Commit: 2d37c6cf645f282661995c21106920ef587b1034 - hardware-libaudio (git)

hardware/libaudio

Commit MetaInfo

• Revisión: 2d37c6cf645f282661995c21106920ef587b1034 (tree)
• Tiempo: 2012-01-23 04:20:37
• Autor: Chih-Wei Huang <cwhuang@linu...>
• Commiter: Chih-Wei Huang

Log Message

audio hal for ics-x86

Modified from Linaro project.

Cambiar Resumen

• modified: Android.mk
• add: AudioHardware.cpp
• add: AudioHardware.h
• add: alsa.mk
• add: alsa_audio.h
• add: alsa_mixer.c
• add: alsa_pcm.c
• add: asound.h

Diferencia incremental

--- a/Android.mk
+++ b/Android.mk
@@ -11,3 +11,35 @@
 # 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.
+
+ifeq ($(strip $(BOARD_USES_ALSA_AUDIO)),true)
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE := audio.primary.default
+LOCAL_MODULE_TAGS := optional
+LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/hw
+
+LOCAL_SHARED_LIBRARIES := \
+    libdl \
+    libmedia \
+    libutils \
+    libcutils \
+    libhardware_legacy
+
+LOCAL_SRC_FILES := \
+    alsa_pcm.c \
+    alsa_mixer.c \
+    AudioHardware.cpp
+
+LOCAL_STATIC_LIBRARIES := \
+    libmedia_helper
+
+LOCAL_WHOLE_STATIC_LIBRARIES := \
+    libaudiohw_legacy
+
+include $(BUILD_SHARED_LIBRARY)
+
+endif

