Eclair ALSA version including new ALSA module mechanism.

12 files changed, 4370 insertions, 1745 deletions

diff --git a/ALSAControl.cpp b/ALSAControl.cpp
new file mode 100644
index 0000000..65fd86f
--- /dev/null
+++ b/ALSAControl.cpp
@@ -0,0 +1,217 @@
+/* ALSAControl.cpp
+ **
+ ** Copyright 2008-2009 Wind River Systems
+ **
+ ** Licensed under the Apache License, Version 2.0 (the "License");
+ ** you may not use this file except in compliance with the License.
+ ** You may obtain a copy of the License at
+ **
+ **     http://www.apache.org/licenses/LICENSE-2.0
+ **
+ ** Unless required by applicable law or agreed to in writing, software
+ ** distributed under the License is distributed on an "AS IS" BASIS,
+ ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ ** See the License for the specific language governing permissions and
+ ** limitations under the License.
+ */
+
+#include <errno.h>
+#include <stdarg.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <dlfcn.h>
+
+#define LOG_TAG "ALSAControl"
+#include <utils/Log.h>
+#include <utils/String8.h>
+
+#include <cutils/properties.h>
+#include <media/AudioRecord.h>
+#include <hardware_legacy/power.h>
+
+#include "AudioHardwareALSA.h"
+
+namespace android
+{
+
+ALSAControl::ALSAControl(const char *device)
+{
+    snd_ctl_open(&mHandle, device, 0);
+}
+
+ALSAControl::~ALSAControl()
+{
+    if (mHandle) snd_ctl_close(mHandle);
+}
+
+status_t ALSAControl::get(const char *name, unsigned int &value, int index)
+{
+    if (!mHandle) {
+        LOGE("Control not initialized");
+        return NO_INIT;
+    }
+
+    snd_ctl_elem_id_t *id;
+    snd_ctl_elem_info_t *info;
+    snd_ctl_elem_value_t *control;
+
+    snd_ctl_elem_id_alloca(&id);
+    snd_ctl_elem_info_alloca(&info);
+    snd_ctl_elem_value_alloca(&control);
+
+    snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER);
+    snd_ctl_elem_id_set_name(id, name);
+    snd_ctl_elem_info_set_id(info, id);
+
+    int ret = snd_ctl_elem_info(mHandle, info);
+    if (ret < 0) {
+        LOGE("Control '%s' cannot get element info: %d", name, ret);
+        return BAD_VALUE;
+    }
+
+    int count = snd_ctl_elem_info_get_count(info);
+    if (index >= count) {
+        LOGE("Control '%s' index is out of range (%d >= %d)", name, index, count);
+        return BAD_VALUE;
+    }
+
+    snd_ctl_elem_info_get_id(info, id);
+    snd_ctl_elem_value_set_id(control, id);
+
+    ret = snd_ctl_elem_read(mHandle, control);
+    if (ret < 0) {
+        LOGE("Control '%s' cannot read element value: %d", name, ret);
+        return BAD_VALUE;
+    }
+
+    snd_ctl_elem_type_t type = snd_ctl_elem_info_get_type(info);
+    switch (type) {
+        case SND_CTL_ELEM_TYPE_BOOLEAN:
+            value = snd_ctl_elem_value_get_boolean(control, index);
+            break;
+        case SND_CTL_ELEM_TYPE_INTEGER:
+            value = snd_ctl_elem_value_get_integer(control, index);
+            break;
+        case SND_CTL_ELEM_TYPE_INTEGER64:
+            value = snd_ctl_elem_value_get_integer64(control, index);
+            break;
+        case SND_CTL_ELEM_TYPE_ENUMERATED:
+            value = snd_ctl_elem_value_get_enumerated(control, index);
+            break;
+        case SND_CTL_ELEM_TYPE_BYTES:
+            value = snd_ctl_elem_value_get_byte(control, index);
+            break;
+        default:
+            return BAD_VALUE;
+    }
+
+    return NO_ERROR;
+}
+
+status_t ALSAControl::set(const char *name, unsigned int value, int index)
+{
+    if (!mHandle) {
+        LOGE("Control not initialized");
+        return NO_INIT;
+    }
+
+    snd_ctl_elem_id_t *id;
+    snd_ctl_elem_info_t *info;
+
+    snd_ctl_elem_id_alloca(&id);
+    snd_ctl_elem_info_alloca(&info);
+
+    snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER);
+    snd_ctl_elem_id_set_name(id, name);
+    snd_ctl_elem_info_set_id(info, id);
+
+    int ret = snd_ctl_elem_info(mHandle, info);
+    if (ret < 0) {
+        LOGE("Control '%s' cannot get element info: %d", name, ret);
+        return BAD_VALUE;
+    }
+
+    int count = snd_ctl_elem_info_get_count(info);
+    if (index >= count) {
+        LOGE("Control '%s' index is out of range (%d >= %d)", name, index, count);
+        return BAD_VALUE;
+    }
+
+    if (index == -1)
+        index = 0; // Range over all of them
+    else
+        count = index + 1; // Just do the one specified
+
+    snd_ctl_elem_type_t type = snd_ctl_elem_info_get_type(info);
+
+    snd_ctl_elem_value_t *control;
+    snd_ctl_elem_value_alloca(&control);
+
+    snd_ctl_elem_info_get_id(info, id);
+    snd_ctl_elem_value_set_id(control, id);
+
+    for (int i = index; i < count; i++)
+        switch (type) {
+            case SND_CTL_ELEM_TYPE_BOOLEAN:
+                snd_ctl_elem_value_set_boolean(control, i, value);
+                break;
+            case SND_CTL_ELEM_TYPE_INTEGER:
+                snd_ctl_elem_value_set_integer(control, i, value);
+                break;
+            case SND_CTL_ELEM_TYPE_INTEGER64:
+                snd_ctl_elem_value_set_integer64(control, i, value);
+                break;
+            case SND_CTL_ELEM_TYPE_ENUMERATED:
+                snd_ctl_elem_value_set_enumerated(control, i, value);
+                break;
+            case SND_CTL_ELEM_TYPE_BYTES:
+                snd_ctl_elem_value_set_byte(control, i, value);
+                break;
+            default:
+                break;
+        }
+
+    ret = snd_ctl_elem_write(mHandle, control);
+    return (ret < 0) ? BAD_VALUE : NO_ERROR;
+}
+
+status_t ALSAControl::set(const char *name, const char *value)
+{
+    if (!mHandle) {
+        LOGE("Control not initialized");
+        return NO_INIT;
+    }
+
+    snd_ctl_elem_id_t *id;
+    snd_ctl_elem_info_t *info;
+
+    snd_ctl_elem_id_alloca(&id);
+    snd_ctl_elem_info_alloca(&info);
+
+    snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER);
+    snd_ctl_elem_id_set_name(id, name);
+    snd_ctl_elem_info_set_id(info, id);
+
+    int ret = snd_ctl_elem_info(mHandle, info);
+    if (ret < 0) {
+        LOGE("Control '%s' cannot get element info: %d", name, ret);
+        return BAD_VALUE;
+    }
+
+    int items = snd_ctl_elem_info_get_items(info);
+    for (int i = 0; i < items; i++) {
+        snd_ctl_elem_info_set_item(info, i);
+        ret = snd_ctl_elem_info(mHandle, info);
+        if (ret < 0) continue;
+        if (strcmp(value, snd_ctl_elem_info_get_item_name(info)) == 0)
+            return set(name, i, -1);
+    }
+
+    LOGE("Control '%s' has no enumerated value of '%s'", name, value);
+
+    return BAD_VALUE;
+}
+
+};        // namespace android
diff --git a/ALSAMixer.cpp b/ALSAMixer.cpp
new file mode 100644
index 0000000..883edf1
--- /dev/null
+++ b/ALSAMixer.cpp
@@ -0,0 +1,432 @@
+/* ALSAMixer.cpp
+ **
+ ** Copyright 2008-2009 Wind River Systems
+ **
+ ** Licensed under the Apache License, Version 2.0 (the "License");
+ ** you may not use this file except in compliance with the License.
+ ** You may obtain a copy of the License at
+ **
+ **     http://www.apache.org/licenses/LICENSE-2.0
+ **
+ ** Unless required by applicable law or agreed to in writing, software
+ ** distributed under the License is distributed on an "AS IS" BASIS,
+ ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ ** See the License for the specific language governing permissions and
+ ** limitations under the License.
+ */
+
+#include <errno.h>
+#include <stdarg.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <dlfcn.h>
+
+#define LOG_TAG "AudioHardwareALSA"
+#include <utils/Log.h>
+#include <utils/String8.h>
+
+#include <cutils/properties.h>
+#include <media/AudioRecord.h>
+#include <hardware_legacy/power.h>
+
+#include "AudioHardwareALSA.h"
+
+#define SND_MIXER_VOL_RANGE_MIN  (0)
+#define SND_MIXER_VOL_RANGE_MAX  (100)
+
+#define ALSA_NAME_MAX 128
+
+#define ALSA_STRCAT(x,y) \
+    if (strlen(x) + strlen(y) < ALSA_NAME_MAX) \
+        strcat(x, y);
+
+namespace android
+{
+
+// ----------------------------------------------------------------------------
+
+struct mixer_info_t;
+
+struct alsa_properties_t
+{
+    const AudioSystem::audio_devices device;
+    const char         *propName;
+    const char         *propDefault;
+    mixer_info_t       *mInfo;
+};
+
+#define ALSA_PROP(dev, name, out, in) \
+    {\
+        {dev, "alsa.mixer.playback." name, out, NULL},\
+        {dev, "alsa.mixer.capture." name, in, NULL}\
+    }
+
+static alsa_properties_t
+mixerMasterProp[SND_PCM_STREAM_LAST+1] =
+        ALSA_PROP(AudioSystem::DEVICE_OUT_ALL, "master", "PCM", "Capture");
+
+static alsa_properties_t
+mixerProp[][SND_PCM_STREAM_LAST+1] = {
+    ALSA_PROP(AudioSystem::DEVICE_OUT_EARPIECE, "earpiece", "Earpiece", "Capture"),
+    ALSA_PROP(AudioSystem::DEVICE_OUT_SPEAKER, "speaker", "Speaker",  ""),
+    ALSA_PROP(AudioSystem::DEVICE_OUT_WIRED_HEADSET, "headset", "Headphone", "Capture"),
+    ALSA_PROP(AudioSystem::DEVICE_OUT_BLUETOOTH_SCO, "bluetooth.sco", "Bluetooth", "Bluetooth Capture"),
+    ALSA_PROP(AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP, "bluetooth.a2dp", "Bluetooth A2DP", "Bluetooth A2DP Capture"),
+    ALSA_PROP(AudioSystem::DEVICE_OUT_FM_HEADPHONE, "fm", "FM", ""),
+    ALSA_PROP(static_cast<AudioSystem::audio_devices>(0), "", NULL, NULL)
+};
+
+struct mixer_info_t
+{
+    mixer_info_t() :
+        elem(0),
+        min(SND_MIXER_VOL_RANGE_MIN),
+        max(SND_MIXER_VOL_RANGE_MAX),
+        mute(false)
+    {
+    }
+
+    snd_mixer_elem_t *elem;
+    long              min;
+    long              max;
+    long              volume;
+    bool              mute;
+    char              name[ALSA_NAME_MAX];
+};
+
+static int initMixer (snd_mixer_t **mixer, const char *name)
+{
+    int err;
+
+    if ((err = snd_mixer_open(mixer, 0)) < 0) {
+        LOGE("Unable to open mixer: %s", snd_strerror(err));
+        return err;
+    }
+
+    if ((err = snd_mixer_attach(*mixer, name)) < 0) {
+        LOGW("Unable to attach mixer to device %s: %s",
+            name, snd_strerror(err));
+
+        if ((err = snd_mixer_attach(*mixer, "hw:00")) < 0) {
+            LOGE("Unable to attach mixer to device default: %s",
+                snd_strerror(err));
+
+            snd_mixer_close (*mixer);
+            *mixer = NULL;
+            return err;
+        }
+    }
+
+    if ((err = snd_mixer_selem_register(*mixer, NULL, NULL)) < 0) {
+        LOGE("Unable to register mixer elements: %s", snd_strerror(err));
+        snd_mixer_close (*mixer);
+        *mixer = NULL;
+        return err;
+    }
+
+    // Get the mixer controls from the kernel
+    if ((err = snd_mixer_load(*mixer)) < 0) {
+        LOGE("Unable to load mixer elements: %s", snd_strerror(err));
+        snd_mixer_close (*mixer);
+        *mixer = NULL;
+        return err;
+    }
+
+    return 0;
+}
+
+typedef int (*hasVolume_t)(snd_mixer_elem_t*);
+
+static const hasVolume_t hasVolume[] = {
+    snd_mixer_selem_has_playback_volume,
+    snd_mixer_selem_has_capture_volume
+};
+
+typedef int (*getVolumeRange_t)(snd_mixer_elem_t*, long int*, long int*);
+
+static const getVolumeRange_t getVolumeRange[] = {
+    snd_mixer_selem_get_playback_volume_range,
+    snd_mixer_selem_get_capture_volume_range
+};
+
+typedef int (*setVolume_t)(snd_mixer_elem_t*, long int);
+
+static const setVolume_t setVol[] = {
+    snd_mixer_selem_set_playback_volume_all,
+    snd_mixer_selem_set_capture_volume_all
+};
+
+ALSAMixer::ALSAMixer()
+{
+    int err;
+
+    initMixer (&mMixer[SND_PCM_STREAM_PLAYBACK], "AndroidOut");
+    initMixer (&mMixer[SND_PCM_STREAM_CAPTURE], "AndroidIn");
+
+    snd_mixer_selem_id_t *sid;
+    snd_mixer_selem_id_alloca(&sid);
+
+    for (int i = 0; i <= SND_PCM_STREAM_LAST; i++) {
+
+        mixer_info_t *info = mixerMasterProp[i].mInfo = new mixer_info_t;
+
+        property_get (mixerMasterProp[i].propName,
+                      info->name,
+                      mixerMasterProp[i].propDefault);
+
+        for (snd_mixer_elem_t *elem = snd_mixer_first_elem(mMixer[i]);
+             elem;
+             elem = snd_mixer_elem_next(elem)) {
+
+            if (!snd_mixer_selem_is_active(elem))
+                continue;
+
+            snd_mixer_selem_get_id(elem, sid);
+
+            // Find PCM playback volume control element.
+            const char *elementName = snd_mixer_selem_id_get_name(sid);
+
+            if (info->elem == NULL &&
+                strcmp(elementName, info->name) == 0 &&
+                hasVolume[i] (elem)) {
+
+                info->elem = elem;
+                getVolumeRange[i] (elem, &info->min, &info->max);
+                info->volume = info->max;
+                setVol[i] (elem, info->volume);
+                if (i == SND_PCM_STREAM_PLAYBACK &&
+                    snd_mixer_selem_has_playback_switch (elem))
+                    snd_mixer_selem_set_playback_switch_all (elem, 1);
+                break;
+            }
+        }
+
+        LOGV("Mixer: master '%s' %s.", info->name, info->elem ? "found" : "not found");
+
+        for (int j = 0; mixerProp[j][i].device; j++) {
+
+            mixer_info_t *info = mixerProp[j][i].mInfo = new mixer_info_t;
+
+            property_get (mixerProp[j][i].propName,
+                          info->name,
+                          mixerProp[j][i].propDefault);
+
+            for (snd_mixer_elem_t *elem = snd_mixer_first_elem(mMixer[i]);
+                 elem;
+                 elem = snd_mixer_elem_next(elem)) {
+
+                if (!snd_mixer_selem_is_active(elem))
+                    continue;
+
+                snd_mixer_selem_get_id(elem, sid);
+
+                // Find PCM playback volume control element.
+                const char *elementName = snd_mixer_selem_id_get_name(sid);
+
+               if (info->elem == NULL &&
+                    strcmp(elementName, info->name) == 0 &&
+                    hasVolume[i] (elem)) {
+
+                    info->elem = elem;
+                    getVolumeRange[i] (elem, &info->min, &info->max);
+                    info->volume = info->max;
+                    setVol[i] (elem, info->volume);
+                    if (i == SND_PCM_STREAM_PLAYBACK &&
+                        snd_mixer_selem_has_playback_switch (elem))
+                        snd_mixer_selem_set_playback_switch_all (elem, 1);
+                    break;
+                }
