for valid sampling rates.
+    LOGE("AudioStreamInALSA::getBufferSize() invalid sampling rate %d", sampleRate);
+    return 0;
 }
 
 int AudioHardware::AudioStreamInALSA::prepareLock()
@@ -1931,375 +2411,6 @@ void AudioHardware::AudioStreamInALSA::unlock() {
 }
 
 //------------------------------------------------------------------------------
-//  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;
-        const uint32_t step = (uint32_t)(step_float * 32768.0f + 0.5f);  // 17.15 fixed point
-
-        uint32_t in_sample_num = 0;   // 17.15 fixed point
-        for (int j = 0; j < RESAMPLE_16KHZ_SAMPLES_OUT; ++j, in_sample_num += step) {
-            const uint32_t whole = in_sample_num >> 15;
-            const uint32_t frac = (in_sample_num & 0x7fff);  // 0.15 fixed point
-            const int32_t s1 = tmp[whole];
-            const int32_t s2 = tmp[whole + 1];
-            *output++ = clip(s1 + (((s2 - s1) * (int32_t)frac) >> 15));
-        }
-
-    }
-
-    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
 //------------------------------------------------------------------------------
 
@@ -2307,4 +2418,4 @@ extern "C" AudioHardwareInterface* createAudioHardware(void) {
     return new AudioHardware();
 }
 
-}; // namespace android_audio_legacy
+}; // namespace android
diff --git a/libaudio/AudioHardware.h b/libaudio/AudioHardware.h
index 69bc2b4..d191f3f 100644
--- a/libaudio/AudioHardware.h
+++ b/libaudio/AudioHardware.h
@@ -25,9 +25,13 @@
 
 #include <hardware_legacy/AudioHardwareBase.h>
 #include <media/mediarecorder.h>
+#include <hardware/audio_effect.h>
 
 #include "secril-client.h"
 
+#include <audio_utils/resampler.h>
+#include <audio_utils/echo_reference.h>
+
 extern "C" {
     struct pcm;
     struct mixer;
@@ -35,8 +39,13 @@ extern "C" {
 };
 
 namespace android_audio_legacy {
-
-using namespace android;
+    using android::AutoMutex;
+    using android::Mutex;
+    using android::RefBase;
+    using android::SortedVector;
+    using android::sp;
+    using android::String16;
+    using android::Vector;
 
 // TODO: determine actual audio DSP and hardware latency
 // Additionnal latency introduced by audio DSP and hardware in ms
@@ -48,24 +57,20 @@ using namespace android;
 // 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_SZ 1024
 #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
+#define AUDIO_HW_IN_SAMPLERATE 44100
 // 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
+#define AUDIO_HW_IN_PERIOD_SZ 1024
+#define AUDIO_HW_IN_PERIOD_CNT 4
 // Default audio input buffer size in bytes (8kHz mono)
 #define AUDIO_HW_IN_PERIOD_BYTES ((AUDIO_HW_IN_PERIOD_SZ*sizeof(int16_t))/8)
 
@@ -74,13 +79,13 @@ class AudioHardware : public AudioHardwareBase
 {
     class AudioStreamOutALSA;
     class AudioStreamInALSA;
+
 public:
 
     // input path names used to translate from input sources to driver paths
     static const char *inputPathNameDefault;
     static const char *inputPathNameCamcorder;
     static const char *inputPathNameVoiceRecognition;
-    static const char *inputPathNameVoiceCommunication;
 
     AudioHardware();
     virtual ~AudioHardware();
@@ -130,6 +135,8 @@ public:
 
             status_t setInputSource_l(audio_source source);
 
+            void setVoiceVolume_l(float volume);
+
     static uint32_t    getInputSampleRate(uint32_t sampleRate);
            sp <AudioStreamInALSA> getActiveInput_l();
 