--- /dev/null
+++ b/AudioHardware.cpp
@@ -0,0 +1,1811 @@
+/*
+** Copyright 2011, The Android Open-Source Project
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+**     http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+*/
+
+#include <math.h>
+
+//#define LOG_NDEBUG 0
+
+#define LOG_TAG "AudioHardware"
+
+#include <utils/Log.h>
+#include <utils/String8.h>
+
+#include <stdio.h>
+#include <unistd.h>
+#include <sys/ioctl.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <sys/resource.h>
+#include <dlfcn.h>
+#include <fcntl.h>
+
+#include "AudioHardware.h"
+#include <media/AudioRecord.h>
+#include <hardware_legacy/power.h>
+
+#include "alsa_audio.h"
+
+
+namespace android_audio_legacy {
+
+const uint32_t AudioHardware::inputSamplingRates[] = {
+    8000, 11025, 16000, 22050, 44100
+};
+
+//  trace driver operations for dump
+//
+#define DRIVER_TRACE
+
+enum {
+    DRV_NONE,
+    DRV_PCM_OPEN,
+    DRV_PCM_CLOSE,
+    DRV_PCM_WRITE,
+    DRV_PCM_READ,
+    DRV_MIXER_OPEN,
+    DRV_MIXER_CLOSE,
+    DRV_MIXER_GET,
+    DRV_MIXER_SEL
+};
+
+#ifdef DRIVER_TRACE
+#define TRACE_DRIVER_IN(op) mDriverOp = op;
+#define TRACE_DRIVER_OUT mDriverOp = DRV_NONE;
+#else
+#define TRACE_DRIVER_IN(op)
+#define TRACE_DRIVER_OUT
+#endif
+
+// ----------------------------------------------------------------------------
+
+AudioHardware::AudioHardware() :
+    mInit(false),
+    mMicMute(false),
+    mPcm(NULL),
+    mMixer(NULL),
+    mPcmOpenCnt(0),
+    mMixerOpenCnt(0),
+    mInCallAudioMode(false),
+    mInputSource("Default"),
+    mBluetoothNrec(true),
+    mActivatedCP(false),
+    mDriverOp(DRV_NONE)
+{
+    mInit = true;
+}
+
+AudioHardware::~AudioHardware()
+{
+    for (size_t index = 0; index < mInputs.size(); index++) {
+        closeInputStream(mInputs[index].get());
+    }
+    mInputs.clear();
+    closeOutputStream((AudioStreamOut*)mOutput.get());
+
+    if (mMixer) {
+        TRACE_DRIVER_IN(DRV_MIXER_CLOSE)
+        mixer_close(mMixer);
+        TRACE_DRIVER_OUT
+    }
+    if (mPcm) {
+        TRACE_DRIVER_IN(DRV_PCM_CLOSE)
+        pcm_close(mPcm);
+        TRACE_DRIVER_OUT
+    }
+
+    mInit = false;
+}
+
+status_t AudioHardware::initCheck()
+{
+    return mInit ? NO_ERROR : NO_INIT;
+}
+
+AudioStreamOut* AudioHardware::openOutputStream(
+    uint32_t devices, int *format, uint32_t *channels,
+    uint32_t *sampleRate, status_t *status)
+{
+    sp <AudioStreamOutALSA> out;
+    status_t rc;
+
+    { // scope for the lock
+        Mutex::Autolock lock(mLock);
+
+        // only one output stream allowed
+        if (mOutput != 0) {
+            if (status) {
+                *status = INVALID_OPERATION;
+            }
+            return NULL;
+        }
+
+        out = new AudioStreamOutALSA();
+
+        rc = out->set(this, devices, format, channels, sampleRate);
+        if (rc == NO_ERROR) {
+            mOutput = out;
+        }
+    }
+
+    if (rc != NO_ERROR) {
+        if (out != 0) {
+            out.clear();
+        }
+    }
+    if (status) {
+        *status = rc;
+    }
+
+    return out.get();
+}
+
+void AudioHardware::closeOutputStream(AudioStreamOut* out) {
+    sp <AudioStreamOutALSA> spOut;
+    {
+        Mutex::Autolock lock(mLock);
+        if (mOutput == 0 || mOutput.get() != out) {
+            LOGW("Attempt to close invalid output stream");
+            return;
+        }
+        spOut = mOutput;
+        mOutput.clear();
+    }
+    spOut.clear();
+}
+
+AudioStreamIn* AudioHardware::openInputStream(
+    uint32_t devices, int *format, uint32_t *channels,
+    uint32_t *sampleRate, status_t *status,
+    AudioSystem::audio_in_acoustics acoustic_flags)
+{
+    // check for valid input source
+    if (!AudioSystem::isInputDevice((AudioSystem::audio_devices)devices)) {
+        if (status) {
+            *status = BAD_VALUE;
+        }
+        return NULL;
+    }
+
+    status_t rc = NO_ERROR;
+    sp <AudioStreamInALSA> in;
+
+    { // scope for the lock
+        Mutex::Autolock lock(mLock);
+
+        in = new AudioStreamInALSA();
+        rc = in->set(this, devices, format, channels, sampleRate, acoustic_flags);
+        if (rc == NO_ERROR) {
+            mInputs.add(in);
+        }
+    }
+
+    if (rc != NO_ERROR) {
+        if (in != 0) {
+            in.clear();
+        }
+    }
+    if (status) {
+        *status = rc;
+    }
+
+    LOGV("AudioHardware::openInputStream()%p", in.get());
+    return in.get();
+}
+
+void AudioHardware::closeInputStream(AudioStreamIn* in) {
+
+    sp<AudioStreamInALSA> spIn;
+    {
+        Mutex::Autolock lock(mLock);
+
+        ssize_t index = mInputs.indexOf((AudioStreamInALSA *)in);
+        if (index < 0) {
+            LOGW("Attempt to close invalid input stream");
+            return;
+        }
+        spIn = mInputs[index];
+        mInputs.removeAt(index);
+    }
+    LOGV("AudioHardware::closeInputStream()%p", in);
+    spIn.clear();
+}
+
+
+status_t AudioHardware::setMode(int mode)
+{
+    sp<AudioStreamOutALSA> spOut;
+    sp<AudioStreamInALSA> spIn;
+    status_t status;
+
+    // bump thread priority to speed up mutex acquisition
+    int  priority = getpriority(PRIO_PROCESS, 0);
+    setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_URGENT_AUDIO);
+
+    // Mutex acquisition order is always out -> in -> hw
+    AutoMutex lock(mLock);
+
+    spOut = mOutput;
+    while (spOut != 0) {
+        if (!spOut->checkStandby()) {
+            int cnt = spOut->standbyCnt();
+            mLock.unlock();
+            spOut->lock();
+            mLock.lock();
+            // make sure that another thread did not change output state while the
+            // mutex is released
+            if ((spOut == mOutput) && (cnt == spOut->standbyCnt())) {
+                break;
+            }
+            spOut->unlock();
+            spOut = mOutput;
+        } else {
+            spOut.clear();
+        }
+    }
+    // spOut is not 0 here only if the output is active
+
+    spIn = getActiveInput_l();
+    while (spIn != 0) {
+        int cnt = spIn->standbyCnt();
+        mLock.unlock();
+        spIn->lock();
+        mLock.lock();
+        // make sure that another thread did not change input state while the
+        // mutex is released
+        if ((spIn == getActiveInput_l()) && (cnt == spIn->standbyCnt())) {
+            break;
+        }
+        spIn->unlock();
+        spIn = getActiveInput_l();
+    }
+    // spIn is not 0 here only if the input is active
+
+    setpriority(PRIO_PROCESS, 0, priority);
+
+    int prevMode = mMode;
+    status = AudioHardwareBase::setMode(mode);
+    LOGV("setMode() : new %d, old %d", mMode, prevMode);
+    if (status == NO_ERROR) {
+        if (mMode == AudioSystem::MODE_IN_CALL && !mInCallAudioMode) {
+            if (spOut != 0) {
+                LOGV("setMode() in call force output standby");
+                spOut->doStandby_l();
+            }
+            if (spIn != 0) {
+                LOGV("setMode() in call force input standby");
+                spIn->doStandby_l();
+            }
+
+            LOGV("setMode() openPcmOut_l()");
+            openPcmOut_l();
+            openMixer_l();
+            setInputSource_l(String8("Default"));
+            mInCallAudioMode = true;
+        }
+        if (mMode == AudioSystem::MODE_NORMAL && mInCallAudioMode) {
+            setInputSource_l(mInputSource);
+            if (mMixer != NULL) {
+                TRACE_DRIVER_IN(DRV_MIXER_GET)
+                struct mixer_ctl *ctl= mixer_get_control(mMixer, "Playback Path", 0);
+                TRACE_DRIVER_OUT
+                if (ctl != NULL) {
+                    LOGV("setMode() reset Playback Path to RCV");
+                    TRACE_DRIVER_IN(DRV_MIXER_SEL)
+                    mixer_ctl_select(ctl, "RCV");
+                    TRACE_DRIVER_OUT
+                }
+            }
+            LOGV("setMode() closePcmOut_l()");
+            closeMixer_l();
+            closePcmOut_l();
+
+            if (spOut != 0) {
+                LOGV("setMode() off call force output standby");
+                spOut->doStandby_l();
+            }
+            if (spIn != 0) {
+                LOGV("setMode() off call force input standby");
+                spIn->doStandby_l();
+            }
+
+            mInCallAudioMode = false;
+        }
+
+        if (mMode == AudioSystem::MODE_NORMAL) {
+            if(mActivatedCP)
+                mActivatedCP = false;
+        }
+    }
+
+    if (spIn != 0) {
+        spIn->unlock();
+    }
+    if (spOut != 0) {
+        spOut->unlock();
+    }
+
+    return status;
+}
+
+status_t AudioHardware::setMicMute(bool state)
+{
+    LOGV("setMicMute(%d) mMicMute %d", state, mMicMute);
+    sp<AudioStreamInALSA> spIn;
+    {
+        AutoMutex lock(mLock);
+        if (mMicMute != state) {
+            mMicMute = state;
+            if (mMode != AudioSystem::MODE_IN_CALL) {
+                spIn = getActiveInput_l();
+            }
+        }
+    }
+
+    if (spIn != 0) {
+        spIn->standby();
+    }
+
+    return NO_ERROR;
+}
+
+status_t AudioHardware::getMicMute(bool* state)
+{
+    *state = mMicMute;
+    return NO_ERROR;
+}
+
+status_t AudioHardware::setParameters(const String8& keyValuePairs)
+{
+    AudioParameter param = AudioParameter(keyValuePairs);
+    String8 value;
+    String8 key;
+    const char BT_NREC_KEY[] = "bt_headset_nrec";
+    const char BT_NREC_VALUE_ON[] = "on";
+
+    key = String8(BT_NREC_KEY);
+    if (param.get(key, value) == NO_ERROR) {
+        if (value == BT_NREC_VALUE_ON) {
+            mBluetoothNrec = true;
+        } else {
+            mBluetoothNrec = false;
+            LOGD("Turning noise reduction and echo cancellation off for BT "
+                 "headset");
+        }
+    }
+
+    return NO_ERROR;
+}
+
+String8 AudioHardware::getParameters(const String8& keys)
+{
+    AudioParameter request = AudioParameter(keys);
+    AudioParameter reply = AudioParameter();
+
+    LOGV("getParameters() %s", keys.string());
+
+    return reply.toString();
+}
+
+size_t AudioHardware::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
+{
+    if (format != AudioSystem::PCM_16_BIT) {
+        LOGW("getInputBufferSize bad format: %d", format);
+        return 0;
+    }
+    if (channelCount < 1 || channelCount > 2) {
+        LOGW("getInputBufferSize bad channel count: %d", channelCount);
+        return 0;
+    }
+    if (sampleRate != 8000 && sampleRate != 11025 && sampleRate != 16000 &&
+            sampleRate != 22050 && sampleRate != 44100) {
+        LOGW("getInputBufferSize bad sample rate: %d", sampleRate);
+        return 0;
+    }
+
+    return AudioStreamInALSA::getBufferSize(sampleRate, channelCount);
+}
+
+
+status_t AudioHardware::setVoiceVolume(float volume)
+{
+    LOGD("### setVoiceVolume: %f", volume);
+    return NO_ERROR;
+}
+
+status_t AudioHardware::setMasterVolume(float volume)
+{
+    LOGV("Set master volume to %f.\n", volume);
+    // We return an error code here to let the audioflinger do in-software
+    // volume on top of the maximum volume that we set through the SND API.
+    // return error - software mixer will handle it
+    return -1;
+}
+
+static const int kDumpLockRetries = 50;
+static const int kDumpLockSleep = 20000;
+
+static bool tryLock(Mutex& mutex)
+{
+    bool locked = false;
+    for (int i = 0; i < kDumpLockRetries; ++i) {
+        if (mutex.tryLock() == NO_ERROR) {
+            locked = true;
+            break;
+        }
+        usleep(kDumpLockSleep);
+    }
+    return locked;
+}
+
+status_t AudioHardware::dump(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    bool locked = tryLock(mLock);
+    if (!locked) {
+        snprintf(buffer, SIZE, "\n\tAudioHardware maybe deadlocked\n");
+    } else {
+        mLock.unlock();
+    }
+
+    snprintf(buffer, SIZE, "\tInit %s\n", (mInit) ? "OK" : "Failed");
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tMic Mute %s\n", (mMicMute) ? "ON" : "OFF");
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmPcm: %p\n", mPcm);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmPcmOpenCnt: %d\n", mPcmOpenCnt);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmMixer: %p\n", mMixer);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmMixerOpenCnt: %d\n", mMixerOpenCnt);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tIn Call Audio Mode %s\n",
+             (mInCallAudioMode) ? "ON" : "OFF");
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tInput source %s\n", mInputSource.string());
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tCP %s\n",
+             (mActivatedCP) ? "Activated" : "Deactivated");
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmDriverOp: %d\n", mDriverOp);
+    result.append(buffer);
+
+    snprintf(buffer, SIZE, "\n\tmOutput %p dump:\n", mOutput.get());
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+    if (mOutput != 0) {
+        mOutput->dump(fd, args);
+    }
+
+    snprintf(buffer, SIZE, "\n\t%d inputs opened:\n", mInputs.size());
+    write(fd, buffer, strlen(buffer));
+    for (size_t i = 0; i < mInputs.size(); i++) {
+        snprintf(buffer, SIZE, "\t- input %d dump:\n", i);
+        write(fd, buffer, strlen(buffer));
+        mInputs[i]->dump(fd, args);
+    }
+
+    return NO_ERROR;
+}
+
+status_t AudioHardware::setIncallPath_l(uint32_t device)
+{
+    LOGV("setIncallPath_l: device %x", device);