+            }
+            LOGV("Mixer: route '%s' %s.", info->name, info->elem ? "found" : "not found");
+        }
+    }
+    LOGV("mixer initialized.");
+}
+
+ALSAMixer::~ALSAMixer()
+{
+    for (int i = 0; i <= SND_PCM_STREAM_LAST; i++) {
+        if (mMixer[i]) snd_mixer_close (mMixer[i]);
+        if (mixerMasterProp[i].mInfo) {
+            delete mixerMasterProp[i].mInfo;
+            mixerMasterProp[i].mInfo = NULL;
+        }
+        for (int j = 0; mixerProp[j][i].device; j++) {
+            if (mixerProp[j][i].mInfo) {
+                delete mixerProp[j][i].mInfo;
+                mixerProp[j][i].mInfo = NULL;
+            }
+        }
+    }
+    LOGV("mixer destroyed.");
+}
+
+status_t ALSAMixer::setMasterVolume(float volume)
+{
+    mixer_info_t *info = mixerMasterProp[SND_PCM_STREAM_PLAYBACK].mInfo;
+    if (!info || !info->elem) return INVALID_OPERATION;
+
+    long minVol = info->min;
+    long maxVol = info->max;
+
+    // Make sure volume is between bounds.
+    long vol = minVol + volume * (maxVol - minVol);
+    if (vol > maxVol) vol = maxVol;
+    if (vol < minVol) vol = minVol;
+
+    info->volume = vol;
+    snd_mixer_selem_set_playback_volume_all (info->elem, vol);
+
+    return NO_ERROR;
+}
+
+status_t ALSAMixer::setMasterGain(float gain)
+{
+    mixer_info_t *info = mixerMasterProp[SND_PCM_STREAM_CAPTURE].mInfo;
+    if (!info || !info->elem) return INVALID_OPERATION;
+
+    long minVol = info->min;
+    long maxVol = info->max;
+
+    // Make sure volume is between bounds.
+    long vol = minVol + gain * (maxVol - minVol);
+    if (vol > maxVol) vol = maxVol;
+    if (vol < minVol) vol = minVol;
+
+    info->volume = vol;
+    snd_mixer_selem_set_capture_volume_all (info->elem, vol);
+
+    return NO_ERROR;
+}
+
+status_t ALSAMixer::setVolume(uint32_t device, float left, float right)
+{
+    for (int j = 0; mixerProp[j][SND_PCM_STREAM_PLAYBACK].device; j++)
+        if (mixerProp[j][SND_PCM_STREAM_PLAYBACK].device & device) {
+
+            mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_PLAYBACK].mInfo;
+            if (!info || !info->elem) return INVALID_OPERATION;
+
+            long minVol = info->min;
+            long maxVol = info->max;
+
+            // Make sure volume is between bounds.
+            long vol = minVol + left * (maxVol - minVol);
+            if (vol > maxVol) vol = maxVol;
+            if (vol < minVol) vol = minVol;
+
+            info->volume = vol;
+            snd_mixer_selem_set_playback_volume_all (info->elem, vol);
+        }
+
+    return NO_ERROR;
+}
+
+status_t ALSAMixer::setGain(uint32_t device, float gain)
+{
+    for (int j = 0; mixerProp[j][SND_PCM_STREAM_CAPTURE].device; j++)
+        if (mixerProp[j][SND_PCM_STREAM_CAPTURE].device & device) {
+
+            mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_CAPTURE].mInfo;
+            if (!info || !info->elem) return INVALID_OPERATION;
+
+            long minVol = info->min;
+            long maxVol = info->max;
+
+            // Make sure volume is between bounds.
+            long vol = minVol + gain * (maxVol - minVol);
+            if (vol > maxVol) vol = maxVol;
+            if (vol < minVol) vol = minVol;
+
+            info->volume = vol;
+            snd_mixer_selem_set_capture_volume_all (info->elem, vol);
+        }
+
+    return NO_ERROR;
+}
+
+status_t ALSAMixer::setCaptureMuteState(uint32_t device, bool state)
+{
+    for (int j = 0; mixerProp[j][SND_PCM_STREAM_CAPTURE].device; j++)
+        if (mixerProp[j][SND_PCM_STREAM_CAPTURE].device & device) {
+
+            mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_CAPTURE].mInfo;
+            if (!info || !info->elem) return INVALID_OPERATION;
+
+            if (snd_mixer_selem_has_capture_switch (info->elem)) {
+
+                int err = snd_mixer_selem_set_capture_switch_all (info->elem, static_cast<int>(!state));
+                if (err < 0) {
+                    LOGE("Unable to %s capture mixer switch %s",
+                        state ? "enable" : "disable", info->name);
+                    return INVALID_OPERATION;
+                }
+            }
+
+            info->mute = state;
+        }
+
+    return NO_ERROR;
+}
+
+status_t ALSAMixer::getCaptureMuteState(uint32_t device, bool *state)
+{
+    if (!state) return BAD_VALUE;
+
+    for (int j = 0; mixerProp[j][SND_PCM_STREAM_CAPTURE].device; j++)
+        if (mixerProp[j][SND_PCM_STREAM_CAPTURE].device & device) {
+
+            mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_CAPTURE].mInfo;
+            if (!info || !info->elem) return INVALID_OPERATION;
+
+            *state = info->mute;
+            return NO_ERROR;
+        }
+
+    return BAD_VALUE;
+}
+
+status_t ALSAMixer::setPlaybackMuteState(uint32_t device, bool state)
+{
+    for (int j = 0; mixerProp[j][SND_PCM_STREAM_PLAYBACK].device; j++)
+        if (mixerProp[j][SND_PCM_STREAM_PLAYBACK].device & device) {
+
+            mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_PLAYBACK].mInfo;
+            if (!info || !info->elem) return INVALID_OPERATION;
+
+            if (snd_mixer_selem_has_playback_switch (info->elem)) {
+
+                int err = snd_mixer_selem_set_playback_switch_all (info->elem, static_cast<int>(!state));
+                if (err < 0) {
+                    LOGE("Unable to %s playback mixer switch %s",
+                        state ? "enable" : "disable", info->name);
+                    return INVALID_OPERATION;
+                }
+            }
+
+            info->mute = state;
+        }
+
+    return NO_ERROR;
+}
+
+status_t ALSAMixer::getPlaybackMuteState(uint32_t device, bool *state)
+{
+    if (!state) return BAD_VALUE;
+
+    for (int j = 0; mixerProp[j][SND_PCM_STREAM_PLAYBACK].device; j++)
+        if (mixerProp[j][SND_PCM_STREAM_PLAYBACK].device & device) {
+
+            mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_PLAYBACK].mInfo;
+            if (!info || !info->elem) return INVALID_OPERATION;
+
+            *state = info->mute;
+            return NO_ERROR;
+        }
+
+    return BAD_VALUE;
+}
+
+};        // namespace android
diff --git a/ALSAStreamOps.cpp b/ALSAStreamOps.cpp
new file mode 100644
index 0000000..0d14971
--- /dev/null
+++ b/ALSAStreamOps.cpp
@@ -0,0 +1,291 @@
+/* ALSAStreamOps.cpp
+ **
+ ** Copyright 2008-2009 Wind River Systems
+ **
+ ** Licensed under the Apache License, Version 2.0 (the "License");
+ ** you may not use this file except in compliance with the License.
+ ** You may obtain a copy of the License at
+ **
+ **     http://www.apache.org/licenses/LICENSE-2.0
+ **
+ ** Unless required by applicable law or agreed to in writing, software
+ ** distributed under the License is distributed on an "AS IS" BASIS,
+ ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ ** See the License for the specific language governing permissions and
+ ** limitations under the License.
+ */
+
+#include <errno.h>
+#include <stdarg.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <dlfcn.h>
+
+#define LOG_TAG "AudioHardwareALSA"
+#include <utils/Log.h>
+#include <utils/String8.h>
+
+#include <cutils/properties.h>
+#include <media/AudioRecord.h>
+#include <hardware_legacy/power.h>
+
+#include "AudioHardwareALSA.h"
+
+namespace android
+{
+
+// ----------------------------------------------------------------------------
+
+ALSAStreamOps::ALSAStreamOps(AudioHardwareALSA *parent, alsa_handle_t *handle) :
+    mParent(parent),
+    mHandle(handle),
+    mPowerLock(false)
+{
+}
+
+ALSAStreamOps::~ALSAStreamOps()
+{
+    AutoMutex lock(mLock);
+
+    close();
+}
+
+// use emulated popcount optimization
+// http://www.df.lth.se/~john_e/gems/gem002d.html
+static inline uint32_t popCount(uint32_t u)
+{
+    u = ((u&0x55555555) + ((u>>1)&0x55555555));
+    u = ((u&0x33333333) + ((u>>2)&0x33333333));
+    u = ((u&0x0f0f0f0f) + ((u>>4)&0x0f0f0f0f));
+    u = ((u&0x00ff00ff) + ((u>>8)&0x00ff00ff));
+    u = ( u&0x0000ffff) + (u>>16);
+    return u;
+}
+
+acoustic_device_t *ALSAStreamOps::acoustics()
+{
+    return mParent->mAcousticDevice;
+}
+
+ALSAMixer *ALSAStreamOps::mixer()
+{
+    return mParent->mMixer;
+}
+
+status_t ALSAStreamOps::set(int      *format,
+                            uint32_t *channels,
+                            uint32_t *rate)
+{
+    if (channels && *channels != 0) {
+        if (mHandle->channels != popCount(*channels))
+            return BAD_VALUE;
+    } else if (channels) {
+        *channels = 0;
+        if (mHandle->devices & AudioSystem::DEVICE_OUT_ALL)
+            switch(mHandle->channels) {
+                case 4:
+                    *channels |= AudioSystem::CHANNEL_OUT_BACK_LEFT;
+                    *channels |= AudioSystem::CHANNEL_OUT_BACK_RIGHT;
+                    // Fall through...
+                default:
+                case 2:
+                    *channels |= AudioSystem::CHANNEL_OUT_FRONT_RIGHT;
+                    // Fall through...
+                case 1:
+                    *channels |= AudioSystem::CHANNEL_OUT_FRONT_LEFT;
+                    break;
+            }
+        else
+            switch(mHandle->channels) {
+                default:
+                case 2:
+                    *channels |= AudioSystem::CHANNEL_IN_RIGHT;
+                    // Fall through...
+                case 1:
+                    *channels |= AudioSystem::CHANNEL_IN_LEFT;
+                    break;
+            }
+    }
+
+    if (rate && *rate > 0) {
+        if (mHandle->sampleRate != *rate)
+            return BAD_VALUE;
+    } else if (rate)
+        *rate = mHandle->sampleRate;
+
+    snd_pcm_format_t iformat = mHandle->format;
+
+    if (format) {
+        switch(*format) {
+            case AudioSystem::FORMAT_DEFAULT:
+                break;
+
+            case AudioSystem::PCM_16_BIT:
+                iformat = SND_PCM_FORMAT_S16_LE;
+                break;
+
+            case AudioSystem::PCM_8_BIT:
+                iformat = SND_PCM_FORMAT_S8;
+                break;
+
+            default:
+                LOGE("Unknown PCM format %i. Forcing default", *format);
+                break;
+        }
+
+        if (mHandle->format != iformat)
+            return BAD_VALUE;
+
+        switch(iformat) {
+            default:
+            case SND_PCM_FORMAT_S16_LE:
+                *format = AudioSystem::PCM_16_BIT;
+                break;
+            case SND_PCM_FORMAT_S8:
+                *format = AudioSystem::PCM_8_BIT;
+                break;
+        }
+    }
+
+    return NO_ERROR;
+}
+
+status_t ALSAStreamOps::setParameters(const String8& keyValuePairs)
+{
+    AudioParameter param = AudioParameter(keyValuePairs);
+    String8 key = String8(AudioParameter::keyRouting);
+    status_t status = NO_ERROR;
+    int device;
+    LOGV("setParameters() %s", keyValuePairs.string());
+
+    if (param.getInt(key, device) == NO_ERROR) {
+        mParent->mALSADevice->route(mHandle, (uint32_t)device, mParent->mode());
+        param.remove(key);
+    }
+
+    if (param.size()) {
+        status = BAD_VALUE;
+    }
+    return status;
+}
+
+String8 ALSAStreamOps::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)mHandle->curDev);
+    }
+
+    LOGV("getParameters() %s", param.toString().string());
+    return param.toString();
+}
+
+uint32_t ALSAStreamOps::sampleRate() const
+{
+    return mHandle->sampleRate;
+}
+
+//
+// Return the number of bytes (not frames)
+//
+size_t ALSAStreamOps::bufferSize() const
+{
+    snd_pcm_uframes_t bufferSize = mHandle->bufferSize;
+    snd_pcm_uframes_t periodSize;
+
+    snd_pcm_get_params(mHandle->handle, &bufferSize, &periodSize);
+
+    size_t bytes = static_cast<size_t>(snd_pcm_frames_to_bytes(mHandle->handle, bufferSize));
+
+    // Not sure when this happened, but unfortunately it now
+    // appears that the bufferSize must be reported as a
+    // power of 2. This might be for OSS compatibility.
+    for (size_t i = 1; (bytes & ~i) != 0; i<<=1)
+        bytes &= ~i;
+
+    return bytes;
+}
+
+int ALSAStreamOps::format() const
+{
+    int pcmFormatBitWidth;
+    int audioSystemFormat;
+
+    snd_pcm_format_t ALSAFormat = mHandle->format;
+
+    pcmFormatBitWidth = snd_pcm_format_physical_width(ALSAFormat);
+    switch(pcmFormatBitWidth) {
+        case 8:
+            audioSystemFormat = AudioSystem::PCM_8_BIT;
+            break;
+
+        default:
+            LOG_FATAL("Unknown AudioSystem bit width %i!", pcmFormatBitWidth);
+
+        case 16:
+            audioSystemFormat = AudioSystem::PCM_16_BIT;
+            break;
+    }
+
+    return audioSystemFormat;
+}
+
+uint32_t ALSAStreamOps::channels() const
+{
+    unsigned int count = mHandle->channels;
+    uint32_t channels = 0;
+
+    if (mHandle->curDev & AudioSystem::DEVICE_OUT_ALL)
+        switch(count) {
+            case 4:
+                channels |= AudioSystem::CHANNEL_OUT_BACK_LEFT;
+                channels |= AudioSystem::CHANNEL_OUT_BACK_RIGHT;
+                // Fall through...
+            default:
+            case 2:
+                channels |= AudioSystem::CHANNEL_OUT_FRONT_RIGHT;
+                // Fall through...
+            case 1:
+                channels |= AudioSystem::CHANNEL_OUT_FRONT_LEFT;
+                break;
+        }
+    else
+        switch(count) {
+            default:
+            case 2:
+                channels |= AudioSystem::CHANNEL_IN_RIGHT;
+                // Fall through...
+            case 1:
+                channels |= AudioSystem::CHANNEL_IN_LEFT;
+                break;
+        }
+
+    return channels;
+}
+
+void ALSAStreamOps::close()
+{
+    mParent->mALSADevice->close(mHandle);
+}
+
+//
+// Set playback or capture PCM device.  It's possible to support audio output
+// or input from multiple devices by using the ALSA plugins, but this is
+// not supported for simplicity.
+//
+// The AudioHardwareALSA API does not allow one to set the input routing.
+//
+// If the "routes" value does not map to a valid device, the default playback
+// device is used.
+//
+status_t ALSAStreamOps::open(int mode)
+{
+    return mParent->mALSADevice->open(mHandle, mHandle->curDev, mode);
+}
+
+}       // namespace android
diff --git a/Android.mk b/Android.mk
index 9a171dc..5eee4a1 100644
--- a/Android.mk
+++ b/Android.mk
@@ -11,42 +11,88 @@ ifeq ($(strip $(BOARD_USES_ALSA_AUDIO)),true)
 