@@ -143,6 +150,11 @@ public:
 
            sp <AudioStreamOutALSA>  output() { return mOutput; }
 
+           struct echo_reference_itfe *getEchoReference(audio_format_t format,
+                                          uint32_t channelCount,
+                                          uint32_t samplingRate);
+           void releaseEchoReference(struct echo_reference_itfe *reference);
+
 protected:
     virtual status_t dump(int fd, const Vector<String16>& args);
 
@@ -165,6 +177,7 @@ private:
     uint32_t        mPcmOpenCnt;
     uint32_t        mMixerOpenCnt;
     bool            mInCallAudioMode;
+    float           mVoiceVol;
 
     audio_source    mInputSource;
     bool            mBluetoothNrec;
@@ -183,6 +196,7 @@ private:
     int             (*setCallClockSync)(HRilClient, SoundClockCondition);
     void            loadRILD(void);
     status_t        connectRILDIfRequired(void);
+    struct echo_reference_itfe *mEchoReference;
 
 #ifdef HAVE_FM_RADIO
     int             mFmFd;
@@ -194,7 +208,17 @@ private:
     int             mDriverOp;
 
     static uint32_t         checkInputSampleRate(uint32_t sampleRate);
-    static const uint32_t   inputSamplingRates[];
+
+    // column index in inputConfigTable[][]
+    enum {
+        INPUT_CONFIG_SAMPLE_RATE,
+        INPUT_CONFIG_BUFFER_RATIO,
+        INPUT_CONFIG_CNT
+    };
+
+    // contains the list of valid sampling rates for input streams as well as the ratio
+    // between the kernel buffer size and audio hal buffer size for each sampling rate
+    static const uint32_t  inputConfigTable[][INPUT_CONFIG_CNT];
 
     class AudioStreamOutALSA : public AudioStreamOut, public RefBase
     {
@@ -239,8 +263,14 @@ private:
                 void lock();
                 void unlock();
 
+                void addEchoReference(struct echo_reference_itfe *reference);
+                void removeEchoReference(struct echo_reference_itfe *reference);
+
     private:
 
+                int computeEchoReferenceDelay(size_t frames, struct timespec *echoRefRenderTime);
+                int getPlaybackDelay(size_t frames, struct echo_reference_buffer *buffer);
+
         Mutex mLock;
         AudioHardware* mHardware;
         struct pcm *mPcm;
@@ -256,65 +286,10 @@ private:
         int mDriverOp;
         int mStandbyCnt;
         bool mSleepReq;
+        struct echo_reference_itfe *mEchoReference;
     };
 
-    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
+    class AudioStreamInALSA : public AudioStreamIn, public RefBase
     {
 
      public:
@@ -338,6 +313,9 @@ private:
         virtual status_t setParameters(const String8& keyValuePairs);
         virtual String8 getParameters(const String8& keys);
         virtual unsigned int getInputFramesLost() const { return 0; }
+        virtual status_t    addAudioEffect(effect_handle_t effect);
+        virtual status_t    removeAudioEffect(effect_handle_t effect);
+
                 uint32_t device() { return mDevices; }
                 void doStandby_l();
                 void close_l();
@@ -346,19 +324,36 @@ private:
 
         static size_t getBufferSize(uint32_t sampleRate, int channelCount);
 
-        // BufferProvider
-        virtual status_t getNextBuffer(BufferProvider::Buffer* buffer);
-        virtual void releaseBuffer(BufferProvider::Buffer* buffer);
-
-        // Stubs (ICS)
-        virtual status_t addAudioEffect(effect_handle_t effect) { return INVALID_OPERATION; }
-        virtual status_t removeAudioEffect(effect_handle_t effect) { return INVALID_OPERATION; }
+        // resampler_buffer_provider
+        static int getNextBufferStatic(struct resampler_buffer_provider *provider,
+                             struct resampler_buffer* buffer);
+        static void releaseBufferStatic(struct resampler_buffer_provider *provider,
+                             struct resampler_buffer* buffer);
 
         int prepareLock();
         void lock();
         void unlock();
 