+    return NO_ERROR;
+}
+
+struct pcm *AudioHardware::openPcmOut_l()
+{
+    LOGD("openPcmOut_l() mPcmOpenCnt: %d", mPcmOpenCnt);
+    if (mPcmOpenCnt++ == 0) {
+        if (mPcm != NULL) {
+            LOGE("openPcmOut_l() mPcmOpenCnt == 0 and mPcm == %p\n", mPcm);
+            mPcmOpenCnt--;
+            return NULL;
+        }
+        unsigned flags = PCM_OUT;
+
+        flags |= (AUDIO_HW_OUT_PERIOD_MULT - 1) << PCM_PERIOD_SZ_SHIFT;
+        flags |= (AUDIO_HW_OUT_PERIOD_CNT - PCM_PERIOD_CNT_MIN) << PCM_PERIOD_CNT_SHIFT;
+
+        TRACE_DRIVER_IN(DRV_PCM_OPEN)
+        mPcm = pcm_open(flags);
+        TRACE_DRIVER_OUT
+        if (!pcm_ready(mPcm)) {
+            LOGE("openPcmOut_l() cannot open pcm_out driver: %s\n", pcm_error(mPcm));
+            TRACE_DRIVER_IN(DRV_PCM_CLOSE)
+            pcm_close(mPcm);
+            TRACE_DRIVER_OUT
+            mPcmOpenCnt--;
+            mPcm = NULL;
+        }
+    }
+    return mPcm;
+}
+
+void AudioHardware::closePcmOut_l()
+{
+    LOGD("closePcmOut_l() mPcmOpenCnt: %d", mPcmOpenCnt);
+    if (mPcmOpenCnt == 0) {
+        LOGE("closePcmOut_l() mPcmOpenCnt == 0");
+        return;
+    }
+
+    if (--mPcmOpenCnt == 0) {
+        TRACE_DRIVER_IN(DRV_PCM_CLOSE)
+        pcm_close(mPcm);
+        TRACE_DRIVER_OUT
+        mPcm = NULL;
+    }
+}
+
+struct mixer *AudioHardware::openMixer_l()
+{
+    LOGV("openMixer_l() mMixerOpenCnt: %d", mMixerOpenCnt);
+    if (mMixerOpenCnt++ == 0) {
+        if (mMixer != NULL) {
+            LOGE("openMixer_l() mMixerOpenCnt == 0 and mMixer == %p\n", mMixer);
+            mMixerOpenCnt--;
+            return NULL;
+        }
+        TRACE_DRIVER_IN(DRV_MIXER_OPEN)
+        mMixer = mixer_open();
+        TRACE_DRIVER_OUT
+        if (mMixer == NULL) {
+            LOGE("openMixer_l() cannot open mixer");
+            mMixerOpenCnt--;
+            return NULL;
+        }
+    }
+    return mMixer;
+}
+
+void AudioHardware::closeMixer_l()
+{
+    LOGV("closeMixer_l() mMixerOpenCnt: %d", mMixerOpenCnt);
+    if (mMixerOpenCnt == 0) {
+        LOGE("closeMixer_l() mMixerOpenCnt == 0");
+        return;
+    }
+
+    if (--mMixerOpenCnt == 0) {
+        TRACE_DRIVER_IN(DRV_MIXER_CLOSE)
+        mixer_close(mMixer);
+        TRACE_DRIVER_OUT
+        mMixer = NULL;
+    }
+}
+
+const char *AudioHardware::getOutputRouteFromDevice(uint32_t device)
+{
+    switch (device) {
+    case AudioSystem::DEVICE_OUT_EARPIECE:
+        return "RCV";
+    case AudioSystem::DEVICE_OUT_SPEAKER:
+        if (mMode == AudioSystem::MODE_RINGTONE) return "RING_SPK";
+        else return "SPK";
+    case AudioSystem::DEVICE_OUT_WIRED_HEADPHONE:
+        if (mMode == AudioSystem::MODE_RINGTONE) return "RING_NO_MIC";
+        else return "HP_NO_MIC";
+    case AudioSystem::DEVICE_OUT_WIRED_HEADSET:
+        if (mMode == AudioSystem::MODE_RINGTONE) return "RING_HP";
+        else return "HP";
+    case (AudioSystem::DEVICE_OUT_SPEAKER|AudioSystem::DEVICE_OUT_WIRED_HEADPHONE):
+    case (AudioSystem::DEVICE_OUT_SPEAKER|AudioSystem::DEVICE_OUT_WIRED_HEADSET):
+        if (mMode == AudioSystem::MODE_RINGTONE) return "RING_SPK_HP";
+        else return "SPK_HP";
+    case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO:
+    case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
+    case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
+        return "BT";
+    default:
+        return "OFF";
+    }
+}
+
+const char *AudioHardware::getVoiceRouteFromDevice(uint32_t device)
+{
+    switch (device) {
+    case AudioSystem::DEVICE_OUT_EARPIECE:
+        return "RCV";
+    case AudioSystem::DEVICE_OUT_SPEAKER:
+        return "SPK";
+    case AudioSystem::DEVICE_OUT_WIRED_HEADPHONE:
+        return "HP_NO_MIC";
+    case AudioSystem::DEVICE_OUT_WIRED_HEADSET:
+        return "HP";
+    case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO:
+    case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
+    case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
+        return "BT";
+    default:
+        return "OFF";
+    }
+}
+
+const char *AudioHardware::getInputRouteFromDevice(uint32_t device)
+{
+    if (mMicMute) {
+        return "MIC OFF";
+    }
+
+    switch (device) {
+    case AudioSystem::DEVICE_IN_BUILTIN_MIC:
+        return "Main Mic";
+    case AudioSystem::DEVICE_IN_WIRED_HEADSET:
+        return "Hands Free Mic";
+    case AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET:
+        return "BT Sco Mic";
+    default:
+        return "MIC OFF";
+    }
+}
+
+uint32_t AudioHardware::getInputSampleRate(uint32_t sampleRate)
+{
+    uint32_t i;
+    uint32_t prevDelta;
+    uint32_t delta;
+
+    for (i = 0, prevDelta = 0xFFFFFFFF; i < sizeof(inputSamplingRates)/sizeof(uint32_t); i++, prevDelta = delta) {
+        delta = abs(sampleRate - inputSamplingRates[i]);
+        if (delta > prevDelta) break;
+    }
+    // i is always > 0 here
+    return inputSamplingRates[i-1];
+}
+
+// getActiveInput_l() must be called with mLock held
+sp <AudioHardware::AudioStreamInALSA> AudioHardware::getActiveInput_l()
+{
+    sp< AudioHardware::AudioStreamInALSA> spIn;
+
+    for (size_t i = 0; i < mInputs.size(); i++) {
+        // return first input found not being in standby mode
+        // as only one input can be in this state
+        if (!mInputs[i]->checkStandby()) {
+            spIn = mInputs[i];
+            break;
+        }
+    }
+
+    return spIn;
+}
+
+status_t AudioHardware::setInputSource_l(String8 source)
+{
+     LOGV("setInputSource_l(%s)", source.string());
+     if (source != mInputSource) {
+         if ((source == "Default") || (mMode != AudioSystem::MODE_IN_CALL)) {
+             if (mMixer) {
+                 TRACE_DRIVER_IN(DRV_MIXER_GET)
+                 struct mixer_ctl *ctl= mixer_get_control(mMixer, "Input Source", 0);
+                 TRACE_DRIVER_OUT
+                 if (ctl == NULL) {
+                     return NO_INIT;
+                 }
+                 LOGV("mixer_ctl_select, Input Source, (%s)", source.string());
+                 TRACE_DRIVER_IN(DRV_MIXER_SEL)
+                 mixer_ctl_select(ctl, source.string());
+                 TRACE_DRIVER_OUT
+             }
+         }
+         mInputSource = source;
+     }
+
+     return NO_ERROR;
+}
+
+
+//------------------------------------------------------------------------------
+//  AudioStreamOutALSA
+//------------------------------------------------------------------------------
+
+AudioHardware::AudioStreamOutALSA::AudioStreamOutALSA() :
+    mHardware(0), mPcm(0), mMixer(0), mRouteCtl(0),
+    mStandby(true), mDevices(0), mChannels(AUDIO_HW_OUT_CHANNELS),
+    mSampleRate(AUDIO_HW_OUT_SAMPLERATE), mBufferSize(AUDIO_HW_OUT_PERIOD_BYTES),
+    mDriverOp(DRV_NONE), mStandbyCnt(0)
+{
+}
+
+status_t AudioHardware::AudioStreamOutALSA::set(
+    AudioHardware* hw, uint32_t devices, int *pFormat,
+    uint32_t *pChannels, uint32_t *pRate)
+{
+    int lFormat = pFormat ? *pFormat : 0;
+    uint32_t lChannels = pChannels ? *pChannels : 0;
+    uint32_t lRate = pRate ? *pRate : 0;
+
+    mHardware = hw;
+    mDevices = devices;
+
+    // fix up defaults
+    if (lFormat == 0) lFormat = format();
+    if (lChannels == 0) lChannels = channels();
+    if (lRate == 0) lRate = sampleRate();
+
+    // check values
+    if ((lFormat != format()) ||
+        (lChannels != channels()) ||
+        (lRate != sampleRate())) {
+        if (pFormat) *pFormat = format();
+        if (pChannels) *pChannels = channels();
+        if (pRate) *pRate = sampleRate();
+        return BAD_VALUE;
+    }
+
+    if (pFormat) *pFormat = lFormat;
+    if (pChannels) *pChannels = lChannels;
+    if (pRate) *pRate = lRate;
+
+    mChannels = lChannels;
+    mSampleRate = lRate;
+    mBufferSize = AUDIO_HW_OUT_PERIOD_BYTES;
+
+    return NO_ERROR;
+}
+
+AudioHardware::AudioStreamOutALSA::~AudioStreamOutALSA()
+{
+    standby();
+}
+
+ssize_t AudioHardware::AudioStreamOutALSA::write(const void* buffer, size_t bytes)
+{
+    //    LOGV("AudioStreamOutALSA::write(%p, %u)", buffer, bytes);
+    status_t status = NO_INIT;
+    const uint8_t* p = static_cast<const uint8_t*>(buffer);
+    int ret;
+
+    if (mHardware == NULL) return NO_INIT;
+
+    { // scope for the lock
+
+        AutoMutex lock(mLock);
+
+        if (mStandby) {
+            AutoMutex hwLock(mHardware->lock());
+
+            LOGD("AudioHardware pcm playback is exiting standby.");
+            acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioOutLock");
+
+            sp<AudioStreamInALSA> spIn = mHardware->getActiveInput_l();
+            while (spIn != 0) {
+                int cnt = spIn->standbyCnt();
+                mHardware->lock().unlock();
+                // Mutex acquisition order is always out -> in -> hw
+                spIn->lock();
+                mHardware->lock().lock();
+                // make sure that another thread did not change input state
+                // while the mutex is released
+                if ((spIn == mHardware->getActiveInput_l()) &&
+                        (cnt == spIn->standbyCnt())) {
+                    LOGV("AudioStreamOutALSA::write() force input standby");
+                    spIn->close_l();
+                    break;
+                }
+                spIn->unlock();
+                spIn = mHardware->getActiveInput_l();
+            }
+            // spIn is not 0 here only if the input was active and has been
+            // closed above
+
+            // open output before input
+            open_l();
+
+            if (spIn != 0) {
+                if (spIn->open_l() != NO_ERROR) {
+                    spIn->doStandby_l();
+                }
+                spIn->unlock();
+            }
+            if (mPcm == NULL) {
+                release_wake_lock("AudioOutLock");
+                goto Error;
+            }
+            mStandby = false;
+        }
+
+        TRACE_DRIVER_IN(DRV_PCM_WRITE)
+        ret = pcm_write(mPcm,(void*) p, bytes);
+        TRACE_DRIVER_OUT
+
+        if (ret == 0) {
+            return bytes;
+        }
+        LOGW("write error: %d", errno);
+        status = -errno;
+    }
+Error:
+
+    standby();
+
+    // Simulate audio output timing in case of error
+    usleep((((bytes * 1000) / frameSize()) * 1000) / sampleRate());
+
+    return status;
+}
+
+status_t AudioHardware::AudioStreamOutALSA::standby()
+{
+    if (mHardware == NULL) return NO_INIT;
+
+    AutoMutex lock(mLock);
+
+    { // scope for the AudioHardware lock
+        AutoMutex hwLock(mHardware->lock());
+
+        doStandby_l();
+    }
+
+    return NO_ERROR;
+}
+
+void AudioHardware::AudioStreamOutALSA::doStandby_l()
+{
+    mStandbyCnt++;
+
+    if (!mStandby) {
+        LOGD("AudioHardware pcm playback is going to standby.");
+        release_wake_lock("AudioOutLock");
+        mStandby = true;
+    }
+
+    close_l();
+}
+
+void AudioHardware::AudioStreamOutALSA::close_l()
+{
+    if (mMixer) {
+        mHardware->closeMixer_l();
+        mMixer = NULL;
+        mRouteCtl = NULL;
+    }
+    if (mPcm) {
+        mHardware->closePcmOut_l();
+        mPcm = NULL;
+    }
+}
+
+status_t AudioHardware::AudioStreamOutALSA::open_l()
+{
+    LOGV("open pcm_out driver");
+    mPcm = mHardware->openPcmOut_l();
+    if (mPcm == NULL) {
+        return NO_INIT;
+    }
+
+    mMixer = mHardware->openMixer_l();
+    if (mMixer) {
+        LOGV("open playback normal");
+        TRACE_DRIVER_IN(DRV_MIXER_GET)
+        mRouteCtl = mixer_get_control(mMixer, "Playback Path", 0);
+        TRACE_DRIVER_OUT
+    }
+    if (mHardware->mode() != AudioSystem::MODE_IN_CALL) {
+        const char *route = mHardware->getOutputRouteFromDevice(mDevices);
+        LOGV("write() wakeup setting route %s", route);
+        if (mRouteCtl) {
+            TRACE_DRIVER_IN(DRV_MIXER_SEL)
+            mixer_ctl_select(mRouteCtl, route);
+            TRACE_DRIVER_OUT
+        }
+    }
+    return NO_ERROR;
+}
+
+status_t AudioHardware::AudioStreamOutALSA::dump(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    bool locked = tryLock(mLock);
+    if (!locked) {
+        snprintf(buffer, SIZE, "\n\t\tAudioStreamOutALSA maybe deadlocked\n");
+    } else {
+        mLock.unlock();
+    }
+
+    snprintf(buffer, SIZE, "\t\tmHardware: %p\n", mHardware);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmPcm: %p\n", mPcm);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmMixer: %p\n", mMixer);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmRouteCtl: %p\n", mRouteCtl);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tStandby %s\n", (mStandby) ? "ON" : "OFF");
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmDevices: 0x%08x\n", mDevices);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmChannels: 0x%08x\n", mChannels);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmSampleRate: %d\n", mSampleRate);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmBufferSize: %d\n", mBufferSize);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmDriverOp: %d\n", mDriverOp);
+    result.append(buffer);
+
+    ::write(fd, result.string(), result.size());
+
+    return NO_ERROR;
+}
+
+bool AudioHardware::AudioStreamOutALSA::checkStandby()
+{
+    return mStandby;
+}
+
+status_t AudioHardware::AudioStreamOutALSA::setParameters(const String8& keyValuePairs)
+{
+    AudioParameter param = AudioParameter(keyValuePairs);