   LOCAL_ARM_MODE := arm
   LOCAL_CFLAGS := -D_POSIX_SOURCE
-  LOCAL_WHOLE_STATIC_LIBRARIES := libasound
-
-  ifneq ($(ALSA_DEFAULT_SAMPLE_RATE),)
-    LOCAL_CFLAGS += -DALSA_DEFAULT_SAMPLE_RATE=$(ALSA_DEFAULT_SAMPLE_RATE)
-  endif
-
-  ifeq ($(strip $(BOARD_HAVE_FM_ROUTING)),true)
-    LOCAL_CFLAGS += -DFM_ROUTE_SUPPORT
-  endif
 
   LOCAL_C_INCLUDES += external/alsa-lib/include
 
-  LOCAL_SRC_FILES := AudioHardwareALSA.cpp
+  LOCAL_SRC_FILES := \
+	AudioHardwareALSA.cpp \
+	AudioStreamOutALSA.cpp \
+	AudioStreamInALSA.cpp \
+	ALSAStreamOps.cpp \
+	ALSAMixer.cpp \
+	ALSAControl.cpp
 
   LOCAL_MODULE := libaudio
 
   LOCAL_STATIC_LIBRARIES += libaudiointerface
 
   LOCAL_SHARED_LIBRARIES := \
+    libasound \
     libcutils \
     libutils \
     libmedia \
     libhardware \
     libhardware_legacy \
-    libdl \
     libc
 
+ifeq ($(BOARD_HAVE_BLUETOOTH),true)
+  LOCAL_SHARED_LIBRARIES += liba2dp
+endif
+
+  include $(BUILD_SHARED_LIBRARY)
+
+# This is the ALSA audio policy manager
+
+  include $(CLEAR_VARS)
+
+  LOCAL_CFLAGS := -D_POSIX_SOURCE
+
+  LOCAL_SRC_FILES := AudioPolicyManagerALSA.cpp
+
+  LOCAL_MODULE := libaudiopolicy
+
+  LOCAL_SHARED_LIBRARIES := \
+    libcutils \
+    libutils \
+    libmedia
+
   include $(BUILD_SHARED_LIBRARY)
 
+# This is the default ALSA module which behaves closely like the original
+
   include $(CLEAR_VARS)
 
   LOCAL_PRELINK_MODULE := false
 
   LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/hw
 
-  LOCAL_CFLAGS := -D_POSIX_SOURCE
+  LOCAL_CFLAGS := -D_POSIX_SOURCE -Wno-multichar
+
+ifneq ($(ALSA_DEFAULT_SAMPLE_RATE),)
+    LOCAL_CFLAGS += -DALSA_DEFAULT_SAMPLE_RATE=$(ALSA_DEFAULT_SAMPLE_RATE)
+endif
+
+  LOCAL_C_INCLUDES += external/alsa-lib/include
+
+  LOCAL_SRC_FILES:= alsa_default.cpp
+
+  LOCAL_SHARED_LIBRARIES := \
+  	libasound \
+  	liblog
+
+  LOCAL_MODULE:= alsa.default
+
+  include $(BUILD_SHARED_LIBRARY)
+
+# This is the default Acoustics module which is essentially a stub
+
+  include $(CLEAR_VARS)
+
+  LOCAL_PRELINK_MODULE := false
+
+  LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/hw
+
+  LOCAL_CFLAGS := -D_POSIX_SOURCE -Wno-multichar
 
   LOCAL_C_INCLUDES += external/alsa-lib/include
 
diff --git a/AudioHardwareALSA.cpp b/AudioHardwareALSA.cpp
index 684ee63..46dfd4e 100644
--- a/AudioHardwareALSA.cpp
+++ b/AudioHardwareALSA.cpp
@@ -33,61 +33,21 @@
 
 #include "AudioHardwareALSA.h"
 
-#undef DISABLE_HARWARE_RESAMPLING
-
-#ifndef ALSA_DEFAULT_SAMPLE_RATE
-#define ALSA_DEFAULT_SAMPLE_RATE 44100 // in Hz
-#endif
-
-#define SND_MIXER_VOL_RANGE_MIN  (0)
-#define SND_MIXER_VOL_RANGE_MAX  (100)
-
-#define ALSA_NAME_MAX 128
-
-#define ALSA_STRCAT(x,y) \
-    if (strlen(x) + strlen(y) < ALSA_NAME_MAX) \
-        strcat(x, y);
-
 extern "C"
 {
     //
-    // Make sure this prototype is consistent with what's in
-    // external/libasound/alsa-lib-1.0.16/src/pcm/pcm_null.c!
-    //
-    extern int snd_pcm_null_open(snd_pcm_t **pcmp,
-                                 const char *name,
-                                 snd_pcm_stream_t stream,
-                                 int mode);
-
-    //
     // Function for dlsym() to look up for creating a new AudioHardwareInterface.
     //
     android::AudioHardwareInterface *createAudioHardware(void) {
-        return new android::AudioHardwareALSA();
+        return android::AudioHardwareALSA::create();
     }
 }         // extern "C"
 
 namespace android
 {
 
-typedef AudioSystem::audio_routes audio_routes;
-
-#define ROUTE_ALL            AudioSystem::ROUTE_ALL
-#define ROUTE_EARPIECE       AudioSystem::ROUTE_EARPIECE
-#define ROUTE_SPEAKER        AudioSystem::ROUTE_SPEAKER
-#define ROUTE_BLUETOOTH_SCO  AudioSystem::ROUTE_BLUETOOTH_SCO
-#define ROUTE_HEADSET        AudioSystem::ROUTE_HEADSET
-#define ROUTE_BLUETOOTH_A2DP AudioSystem::ROUTE_BLUETOOTH_A2DP
-#ifdef FM_ROUTE_SUPPORT
-#define ROUTE_FM             AudioSystem::ROUTE_FM
-#endif
-
 // ----------------------------------------------------------------------------
 
-static const int DEFAULT_SAMPLE_RATE = ALSA_DEFAULT_SAMPLE_RATE;
-
-static const char _nullALSADeviceName[] = "NULL_Device";
-
 static void ALSAErrorHandler(const char *file,
                              int line,
                              const char *function,
@@ -107,108 +67,57 @@ static void ALSAErrorHandler(const char *file,
     va_end(arg);
 }
 
-// ----------------------------------------------------------------------------
-
-/* The following table(s) need to match in order of the route bits
- */
-static const char *deviceSuffix[] = {
-    /* ROUTE_EARPIECE       */ "_Earpiece",
-    /* ROUTE_SPEAKER        */ "_Speaker",
-    /* ROUTE_BLUETOOTH_SCO  */ "_Bluetooth",
-    /* ROUTE_HEADSET        */ "_Headset",
-    /* ROUTE_BLUETOOTH_A2DP */ "_Bluetooth-A2DP",
-#ifdef FM_ROUTE_SUPPORT
-    /* ROUTE_FM             */ "_FM",
-#endif
-};
-
-static const int deviceSuffixLen = (sizeof(deviceSuffix) / sizeof(char *));
-
-struct mixer_info_t;
-
-struct alsa_properties_t
-{
-    const audio_routes  routes;
-    const char         *propName;
-    const char         *propDefault;
-    mixer_info_t       *mInfo;
-};
-
-static alsa_properties_t
-mixerMasterProp[SND_PCM_STREAM_LAST+1] = {
-    { ROUTE_ALL, "alsa.mixer.playback.master",  "PCM",     NULL},
-    { ROUTE_ALL, "alsa.mixer.capture.master",   "Capture", NULL}
-};
-
-static alsa_properties_t
-mixerProp[][SND_PCM_STREAM_LAST+1] = {
-    {
-        {ROUTE_EARPIECE,       "alsa.mixer.playback.earpiece",       "Earpiece", NULL},
-        {ROUTE_EARPIECE,       "alsa.mixer.capture.earpiece",        "Capture",  NULL}
-    },
-    {
-        {ROUTE_SPEAKER,        "alsa.mixer.playback.speaker",        "Speaker", NULL},
-        {ROUTE_SPEAKER,        "alsa.mixer.capture.speaker",         "",        NULL}
-    },
-    {
-        {ROUTE_BLUETOOTH_SCO,  "alsa.mixer.playback.bluetooth.sco",  "Bluetooth",         NULL},
-        {ROUTE_BLUETOOTH_SCO,  "alsa.mixer.capture.bluetooth.sco",   "Bluetooth Capture", NULL}
-    },
-    {
-        {ROUTE_HEADSET,        "alsa.mixer.playback.headset",        "Headphone", NULL},
-        {ROUTE_HEADSET,        "alsa.mixer.capture.headset",         "Capture",   NULL}
-    },
-    {
-        {ROUTE_BLUETOOTH_A2DP, "alsa.mixer.playback.bluetooth.a2dp", "Bluetooth A2DP",         NULL},
-        {ROUTE_BLUETOOTH_A2DP, "alsa.mixer.capture.bluetooth.a2dp",  "Bluetooth A2DP Capture", NULL}
-    },
-#ifdef FM_ROUTE_SUPPORT
-    {
-        {ROUTE_FM,             "alsa.mixer.playback.fm",             "FM", NULL},
-        {ROUTE_FM,             "alsa.mixer.capture.fm",              "",   NULL}
-    },
-#endif
-    {
-        {static_cast<audio_routes>(0), NULL, NULL, NULL},
-        {static_cast<audio_routes>(0), NULL, NULL, NULL}
-    }
-};
-
-// ----------------------------------------------------------------------------
+AudioHardwareInterface *AudioHardwareALSA::create() {
+    return new AudioHardwareALSA();
+}
 
 AudioHardwareALSA::AudioHardwareALSA() :
-    mOutput(0),
-    mInput(0),
+    mALSADevice(0),
     mAcousticDevice(0)
 {
     snd_lib_error_set_handler(&ALSAErrorHandler);
     mMixer = new ALSAMixer;
 
     hw_module_t *module;
-    int err = hw_get_module(ACOUSTICS_HARDWARE_MODULE_ID,
+    int err = hw_get_module(ALSA_HARDWARE_MODULE_ID,
             (hw_module_t const**)&module);
 
     if (err == 0) {
         hw_device_t* device;
-        err = module->methods->open(module, ACOUSTICS_HARDWARE_NAME, &device);
+        err = module->methods->open(module, ALSA_HARDWARE_NAME, &device);
         if (err == 0) {
+            mALSADevice = (alsa_device_t *)device;
+            mALSADevice->init(mALSADevice, mDeviceList);
+        } else
+            LOGE("ALSA Module could not be opened!!!");
+    } else
+        LOGE("ALSA Module not found!!!");
+
+    err = hw_get_module(ACOUSTICS_HARDWARE_MODULE_ID,
+            (hw_module_t const**)&module);
+
+    if (err == 0) {
+        hw_device_t* device;
+        err = module->methods->open(module, ACOUSTICS_HARDWARE_NAME, &device);
+        if (err == 0)
             mAcousticDevice = (acoustic_device_t *)device;
-        }
+        else
+            LOGE("Acoustics Module not found.");
     }
 }
 
 AudioHardwareALSA::~AudioHardwareALSA()
 {
-    if (mOutput) delete mOutput;
-    if (mInput) delete mInput;
     if (mMixer) delete mMixer;
+    if (mALSADevice)
+        mALSADevice->common.close(&mALSADevice->common);
     if (mAcousticDevice)
         mAcousticDevice->common.close(&mAcousticDevice->common);
 }
 
 status_t AudioHardwareALSA::initCheck()
 {
-    if (mMixer && mMixer->isValid())
+    if (mALSADevice && mMixer && mMixer->isValid())
         return NO_ERROR;
     else
         return NO_INIT;
@@ -218,7 +127,7 @@ status_t AudioHardwareALSA::setVoiceVolume(float volume)
 {
     // The voice volume is used by the VOICE_CALL audio stream.
     if (mMixer)
-        return mMixer->setVolume(ROUTE_EARPIECE, volume);
+        return mMixer->setVolume(AudioSystem::DEVICE_OUT_EARPIECE, volume, volume);
     else
         return INVALID_OPERATION;
 }
@@ -231,1363 +140,128 @@ status_t AudioHardwareALSA::setMasterVolume(float volume)
         return INVALID_OPERATION;
 }
 