-    private:
+     private:
+
+        struct ResamplerBufferProvider {
+            struct resampler_buffer_provider mProvider;
+            AudioStreamInALSA *mInputStream;
+        };
+
+        ssize_t readFrames(void* buffer, ssize_t frames);
+        ssize_t processFrames(void* buffer, ssize_t frames);
+        int32_t updateEchoReference(size_t frames);
+        void pushEchoReference(size_t frames);
+        void updateEchoDelay(size_t frames, struct timespec *echoRefRenderTime);
+        void getCaptureDelay(size_t frames, struct echo_reference_buffer *buffer);
+        status_t setPreProcessorEchoDelay(effect_handle_t handle, int32_t delayUs);
+        status_t setPreprocessorParam(effect_handle_t handle, effect_param_t *param);
+
+        // BufferProvider
+        status_t getNextBuffer(struct resampler_buffer* buffer);
+        void releaseBuffer(struct resampler_buffer* buffer);
+
         Mutex mLock;
         AudioHardware* mHardware;
         struct pcm *mPcm;
@@ -371,18 +366,28 @@ private:
         uint32_t mChannelCount;
         uint32_t mSampleRate;
         size_t mBufferSize;
-        DownSampler *mDownSampler;
+        struct resampler_itfe *mDownSampler;
+        struct ResamplerBufferProvider mBufferProvider;
         status_t mReadStatus;
-        size_t mInPcmInBuf;
-        int16_t *mPcmIn;
+        size_t mInputFramesIn;
+        int16_t *mInputBuf;
         //  trace driver operations for dump
         int mDriverOp;
         int mStandbyCnt;
         bool mSleepReq;
+        SortedVector<effect_handle_t> mPreprocessors;
+        int16_t *mProcBuf;
+        size_t mProcBufSize;
+        size_t mProcFramesIn;
+        int16_t *mRefBuf;
+        size_t mRefBufSize;
+        size_t mRefFramesIn;
+        struct echo_reference_itfe *mEchoReference;
+        bool mNeedEchoReference;
     };
 
 };
 
-}; // namespace android_audio_legacy
+}; // namespace android
 
 #endif
diff --git a/libaudio/AudioPolicyManager.cpp b/libaudio/AudioPolicyManager.cpp
index 4572729..bfa6c00 100644
--- a/libaudio/AudioPolicyManager.cpp
+++ b/libaudio/AudioPolicyManager.cpp
@@ -43,4 +43,4 @@ extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
 }
 
 
-}; // namespace android_audio_legacy
+}; // namespace android
diff --git a/libaudio/alsa_audio.h b/libaudio/alsa_audio.h
deleted file mode 100644
index 3cb86d9..0000000
--- a/libaudio/alsa_audio.h
+++ /dev/null
@@ -1,77 +0,0 @@
-/*
-** 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
deleted file mode 100644
index 3036ef8..0000000
--- a/libaudio/alsa_mixer.c
+++ /dev/null
@@ -1,371 +0,0 @@
-/*
-** 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
deleted file mode 100644
index 5673391..0000000
--- a/libaudio/alsa_pcm.c
+++ /dev/null
@@ -1,405 +0,0 @@
-/*
-** 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
deleted file mode 100644
index d978caa..0000000
--- a/libaudio/amix.c
+++ /dev/null
@@ -1,78 +0,0 @@
-/*
-** 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
deleted file mode 100644
index 0ac0ac0..0000000
--- a/libaudio/aplay.c
+++ /dev/null
@@ -1,140 +0,0 @@
-/*
-** 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
deleted file mode 100644
index b1e9eda..0000000
--- a/libaudio/arec.c
+++ /dev/null
@@ -1,128 +0,0 @@
-/*
-** 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
deleted file mode 100644
index 6a17f29..0000000
--- a/libaudio/asound.h
+++ /dev/null
@@ -1,814 +0,0 @@
-/****************************************************************************
- ****************************************************************************
- ***
- ***   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
-