+    status_t status = NO_ERROR;
+    int device;
+    LOGD("AudioStreamOutALSA::setParameters() %s", keyValuePairs.string());
+
+    if (mHardware == NULL) return NO_INIT;
+
+    {
+        AutoMutex lock(mLock);
+
+        if (param.getInt(String8(AudioParameter::keyRouting), device) == NO_ERROR)
+        {
+            AutoMutex hwLock(mHardware->lock());
+
+            if (mDevices != (uint32_t)device) {
+                mDevices = (uint32_t)device;
+                if (mHardware->mode() != AudioSystem::MODE_IN_CALL) {
+                    doStandby_l();
+                }
+            }
+            if (mHardware->mode() == AudioSystem::MODE_IN_CALL) {
+                mHardware->setIncallPath_l(device);
+            }
+            param.remove(String8(AudioParameter::keyRouting));
+        }
+    }
+
+    if (param.size()) {
+        status = BAD_VALUE;
+    }
+
+
+    return status;
+
+}
+
+String8 AudioHardware::AudioStreamOutALSA::getParameters(const String8& keys)
+{
+    AudioParameter param = AudioParameter(keys);
+    String8 value;
+    String8 key = String8(AudioParameter::keyRouting);
+
+    if (param.get(key, value) == NO_ERROR) {
+        param.addInt(key, (int)mDevices);
+    }
+
+    LOGV("AudioStreamOutALSA::getParameters() %s", param.toString().string());
+    return param.toString();
+}
+
+status_t AudioHardware::AudioStreamOutALSA::getRenderPosition(uint32_t *dspFrames)
+{
+    //TODO
+    return INVALID_OPERATION;
+}
+
+//------------------------------------------------------------------------------
+//  AudioStreamInALSA
+//------------------------------------------------------------------------------
+
+AudioHardware::AudioStreamInALSA::AudioStreamInALSA() :
+    mHardware(0), mPcm(0), mMixer(0), mRouteCtl(0),
+    mStandby(true), mDevices(0), mChannels(AUDIO_HW_IN_CHANNELS), mChannelCount(1),
+    mSampleRate(AUDIO_HW_IN_SAMPLERATE), mBufferSize(AUDIO_HW_IN_PERIOD_BYTES),
+    mDownSampler(NULL), mReadStatus(NO_ERROR), mDriverOp(DRV_NONE),
+    mStandbyCnt(0)
+{
+}
+
+status_t AudioHardware::AudioStreamInALSA::set(
+    AudioHardware* hw, uint32_t devices, int *pFormat,
+    uint32_t *pChannels, uint32_t *pRate, AudioSystem::audio_in_acoustics acoustics)
+{
+    if (pFormat == 0 || *pFormat != AUDIO_HW_IN_FORMAT) {
+        *pFormat = AUDIO_HW_IN_FORMAT;
+        return BAD_VALUE;
+    }
+    if (pRate == 0) {
+        return BAD_VALUE;
+    }
+    uint32_t rate = AudioHardware::getInputSampleRate(*pRate);
+    if (rate != *pRate) {
+        *pRate = rate;
+        return BAD_VALUE;
+    }
+
+    if (pChannels == 0 || (*pChannels != AudioSystem::CHANNEL_IN_MONO &&
+        *pChannels != AudioSystem::CHANNEL_IN_STEREO)) {
+        *pChannels = AUDIO_HW_IN_CHANNELS;
+        return BAD_VALUE;
+    }
+
+    mHardware = hw;
+
+    LOGV("AudioStreamInALSA::set(%d, %d, %u)", *pFormat, *pChannels, *pRate);
+
+    mBufferSize = getBufferSize(*pRate, AudioSystem::popCount(*pChannels));
+    mDevices = devices;
+    mChannels = *pChannels;
+    mChannelCount = AudioSystem::popCount(mChannels);
+    mSampleRate = rate;
+    if (mSampleRate != AUDIO_HW_OUT_SAMPLERATE) {
+        mDownSampler = new AudioHardware::DownSampler(mSampleRate,
+                                                  mChannelCount,
+                                                  AUDIO_HW_IN_PERIOD_SZ,
+                                                  this);
+        status_t status = mDownSampler->initCheck();
+        if (status != NO_ERROR) {
+            delete mDownSampler;
+            LOGW("AudioStreamInALSA::set() downsampler init failed: %d", status);
+            return status;
+        }
+
+        mPcmIn = new int16_t[AUDIO_HW_IN_PERIOD_SZ * mChannelCount];
+    }
+    return NO_ERROR;
+}
+
+AudioHardware::AudioStreamInALSA::~AudioStreamInALSA()
+{
+    standby();
+    if (mDownSampler != NULL) {
+        delete mDownSampler;
+        if (mPcmIn != NULL) {
+            delete[] mPcmIn;
+        }
+    }
+}
+
+ssize_t AudioHardware::AudioStreamInALSA::read(void* buffer, ssize_t bytes)
+{
+    //    LOGV("AudioStreamInALSA::read(%p, %u)", buffer, bytes);
+    status_t status = NO_INIT;
+    int ret;
+
+    if (mHardware == NULL) return NO_INIT;
+
+    { // scope for the lock
+        AutoMutex lock(mLock);
+
+        if (mStandby) {
+            AutoMutex hwLock(mHardware->lock());
+
+            LOGD("AudioHardware pcm capture is exiting standby.");
+            acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioInLock");
+
+            sp<AudioStreamOutALSA> spOut = mHardware->output();
+            while (spOut != 0) {
+                if (!spOut->checkStandby()) {
+                    int cnt = spOut->standbyCnt();
+                    mHardware->lock().unlock();
+                    mLock.unlock();
+                    // Mutex acquisition order is always out -> in -> hw
+                    spOut->lock();
+                    mLock.lock();
+                    mHardware->lock().lock();
+                    // make sure that another thread did not change output state
+                    // while the mutex is released
+                    if ((spOut == mHardware->output()) && (cnt == spOut->standbyCnt())) {
+                        LOGV("AudioStreamInALSA::read() force output standby");
+                        spOut->close_l();
+                        break;
+                    }
+                    spOut->unlock();
+                    spOut = mHardware->output();
+                } else {
+                    spOut.clear();
+                }
+            }
+            // spOut is not 0 here only if the output was active and has been
+            // closed above
+
+            // open output before input
+            if (spOut != 0) {
+                if (spOut->open_l() != NO_ERROR) {
+                    spOut->doStandby_l();
+                }
+                spOut->unlock();
+            }
+
+            open_l();
+
+            if (mPcm == NULL) {
+                release_wake_lock("AudioInLock");
+                goto Error;
+            }
+            mStandby = false;
+        }
+
+
+        if (mDownSampler != NULL) {
+            size_t frames = bytes / frameSize();
+            size_t framesIn = 0;
+            mReadStatus = 0;
+            do {
+                size_t outframes = frames - framesIn;
+                mDownSampler->resample(
+                        (int16_t *)buffer + (framesIn * mChannelCount),
+                        &outframes);
+                framesIn += outframes;
+            } while ((framesIn < frames) && mReadStatus == 0);
+            ret = mReadStatus;
+            bytes = framesIn * frameSize();
+        } else {
+            TRACE_DRIVER_IN(DRV_PCM_READ)
+            ret = pcm_read(mPcm, buffer, bytes);
+            TRACE_DRIVER_OUT
+        }
+
+        if (ret == 0) {
+            return bytes;
+        }
+
+        LOGW("read error: %d", ret);
+        status = ret;
+    }
+
+Error:
+
+    standby();
+
+    // Simulate audio output timing in case of error
+    usleep((((bytes * 1000) / frameSize()) * 1000) / sampleRate());
+
+    return status;
+}
+
+status_t AudioHardware::AudioStreamInALSA::standby()
+{
+    if (mHardware == NULL) return NO_INIT;
+
+    AutoMutex lock(mLock);
+
+    { // scope for AudioHardware lock
+        AutoMutex hwLock(mHardware->lock());
+
+        doStandby_l();
+    }
+    return NO_ERROR;
+}
+
+void AudioHardware::AudioStreamInALSA::doStandby_l()
+{
+    mStandbyCnt++;
+
+    if (!mStandby) {
+        LOGD("AudioHardware pcm capture is going to standby.");
+        release_wake_lock("AudioInLock");
+        mStandby = true;
+    }
+    close_l();
+}
+
+void AudioHardware::AudioStreamInALSA::close_l()
+{
+    if (mMixer) {
+        mHardware->closeMixer_l();
+        mMixer = NULL;
+        mRouteCtl = NULL;
+    }
+
+    if (mPcm) {
+        TRACE_DRIVER_IN(DRV_PCM_CLOSE)
+        pcm_close(mPcm);
+        TRACE_DRIVER_OUT
+        mPcm = NULL;
+    }
+}
+
+status_t AudioHardware::AudioStreamInALSA::open_l()
+{
+    unsigned flags = PCM_IN;
+    if (mChannels == AudioSystem::CHANNEL_IN_MONO) {
+        flags |= PCM_MONO;
+    }
+    flags |= (AUDIO_HW_IN_PERIOD_MULT - 1) << PCM_PERIOD_SZ_SHIFT;
+    flags |= (AUDIO_HW_IN_PERIOD_CNT - PCM_PERIOD_CNT_MIN)
+            << PCM_PERIOD_CNT_SHIFT;
+
+    LOGV("open pcm_in driver");
+    TRACE_DRIVER_IN(DRV_PCM_OPEN)
+    mPcm = pcm_open(flags);
+    TRACE_DRIVER_OUT
+    if (!pcm_ready(mPcm)) {
+        LOGE("cannot open pcm_in driver: %s\n", pcm_error(mPcm));
+        TRACE_DRIVER_IN(DRV_PCM_CLOSE)
+        pcm_close(mPcm);
+        TRACE_DRIVER_OUT
+        mPcm = NULL;
+        return NO_INIT;
+    }
+
+    if (mDownSampler != NULL) {
+        mInPcmInBuf = 0;
+        mDownSampler->reset();
+    }
+
+    mMixer = mHardware->openMixer_l();
+    if (mMixer) {
+        TRACE_DRIVER_IN(DRV_MIXER_GET)
+        mRouteCtl = mixer_get_control(mMixer, "Capture MIC Path", 0);
+        TRACE_DRIVER_OUT
+    }
+
+    if (mHardware->mode() != AudioSystem::MODE_IN_CALL) {
+        const char *route = mHardware->getInputRouteFromDevice(mDevices);
+        LOGV("read() wakeup setting route %s", route);
+        if (mRouteCtl) {
+            TRACE_DRIVER_IN(DRV_MIXER_SEL)
+            mixer_ctl_select(mRouteCtl, route);
+            TRACE_DRIVER_OUT
+        }
+    }
+
+    return NO_ERROR;
+}
+
+status_t AudioHardware::AudioStreamInALSA::dump(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    bool locked = tryLock(mLock);
+    if (!locked) {
+        snprintf(buffer, SIZE, "\n\t\tAudioStreamInALSA maybe deadlocked\n");
+    } else {
+        mLock.unlock();
+    }
+
+    snprintf(buffer, SIZE, "\t\tmHardware: %p\n", mHardware);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmPcm: %p\n", mPcm);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmMixer: %p\n", mMixer);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tStandby %s\n", (mStandby) ? "ON" : "OFF");
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmDevices: 0x%08x\n", mDevices);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmChannels: 0x%08x\n", mChannels);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmSampleRate: %d\n", mSampleRate);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmBufferSize: %d\n", mBufferSize);
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\t\tmDriverOp: %d\n", mDriverOp);
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+
+    return NO_ERROR;
+}
+
+bool AudioHardware::AudioStreamInALSA::checkStandby()
+{
+    return mStandby;
+}
+
+status_t AudioHardware::AudioStreamInALSA::setParameters(const String8& keyValuePairs)
+{
+    AudioParameter param = AudioParameter(keyValuePairs);
+    status_t status = NO_ERROR;
+    int value;
+    String8 source;
+
+    LOGD("AudioStreamInALSA::setParameters() %s", keyValuePairs.string());
+
+    if (mHardware == NULL) return NO_INIT;
+
+    {
+        AutoMutex lock(mLock);
+
+        if (param.get(String8(INPUT_SOURCE_KEY), source) == NO_ERROR) {
+            AutoMutex hwLock(mHardware->lock());
+
+            mHardware->openMixer_l();
+            mHardware->setInputSource_l(source);
+            mHardware->closeMixer_l();
+
+            param.remove(String8(INPUT_SOURCE_KEY));
+        }
+
+        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR)
+        {
+            if (value != 0) {
+                AutoMutex hwLock(mHardware->lock());
+
+                if (mDevices != (uint32_t)value) {
+                    mDevices = (uint32_t)value;
+                    if (mHardware->mode() != AudioSystem::MODE_IN_CALL) {
+                        doStandby_l();
+                    }
+                }
+            }
+            param.remove(String8(AudioParameter::keyRouting));
+        }
+    }
+
+
+    if (param.size()) {
+        status = BAD_VALUE;
+    }
+
+    return status;
+
+}
+
+String8 AudioHardware::AudioStreamInALSA::getParameters(const String8& keys)
+{
+    AudioParameter param = AudioParameter(keys);
+    String8 value;
+    String8 key = String8(AudioParameter::keyRouting);
+
+    if (param.get(key, value) == NO_ERROR) {
+        param.addInt(key, (int)mDevices);
+    }
+
+    LOGV("AudioStreamInALSA::getParameters() %s", param.toString().string());
+    return param.toString();
+}
+
+status_t AudioHardware::AudioStreamInALSA::getNextBuffer(AudioHardware::BufferProvider::Buffer* buffer)
+{
+    if (mPcm == NULL) {
+        buffer->raw = NULL;
+        buffer->frameCount = 0;
+        mReadStatus = NO_INIT;
+        return NO_INIT;
+    }
+
+    if (mInPcmInBuf == 0) {
+        TRACE_DRIVER_IN(DRV_PCM_READ)
+        mReadStatus = pcm_read(mPcm,(void*) mPcmIn, AUDIO_HW_IN_PERIOD_SZ * frameSize());
+        TRACE_DRIVER_OUT
+        if (mReadStatus != 0) {
+            buffer->raw = NULL;
+            buffer->frameCount = 0;
+            return mReadStatus;
+        }
+        mInPcmInBuf = AUDIO_HW_IN_PERIOD_SZ;
+    }
+
+    buffer->frameCount = (buffer->frameCount > mInPcmInBuf) ? mInPcmInBuf : buffer->frameCount;
+    buffer->i16 = mPcmIn + (AUDIO_HW_IN_PERIOD_SZ - mInPcmInBuf) * mChannelCount;
+
+    return mReadStatus;
+}
+
+void AudioHardware::AudioStreamInALSA::releaseBuffer(Buffer* buffer)
+{
+    mInPcmInBuf -= buffer->frameCount;
+}
+
+size_t AudioHardware::AudioStreamInALSA::getBufferSize(uint32_t sampleRate, int channelCount)
+{