-AudioStreamOut *
-AudioHardwareALSA::openOutputStream(int format,
-                                    int channelCount,
-                                    uint32_t sampleRate,
-                                    status_t *status)
-{
-    AutoMutex lock(mLock);
-
-    // only one output stream allowed
-    if (mOutput) {
-        *status = ALREADY_EXISTS;
-        return 0;
-    }
-
-    AudioStreamOutALSA *out = new AudioStreamOutALSA(this);
-
-    *status = out->set(format, channelCount, sampleRate);
-
-    if (*status == NO_ERROR) {
-        mOutput = out;
-        // Some information is expected to be available immediately after
-        // the device is open.
-        *status = mOutput->setDevice(mMode, mRoutes[mMode]);
-    }
-    else {
-        delete out;
-    }
-
-    return mOutput;
-}
-
-AudioStreamIn *
-AudioHardwareALSA::openInputStream(int      inputSource,
-                                   int      format,
-                                   int      channelCount,
-                                   uint32_t sampleRate,
-                                   status_t *status,
-                                   AudioSystem::audio_in_acoustics acoustics)
-{
-    // check for valid input source
-    if ((inputSource < AudioRecord::DEFAULT_INPUT) ||
-        (inputSource >= AudioRecord::NUM_INPUT_SOURCES)) {
-        return 0;
-    }
-
-    AutoMutex lock(mLock);
-
-    // only one input stream allowed
-    if (mInput) {
-        *status = ALREADY_EXISTS;
-        return 0;
-    }
-
-    AudioStreamInALSA *in = new AudioStreamInALSA(this, acoustics);
-
-    *status = in->set(format, channelCount, sampleRate);
-    if (*status == NO_ERROR) {
-        mInput = in;
-        // Some information is expected to be available immediately after
-        // the device is open.
-        *status = mInput->setDevice(mMode, mRoutes[mMode]);
-    }
-    else {
-        delete in;
-    }
-
-    return mInput;
-}
-
-status_t AudioHardwareALSA::doRouting()
-{
-    AutoMutex lock(mLock);
-
-    if (!mOutput)
-        return NO_INIT;
-
-    return mOutput->setDevice(mMode, mRoutes[mMode]);
-}
-
-status_t AudioHardwareALSA::setMicMute(bool state)
-{
-    if (mMixer)
-        return mMixer->setCaptureMuteState(ROUTE_EARPIECE, state);
-
-    return NO_INIT;
-}
-
-status_t AudioHardwareALSA::getMicMute(bool *state)
-{
-    if (mMixer)
-        return mMixer->getCaptureMuteState(ROUTE_EARPIECE, state);
-
-    return NO_ERROR;
-}
-
-status_t AudioHardwareALSA::dump(int fd, const Vector<String16>& args)
-{
-    return NO_ERROR;
-}
-
-// ----------------------------------------------------------------------------
-
-ALSAStreamOps::ALSAStreamOps(AudioHardwareALSA *parent) :
-    mParent(parent),
-    mHandle(0),
-    mHardwareParams(0),
-    mSoftwareParams(0),
-    mMode(-1),
-    mDevice(0),
-    mPowerLock(false)
-{
-    if (snd_pcm_hw_params_malloc(&mHardwareParams) < 0) {
-        LOG_ALWAYS_FATAL("Failed to allocate ALSA hardware parameters!");
-    }
-
-    if (snd_pcm_sw_params_malloc(&mSoftwareParams) < 0) {
-        LOG_ALWAYS_FATAL("Failed to allocate ALSA software parameters!");
-    }
-}
-
-ALSAStreamOps::~ALSAStreamOps()
-{
-    AutoMutex lock(mLock);
-
-    close();
-
-    if (mHardwareParams)
-        snd_pcm_hw_params_free(mHardwareParams);
-
-    if (mSoftwareParams)
-        snd_pcm_sw_params_free(mSoftwareParams);
-}
-
-status_t ALSAStreamOps::set(int      format,
-                            int      channels,
-                            uint32_t rate)
-{
-    if (channels > 0)
-        mDefaults->channels = channels;
-
-    if (rate > 0)
-        mDefaults->sampleRate = rate;
-
-    switch(format) {
-      // format == 0
-        case AudioSystem::DEFAULT:
-            break;
-
-        case AudioSystem::PCM_16_BIT:
-            mDefaults->format = SND_PCM_FORMAT_S16_LE;
-            break;
-
-        case AudioSystem::PCM_8_BIT:
-            mDefaults->format = SND_PCM_FORMAT_S8;
-            break;
-
-        default:
-            LOGE("Unknown PCM format %i. Forcing default", format);
-            break;
-    }
-
-    return NO_ERROR;
-}
-
-uint32_t ALSAStreamOps::sampleRate() const
-{
-    unsigned int rate;
-    int err;
-
-    if (!mHandle)
-        return mDefaults->sampleRate;
-
-    return snd_pcm_hw_params_get_rate(mHardwareParams, &rate, 0) < 0
-        ? 0 : static_cast<uint32_t>(rate);
-}
-
-status_t ALSAStreamOps::sampleRate(uint32_t rate)
-{
-    const char *stream;
-    unsigned int requestedRate;
-    int err;
-
-    if (!mHandle)
-        return NO_INIT;
-
-    stream = streamName();
-    requestedRate = rate;
-    err = snd_pcm_hw_params_set_rate_near(mHandle,
-                                          mHardwareParams,
-                                          &requestedRate,
-                                          0);
-
-    if (err < 0) {
-        LOGE("Unable to set %s sample rate to %u: %s",
-            stream, rate, snd_strerror(err));
-        return BAD_VALUE;
-    }
-    if (requestedRate != rate) {
-        // Some devices have a fixed sample rate, and can not be changed.
-        // This may cause resampling problems; i.e. PCM playback will be too
-        // slow or fast.
-        LOGW("Requested rate (%u HZ) does not match actual rate (%u HZ)",
-            rate, requestedRate);
-    }
-    else {
-        LOGV("Set %s sample rate to %u HZ", stream, requestedRate);
-    }
-    return NO_ERROR;
-}
-
-//
-// Return the number of bytes (not frames)
-//
-size_t ALSAStreamOps::bufferSize() const
+status_t AudioHardwareALSA::setMode(int mode)
 {
-    if (!mHandle)
-        return NO_INIT;
-
-    snd_pcm_uframes_t bufferSize = 0;
-    snd_pcm_uframes_t periodSize = 0;
-    int err;
-
-    err = snd_pcm_get_params(mHandle, &bufferSize, &periodSize);
-
-    if (err < 0)
-        return -1;
-
-    size_t bytes = static_cast<size_t>(snd_pcm_frames_to_bytes(mHandle, bufferSize));
-
-    // Not sure when this happened, but unfortunately it now
-    // appears that the bufferSize must be reported as a
-    // power of 2. This might be the fault of 3rd party
-    // users.
-    for (size_t i = 1; (bytes & ~i) != 0; i<<=1)
-        bytes &= ~i;
-
-    return bytes;
-}
-
-int ALSAStreamOps::format() const
-{
-    snd_pcm_format_t ALSAFormat;
-    int pcmFormatBitWidth;
-    int audioSystemFormat;
-
-    if (!mHandle)
-        return -1;
-
-    if (snd_pcm_hw_params_get_format(mHardwareParams, &ALSAFormat) < 0) {
-        return -1;
-    }
-
-    pcmFormatBitWidth = snd_pcm_format_physical_width(ALSAFormat);
-    audioSystemFormat = AudioSystem::DEFAULT;
-    switch(pcmFormatBitWidth) {
-        case 8:
-            audioSystemFormat = AudioSystem::PCM_8_BIT;
-            break;
-
-        case 16:
-            audioSystemFormat = AudioSystem::PCM_16_BIT;
-            break;
-
-        default:
-            LOG_FATAL("Unknown AudioSystem bit width %i!", pcmFormatBitWidth);
-    }
-
-    return audioSystemFormat;
-}
-
-int ALSAStreamOps::channelCount() const
-{
-    if (!mHandle)
-        return mDefaults->channels;
-
-    unsigned int val;
-    int err;
-
-    err = snd_pcm_hw_params_get_channels(mHardwareParams, &val);
-    if (err < 0) {
-        LOGE("Unable to get device channel count: %s",
-            snd_strerror(err));
-        return mDefaults->channels;
-    }
-
-    return val;
-}
-
-status_t ALSAStreamOps::channelCount(int channels) {
-    int err;
-
-    if (!mHandle)
-        return NO_INIT;
-
-    err = snd_pcm_hw_params_set_channels(mHandle, mHardwareParams, channels);
-    if (err < 0) {
-        LOGE("Unable to set channel count to %i: %s",
-            channels, snd_strerror(err));
-        return BAD_VALUE;
-    }
-
-    LOGV("Using %i %s for %s.",
-        channels, channels == 1 ? "channel" : "channels", streamName());
+    status_t status = NO_ERROR;
 
-    return NO_ERROR;
-}
-
-status_t ALSAStreamOps::open(int mode, uint32_t device)
-{
-    const char *stream = streamName();
-    const char *devName = deviceName(mode, device);
-
-    int         err;
-
-    for(;;) {
-        // The PCM stream is opened in blocking mode, per ALSA defaults.  The
-        // AudioFlinger seems to assume blocking mode too, so asynchronous mode
-        // should not be used.
-        err = snd_pcm_open(&mHandle, devName, mDefaults->direction, 0);
-        if (err == 0) break;
-
-        // See if there is a less specific name we can try.
-        // Note: We are changing the contents of a const char * here.
-        char *tail = strrchr(devName, '_');
-        if (!tail) break;
-        *tail = 0;
-    }
-
-    if (err < 0) {
-        // None of the Android defined audio devices exist. Open a generic one.
-        devName = "default";
-        err = snd_pcm_open(&mHandle, devName, mDefaults->direction, 0);
-        if (err < 0) {
-            // Last resort is the NULL device (i.e. the bit bucket).
-            devName = _nullALSADeviceName;
-            err = snd_pcm_open(&mHandle, devName, mDefaults->direction, 0);
-        }
-    }
-
-    mMode   = mode;
-    mDevice = device;
-
-    LOGI("Initialized ALSA %s device %s", stream, devName);
-    return err;
-}
-
-void ALSAStreamOps::close()
-{
-    snd_pcm_t *handle = mHandle;
-    mHandle = NULL;
-
-    if (handle)
-        snd_pcm_close(handle);
-}
-
-status_t ALSAStreamOps::setSoftwareParams()
-{
-    if (!mHandle)
-        return NO_INIT;
-
-    int err;
-
-    // Get the current software parameters
-    err = snd_pcm_sw_params_current(mHandle, mSoftwareParams);
-    if (err < 0) {
-        LOGE("Unable to get software parameters: %s", snd_strerror(err));
-        return NO_INIT;
-    }
-
-    snd_pcm_uframes_t bufferSize = 0;
-    snd_pcm_uframes_t periodSize = 0;
-    snd_pcm_uframes_t startThreshold, stopThreshold;
+    if (mode != mMode) {
+        status = AudioHardwareBase::setMode(mode);
 
-    // Configure ALSA to start the transfer when the buffer is almost full.
-    snd_pcm_get_params(mHandle, &bufferSize, &periodSize);
-
-    if (mDefaults->direction == SND_PCM_STREAM_PLAYBACK) {
-        // For playback, configure ALSA to start the transfer when the
-        // buffer is full.
-        startThreshold = bufferSize - periodSize;
-        stopThreshold = bufferSize;
-    }
-    else {
-        // For recording, configure ALSA to start the transfer on the
-        // first frame.
-        startThreshold = 1;
-        stopThreshold = bufferSize;
-}
-
-    err = snd_pcm_sw_params_set_start_threshold(mHandle,
-                                                mSoftwareParams,
-                                                startThreshold);
-    if (err < 0) {
-        LOGE("Unable to set start threshold to %lu frames: %s",
-            startThreshold, snd_strerror(err));
-        return NO_INIT;
-    }
-
-    err = snd_pcm_sw_params_set_stop_threshold(mHandle,
-                                               mSoftwareParams,
-                                               stopThreshold);
-    if (err < 0) {
-        LOGE("Unable to set stop threshold to %lu frames: %s",
-            stopThreshold, snd_strerror(err));
-        return NO_INIT;
-    }
-
-    // Allow the transfer to start when at least periodSize samples can be
-    // processed.
-    err = snd_pcm_sw_params_set_avail_min(mHandle,
-                                          mSoftwareParams,
-                                          periodSize);
-    if (err < 0) {
-        LOGE("Unable to configure available minimum to %lu: %s",
-            periodSize, snd_strerror(err));
-        return NO_INIT;
-    }
-
-    // Commit the software parameters back to the device.
-    err = snd_pcm_sw_params(mHandle, mSoftwareParams);
-    if (err < 0) {
-        LOGE("Unable to configure software parameters: %s",
-            snd_strerror(err));
-        return NO_INIT;
-    }
-
-    return NO_ERROR;
-}
-
-status_t ALSAStreamOps::setPCMFormat(snd_pcm_format_t format)
-{
-    const char *formatDesc;
-    const char *formatName;
-    bool validFormat;
-    int err;
-
-    // snd_pcm_format_description() and snd_pcm_format_name() do not perform
-    // proper bounds checking.
-    validFormat = (static_cast<int>(format) > SND_PCM_FORMAT_UNKNOWN) &&
-        (static_cast<int>(format) <= SND_PCM_FORMAT_LAST);
-    formatDesc = validFormat ?
-        snd_pcm_format_description(format) : "Invalid Format";
-    formatName = validFormat ?
-        snd_pcm_format_name(format) : "UNKNOWN";
-
-    err = snd_pcm_hw_params_set_format(mHandle, mHardwareParams, format);
-    if (err < 0) {
-        LOGE("Unable to configure PCM format %s (%s): %s",
-            formatName, formatDesc, snd_strerror(err));
-        return NO_INIT;
-    }
-
-    LOGV("Set %s PCM format to %s (%s)", streamName(), formatName, formatDesc);
-    return NO_ERROR;
-}
-
-status_t ALSAStreamOps::setHardwareResample(bool resample)
-{
-    int err;
-
-    err = snd_pcm_hw_params_set_rate_resample(mHandle,
-                                              mHardwareParams,
-                                              static_cast<int>(resample));
-    if (err < 0) {
-        LOGE("Unable to %s hardware resampling: %s",
-            resample ? "enable" : "disable",
-            snd_strerror(err));
-        return NO_INIT;
-    }
-    return NO_ERROR;
-}
-
-const char *ALSAStreamOps::streamName()
-{
-    // Don't use snd_pcm_stream(mHandle), as the PCM stream may not be
-    // opened yet.  In such case, snd_pcm_stream() will abort().
-    return snd_pcm_stream_name(mDefaults->direction);
-}
-
-//
-// Set playback or capture PCM device.  It's possible to support audio output
-// or input from multiple devices by using the ALSA plugins, but this is
-// not supported for simplicity.
-//
-// The AudioHardwareALSA API does not allow one to set the input routing.
-//
-// If the "routes" value does not map to a valid device, the default playback
-// device is used.
-//
-status_t ALSAStreamOps::setDevice(int mode, uint32_t device)
-{
-    // Close off previously opened device.
-    // It would be nice to determine if the underlying device actually
-    // changes, but we might be manipulating mixer settings (see asound.conf).
-    //
-    close();
-
-    const char *stream = streamName();
-
-    status_t    status = open (mode, device);
-    int     err;
-
-    if (status != NO_ERROR)
-        return status;
-
-    err = snd_pcm_hw_params_any(mHandle, mHardwareParams);
-    if (err < 0) {
-        LOGE("Unable to configure hardware: %s", snd_strerror(err));
-        return NO_INIT;
-    }
-
-    // Set the interleaved read and write format.
-    err = snd_pcm_hw_params_set_access(mHandle, mHardwareParams,
-                                       SND_PCM_ACCESS_RW_INTERLEAVED);
-    if (err < 0) {
-        LOGE("Unable to configure PCM read/write format: %s",
-            snd_strerror(err));
-        return NO_INIT;
-    }
-
-    status = setPCMFormat(mDefaults->format);
-
-    //
-    // Some devices do not have the default two channels.  Force an error to
-    // prevent AudioMixer from crashing and taking the whole system down.
-    //
-    // Note that some devices will return an -EINVAL if the channel count
-    // is queried before it has been set.  i.e. calling channelCount()
-    // before channelCount(channels) may return -EINVAL.
-    //
-    status = channelCount(mDefaults->channels);
-    if (status != NO_ERROR)
-        return status;
-
-    // Don't check for failure; some devices do not support the default
-    // sample rate.
-    sampleRate(mDefaults->sampleRate);
-
-#ifdef DISABLE_HARWARE_RESAMPLING
-    // Disable hardware resampling.
-    status = setHardwareResample(false);
-    if (status != NO_ERROR)
-        return status;
-#endif
-
-    snd_pcm_uframes_t bufferSize = mDefaults->bufferSize;
-
-    // Make sure we have at least the size we originally wanted
-    err = snd_pcm_hw_params_set_buffer_size(mHandle, mHardwareParams, bufferSize);
-    if (err < 0) {
-        LOGE("Unable to set buffer size to %d:  %s",
-             (int)bufferSize, snd_strerror(err));
-        return NO_INIT;
-    }
-
-    unsigned int latency = mDefaults->latency;
-
-    // Setup buffers for latency
-    err = snd_pcm_hw_params_set_buffer_time_near (mHandle, mHardwareParams,
-                                                  &latency, NULL);
-    if (err < 0) {
-        /* That didn't work, set the period instead */
-        unsigned int periodTime = latency / 4;
-        err = snd_pcm_hw_params_set_period_time_near (mHandle, mHardwareParams,
-                                                      &periodTime, NULL);
-        if (err < 0) {
-            LOGE("Unable to set the period time for latency: %s", snd_strerror(err));
-            return NO_INIT;
-        }
-        snd_pcm_uframes_t periodSize;
-        err = snd_pcm_hw_params_get_period_size (mHardwareParams, &periodSize, NULL);
-        if (err < 0) {
-            LOGE("Unable to get the period size for latency: %s", snd_strerror(err));
-            return NO_INIT;
-        }
-        bufferSize = periodSize * 4;
-        if (bufferSize < mDefaults->bufferSize)
-            bufferSize = mDefaults->bufferSize;
-        err = snd_pcm_hw_params_set_buffer_size_near (mHandle, mHardwareParams, &bufferSize);
-        if (err < 0) {
-            LOGE("Unable to set the buffer size for latency: %s", snd_strerror(err));
-            return NO_INIT;
-        }
-    } else {
-        // OK, we got buffer time near what we expect. See what that did for bufferSize.
-        err = snd_pcm_hw_params_get_buffer_size (mHardwareParams, &bufferSize);
-        if (err < 0) {
-            LOGE("Unable to get the buffer size for latency: %s", snd_strerror(err));
-            return NO_INIT;
-        }
-        // Does set_buffer_time_near change the passed value? It should.
-        err = snd_pcm_hw_params_get_buffer_time (mHardwareParams, &latency, NULL);
-        if (err < 0) {
-            LOGE("Unable to get the buffer time for latency: %s", snd_strerror(err));
-            return NO_INIT;
-        }
-        unsigned int periodTime = latency / 4;
-        err = snd_pcm_hw_params_set_period_time_near (mHandle, mHardwareParams,
-                                                      &periodTime, NULL);
-        if (err < 0) {
-            LOGE("Unable to set the period time for latency: %s", snd_strerror(err));
-            return NO_INIT;
+        if (status == NO_ERROR) {
+            // take care of mode change.
+            for(ALSAHandleList::iterator it = mDeviceList.begin();
+                it != mDeviceList.end(); ++it) {
+                status = mALSADevice->route(&(*it), it->curDev, mode);
+                if (status != NO_ERROR)
+                    break;
+            }
         }
     }
 