+    size_t ratio;
+
+    switch (sampleRate) {
+    case 8000:
+    case 11025:
+        ratio = 4;
+        break;
+    case 16000:
+    case 22050:
+        ratio = 2;
+        break;
+    case 44100:
+    default:
+        ratio = 1;
+        break;
+    }
+
+    return (AUDIO_HW_IN_PERIOD_SZ*channelCount*sizeof(int16_t)) / ratio ;
+}
+
+//------------------------------------------------------------------------------
+//  DownSampler
+//------------------------------------------------------------------------------
+
+/*
+ * 2.30 fixed point FIR filter coefficients for conversion 44100 -> 22050.
+ * (Works equivalently for 22010 -> 11025 or any other halving, of course.)
+ *
+ * Transition band from about 18 kHz, passband ripple < 0.1 dB,
+ * stopband ripple at about -55 dB, linear phase.
+ *
+ * Design and display in MATLAB or Octave using:
+ *
+ * filter = fir1(19, 0.5); filter = round(filter * 2**30); freqz(filter * 2**-30);
+ */
+static const int32_t filter_22khz_coeff[] = {
+    2089257, 2898328, -5820678, -10484531,
+    19038724, 30542725, -50469415, -81505260,
+    152544464, 478517512, 478517512, 152544464,
+    -81505260, -50469415, 30542725, 19038724,
+    -10484531, -5820678, 2898328, 2089257,
+};
+#define NUM_COEFF_22KHZ (sizeof(filter_22khz_coeff) / sizeof(filter_22khz_coeff[0]))
+#define OVERLAP_22KHZ (NUM_COEFF_22KHZ - 2)
+
+/*
+ * Convolution of signals A and reverse(B). (In our case, the filter response
+ * is symmetric, so the reversing doesn't matter.)
+ * A is taken to be in 0.16 fixed-point, and B is taken to be in 2.30 fixed-point.
+ * The answer will be in 16.16 fixed-point, unclipped.
+ *
+ * This function would probably be the prime candidate for SIMD conversion if
+ * you want more speed.
+ */
+int32_t fir_convolve(const int16_t* a, const int32_t* b, int num_samples)
+{
+        int32_t sum = 1 << 13;
+        for (int i = 0; i < num_samples; ++i) {
+                sum += a[i] * (b[i] >> 16);
+        }
+        return sum >> 14;
+}
+
+/* Clip from 16.16 fixed-point to 0.16 fixed-point. */
+int16_t clip(int32_t x)
+{
+    if (x < -32768) {
+        return -32768;
+    } else if (x > 32767) {
+        return 32767;
+    } else {
+        return x;
+    }
+}
+
+/*
+ * Convert a chunk from 44 kHz to 22 kHz. Will update num_samples_in and num_samples_out
+ * accordingly, since it may leave input samples in the buffer due to overlap.
+ *
+ * Input and output are taken to be in 0.16 fixed-point.
+ */
+void resample_2_1(int16_t* input, int16_t* output, int* num_samples_in, int* num_samples_out)
+{
+    if (*num_samples_in < (int)NUM_COEFF_22KHZ) {
+        *num_samples_out = 0;
+        return;
+    }
+
+    int odd_smp = *num_samples_in & 0x1;
+    int num_samples = *num_samples_in - odd_smp - OVERLAP_22KHZ;
+
+    for (int i = 0; i < num_samples; i += 2) {
+            output[i / 2] = clip(fir_convolve(input + i, filter_22khz_coeff, NUM_COEFF_22KHZ));
+    }
+
+    memmove(input, input + num_samples, (OVERLAP_22KHZ + odd_smp) * sizeof(*input));
+    *num_samples_out = num_samples / 2;
+    *num_samples_in = OVERLAP_22KHZ + odd_smp;
+}
+
+/*
+ * 2.30 fixed point FIR filter coefficients for conversion 22050 -> 16000,
+ * or 11025 -> 8000.
+ *
+ * Transition band from about 14 kHz, passband ripple < 0.1 dB,
+ * stopband ripple at about -50 dB, linear phase.
+ *
+ * Design and display in MATLAB or Octave using:
+ *
+ * filter = fir1(23, 16000 / 22050); filter = round(filter * 2**30); freqz(filter * 2**-30);
+ */
+static const int32_t filter_16khz_coeff[] = {
+    2057290, -2973608, 1880478, 4362037,
+    -14639744, 18523609, -1609189, -38502470,
+    78073125, -68353935, -59103896, 617555440,
+    617555440, -59103896, -68353935, 78073125,
+    -38502470, -1609189, 18523609, -14639744,
+    4362037, 1880478, -2973608, 2057290,
+};
+#define NUM_COEFF_16KHZ (sizeof(filter_16khz_coeff) / sizeof(filter_16khz_coeff[0]))
+#define OVERLAP_16KHZ (NUM_COEFF_16KHZ - 1)
+
+/*
+ * Convert a chunk from 22 kHz to 16 kHz. Will update num_samples_in and
+ * num_samples_out accordingly, since it may leave input samples in the buffer
+ * due to overlap.
+ *
+ * This implementation is rather ad-hoc; it first low-pass filters the data
+ * into a temporary buffer, and then converts chunks of 441 input samples at a
+ * time into 320 output samples by simple linear interpolation. A better
+ * implementation would use a polyphase filter bank to do these two operations
+ * in one step.
+ *
+ * Input and output are taken to be in 0.16 fixed-point.
+ */
+
+#define RESAMPLE_16KHZ_SAMPLES_IN 441
+#define RESAMPLE_16KHZ_SAMPLES_OUT 320
+
+void resample_441_320(int16_t* input, int16_t* output, int* num_samples_in, int* num_samples_out)
+{
+    const int num_blocks = (*num_samples_in - OVERLAP_16KHZ) / RESAMPLE_16KHZ_SAMPLES_IN;
+    if (num_blocks < 1) {
+        *num_samples_out = 0;
+        return;
+    }
+
+    for (int i = 0; i < num_blocks; ++i) {
+        uint32_t tmp[RESAMPLE_16KHZ_SAMPLES_IN];
+        for (int j = 0; j < RESAMPLE_16KHZ_SAMPLES_IN; ++j) {
+            tmp[j] = fir_convolve(input + i * RESAMPLE_16KHZ_SAMPLES_IN + j,
+                          filter_16khz_coeff,
+                          NUM_COEFF_16KHZ);
+        }
+
+        const float step_float = (float)RESAMPLE_16KHZ_SAMPLES_IN / (float)RESAMPLE_16KHZ_SAMPLES_OUT;
+
+        uint32_t in_sample_num = 0;   // 16.16 fixed point
+        const uint32_t step = (uint32_t)(step_float * 65536.0f + 0.5f);  // 16.16 fixed point
+        for (int j = 0; j < RESAMPLE_16KHZ_SAMPLES_OUT; ++j, in_sample_num += step) {
+            const uint32_t whole = in_sample_num >> 16;
+            const uint32_t frac = (in_sample_num & 0xffff);  // 0.16 fixed point
+            const int32_t s1 = tmp[whole];
+            const int32_t s2 = tmp[whole + 1];
+            *output++ = clip(s1 + (((s2 - s1) * (int32_t)frac) >> 16));
+        }
+    }
+
+    const int samples_consumed = num_blocks * RESAMPLE_16KHZ_SAMPLES_IN;
+    memmove(input, input + samples_consumed, (*num_samples_in - samples_consumed) * sizeof(*input));
+    *num_samples_in -= samples_consumed;
+    *num_samples_out = RESAMPLE_16KHZ_SAMPLES_OUT * num_blocks;
+}
+
+
+AudioHardware::DownSampler::DownSampler(uint32_t outSampleRate,
+                                    uint32_t channelCount,
+                                    uint32_t frameCount,
+                                    AudioHardware::BufferProvider* provider)
+    :  mStatus(NO_INIT), mProvider(provider), mSampleRate(outSampleRate),
+       mChannelCount(channelCount), mFrameCount(frameCount),
+       mInLeft(NULL), mInRight(NULL), mTmpLeft(NULL), mTmpRight(NULL),
+       mTmp2Left(NULL), mTmp2Right(NULL), mOutLeft(NULL), mOutRight(NULL)
+
+{
+    LOGV("AudioHardware::DownSampler() cstor %p SR %d channels %d frames %d",
+         this, mSampleRate, mChannelCount, mFrameCount);
+
+    if (mSampleRate != 8000 && mSampleRate != 11025 && mSampleRate != 16000 &&
+            mSampleRate != 22050) {
+        LOGW("AudioHardware::DownSampler cstor: bad sampling rate: %d", mSampleRate);
+        return;
+    }
+
+    mInLeft = new int16_t[mFrameCount];
+    mInRight = new int16_t[mFrameCount];
+    mTmpLeft = new int16_t[mFrameCount];
+    mTmpRight = new int16_t[mFrameCount];
+    mTmp2Left = new int16_t[mFrameCount];
+    mTmp2Right = new int16_t[mFrameCount];
+    mOutLeft = new int16_t[mFrameCount];
+    mOutRight = new int16_t[mFrameCount];
+
+    mStatus = NO_ERROR;
+}
+
+AudioHardware::DownSampler::~DownSampler()
+{
+    if (mInLeft) delete[] mInLeft;
+    if (mInRight) delete[] mInRight;
+    if (mTmpLeft) delete[] mTmpLeft;
+    if (mTmpRight) delete[] mTmpRight;
+    if (mTmp2Left) delete[] mTmp2Left;
+    if (mTmp2Right) delete[] mTmp2Right;
+    if (mOutLeft) delete[] mOutLeft;
+    if (mOutRight) delete[] mOutRight;
+}
+
+void AudioHardware::DownSampler::reset()
+{
+    mInInBuf = 0;
+    mInTmpBuf = 0;
+    mInTmp2Buf = 0;
+    mOutBufPos = 0;
+    mInOutBuf = 0;
+}
+
+
+int AudioHardware::DownSampler::resample(int16_t* out, size_t *outFrameCount)
+{
+    if (mStatus != NO_ERROR) {
+        return mStatus;
+    }
+
+    if (out == NULL || outFrameCount == NULL) {
+        return BAD_VALUE;
+    }
+
+    int16_t *outLeft = mTmp2Left;
+    int16_t *outRight = mTmp2Left;
+    if (mSampleRate == 22050) {
+        outLeft = mTmpLeft;
+        outRight = mTmpRight;
+    } else if (mSampleRate == 8000){
+        outLeft = mOutLeft;
+        outRight = mOutRight;
+    }
+
+    int outFrames = 0;
+    int remaingFrames = *outFrameCount;
+
+    if (mInOutBuf) {
+        int frames = (remaingFrames > mInOutBuf) ? mInOutBuf : remaingFrames;
+
+        for (int i = 0; i < frames; ++i) {
+            out[i] = outLeft[mOutBufPos + i];
+        }
+        if (mChannelCount == 2) {
+            for (int i = 0; i < frames; ++i) {
+                out[i * 2] = outLeft[mOutBufPos + i];
+                out[i * 2 + 1] = outRight[mOutBufPos + i];
+            }
+        }
+        remaingFrames -= frames;
+        mInOutBuf -= frames;
+        mOutBufPos += frames;
+        outFrames += frames;
+    }
+
+    while (remaingFrames) {
+        LOGW_IF((mInOutBuf != 0), "mInOutBuf should be 0 here");
+
+        AudioHardware::BufferProvider::Buffer buf;
+        buf.frameCount =  mFrameCount - mInInBuf;
+        int ret = mProvider->getNextBuffer(&buf);
+        if (buf.raw == NULL) {
+            *outFrameCount = outFrames;
+            return ret;
+        }
+
+        for (size_t i = 0; i < buf.frameCount; ++i) {
+            mInLeft[i + mInInBuf] = buf.i16[i];
+        }
+        if (mChannelCount == 2) {
+            for (size_t i = 0; i < buf.frameCount; ++i) {
+                mInLeft[i + mInInBuf] = buf.i16[i * 2];
+                mInRight[i + mInInBuf] = buf.i16[i * 2 + 1];
+            }
+        }
+        mInInBuf += buf.frameCount;
+        mProvider->releaseBuffer(&buf);
+
+        /* 44010 -> 22050 */
+        {
+            int samples_in_left = mInInBuf;
+            int samples_out_left;
+            resample_2_1(mInLeft, mTmpLeft + mInTmpBuf, &samples_in_left, &samples_out_left);
+
+            if (mChannelCount == 2) {
+                int samples_in_right = mInInBuf;
+                int samples_out_right;
+                resample_2_1(mInRight, mTmpRight + mInTmpBuf, &samples_in_right, &samples_out_right);
+            }
+
+            mInInBuf = samples_in_left;
+            mInTmpBuf += samples_out_left;
+            mInOutBuf = samples_out_left;
+        }
+
+        if (mSampleRate == 11025 || mSampleRate == 8000) {
+            /* 22050 - > 11025 */
+            int samples_in_left = mInTmpBuf;
+            int samples_out_left;
+            resample_2_1(mTmpLeft, mTmp2Left + mInTmp2Buf, &samples_in_left, &samples_out_left);
+
+            if (mChannelCount == 2) {
+                int samples_in_right = mInTmpBuf;
+                int samples_out_right;
+                resample_2_1(mTmpRight, mTmp2Right + mInTmp2Buf, &samples_in_right, &samples_out_right);
+            }
+
+
+            mInTmpBuf = samples_in_left;
+            mInTmp2Buf += samples_out_left;
+            mInOutBuf = samples_out_left;
+
+            if (mSampleRate == 8000) {
+                /* 11025 -> 8000*/
+                int samples_in_left = mInTmp2Buf;
+                int samples_out_left;
+                resample_441_320(mTmp2Left, mOutLeft, &samples_in_left, &samples_out_left);
+
+                if (mChannelCount == 2) {
+                    int samples_in_right = mInTmp2Buf;
+                    int samples_out_right;
+                    resample_441_320(mTmp2Right, mOutRight, &samples_in_right, &samples_out_right);
+                }
+
+                mInTmp2Buf = samples_in_left;
+                mInOutBuf = samples_out_left;
+            } else {
+                mInTmp2Buf = 0;
+            }
+
+        } else if (mSampleRate == 16000) {
+            /* 22050 -> 16000*/
+            int samples_in_left = mInTmpBuf;
+            int samples_out_left;
+            resample_441_320(mTmpLeft, mTmp2Left, &samples_in_left, &samples_out_left);
+
+            if (mChannelCount == 2) {
+                int samples_in_right = mInTmpBuf;
+                int samples_out_right;
+                resample_441_320(mTmpRight, mTmp2Right, &samples_in_right, &samples_out_right);
+            }
+
+            mInTmpBuf = samples_in_left;
+            mInOutBuf = samples_out_left;
+        } else {
+            mInTmpBuf = 0;
+        }
+
+        int frames = (remaingFrames > mInOutBuf) ? mInOutBuf : remaingFrames;
+
+        for (int i = 0; i < frames; ++i) {
+            out[outFrames + i] = outLeft[i];
+        }
+        if (mChannelCount == 2) {
+            for (int i = 0; i < frames; ++i) {
+                out[(outFrames + i) * 2] = outLeft[i];
+                out[(outFrames + i) * 2 + 1] = outRight[i];
+            }
+        }
+        remaingFrames -= frames;
+        outFrames += frames;
+        mOutBufPos = frames;
+        mInOutBuf -= frames;
+    }
+
+    return 0;
+}
+
+
+
+
+
+
+
+//------------------------------------------------------------------------------
+//  Factory
+//------------------------------------------------------------------------------
+
+extern "C" AudioHardwareInterface* createAudioHardware(void) {
+    return new AudioHardware();
+}
+
+}; // namespace android

--- /dev/null
+++ b/AudioHardware.h
@@ -0,0 +1,331 @@
+/*
+** Copyright 2011, The Android Open-Source Project
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+**     http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+*/
+
+#ifndef ANDROID_AUDIO_HARDWARE_H
+#define ANDROID_AUDIO_HARDWARE_H
+
+#include <stdint.h>
+#include <sys/types.h>
+
+#include <utils/threads.h>
+#include <utils/SortedVector.h>
+
+#include <hardware_legacy/AudioHardwareBase.h>
+
+extern "C" {
+    struct pcm;
+    struct mixer;
+    struct mixer_ctl;
+};
+
+namespace android_audio_legacy {
+
+using android::sp;
+using android::Mutex;
+using android::AutoMutex;
+using android::SortedVector;
+using android::RefBase;
+
+// TODO: determine actual audio DSP and hardware latency
+// Additionnal latency introduced by audio DSP and hardware in ms
+#define AUDIO_HW_OUT_LATENCY_MS 0
+// Default audio output sample rate
+#define AUDIO_HW_OUT_SAMPLERATE 44100
+// Default audio output channel mask
+#define AUDIO_HW_OUT_CHANNELS (AudioSystem::CHANNEL_OUT_STEREO)
+// Default audio output sample format
+#define AUDIO_HW_OUT_FORMAT (AudioSystem::PCM_16_BIT)
+// Kernel pcm out buffer size in frames at 44.1kHz
+#define AUDIO_HW_OUT_PERIOD_MULT 8 // (8 * 128 = 1024 frames)
+#define AUDIO_HW_OUT_PERIOD_SZ (PCM_PERIOD_SZ_MIN * AUDIO_HW_OUT_PERIOD_MULT)
+#define AUDIO_HW_OUT_PERIOD_CNT 4
+// Default audio output buffer size in bytes
+#define AUDIO_HW_OUT_PERIOD_BYTES (AUDIO_HW_OUT_PERIOD_SZ * 2 * sizeof(int16_t))
+
+// Default audio input sample rate
+#define AUDIO_HW_IN_SAMPLERATE 8000
+// Default audio input channel mask
+#define AUDIO_HW_IN_CHANNELS (AudioSystem::CHANNEL_IN_MONO)
+// Default audio input sample format
+#define AUDIO_HW_IN_FORMAT (AudioSystem::PCM_16_BIT)
+// Number of buffers in audio driver for input
+#define AUDIO_HW_NUM_IN_BUF 2
+// Kernel pcm in buffer size in frames at 44.1kHz (before resampling)
+#define AUDIO_HW_IN_PERIOD_MULT 16  // (16 * 128 = 2048 frames)
+#define AUDIO_HW_IN_PERIOD_SZ (PCM_PERIOD_SZ_MIN * AUDIO_HW_IN_PERIOD_MULT)
+#define AUDIO_HW_IN_PERIOD_CNT 2
+// Default audio input buffer size in bytes (8kHz mono)
+#define AUDIO_HW_IN_PERIOD_BYTES ((AUDIO_HW_IN_PERIOD_SZ*sizeof(int16_t))/8)
+
+#define INPUT_SOURCE_KEY "Input Source"
+
+class AudioHardware : public AudioHardwareBase {
+    class AudioStreamOutALSA;
+    class AudioStreamInALSA;
+
+public:
+
+    AudioHardware();
+    virtual ~AudioHardware();
+    virtual status_t initCheck();
+
+    virtual status_t setVoiceVolume(float volume);
+    virtual status_t setMasterVolume(float volume);
+
+    virtual status_t setMode(int mode);
+
+    virtual status_t setMicMute(bool state);
+    virtual status_t getMicMute(bool* state);
+
+    virtual status_t setParameters(const String8& keyValuePairs);
+    virtual String8 getParameters(const String8& keys);
+
+    virtual AudioStreamOut* openOutputStream(
+        uint32_t devices, int *format=0, uint32_t *channels=0,
+        uint32_t *sampleRate=0, status_t *status=0);
+
+    virtual AudioStreamIn* openInputStream(
+        uint32_t devices, int *format, uint32_t *channels,
+        uint32_t *sampleRate, status_t *status,
+        AudioSystem::audio_in_acoustics acoustics);
+
+    virtual void closeOutputStream(AudioStreamOut* out);
+    virtual void closeInputStream(AudioStreamIn* in);
+
+    virtual size_t getInputBufferSize(
+        uint32_t sampleRate, int format, int channelCount);
+
+    int  mode() { return mMode; }
+    const char *getOutputRouteFromDevice(uint32_t device);
+    const char *getInputRouteFromDevice(uint32_t device);
+    const char *getVoiceRouteFromDevice(uint32_t device);
+
+    status_t setIncallPath_l(uint32_t device);
+
+    status_t setInputSource_l(String8 source);
+
+    static uint32_t getInputSampleRate(uint32_t sampleRate);
+    sp <AudioStreamInALSA> getActiveInput_l();
+
+    Mutex& lock() { return mLock; }
+
+    struct pcm *openPcmOut_l();
+    void closePcmOut_l();
+
+    struct mixer *openMixer_l();
+    void closeMixer_l();
+
+    sp <AudioStreamOutALSA> output() { return mOutput; }
+
+protected:
+    virtual status_t dump(int fd, const Vector<String16>& args);
+
+private:
+
+    bool            mInit;
+    bool            mMicMute;
+    sp <AudioStreamOutALSA>                mOutput;
+    SortedVector < sp<AudioStreamInALSA> > mInputs;
+    Mutex           mLock;
+    struct pcm*     mPcm;
+    struct mixer*   mMixer;
+    uint32_t        mPcmOpenCnt;
+    uint32_t        mMixerOpenCnt;
+    bool            mInCallAudioMode;
+
+    String8         mInputSource;
+    bool            mBluetoothNrec;
+    bool            mActivatedCP;
+    //  trace driver operations for dump
+    int             mDriverOp;
+
+    static uint32_t         checkInputSampleRate(uint32_t sampleRate);
+    static const uint32_t   inputSamplingRates[];
+
+    class AudioStreamOutALSA : public AudioStreamOut, public RefBase {
+    public:
+        AudioStreamOutALSA();
+        virtual ~AudioStreamOutALSA();
+        status_t set(AudioHardware* mHardware,
+                     uint32_t devices,
+                     int *pFormat,
+                     uint32_t *pChannels,
+                     uint32_t *pRate);
+        virtual uint32_t sampleRate() const { return mSampleRate; }
+        virtual size_t bufferSize() const { return mBufferSize; }
+        virtual uint32_t channels() const { return mChannels; }
+        virtual int format() const { return AUDIO_HW_OUT_FORMAT; }
+        virtual uint32_t latency() const
+        {
+            return (1000 * AUDIO_HW_OUT_PERIOD_CNT *
+                   (bufferSize()/frameSize()))/sampleRate() + AUDIO_HW_OUT_LATENCY_MS;
+        }
+        virtual status_t setVolume(float left, float right) { return INVALID_OPERATION; }
+        virtual ssize_t write(const void* buffer, size_t bytes);
+        virtual status_t standby();
+        bool checkStandby();
+
+        virtual status_t dump(int fd, const Vector<String16>& args);
+        virtual status_t setParameters(const String8& keyValuePairs);
+        virtual String8 getParameters(const String8& keys);
+        uint32_t device() { return mDevices; }
+        virtual status_t getRenderPosition(uint32_t *dspFrames);
+
+        void doStandby_l();
+        void close_l();
+        status_t open_l();
+        int standbyCnt() { return mStandbyCnt; }
+
+        void lock() { mLock.lock(); }
+        void unlock() { mLock.unlock(); }
+
+    private:
+        Mutex mLock;
+        AudioHardware *mHardware;
+        struct pcm *mPcm;
+        struct mixer *mMixer;
+        struct mixer_ctl *mRouteCtl;
+        const char *next_route;
+        bool mStandby;
+        uint32_t mDevices;
+        uint32_t mChannels;
+        uint32_t mSampleRate;
+        size_t mBufferSize;
+        //  trace driver operations for dump
+        int mDriverOp;
+        int mStandbyCnt;
+    };
+
+    class BufferProvider {
+    public:
+
+        struct Buffer {
+            union {
+                void*       raw;
+                short*      i16;
+                int8_t*     i8;
+            };
+            size_t frameCount;
+        };
+
+        virtual ~BufferProvider() {}
+
+        virtual status_t getNextBuffer(Buffer* buffer) = 0;
+        virtual void releaseBuffer(Buffer* buffer) = 0;
+    };
+
+    class DownSampler {
+    public:
+        DownSampler(uint32_t outSampleRate,
+                    uint32_t channelCount,
+                    uint32_t frameCount,
+                    BufferProvider *provider);
+
+        virtual ~DownSampler();
+
+        void reset();
+        status_t initCheck() { return mStatus; }
+        int resample(int16_t *out, size_t *outFrameCount);
+
+    private:
+        status_t mStatus;
+        BufferProvider *mProvider;
+        uint32_t mSampleRate;
+        uint32_t mChannelCount;
+        uint32_t mFrameCount;
+        int16_t *mInLeft;
+        int16_t *mInRight;
+        int16_t *mTmpLeft;
+        int16_t *mTmpRight;
+        int16_t *mTmp2Left;
+        int16_t *mTmp2Right;
+        int16_t *mOutLeft;
+        int16_t *mOutRight;
+        int mInInBuf;
+        int mInTmpBuf;
+        int mInTmp2Buf;
+        int mOutBufPos;
+        int mInOutBuf;
+    };
+
+
+    class AudioStreamInALSA : public AudioStreamIn, public BufferProvider, public RefBase {
+    public:
+        AudioStreamInALSA();
+        virtual ~AudioStreamInALSA();
+        status_t set(AudioHardware* hw,
+                    uint32_t devices,
+                    int *pFormat,
+                    uint32_t *pChannels,
+                    uint32_t *pRate,
+                    AudioSystem::audio_in_acoustics acoustics);
+        virtual size_t bufferSize() const { return mBufferSize; }
+        virtual uint32_t channels() const { return mChannels; }
+        virtual int format() const { return AUDIO_HW_IN_FORMAT; }
+        virtual uint32_t sampleRate() const { return mSampleRate; }
+        virtual status_t setGain(float gain) { return INVALID_OPERATION; }
+        virtual ssize_t read(void* buffer, ssize_t bytes);
+        virtual status_t dump(int fd, const Vector<String16>& args);
+        virtual status_t standby();
+        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) { return NO_ERROR; }
+        virtual status_t removeAudioEffect(effect_handle_t effect) { return NO_ERROR; }
+
+        bool checkStandby();
+        uint32_t device() { return mDevices; }
+        void doStandby_l();
+        void close_l();
+        status_t open_l();
+        int standbyCnt() { return mStandbyCnt; }
+
+        static size_t getBufferSize(uint32_t sampleRate, int channelCount);
+
+        // BufferProvider
+        virtual status_t getNextBuffer(BufferProvider::Buffer* buffer);
+        virtual void releaseBuffer(BufferProvider::Buffer* buffer);
+
+        void lock() { mLock.lock(); }
+        void unlock() { mLock.unlock(); }
+
+    private:
+        Mutex mLock;
+        AudioHardware *mHardware;
+        struct pcm *mPcm;
+        struct mixer *mMixer;
+        struct mixer_ctl *mRouteCtl;
+        const char *next_route;
+        bool mStandby;
+        uint32_t mDevices;
+        uint32_t mChannels;
+        uint32_t mChannelCount;
+        uint32_t mSampleRate;
+        size_t mBufferSize;
+        DownSampler *mDownSampler;
+        status_t mReadStatus;
+        size_t mInPcmInBuf;
+        int16_t *mPcmIn;
+        //  trace driver operations for dump
+        int mDriverOp;
+        int mStandbyCnt;
+    };
+
+};
+
+}; // namespace android
+
+#endif