-    LOGV("Buffer size: %d", (int)bufferSize);
-    LOGV("Latency: %d", (int)latency);
-
-    mDefaults->bufferSize = bufferSize;
-    mDefaults->latency = latency;
-
-    // Commit the hardware parameters back to the device.
-    err = snd_pcm_hw_params(mHandle, mHardwareParams);
-    if (err < 0) {
-        LOGE("Unable to set hardware parameters: %s", snd_strerror(err));
-        return NO_INIT;
-    }
-
-    status = setSoftwareParams();
-
     return status;
 }
 
-const char *ALSAStreamOps::deviceName(int mode, uint32_t device)
-{
-    static char devString[ALSA_NAME_MAX];
-    int hasDevExt = 0;
-
-    strcpy (devString, mDefaults->devicePrefix);
-
-    for (int dev=0; device; dev++)
-        if (device & (1 << dev)) {
-            /* Don't go past the end of our list */
-            if (dev >= deviceSuffixLen)
-                break;
-            ALSA_STRCAT (devString, deviceSuffix[dev]);
-            device &= ~(1 << dev);
-            hasDevExt = 1;
-        }
-
-    if (hasDevExt)
-        switch (mode) {
-            case AudioSystem::MODE_NORMAL:
-                ALSA_STRCAT (devString, "_normal");
-                break;
-            case AudioSystem::MODE_RINGTONE:
-                ALSA_STRCAT (devString, "_ringtone");
-                break;
-            case AudioSystem::MODE_IN_CALL:
-                ALSA_STRCAT (devString, "_incall");
-                break;
-        };
-
-    return devString;
-}
-
-// ----------------------------------------------------------------------------
-
-AudioStreamOutALSA::AudioStreamOutALSA(AudioHardwareALSA *parent) :
-    ALSAStreamOps(parent)
-{
-    static StreamDefaults _defaults = {
-        devicePrefix   : "AndroidPlayback",
-        direction      : SND_PCM_STREAM_PLAYBACK,
-        format         : SND_PCM_FORMAT_S16_LE,   // AudioSystem::PCM_16_BIT
-        channels       : 2,
-        sampleRate     : DEFAULT_SAMPLE_RATE,
-        latency        : 200000,                  // Desired Delay in usec
-        bufferSize     : DEFAULT_SAMPLE_RATE / 5, // Desired Number of samples
-        };
-
-    setStreamDefaults(&_defaults);
-
-    snd_pcm_uframes_t bufferSize = mDefaults->bufferSize;
-
-    // See comment in bufferSize() method.
-    for (size_t i = 1; (bufferSize & ~i) != 0; i<<=1)
-        bufferSize &= ~i;
-
-    mDefaults->bufferSize = bufferSize;
-}
-
-AudioStreamOutALSA::~AudioStreamOutALSA()
-{
-    standby();
-    mParent->mOutput = NULL;
-}
-
-int AudioStreamOutALSA::channelCount() const
-{
-    int c = ALSAStreamOps::channelCount();
-
-    // AudioMixer will seg fault if it doesn't have two channels.
-    LOGW_IF(c != 2,
-        "AudioMixer expects two channels, but only %i found!", c);
-    return c;
-}
-
-status_t AudioStreamOutALSA::setVolume(float volume)
-{
-    if (!mParent->mMixer || !mDevice)
-        return NO_INIT;
-
-    return mParent->mMixer->setVolume (mDevice, volume);
-}
-
-ssize_t AudioStreamOutALSA::write(const void *buffer, size_t bytes)
-{
-    snd_pcm_sframes_t n;
-    size_t            sent = 0;
-    status_t          err;
-
-    AutoMutex lock(mLock);
-
-    if (!mPowerLock) {
-        acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioOutLock");
-        mPowerLock = true;
-    }
-
-    if (!mHandle)
-        ALSAStreamOps::setDevice(mMode, mDevice);
-
-    do {
-        n = snd_pcm_writei(mHandle,
-                           (char *)buffer + sent,
-                           snd_pcm_bytes_to_frames(mHandle, bytes));
-        if (n == -EBADFD) {
-            // Somehow the stream is in a bad state. The driver probably
-            // has a bug and snd_pcm_recover() doesn't seem to handle this.
-            ALSAStreamOps::setDevice(mMode, mDevice);
-        }
-        else if (n < 0) {
-            if (mHandle) {
-                // snd_pcm_recover() will return 0 if successful in recovering from
-                // an error, or -errno if the error was unrecoverable.
-                n = snd_pcm_recover(mHandle, n, 1);
-                if (n)
-                    return static_cast<ssize_t>(n);
-            }
-        }
-        else
-            sent += static_cast<ssize_t>(snd_pcm_frames_to_bytes(mHandle, n));
-
-    } while (mHandle && sent < bytes);
-
-    return sent;
-}
-
-status_t AudioStreamOutALSA::dump(int fd, const Vector<String16>& args)
-{
-    return NO_ERROR;
-}
-
-status_t AudioStreamOutALSA::setDevice(int mode, uint32_t newDevice)
+AudioStreamOut *
+AudioHardwareALSA::openOutputStream(uint32_t devices,
+                                    int *format,
+                                    uint32_t *channels,
+                                    uint32_t *sampleRate,
+                                    status_t *status)
 {
     AutoMutex lock(mLock);
 
-    return ALSAStreamOps::setDevice(mode, newDevice);
-}
-
-status_t AudioStreamOutALSA::standby()
-{
-    AutoMutex lock(mLock);
+    LOGD("openOutputStream called for devices: 0x%08x", devices);
 
-    if (mHandle) {
-        snd_pcm_drain (mHandle);
-        close();
-    }
+    status_t err = BAD_VALUE;
+    AudioStreamOutALSA *out = 0;
 
-    if (mPowerLock) {
-        release_wake_lock ("AudioOutLock");
-        mPowerLock = false;
+    if (devices & (devices - 1)) {
+        if (status) *status = err;
+        LOGD("openOutputStream called with bad devices");
+        return out;
     }
 
-    return NO_ERROR;
-}
-
-#define USEC_TO_MSEC(x) ((x + 999) / 1000)
-
-uint32_t AudioStreamOutALSA::latency() const
-{
-    // Android wants latency in milliseconds.
-    return USEC_TO_MSEC (mDefaults->latency);
-}
-
-// ----------------------------------------------------------------------------
-
-AudioStreamInALSA::AudioStreamInALSA(AudioHardwareALSA *parent,
-                                     AudioSystem::audio_in_acoustics acoustics) :
-    ALSAStreamOps(parent),
-    mAcoustics(acoustics)
-{
-    static StreamDefaults _defaults = {
-        devicePrefix   : "AndroidRecord",
-        direction      : SND_PCM_STREAM_CAPTURE,
-        format         : SND_PCM_FORMAT_S16_LE,   // AudioSystem::PCM_16_BIT
-        channels       : 1,
-        sampleRate     : AudioRecord::DEFAULT_SAMPLE_RATE,
-        latency        : 250000,                  // Desired Delay in usec
-        bufferSize     : 2048,                    // Desired Number of samples
-        };
-
-    setStreamDefaults(&_defaults);
-}
-
-AudioStreamInALSA::~AudioStreamInALSA()
-{
-    standby();
-    mParent->mInput = NULL;
-}
-
-status_t AudioStreamInALSA::setGain(float gain)
-{
-    if (mParent->mMixer)
-        return mParent->mMixer->setMasterGain (gain);
-    else
-        return NO_INIT;
-}
-
-ssize_t AudioStreamInALSA::read(void *buffer, ssize_t bytes)
-{
-    snd_pcm_sframes_t n, frames = snd_pcm_bytes_to_frames(mHandle, bytes);
-    status_t          err;
-
-    AutoMutex lock(mLock);
-
-    if (!mPowerLock) {
-        acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioInLock");
-        mPowerLock = true;
-    }
-
-    if (!mHandle)
-        ALSAStreamOps::setDevice(mMode, mDevice);
-
-    n = snd_pcm_readi(mHandle, buffer, frames);
-    if (n < frames) {
-        if (mHandle) {
-            if (n < 0)
-                n = snd_pcm_recover(mHandle, n, 0);
-            else
-                n = snd_pcm_prepare(mHandle);
+    // Find the appropriate alsa device
+    for(ALSAHandleList::iterator it = mDeviceList.begin();
+        it != mDeviceList.end(); ++it)
+        if (it->devices & devices) {
+            err = mALSADevice->open(&(*it), devices, mode());
+            if (err) break;
+            out = new AudioStreamOutALSA(this, &(*it));
+            err = out->set(format, channels, sampleRate);
+            break;
         }
-        return static_cast<ssize_t>(n);
-    }
 
-    if (mParent->mAcousticDevice &&
-        mParent->mAcousticDevice->filter) {
-        n = mParent->mAcousticDevice->filter(mHandle, buffer, frames);
-        if (n < 0)
-            return static_cast<ssize_t>(n);
-    }
-
-    return static_cast<ssize_t>(snd_pcm_frames_to_bytes(mHandle, n));
+    if (status) *status = err;
+    return out;
 }
 
-status_t AudioStreamInALSA::dump(int fd, const Vector<String16>& args)
-{
-    return NO_ERROR;
-}
-
-status_t AudioStreamInALSA::setDevice(int mode, uint32_t newDevice)
+void
+AudioHardwareALSA::closeOutputStream(AudioStreamOut* out)
 {
     AutoMutex lock(mLock);
-
-    status_t status = ALSAStreamOps::setDevice(mode, newDevice);
-
-    if (status == NO_ERROR && mParent->mAcousticDevice)
-        status = mParent->mAcousticDevice->set_acoustics(mHandle, mAcoustics);
-
-    return status;
+    delete out;
 }
 
-status_t AudioStreamInALSA::standby()
+AudioStreamIn *
+AudioHardwareALSA::openInputStream(uint32_t devices,
+                                   int *format,
+                                   uint32_t *channels,
+                                   uint32_t *sampleRate,
+                                   status_t *status,
+                                   AudioSystem::audio_in_acoustics acoustics)
 {
     AutoMutex lock(mLock);
 
-    close();
-
-    if (mPowerLock) {
-        release_wake_lock ("AudioInLock");
-        mPowerLock = false;
-    }
-
-    return NO_ERROR;
-}
-
-// ----------------------------------------------------------------------------
-
-struct mixer_info_t
-{
-    mixer_info_t() :
-        elem(0),
-        min(SND_MIXER_VOL_RANGE_MIN),
-        max(SND_MIXER_VOL_RANGE_MAX),
-        mute(false)
-    {
-    }
-
-    snd_mixer_elem_t *elem;
-    long              min;
-    long              max;
-    long              volume;
-    bool              mute;
-    char              name[ALSA_NAME_MAX];
-};
-
-static int initMixer (snd_mixer_t **mixer, const char *name)
-{
-    int err;
-
-    if ((err = snd_mixer_open(mixer, 0)) < 0) {
-        LOGE("Unable to open mixer: %s", snd_strerror(err));
-        return err;
-    }
-
-    if ((err = snd_mixer_attach(*mixer, name)) < 0) {
-        LOGE("Unable to attach mixer to device %s: %s",
-            name, snd_strerror(err));
-
-        if ((err = snd_mixer_attach(*mixer, "hw:00")) < 0) {
-            LOGE("Unable to attach mixer to device default: %s",
-                snd_strerror(err));
-
-            snd_mixer_close (*mixer);
-            *mixer = NULL;
-            return err;
-        }
-    }
-
-    if ((err = snd_mixer_selem_register(*mixer, NULL, NULL)) < 0) {
-        LOGE("Unable to register mixer elements: %s", snd_strerror(err));
-        snd_mixer_close (*mixer);
-        *mixer = NULL;
-        return err;
-    }
-
-    // Get the mixer controls from the kernel
-    if ((err = snd_mixer_load(*mixer)) < 0) {
-        LOGE("Unable to load mixer elements: %s", snd_strerror(err));
-        snd_mixer_close (*mixer);
-        *mixer = NULL;
-        return err;
-    }
-
-    return 0;
-}
-
-typedef int (*hasVolume_t)(snd_mixer_elem_t*);
-
-static const hasVolume_t hasVolume[] = {
-    snd_mixer_selem_has_playback_volume,
-    snd_mixer_selem_has_capture_volume
-};
-
-typedef int (*getVolumeRange_t)(snd_mixer_elem_t*, long int*, long int*);
-
-static const getVolumeRange_t getVolumeRange[] = {
-    snd_mixer_selem_get_playback_volume_range,
-    snd_mixer_selem_get_capture_volume_range
-};
-
-typedef int (*setVolume_t)(snd_mixer_elem_t*, long int);
-
-static const setVolume_t setVol[] = {
-    snd_mixer_selem_set_playback_volume_all,
-    snd_mixer_selem_set_capture_volume_all
-};
-
-ALSAMixer::ALSAMixer()
-{
-    int err;
-
-    initMixer (&mMixer[SND_PCM_STREAM_PLAYBACK], "AndroidPlayback");
-    initMixer (&mMixer[SND_PCM_STREAM_CAPTURE], "AndroidRecord");
-
-    snd_mixer_selem_id_t *sid;
-    snd_mixer_selem_id_alloca(&sid);
-
-    for (int i = 0; i <= SND_PCM_STREAM_LAST; i++) {
+    status_t err = BAD_VALUE;
+    AudioStreamInALSA *in = 0;
 
-        mixer_info_t *info = mixerMasterProp[i].mInfo = new mixer_info_t;
-
-        property_get (mixerMasterProp[i].propName,
-                      info->name,
-                      mixerMasterProp[i].propDefault);
-
-        for (snd_mixer_elem_t *elem = snd_mixer_first_elem(mMixer[i]);
-             elem;
-             elem = snd_mixer_elem_next(elem)) {
-
-            if (!snd_mixer_selem_is_active(elem))
-                continue;
-
-            snd_mixer_selem_get_id(elem, sid);
-
-            // Find PCM playback volume control element.
-            const char *elementName = snd_mixer_selem_id_get_name(sid);
-
-            if (info->elem == NULL &&
-                strcmp(elementName, info->name) == 0 &&
-                hasVolume[i] (elem)) {
-
-                info->elem = elem;
-                getVolumeRange[i] (elem, &info->min, &info->max);
-                info->volume = info->max;
-                setVol[i] (elem, info->volume);
-                if (i == SND_PCM_STREAM_PLAYBACK &&
-                    snd_mixer_selem_has_playback_switch (elem))
-                    snd_mixer_selem_set_playback_switch_all (elem, 1);
-                break;
-            }
-        }
-
-        LOGV("Mixer: master '%s' %s.", info->name, info->elem ? "found" : "not found");
-
-        for (int j = 0; mixerProp[j][i].routes; j++) {
-
-            mixer_info_t *info = mixerProp[j][i].mInfo = new mixer_info_t;
-
-            property_get (mixerProp[j][i].propName,
-                          info->name,
-                          mixerProp[j][i].propDefault);
-
-            for (snd_mixer_elem_t *elem = snd_mixer_first_elem(mMixer[i]);
-                 elem;
-                 elem = snd_mixer_elem_next(elem)) {
-
-                if (!snd_mixer_selem_is_active(elem))
-                    continue;
-
-                snd_mixer_selem_get_id(elem, sid);
-
-                // Find PCM playback volume control element.
-                const char *elementName = snd_mixer_selem_id_get_name(sid);
-
-               if (info->elem == NULL &&
-                    strcmp(elementName, info->name) == 0 &&
-                    hasVolume[i] (elem)) {
-
-                    info->elem = elem;
-                    getVolumeRange[i] (elem, &info->min, &info->max);
-                    info->volume = info->max;
-                    setVol[i] (elem, info->volume);
-                    if (i == SND_PCM_STREAM_PLAYBACK &&
-                        snd_mixer_selem_has_playback_switch (elem))
-                        snd_mixer_selem_set_playback_switch_all (elem, 1);
-                    break;
-                }
-            }
-            LOGV("Mixer: route '%s' %s.", info->name, info->elem ? "found" : "not found");
-        }
+    if (devices & (devices - 1)) {
+        if (status) *status = err;
+        return in;
     }
-    LOGV("mixer initialized.");
-}
-
-ALSAMixer::~ALSAMixer()
-{
-    for (int i = 0; i <= SND_PCM_STREAM_LAST; i++) {
-        if (mMixer[i]) snd_mixer_close (mMixer[i]);
-        if (mixerMasterProp[i].mInfo) {
-            delete mixerMasterProp[i].mInfo;
-            mixerMasterProp[i].mInfo = NULL;
-        }
-        for (int j = 0; mixerProp[j][i].routes; j++) {
-            if (mixerProp[j][i].mInfo) {
-                delete mixerProp[j][i].mInfo;
-                mixerProp[j][i].mInfo = NULL;
-            }
-        }
-    }
-    LOGV("mixer destroyed.");
-}
-
-status_t ALSAMixer::setMasterVolume(float volume)
-{
-    mixer_info_t *info = mixerMasterProp[SND_PCM_STREAM_PLAYBACK].mInfo;
-    if (!info || !info->elem) return INVALID_OPERATION;
-
-    long minVol = info->min;
-    long maxVol = info->max;
-
-    // Make sure volume is between bounds.
-    long vol = minVol + volume * (maxVol - minVol);
-    if (vol > maxVol) vol = maxVol;
-    if (vol < minVol) vol = minVol;
 
-    info->volume = vol;
-    snd_mixer_selem_set_playback_volume_all (info->elem, vol);
-
-    return NO_ERROR;
-}
-
-status_t ALSAMixer::setMasterGain(float gain)
-{
-    mixer_info_t *info = mixerMasterProp[SND_PCM_STREAM_CAPTURE].mInfo;
-    if (!info || !info->elem) return INVALID_OPERATION;
-
-    long minVol = info->min;
-    long maxVol = info->max;
-
-    // Make sure volume is between bounds.
-    long vol = minVol + gain * (maxVol - minVol);
-    if (vol > maxVol) vol = maxVol;
-    if (vol < minVol) vol = minVol;
-
-    info->volume = vol;
-    snd_mixer_selem_set_capture_volume_all (info->elem, vol);
-
-    return NO_ERROR;
-}
-
-status_t ALSAMixer::setVolume(uint32_t device, float volume)
-{
-    for (int j = 0; mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes; j++)
-        if (mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes & device) {
-
-            mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_PLAYBACK].mInfo;
-            if (!info || !info->elem) return INVALID_OPERATION;
-
-            long minVol = info->min;
-            long maxVol = info->max;
-
-            // Make sure volume is between bounds.
-            long vol = minVol + volume * (maxVol - minVol);
-            if (vol > maxVol) vol = maxVol;
-            if (vol < minVol) vol = minVol;
-
-            info->volume = vol;
-            snd_mixer_selem_set_playback_volume_all (info->elem, vol);
-        }
-
-    return NO_ERROR;
-}
-
-status_t ALSAMixer::setGain(uint32_t device, float gain)
-{
-    for (int j = 0; mixerProp[j][SND_PCM_STREAM_CAPTURE].routes; j++)
-        if (mixerProp[j][SND_PCM_STREAM_CAPTURE].routes & device) {
-
-            mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_CAPTURE].mInfo;
-            if (!info || !info->elem) return INVALID_OPERATION;
-
-            long minVol = info->min;
-            long maxVol = info->max;
-
-            // Make sure volume is between bounds.
-            long vol = minVol + gain * (maxVol - minVol);
-            if (vol > maxVol) vol = maxVol;
-            if (vol < minVol) vol = minVol;
-
-            info->volume = vol;
-            snd_mixer_selem_set_capture_volume_all (info->elem, vol);
+    // Find the appropriate alsa device
+    for(ALSAHandleList::iterator it = mDeviceList.begin();
+        it != mDeviceList.end(); ++it)
+        if (it->devices & devices) {
+            err = mALSADevice->open(&(*it), devices, mode());
+            if (err) break;
+            in = new AudioStreamInALSA(this, &(*it), acoustics);
+            err = in->set(format, channels, sampleRate);
+            break;
         }
 
-    return NO_ERROR;
+    if (status) *status = err;
+    return in;
 }
 
-status_t ALSAMixer::setCaptureMuteState(uint32_t device, bool state)
+void
+AudioHardwareALSA::closeInputStream(AudioStreamIn* in)
 {
-    for (int j = 0; mixerProp[j][SND_PCM_STREAM_CAPTURE].routes; j++)
-        if (mixerProp[j][SND_PCM_STREAM_CAPTURE].routes & device) {
-
-            mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_CAPTURE].mInfo;
-            if (!info || !info->elem) return INVALID_OPERATION;
-
-            if (snd_mixer_selem_has_capture_switch (info->elem)) {
-
-                int err = snd_mixer_selem_set_capture_switch_all (info->elem, static_cast<int>(!state));
-                if (err < 0) {
-                    LOGE("Unable to %s capture mixer switch %s",
-                        state ? "enable" : "disable", info->name);
-                    return INVALID_OPERATION;
-                }
-            }
-
-            info->mute = state;
-        }
-
-    return NO_ERROR;
+    AutoMutex lock(mLock);
+    delete in;
 }
 
-status_t ALSAMixer::getCaptureMuteState(uint32_t device, bool *state)
+status_t AudioHardwareALSA::setMicMute(bool state)
 {
-    if (!state) return BAD_VALUE;
-
-    for (int j = 0; mixerProp[j][SND_PCM_STREAM_CAPTURE].routes; j++)
-        if (mixerProp[j][SND_PCM_STREAM_CAPTURE].routes & device) {
-
-            mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_CAPTURE].mInfo;
-            if (!info || !info->elem) return INVALID_OPERATION;
-
-            *state = info->mute;
-            return NO_ERROR;
-        }
+    if (mMixer)
+        return mMixer->setCaptureMuteState(AudioSystem::DEVICE_OUT_EARPIECE, state);
 
-    return BAD_VALUE;
+    return NO_INIT;
 }
 
-status_t ALSAMixer::setPlaybackMuteState(uint32_t device, bool state)
+status_t AudioHardwareALSA::getMicMute(bool *state)
 {
-    for (int j = 0; mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes; j++)
-        if (mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes & device) {
-
-            mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_PLAYBACK].mInfo;
-            if (!info || !info->elem) return INVALID_OPERATION;
-
-            if (snd_mixer_selem_has_playback_switch (info->elem)) {
-
-                int err = snd_mixer_selem_set_playback_switch_all (info->elem, static_cast<int>(!state));
-                if (err < 0) {
-                    LOGE("Unable to %s playback mixer switch %s",
-                        state ? "enable" : "disable", info->name);
-                    return INVALID_OPERATION;
-                }
-            }
-
-            info->mute = state;
-        }
+    if (mMixer)
+        return mMixer->getCaptureMuteState(AudioSystem::DEVICE_OUT_EARPIECE, state);
 
     return NO_ERROR;
 }
 
-status_t ALSAMixer::getPlaybackMuteState(uint32_t device, bool *state)
-{
-    if (!state) return BAD_VALUE;
-
-    for (int j = 0; mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes; j++)
-        if (mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes & device) {
-
-            mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_PLAYBACK].mInfo;
-            if (!info || !info->elem) return INVALID_OPERATION;
-
-            *state = info->mute;
-            return NO_ERROR;
-        }
-
-    return BAD_VALUE;
-}
-
-// ----------------------------------------------------------------------------
-
-ALSAControl::ALSAControl(const char *device)
-{
-    snd_ctl_open(&mHandle, device, 0);
-}
-
-ALSAControl::~ALSAControl()
-{
-    if (mHandle) snd_ctl_close(mHandle);
-}
-
-status_t ALSAControl::get(const char *name, unsigned int &value, int index)
+status_t AudioHardwareALSA::dump(int fd, const Vector<String16>& args)
 {
-    if (!mHandle) return NO_INIT;
-
-    snd_ctl_elem_id_t *id;
-    snd_ctl_elem_info_t *info;
-    snd_ctl_elem_value_t *control;
-
-    snd_ctl_elem_id_alloca(&id);
-    snd_ctl_elem_info_alloca(&info);
-    snd_ctl_elem_value_alloca(&control);
-
-    snd_ctl_elem_id_set_name(id, name);
-    snd_ctl_elem_info_set_id(info, id);
-
-    int ret = snd_ctl_elem_info(mHandle, info);
-    if (ret < 0) return BAD_VALUE;
-
-    snd_ctl_elem_info_get_id(info, id);
-    snd_ctl_elem_type_t type = snd_ctl_elem_info_get_type(info);
-    unsigned int count = snd_ctl_elem_info_get_count(info);
-    if ((unsigned int)index >= count) return BAD_VALUE;
-
-    snd_ctl_elem_value_set_id(control, id);
-
-    ret = snd_ctl_elem_read(mHandle, control);
-    if (ret < 0) return BAD_VALUE;
-
-    switch (type) {
-        case SND_CTL_ELEM_TYPE_BOOLEAN:
-            value = snd_ctl_elem_value_get_boolean(control, index);
-            break;
-        case SND_CTL_ELEM_TYPE_INTEGER:
-            value = snd_ctl_elem_value_get_integer(control, index);
-            break;
-        case SND_CTL_ELEM_TYPE_INTEGER64:
-            value = snd_ctl_elem_value_get_integer64(control, index);
-            break;
-        case SND_CTL_ELEM_TYPE_ENUMERATED:
-            value = snd_ctl_elem_value_get_enumerated(control, index);
-            break;
-        case SND_CTL_ELEM_TYPE_BYTES:
-            value = snd_ctl_elem_value_get_byte(control, index);
-            break;
-        default:
-            return BAD_VALUE;
-    }
-
     return NO_ERROR;
 }
 
-status_t ALSAControl::set(const char *name, unsigned int value, int index)
-{
-    if (!mHandle) return NO_INIT;
-
-    snd_ctl_elem_id_t *id;
-    snd_ctl_elem_info_t *info;
-    snd_ctl_elem_value_t *control;
-
-    snd_ctl_elem_id_alloca(&id);
-    snd_ctl_elem_info_alloca(&info);
-    snd_ctl_elem_value_alloca(&control);
-
-    snd_ctl_elem_id_set_name(id, name);
-    snd_ctl_elem_info_set_id(info, id);
-
-    int ret = snd_ctl_elem_info(mHandle, info);
-    if (ret < 0) return BAD_VALUE;
-
-    snd_ctl_elem_info_get_id(info, id);
-    snd_ctl_elem_type_t type = snd_ctl_elem_info_get_type(info);
-    unsigned int count = snd_ctl_elem_info_get_count(info);
-    if ((unsigned int)index >= count) return BAD_VALUE;
-
-    if (index == -1)
-        index = 0; // Range over all of them
-    else
-        count = index + 1; // Just do the one specified
-
-    snd_ctl_elem_value_set_id(control, id);
-
-    for (unsigned int i = index; i < count; i++)
-        switch (type) {
-            case SND_CTL_ELEM_TYPE_BOOLEAN:
-                snd_ctl_elem_value_set_boolean(control, i, value);
-                break;
-            case SND_CTL_ELEM_TYPE_INTEGER:
-                snd_ctl_elem_value_set_integer(control, i, value);
-                break;
-            case SND_CTL_ELEM_TYPE_INTEGER64:
-                snd_ctl_elem_value_set_integer64(control, i, value);
-                break;
-            case SND_CTL_ELEM_TYPE_ENUMERATED:
-                snd_ctl_elem_value_set_enumerated(control, i, value);
-                break;
-            case SND_CTL_ELEM_TYPE_BYTES:
-                snd_ctl_elem_value_set_byte(control, i, value);
-                break;
-            default:
-                break;
-        }
-
-    ret = snd_ctl_elem_write(mHandle, control);
-    return (ret < 0) ? BAD_VALUE : NO_ERROR;
-}
-
-// ----------------------------------------------------------------------------
-
-};        // namespace android
+}       // namespace android
diff --git a/AudioHardwareALSA.h b/AudioHardwareALSA.h
index bae5bc7..d2a9bec 100644
--- a/AudioHardwareALSA.h
+++ b/AudioHardwareALSA.h
@@ -18,6 +18,7 @@
 #ifndef ANDROID_AUDIO_HARDWARE_ALSA_H
 #define ANDROID_AUDIO_HARDWARE_ALSA_H
 