--- /dev/null
+++ b/alsa.mk
@@ -0,0 +1,53 @@
+#
+# Copyright (C) 2011 The Android-x86 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
+#
+
+TARGET_ALSA_CONF_DIR := system/usr/share/alsa
+LOCAL_ALSA_CONF_DIR  := external/alsa-lib/src/conf
+
+copy_conf := \
+	alsa.conf \
+	pcm/dsnoop.conf \
+	pcm/modem.conf \
+	pcm/dpl.conf \
+	pcm/default.conf \
+	pcm/surround51.conf \
+	pcm/surround41.conf \
+	pcm/surround50.conf \
+	pcm/dmix.conf \
+	pcm/center_lfe.conf \
+	pcm/surround40.conf \
+	pcm/side.conf \
+	pcm/iec958.conf \
+	pcm/rear.conf \
+	pcm/surround71.conf \
+	pcm/front.conf \
+	cards/aliases.conf
+
+LOCAL_ALSA_INIT_DIR  := external/alsa-utils/alsactl/init
+
+copy_init := \
+	00main \
+	default \
+	hda \
+	help \
+	info \
+	test
+
+PRODUCT_COPY_FILES := \
+	$(foreach f,$(copy_conf),$(LOCAL_ALSA_CONF_DIR)/$(f):$(TARGET_ALSA_CONF_DIR)/$(f)) \
+	$(foreach f,$(copy_init),$(LOCAL_ALSA_INIT_DIR)/$(f):$(TARGET_ALSA_CONF_DIR)/init/$(f))
+
+PRODUCT_PACKAGES := \
+	audio.primary.$(TARGET_PRODUCT) \
+	audio_policy.$(TARGET_PRODUCT) \
+	audio.primary.default \
+	audio.a2dp.default \
+	audio_policy.default \
+	alsa_ctl \