+#include <utils/List.h>
 #include <hardware_legacy/AudioHardwareBase.h>
 
 #include <alsa/asoundlib.h>
@@ -26,289 +27,325 @@
 
 namespace android
 {
-    class AudioHardwareALSA;
 
-    /**
-     * The id of acoustics module
-     */
-#define ACOUSTICS_HARDWARE_MODULE_ID "acoustics"
-#define ACOUSTICS_HARDWARE_NAME "name"
-
-    struct acoustic_device_t {
-        hw_device_t common;
-
-        /**
-         * Set the provided acoustics for a particular ALSA pcm device.
-         *
-         * Returns: 0 on succes, error code on failure.
-         */
-        status_t (*set_acoustics)(snd_pcm_t *, AudioSystem::audio_in_acoustics);
-
-        /**
-         * Read callback with PCM data so that filtering may be applied.
-         *
-         * Returns: frames filtered on success, error code on failure.
-         */
-        ssize_t (*filter)(snd_pcm_t *, void *, ssize_t);
-    };
-
-    // ----------------------------------------------------------------------------
-
-    class ALSAMixer
-    {
-        public:
-                                    ALSAMixer();
-            virtual                ~ALSAMixer();
+class AudioHardwareALSA;
 
-            bool                    isValid() { return !!mMixer[SND_PCM_STREAM_PLAYBACK]; }
-            status_t                setMasterVolume(float volume);
-            status_t                setMasterGain(float gain);
+/**
+ * The id of ALSA module
+ */
+#define ALSA_HARDWARE_MODULE_ID "alsa"
+#define ALSA_HARDWARE_NAME      "alsa"
+
+struct alsa_device_t;
+
+struct alsa_handle_t {
+    alsa_device_t *     module;
+    uint32_t            devices;
+    uint32_t            curDev;
+    int                 curMode;
+    snd_pcm_t *         handle;
+    snd_pcm_format_t    format;
+    uint32_t            channels;
+    uint32_t            sampleRate;
+    unsigned int        latency;         // Delay in usec
+    unsigned int        bufferSize;      // Size of sample buffer
+    void *              modPrivate;
+};
 
-            status_t                setVolume(uint32_t device, float volume);
-            status_t                setGain(uint32_t device, float gain);
+typedef List<alsa_handle_t> ALSAHandleList;
 
-            status_t                setCaptureMuteState(uint32_t device, bool state);
-            status_t                getCaptureMuteState(uint32_t device, bool *state);
-            status_t                setPlaybackMuteState(uint32_t device, bool state);
-            status_t                getPlaybackMuteState(uint32_t device, bool *state);
+struct alsa_device_t {
+    hw_device_t common;
 
-        private:
-            snd_mixer_t            *mMixer[SND_PCM_STREAM_LAST+1];
-    };
+    status_t (*init)(alsa_device_t *, ALSAHandleList &);
+    status_t (*open)(alsa_handle_t *, uint32_t, int);
+    status_t (*close)(alsa_handle_t *);
+    status_t (*route)(alsa_handle_t *, uint32_t, int);
+};
 
-    class ALSAControl
-    {
-        public:
-                                    ALSAControl(const char *device = "default");
-            virtual                ~ALSAControl();
+/**
+ * The id of acoustics module
+ */
+#define ACOUSTICS_HARDWARE_MODULE_ID    "acoustics"
+#define ACOUSTICS_HARDWARE_NAME         "acoustics"
 
-            status_t                get(const char *name, unsigned int &value, int index = 0);
-            status_t                set(const char *name, unsigned int value, int index = -1);
+struct acoustic_device_t {
+    hw_device_t common;
 
-        private:
-            snd_ctl_t              *mHandle;
-    };
+    // Required methods...
+    status_t (*use_handle)(acoustic_device_t *, alsa_handle_t *);
+    status_t (*cleanup)(acoustic_device_t *);
 
-    class ALSAStreamOps
-    {
-        protected:
-            friend class AudioStreamOutALSA;
-            friend class AudioStreamInALSA;
-
-            struct StreamDefaults
-            {
-                const char *        devicePrefix;
-                snd_pcm_stream_t    direction;       // playback or capture
-                snd_pcm_format_t    format;
-                int                 channels;
-                uint32_t            sampleRate;
-                unsigned int        latency;         // Delay in usec
-                unsigned int        bufferSize;      // Size of sample buffer
-            };
-
-                                    ALSAStreamOps(AudioHardwareALSA *parent);
-            virtual                ~ALSAStreamOps();
-
-            status_t                set(int format,
-                                        int channels,
-                                        uint32_t rate);
-            virtual uint32_t        sampleRate() const;
-            status_t                sampleRate(uint32_t rate);
-            virtual size_t          bufferSize() const;
-            virtual int             format() const;
-            virtual int             channelCount() const;
-            status_t                channelCount(int channels);
-
-            status_t                open(int mode, uint32_t device);
-            void                    close();
-            status_t                setSoftwareParams();
-            status_t                setPCMFormat(snd_pcm_format_t format);
-            status_t                setHardwareResample(bool resample);
-
-            status_t                setDevice(int mode, uint32_t device);
-
-            const char             *streamName();
-            const char             *deviceName(int mode, uint32_t device);
-
-            void                    setStreamDefaults(StreamDefaults *dev) {
-                mDefaults = dev;
-            }
-
-        private:
-            AudioHardwareALSA      *mParent;
-            snd_pcm_t              *mHandle;
-            snd_pcm_hw_params_t    *mHardwareParams;
-            snd_pcm_sw_params_t    *mSoftwareParams;
-            int                     mMode;
-            uint32_t                mDevice;
-
-            StreamDefaults         *mDefaults;
-
-            Mutex                   mLock;
-            bool                    mPowerLock;
+    status_t (*set_params)(acoustic_device_t *, AudioSystem::audio_in_acoustics, void *);
+
+    // Optional methods...
+    ssize_t (*read)(acoustic_device_t *, void *, size_t);
+    ssize_t (*write)(acoustic_device_t *, const void *, size_t);
+    status_t (*recover)(acoustic_device_t *, int);
+
+    void *              modPrivate;
+};
+
+// ----------------------------------------------------------------------------
+
+class ALSAMixer
+{
+public:
+    ALSAMixer();
+    virtual                ~ALSAMixer();
+
+    bool                    isValid() { return !!mMixer[SND_PCM_STREAM_PLAYBACK]; }
+    status_t                setMasterVolume(float volume);
+    status_t                setMasterGain(float gain);
+
+    status_t                setVolume(uint32_t device, float left, float right);
+    status_t                setGain(uint32_t device, float gain);
+
+    status_t                setCaptureMuteState(uint32_t device, bool state);
+    status_t                getCaptureMuteState(uint32_t device, bool *state);
+    status_t                setPlaybackMuteState(uint32_t device, bool state);
+    status_t                getPlaybackMuteState(uint32_t device, bool *state);
+
+private:
+    snd_mixer_t *           mMixer[SND_PCM_STREAM_LAST+1];
 };
 
-    // ----------------------------------------------------------------------------
+class ALSAControl
+{
+public:
+    ALSAControl(const char *device = "hw:00");
+    virtual                ~ALSAControl();
+
+    status_t                get(const char *name, unsigned int &value, int index = 0);
+    status_t                set(const char *name, unsigned int value, int index = -1);
+
+    status_t                set(const char *name, const char *);
+
+private:
+    snd_ctl_t *             mHandle;
+};
+
+class ALSAStreamOps
+{
+public:
+    ALSAStreamOps(AudioHardwareALSA *parent, alsa_handle_t *handle);
+    virtual            ~ALSAStreamOps();
 
-    class AudioStreamOutALSA : public AudioStreamOut, public ALSAStreamOps
+    status_t            set(int *format, uint32_t *channels, uint32_t *rate);
+
+    status_t            setParameters(const String8& keyValuePairs);
+    String8             getParameters(const String8& keys);
+
+    uint32_t            sampleRate() const;
+    size_t              bufferSize() const;
+    int                 format() const;
+    uint32_t            channels() const;
+
+    status_t            open(int mode);
+    void                close();
+
+protected:
+    friend class AudioHardwareALSA;
+
+    acoustic_device_t *acoustics();
+    ALSAMixer *mixer();
+
+    AudioHardwareALSA *     mParent;
+    alsa_handle_t *         mHandle;
+
+    Mutex                   mLock;
+    bool                    mPowerLock;
+};
+
+// ----------------------------------------------------------------------------
+
+class AudioStreamOutALSA : public AudioStreamOut, public ALSAStreamOps
+{
+public:
+    AudioStreamOutALSA(AudioHardwareALSA *parent, alsa_handle_t *handle);
+    virtual            ~AudioStreamOutALSA();
+
+    virtual uint32_t    sampleRate() const
     {
-        public:
-                                    AudioStreamOutALSA(AudioHardwareALSA *parent);
-            virtual                ~AudioStreamOutALSA();
+        return ALSAStreamOps::sampleRate();
+    }
 
-            status_t                set(int format          = 0,
-                                        int channelCount    = 0,
-                                        uint32_t sampleRate = 0) {
-                return ALSAStreamOps::set(format, channelCount, sampleRate);
-            }
+    virtual size_t      bufferSize() const
+    {
+        return ALSAStreamOps::bufferSize();
+    }
 
-            virtual uint32_t        sampleRate() const
-            {
-                return ALSAStreamOps::sampleRate();
-            }
+    virtual uint32_t    channels() const;
 
-            virtual size_t          bufferSize() const
-            {
-                return ALSAStreamOps::bufferSize();
-            }
+    virtual int         format() const
+    {
+        return ALSAStreamOps::format();
+    }
 
-            virtual int             channelCount() const;
+    virtual uint32_t    latency() const;
 
-            virtual int             format() const
-            {
-                return ALSAStreamOps::format();
-            }
+    virtual ssize_t     write(const void *buffer, size_t bytes);
+    virtual status_t    dump(int fd, const Vector<String16>& args);
 
-            virtual uint32_t        latency() const;
+    status_t            setVolume(float left, float right);
 
-            virtual ssize_t         write(const void *buffer, size_t bytes);
-            virtual status_t        dump(int fd, const Vector<String16>& args);
+    virtual status_t    standby();
 
-            status_t                setVolume(float volume);
+    virtual status_t    setParameters(const String8& keyValuePairs) {
+        return ALSAStreamOps::setParameters(keyValuePairs);
+    }
 
-            virtual status_t        standby();
+    virtual String8     getParameters(const String8& keys) {
+        return ALSAStreamOps::getParameters(keys);
+    }
 
-        protected:
-            friend class AudioHardwareALSA;
+    status_t            open(int mode);
+    status_t            close();
+};
 
-            status_t                setDevice(int mode, uint32_t newDevice);
-    };
+class AudioStreamInALSA : public AudioStreamIn, public ALSAStreamOps
+{
+public:
+    AudioStreamInALSA(AudioHardwareALSA *parent,
+            alsa_handle_t *handle,
+            AudioSystem::audio_in_acoustics audio_acoustics);
+    virtual            ~AudioStreamInALSA();
 
-    class AudioStreamInALSA : public AudioStreamIn, public ALSAStreamOps
+    virtual uint32_t    sampleRate() const
     {
-        public:
-                                    AudioStreamInALSA(AudioHardwareALSA *parent,
-                                                      AudioSystem::audio_in_acoustics acoustics);
-            virtual                ~AudioStreamInALSA();
+        return ALSAStreamOps::sampleRate();
+    }
 
-            status_t                set(int      format       = 0,
-                                        int      channelCount = 0,
-                                        uint32_t sampleRate   = 0) {
-                return ALSAStreamOps::set(format, channelCount, sampleRate);
-            }
+    virtual size_t      bufferSize() const
+    {
+        return ALSAStreamOps::bufferSize();
+    }
 
-            virtual uint32_t        sampleRate() {
-                return ALSAStreamOps::sampleRate();
-            }
+    virtual uint32_t    channels() const
+    {
+        return ALSAStreamOps::channels();
+    }
+
+    virtual int         format() const
+    {
+        return ALSAStreamOps::format();
+    }
 
-            virtual size_t          bufferSize() const
-            {
-                return ALSAStreamOps::bufferSize();
-            }
+    virtual ssize_t     read(void* buffer, ssize_t bytes);
+    virtual status_t    dump(int fd, const Vector<String16>& args);
 
-            virtual int             channelCount() const
-            {
-                return ALSAStreamOps::channelCount();
-            }
+    virtual status_t    setGain(float gain);
 
-            virtual int             format() const
-            {
-                return ALSAStreamOps::format();
-            }
+    virtual status_t    standby();
 
-            virtual ssize_t         read(void* buffer, ssize_t bytes);
-            virtual status_t        dump(int fd, const Vector<String16>& args);
+    virtual status_t    setParameters(const String8& keyValuePairs)
+    {
+        return ALSAStreamOps::setParameters(keyValuePairs);
+    }
 
-            virtual status_t        setGain(float gain);
+    virtual String8     getParameters(const String8& keys)
+    {
+        return ALSAStreamOps::getParameters(keys);
+    }
 
-            virtual status_t        standby();
+    status_t            setAcousticParams(void* params);
 
-        protected:
-            friend class AudioHardwareALSA;
+    status_t            open(int mode);
+    status_t            close();
 
-            status_t                setDevice(int mode, uint32_t newDevice);
+private:
+    AudioSystem::audio_in_acoustics mAcoustics;
+};
 
-        private:
-            AudioSystem::audio_in_acoustics mAcoustics;
-    };
+class AudioHardwareALSA : public AudioHardwareBase
+{
+public:
+    AudioHardwareALSA();
+    virtual            ~AudioHardwareALSA();
 