--- /dev/null
+++ b/alsa_audio.h
@@ -0,0 +1,85 @@
+/*
+** 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_
+
+#if defined(__cplusplus)
+extern "C" {
+#endif
+
+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);
+
+#if defined(__cplusplus)
+}  /* extern "C" */
+#endif
+
+#endif

--- /dev/null
+++ b/alsa_mixer.c
@@ -0,0 +1,371 @@
+/*
+** Copyright 2010, The Android Open-Source Project
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+**     http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+*/
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <errno.h>
+#include <ctype.h>
+
+#include <linux/ioctl.h>
+#define __force
+#define __bitwise
+#define __user
+#include "asound.h"
+
+#include "alsa_audio.h"
+
+static const char *elem_iface_name(snd_ctl_elem_iface_t n)
+{
+    switch (n) {
+    case SNDRV_CTL_ELEM_IFACE_CARD: return "CARD";
+    case SNDRV_CTL_ELEM_IFACE_HWDEP: return "HWDEP";
+    case SNDRV_CTL_ELEM_IFACE_MIXER: return "MIXER";
+    case SNDRV_CTL_ELEM_IFACE_PCM: return "PCM";
+    case SNDRV_CTL_ELEM_IFACE_RAWMIDI: return "MIDI";
+    case SNDRV_CTL_ELEM_IFACE_TIMER: return "TIMER";
+    case SNDRV_CTL_ELEM_IFACE_SEQUENCER: return "SEQ";
+    default: return "???";
+    }
+}
+
+static const char *elem_type_name(snd_ctl_elem_type_t n)
+{
+    switch (n) {
+    case SNDRV_CTL_ELEM_TYPE_NONE: return "NONE";
+    case SNDRV_CTL_ELEM_TYPE_BOOLEAN: return "BOOL";
+    case SNDRV_CTL_ELEM_TYPE_INTEGER: return "INT32";
+    case SNDRV_CTL_ELEM_TYPE_ENUMERATED: return "ENUM";
+    case SNDRV_CTL_ELEM_TYPE_BYTES: return "BYTES";
+    case SNDRV_CTL_ELEM_TYPE_IEC958: return "IEC958";
+    case SNDRV_CTL_ELEM_TYPE_INTEGER64: return "INT64";
+    default: return "???";
+    }
+}
+
+
+struct mixer_ctl {
+    struct mixer *mixer;
+    struct snd_ctl_elem_info *info;
+    char **ename;
+};
+
+struct mixer {
+    int fd;
+    struct snd_ctl_elem_info *info;
+    struct mixer_ctl *ctl;
+    unsigned count;
+};
+
+void mixer_close(struct mixer *mixer)
+{
+    unsigned n,m;
+
+    if (mixer->fd >= 0)
+        close(mixer->fd);
+
+    if (mixer->ctl) {
+        for (n = 0; n < mixer->count; n++) {
+            if (mixer->ctl[n].ename) {
+                unsigned max = mixer->ctl[n].info->value.enumerated.items;
+                for (m = 0; m < max; m++)
+                    free(mixer->ctl[n].ename[m]);
+                free(mixer->ctl[n].ename);
+            }
+        }
+        free(mixer->ctl);
+    }
+
+    if (mixer->info)
+        free(mixer->info);
+
+    free(mixer);
+}
+
+struct mixer *mixer_open(void)
+{
+    struct snd_ctl_elem_list elist;
+    struct snd_ctl_elem_info tmp;
+    struct snd_ctl_elem_id *eid = NULL;
+    struct mixer *mixer = NULL;
+    unsigned n, m;
+    int fd;
+
+    fd = open("/dev/snd/controlC0", O_RDWR);
+    if (fd < 0)
+        return 0;
+
+    memset(&elist, 0, sizeof(elist));
+    if (ioctl(fd, SNDRV_CTL_IOCTL_ELEM_LIST, &elist) < 0)
+        goto fail;
+
+    mixer = calloc(1, sizeof(*mixer));
+    if (!mixer)
+        goto fail;
+
+    mixer->ctl = calloc(elist.count, sizeof(struct mixer_ctl));
+    mixer->info = calloc(elist.count, sizeof(struct snd_ctl_elem_info));
+    if (!mixer->ctl || !mixer->info)
+        goto fail;
+
+    eid = calloc(elist.count, sizeof(struct snd_ctl_elem_id));
+    if (!eid)
+        goto fail;
+
+    mixer->count = elist.count;
+    mixer->fd = fd;
+    elist.space = mixer->count;
+    elist.pids = eid;
+    if (ioctl(fd, SNDRV_CTL_IOCTL_ELEM_LIST, &elist) < 0)
+        goto fail;
+
+    for (n = 0; n < mixer->count; n++) {
+        struct snd_ctl_elem_info *ei = mixer->info + n;
+        ei->id.numid = eid[n].numid;
+        if (ioctl(fd, SNDRV_CTL_IOCTL_ELEM_INFO, ei) < 0)
+            goto fail;
+        mixer->ctl[n].info = ei;
+        mixer->ctl[n].mixer = mixer;
+        if (ei->type == SNDRV_CTL_ELEM_TYPE_ENUMERATED) {
+            char **enames = calloc(ei->value.enumerated.items, sizeof(char*));
+            if (!enames)
+                goto fail;
+            mixer->ctl[n].ename = enames;
+            for (m = 0; m < ei->value.enumerated.items; m++) {
+                memset(&tmp, 0, sizeof(tmp));
+                tmp.id.numid = ei->id.numid;
+                tmp.value.enumerated.item = m;
+                if (ioctl(fd, SNDRV_CTL_IOCTL_ELEM_INFO, &tmp) < 0)
+                    goto fail;
+                enames[m] = strdup(tmp.value.enumerated.name);
+                if (!enames[m])
+                    goto fail;
+            }
+        }
+    }
+
+    free(eid);
+    return mixer;
+
+fail:
+    if (eid)
+        free(eid);
+    if (mixer)
+        mixer_close(mixer);
+    else if (fd >= 0)
+        close(fd);
+    return 0;
+}
+
+void mixer_dump(struct mixer *mixer)
+{
+    unsigned n, m;
+
+    printf("  id iface dev sub idx num perms     type   name\n");
+    for (n = 0; n < mixer->count; n++) {
+        struct snd_ctl_elem_info *ei = mixer->info + n;
+
+        printf("%4d %5s %3d %3d %3d %3d %c%c%c%c%c%c%c%c%c %-6s %s",
+               ei->id.numid, elem_iface_name(ei->id.iface),
+               ei->id.device, ei->id.subdevice, ei->id.index,
+               ei->count,
+               (ei->access & SNDRV_CTL_ELEM_ACCESS_READ) ? 'r' : ' ',
+               (ei->access & SNDRV_CTL_ELEM_ACCESS_WRITE) ? 'w' : ' ',
+               (ei->access & SNDRV_CTL_ELEM_ACCESS_VOLATILE) ? 'V' : ' ',
+               (ei->access & SNDRV_CTL_ELEM_ACCESS_TIMESTAMP) ? 'T' : ' ',
+               (ei->access & SNDRV_CTL_ELEM_ACCESS_TLV_READ) ? 'R' : ' ',
+               (ei->access & SNDRV_CTL_ELEM_ACCESS_TLV_WRITE) ? 'W' : ' ',
+               (ei->access & SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND) ? 'C' : ' ',
+               (ei->access & SNDRV_CTL_ELEM_ACCESS_INACTIVE) ? 'I' : ' ',
+               (ei->access & SNDRV_CTL_ELEM_ACCESS_LOCK) ? 'L' : ' ',
+               elem_type_name(ei->type),
+               ei->id.name);
+        switch (ei->type) {
+        case SNDRV_CTL_ELEM_TYPE_INTEGER:
+            printf(ei->value.integer.step ?
+                   " { %ld-%ld, %ld }\n" : " { %ld-%ld }",
+                   ei->value.integer.min,
+                   ei->value.integer.max,
+                   ei->value.integer.step);
+            break;
+        case SNDRV_CTL_ELEM_TYPE_INTEGER64:
+            printf(ei->value.integer64.step ?
+                   " { %lld-%lld, %lld }\n" : " { %lld-%lld }",
+                   ei->value.integer64.min,
+                   ei->value.integer64.max,
+                   ei->value.integer64.step);
+            break;
+        case SNDRV_CTL_ELEM_TYPE_ENUMERATED: {
+            unsigned m;
+            printf(" { %s=0", mixer->ctl[n].ename[0]);
+            for (m = 1; m < ei->value.enumerated.items; m++)
+                printf(", %s=%d", mixer->ctl[n].ename[m],m);
+            printf(" }");
+            break;
+        }
+        }
+        printf("\n");
+    }
+}
+
+struct mixer_ctl *mixer_get_control(struct mixer *mixer,
+                                    const char *name, unsigned index)
+{
+    unsigned n;
+    for (n = 0; n < mixer->count; n++) {
+        if (mixer->info[n].id.index == index) {
+            if (!strcmp(name, (char*) mixer->info[n].id.name)) {
+                return mixer->ctl + n;
+            }
+        }
+    }
+    return 0;
+}
+
+struct mixer_ctl *mixer_get_nth_control(struct mixer *mixer, unsigned n)
+{
+    if (n < mixer->count)
+        return mixer->ctl + n;
+    return 0;
+}
+
+void mixer_ctl_print(struct mixer_ctl *ctl)
+{
+    struct snd_ctl_elem_value ev;
+    unsigned n;
+
+    memset(&ev, 0, sizeof(ev));
+    ev.id.numid = ctl->info->id.numid;
+    if (ioctl(ctl->mixer->fd, SNDRV_CTL_IOCTL_ELEM_READ, &ev))
+        return;
+    printf("%s:", ctl->info->id.name);
+
+    switch (ctl->info->type) {
+    case SNDRV_CTL_ELEM_TYPE_BOOLEAN:
+        for (n = 0; n < ctl->info->count; n++)
+            printf(" %s", ev.value.integer.value[n] ? "ON" : "OFF");
+        break;
+    case SNDRV_CTL_ELEM_TYPE_INTEGER: {
+        for (n = 0; n < ctl->info->count; n++)
+            printf(" %ld", ev.value.integer.value[n]);
+        break;
+    }
+    case SNDRV_CTL_ELEM_TYPE_INTEGER64:
+        for (n = 0; n < ctl->info->count; n++)
+            printf(" %lld", ev.value.integer64.value[n]);
+        break;
+    case SNDRV_CTL_ELEM_TYPE_ENUMERATED:
+        for (n = 0; n < ctl->info->count; n++) {
+            unsigned v = ev.value.enumerated.item[n];
+            printf(" %d (%s)", v,
+                   (v < ctl->info->value.enumerated.items) ? ctl->ename[v] : "???");
+        }
+        break;
+    default:
+        printf(" ???");
+    }
+    printf("\n");
+}
+
+static long scale_int(struct snd_ctl_elem_info *ei, unsigned _percent)
+{
+    long percent;
+    long range;
+
+    if (_percent > 100)
+        percent = 100;
+    else
+        percent = (long) _percent;
+
+    range = (ei->value.integer.max - ei->value.integer.min);
+
+    return ei->value.integer.min + (range * percent) / 100LL;
+}
+
+static long long scale_int64(struct snd_ctl_elem_info *ei, unsigned _percent)
+{
+    long long percent;
+    long long range;
+
+    if (_percent > 100)
+        percent = 100;
+    else
+        percent = (long) _percent;
+
+    range = (ei->value.integer.max - ei->value.integer.min) * 100LL;
+
+    return ei->value.integer.min + (range / percent);
+}
+
+int mixer_ctl_set(struct mixer_ctl *ctl, unsigned percent)
+{
+    struct snd_ctl_elem_value ev;
+    unsigned n;
+
+    memset(&ev, 0, sizeof(ev));
+    ev.id.numid = ctl->info->id.numid;
+    switch (ctl->info->type) {
+    case SNDRV_CTL_ELEM_TYPE_BOOLEAN:
+        for (n = 0; n < ctl->info->count; n++)
+            ev.value.integer.value[n] = !!percent;
+        break;
+    case SNDRV_CTL_ELEM_TYPE_INTEGER: {
+        long value = scale_int(ctl->info, percent);
+        for (n = 0; n < ctl->info->count; n++)
+            ev.value.integer.value[n] = value;
+        break;
+    }
+    case SNDRV_CTL_ELEM_TYPE_INTEGER64: {
+        long long value = scale_int64(ctl->info, percent);
+        for (n = 0; n < ctl->info->count; n++)
+            ev.value.integer64.value[n] = value;
+        break;
+    }
+    default:
+        errno = EINVAL;
+        return -1;
+    }
+
+    return ioctl(ctl->mixer->fd, SNDRV_CTL_IOCTL_ELEM_WRITE, &ev);
+}
+
+int mixer_ctl_select(struct mixer_ctl *ctl, const char *value)
+{
+    unsigned n, max;
+    struct snd_ctl_elem_value ev;
+
+    if (ctl->info->type != SNDRV_CTL_ELEM_TYPE_ENUMERATED) {
+        errno = EINVAL;
+        return -1;
+    }
+
+    max = ctl->info->value.enumerated.items;
+    for (n = 0; n < max; n++) {
+        if (!strcmp(value, ctl->ename[n])) {
+            memset(&ev, 0, sizeof(ev));
+            ev.value.enumerated.item[0] = n;
+            ev.id.numid = ctl->info->id.numid;
+            if (ioctl(ctl->mixer->fd, SNDRV_CTL_IOCTL_ELEM_WRITE, &ev) < 0)
+                return -1;
+            return 0;
+        }
+    }
+
+    errno = EINVAL;
+    return -1;
+}

--- /dev/null
+++ b/alsa_pcm.c
@@ -0,0 +1,404 @@
+/*
+** 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)
+{
+    int i;
+    char dname[32];
+    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;
+
+    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;
+    for (i = 0; i < 3; ++i) {
+        sprintf(dname, "/dev/snd/pcmC%dD0%c", i, (flags & PCM_IN) ? 'c' : 'p');
+        pcm->fd = open(dname, O_RDWR);
+        if (pcm->fd >= 0)
+            break;
+        oops(pcm, errno, "cannot open device '%s'", dname);
+    }
+    if (i == 3)
+        return pcm;
+
+    if (ioctl(pcm->fd, SNDRV_PCM_IOCTL_INFO, &info)) {
+        oops(pcm, errno, "cannot get info - %s", dname);
+        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;
+}

--- /dev/null
+++ b/asound.h
@@ -0,0 +1,813 @@
+/****************************************************************************
+ ****************************************************************************
+ ***
+ ***   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