-    class AudioHardwareALSA : public AudioHardwareBase
+    /**
+     * check to see if the audio hardware interface has been initialized.
+     * return status based on values defined in include/utils/Errors.h
+     */
+    virtual status_t    initCheck();
+
+    /** set the audio volume of a voice call. Range is between 0.0 and 1.0 */
+    virtual status_t    setVoiceVolume(float volume);
+
+    /**
+     * set the audio volume for all audio activities other than voice call.
+     * Range between 0.0 and 1.0. If any value other than NO_ERROR is returned,
+     * the software mixer will emulate this capability.
+     */
+    virtual status_t    setMasterVolume(float volume);
+
+    /**
+     * setMode is called when the audio mode changes. NORMAL mode is for
+     * standard audio playback, RINGTONE when a ringtone is playing, and IN_CALL
+     * when a call is in progress.
+     */
+    virtual status_t    setMode(int mode);
+
+    // mic mute
+    virtual status_t    setMicMute(bool state);
+    virtual status_t    getMicMute(bool* state);
+
+    // set/get global audio parameters
+    //virtual status_t    setParameters(const String8& keyValuePairs);
+    //virtual String8     getParameters(const String8& keys);
+
+    // Returns audio input buffer size according to parameters passed or 0 if one of the
+    // parameters is not supported
+    //virtual size_t    getInputBufferSize(uint32_t sampleRate, int format, int channels);
+
+    /** This method creates and opens the audio hardware output stream */
+    virtual AudioStreamOut* openOutputStream(
+            uint32_t devices,
+            int *format=0,
+            uint32_t *channels=0,
+            uint32_t *sampleRate=0,
+            status_t *status=0);
+    virtual    void        closeOutputStream(AudioStreamOut* out);
+
+    /** This method creates and opens the audio hardware input stream */
+    virtual AudioStreamIn* openInputStream(
+            uint32_t devices,
+            int *format,
+            uint32_t *channels,
+            uint32_t *sampleRate,
+            status_t *status,
+            AudioSystem::audio_in_acoustics acoustics);
+    virtual    void        closeInputStream(AudioStreamIn* in);
+
+    /**This method dumps the state of the audio hardware */
+    //virtual status_t dumpState(int fd, const Vector<String16>& args);
+
+    static AudioHardwareInterface* create();
+
+    int                 mode()
     {
-        public:
-                                    AudioHardwareALSA();
-            virtual                ~AudioHardwareALSA();
-
-            /**
-             * check to see if the audio hardware interface has been initialized.
-             * return status based on values defined in include/utils/Errors.h
-             */
-            virtual status_t        initCheck();
-
-            /** set the audio volume of a voice call. Range is between 0.0 and 1.0 */
-            virtual status_t        setVoiceVolume(float volume);
-
-            /**
-             * set the audio volume for all audio activities other than voice call.
-             * Range between 0.0 and 1.0. If any value other than NO_ERROR is returned,
-             * the software mixer will emulate this capability.
-             */
-            virtual status_t        setMasterVolume(float volume);
-
-            // mic mute
-            virtual status_t        setMicMute(bool state);
-            virtual status_t        getMicMute(bool* state);
-
-            /** This method creates and opens the audio hardware output stream */
-            virtual AudioStreamOut* openOutputStream(
-                int format=0,
-                int channelCount=0,
-                uint32_t sampleRate=0,
-                status_t *status=0);
-
-            /** This method creates and opens the audio hardware input stream */
-            virtual AudioStreamIn*  openInputStream(
-                int inputSource,
-                int format,
-                int channelCount,
-                uint32_t sampleRate,
-                status_t *status,
-                AudioSystem::audio_in_acoustics acoustics);
-
-        protected:
-            /**
-             * doRouting actually initiates the routing. A call to setRouting
-             * or setMode may result in a routing change. The generic logic calls
-             * doRouting when required. If the device has any special requirements these
-             * methods can be overriden.
-             */
-            virtual status_t    doRouting();
-
-            virtual status_t    dump(int fd, const Vector<String16>& args);
-
-            friend class AudioStreamOutALSA;
-            friend class AudioStreamInALSA;
-
-            ALSAMixer          *mMixer;
-            AudioStreamOutALSA *mOutput;
-            AudioStreamInALSA  *mInput;
-
-            acoustic_device_t *mAcousticDevice;
-
-        private:
-            Mutex                   mLock;
-    };
-
-    // ----------------------------------------------------------------------------
+        return mMode;
+    }
+
+protected:
+    virtual status_t    dump(int fd, const Vector<String16>& args);
+
+    friend class AudioStreamOutALSA;
+    friend class AudioStreamInALSA;
+    friend class ALSAStreamOps;
+
+    ALSAMixer *         mMixer;
+
+    alsa_device_t *     mALSADevice;
+    acoustic_device_t * mAcousticDevice;
+
+    ALSAHandleList      mDeviceList;
+
+private:
+    Mutex               mLock;
+};
+
+// ----------------------------------------------------------------------------
 
 };        // namespace android
 #endif    // ANDROID_AUDIO_HARDWARE_ALSA_H
diff --git a/AudioPolicyManagerALSA.cpp b/AudioPolicyManagerALSA.cpp
new file mode 100644
index 0000000..b7808c9
--- /dev/null
+++ b/AudioPolicyManagerALSA.cpp
@@ -0,0 +1,1854 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AudioPolicyManagerALSA"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+#include "AudioPolicyManagerALSA.h"
+#include <media/mediarecorder.h>
+
+namespace android {
+
+
+// ----------------------------------------------------------------------------
+// AudioPolicyInterface implementation
+// ----------------------------------------------------------------------------
+
+
+status_t AudioPolicyManagerALSA::setDeviceConnectionState(AudioSystem::audio_devices device,
+                                                  AudioSystem::device_connection_state state,
+                                                  const char *device_address)
+{
+
+    LOGV("setDeviceConnectionState() device: %x, state %d, address %s", device, state, device_address);
+
+    // connect/disconnect only 1 device at a time
+    if (AudioSystem::popCount(device) != 1) return BAD_VALUE;
+
+    if (strlen(device_address) >= MAX_DEVICE_ADDRESS_LEN) {
+        LOGE("setDeviceConnectionState() invalid address: %s", device_address);
+        return BAD_VALUE;
+    }
+
+    // handle output devices
+    if (AudioSystem::isOutputDevice(device)) {
+
+#ifndef WITH_A2DP
+        if (AudioSystem::isA2dpDevice(device)) {
+            LOGE("setDeviceConnectionState() invalid device: %x", device);
+            return BAD_VALUE;
+        }
+#endif
+
+        switch (state)
+        {
+        // handle output device connection
+        case AudioSystem::DEVICE_STATE_AVAILABLE:
+            if (mAvailableOutputDevices & device) {
+                LOGW("setDeviceConnectionState() device already connected: %x", device);
+                return INVALID_OPERATION;
+            }
+            LOGW_IF((getOutputForDevice((uint32_t)device) != 0), "setDeviceConnectionState(): output using unconnected device %x", device);
+
+            LOGV("setDeviceConnectionState() connecting device %x", device);
+
+            // register new device as available
+            mAvailableOutputDevices |= device;
+
+#ifdef WITH_A2DP
+            // handle A2DP device connection
+            if (AudioSystem::isA2dpDevice(device)) {
+                // when an A2DP device is connected, open an A2DP and a duplicated output
+                LOGV("opening A2DP output for device %s", device_address);
+                AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
+                outputDesc->mDevice = device;
+                mA2dpOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
+                                                        &outputDesc->mSamplingRate,
+                                                        &outputDesc->mFormat,
+                                                        &outputDesc->mChannels,
+                                                        &outputDesc->mLatency,
+                                                        outputDesc->mFlags);
+                if (mA2dpOutput) {
+                    // add A2DP output descriptor
+                    mOutputs.add(mA2dpOutput, outputDesc);
+                    // set initial stream volume for A2DP device
+                    applyStreamVolumes(mA2dpOutput, device);
+                    mDuplicatedOutput = mpClientInterface->openDuplicateOutput(mA2dpOutput, mHardwareOutput);
+                    if (mDuplicatedOutput != 0) {
+                        // If both A2DP and duplicated outputs are open, send device address to A2DP hardware
+                        // interface
+                        AudioParameter param;
+                        param.add(String8("a2dp_sink_address"), String8(device_address));
+                        mpClientInterface->setParameters(mA2dpOutput, param.toString());
+                        mA2dpDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
+
+                        // add duplicated output descriptor
+                        AudioOutputDescriptor *dupOutputDesc = new AudioOutputDescriptor();
+                        dupOutputDesc->mOutput1 = mOutputs.valueFor(mHardwareOutput);
+                        dupOutputDesc->mOutput2 = mOutputs.valueFor(mA2dpOutput);
+                        dupOutputDesc->mSamplingRate = outputDesc->mSamplingRate;
+                        dupOutputDesc->mFormat = outputDesc->mFormat;
+                        dupOutputDesc->mChannels = outputDesc->mChannels;
+                        dupOutputDesc->mLatency = outputDesc->mLatency;
+                        mOutputs.add(mDuplicatedOutput, dupOutputDesc);
+                        applyStreamVolumes(mDuplicatedOutput, device);
+                    } else {
+                        LOGW("getOutput() could not open duplicated output for %d and %d",
+                                mHardwareOutput, mA2dpOutput);
+                        mAvailableOutputDevices &= ~device;
+                        delete outputDesc;
+                        return NO_INIT;
+                    }
+                } else {
+                    LOGW("setDeviceConnectionState() could not open A2DP output for device %x", device);
+                    mAvailableOutputDevices &= ~device;
+                    delete outputDesc;
+                    return NO_INIT;
+                }
+                AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput);
+
+                if (mA2dpDeviceAddress == mScoDeviceAddress) {
+                    // It is normal to suspend twice if we are both in call,
+                    // and have the hardware audio output routed to BT SCO
+                    if (mPhoneState != AudioSystem::MODE_NORMAL) {
+                        mpClientInterface->suspendOutput(mA2dpOutput);
+                    }
+                    if (AudioSystem::isBluetoothScoDevice((AudioSystem::audio_devices)hwOutputDesc->device())) {
+                        mpClientInterface->suspendOutput(mA2dpOutput);
+                    }
+                }
+
+                // move streams pertaining to STRATEGY_MEDIA to the newly opened A2DP output
+                if (getDeviceForStrategy(STRATEGY_MEDIA) & device) {
+                    for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+                        if (getStrategy((AudioSystem::stream_type)i) == STRATEGY_MEDIA) {
+                            mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, mA2dpOutput);
+                            outputDesc->mRefCount[i] = hwOutputDesc->mRefCount[i];
+                            hwOutputDesc->mRefCount[i] = 0;
+                        }
+                    }
+
+                }
+                // move streams pertaining to STRATEGY_DTMF to the newly opened A2DP output
+                if (getDeviceForStrategy(STRATEGY_DTMF) & device) {
+                    for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+                        if (getStrategy((AudioSystem::stream_type)i) == STRATEGY_DTMF) {
+                            mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, mA2dpOutput);
+                            outputDesc->mRefCount[i] = hwOutputDesc->mRefCount[i];
+                            hwOutputDesc->mRefCount[i] = 0;
+                        }
+                    }
+
+                }
+                // move streams pertaining to STRATEGY_SONIFICATION to the newly opened duplicated output
+                if (getDeviceForStrategy(STRATEGY_SONIFICATION) & device) {
+                    for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+                        if (getStrategy((AudioSystem::stream_type)i) == STRATEGY_SONIFICATION) {
+                            mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, mDuplicatedOutput);
+                            outputDesc->mRefCount[i] =
+                                hwOutputDesc->mRefCount[i];
+                            mOutputs.valueFor(mDuplicatedOutput)->mRefCount[i] =
+                                hwOutputDesc->mRefCount[i];
+                        }
+                    }
+                }
+            } else
+#endif
+            // handle wired and SCO device connection (accessed via hardware output)
+            {
+
+                uint32_t newDevice = 0;
+                if (AudioSystem::isBluetoothScoDevice(device)) {
+                    LOGV("setDeviceConnectionState() BT SCO  device, address %s", device_address);
+                    // keep track of SCO device address
+                    mScoDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
+                    // if in call and connecting SCO device, check if we must reroute hardware output
+                    if (mPhoneState == AudioSystem::MODE_IN_CALL &&
+                        getDeviceForStrategy(STRATEGY_PHONE) == device) {
+                        newDevice = device;
+                    } else if (mOutputs.valueFor(mHardwareOutput)->isUsedByStrategy(STRATEGY_DTMF) &&
+                               getDeviceForStrategy(STRATEGY_DTMF) == device) {
+                        newDevice = device;
+                    }
+                    if ((mA2dpDeviceAddress == mScoDeviceAddress) &&
+                        (mPhoneState != AudioSystem::MODE_NORMAL)) {
+                        mpClientInterface->suspendOutput(mA2dpOutput);
+                    }
+                } else if (device == AudioSystem::DEVICE_OUT_WIRED_HEADSET ||
+                           device == AudioSystem::DEVICE_OUT_WIRED_HEADPHONE) {
+                    LOGV("setDeviceConnectionState() wired headset device");
+                    // if connecting a wired headset, we check the following by order of priority
+                    // to request a routing change if necessary:
+                    // 1: we are in call or the strategy phone is active on the hardware output:
+                    //      use device for strategy phone
+                    // 2: the strategy sonification is active on the hardware output:
+                    //      use device for strategy sonification
+                    // 3: the strategy media is active on the hardware output:
+                    //      use device for strategy media
+                    // 4: the strategy DTMF is active on the hardware output:
+                    //      use device for strategy DTMF
+                    if (getDeviceForStrategy(STRATEGY_PHONE) == device &&
+                        (mPhoneState == AudioSystem::MODE_IN_CALL ||
+                        mOutputs.valueFor(mHardwareOutput)->isUsedByStrategy(STRATEGY_PHONE))) {
+                        newDevice = device;
+                    } else if ((getDeviceForStrategy(STRATEGY_SONIFICATION) & device) &&
+                               mOutputs.valueFor(mHardwareOutput)->isUsedByStrategy(STRATEGY_SONIFICATION)){
+                        newDevice = getDeviceForStrategy(STRATEGY_SONIFICATION);
+                    } else if ((getDeviceForStrategy(STRATEGY_MEDIA) == device) &&
+                               mOutputs.valueFor(mHardwareOutput)->isUsedByStrategy(STRATEGY_MEDIA)){
+                        newDevice = device;
+                    } else if (getDeviceForStrategy(STRATEGY_DTMF) == device &&
+                            mOutputs.valueFor(mHardwareOutput)->isUsedByStrategy(STRATEGY_DTMF)) {
+                         newDevice = device;
+                    }
+                } else if (device == AudioSystem::DEVICE_OUT_TTY) {
+                    LOGV("setDeviceConnectionState() tty device");
+                    // if connecting a wired headset, we check the following by order of priority
+                    // to request a routing change if necessary:
+                    // 1: we are in call or the strategy phone is active on the hardware output:
+                    //      use device for strategy phone
+                    if (getDeviceForStrategy(STRATEGY_PHONE) == device &&
+                        (mPhoneState == AudioSystem::MODE_IN_CALL ||
+                        mOutputs.valueFor(mHardwareOutput)->isUsedByStrategy(STRATEGY_PHONE))) {
+                        newDevice = device;
+                    }
+                } else if (device == AudioSystem::DEVICE_OUT_FM_SPEAKER ||
+                           device == AudioSystem::DEVICE_OUT_FM_HEADPHONE) {
+                    LOGV("setDeviceConnectionState() no mic headphone device");
+                    // if connecting a wired headset, we check the following by order of priority
+                    // to request a routing change if necessary:
+                    // 1: the strategy sonification is active on the hardware output:
+                    //      use device for strategy sonification
+                    // 2: the strategy media is active on the hardware output:
+                    //      use device for strategy media
+                    if ((getDeviceForStrategy(STRATEGY_SONIFICATION) & device) &&
+                               mOutputs.valueFor(mHardwareOutput)->isUsedByStrategy(STRATEGY_SONIFICATION)){
+                        newDevice = getDeviceForStrategy(STRATEGY_SONIFICATION);
+                    } else if ((getDeviceForStrategy(STRATEGY_MEDIA) == device) &&
+                               mOutputs.valueFor(mHardwareOutput)->isUsedByStrategy(STRATEGY_MEDIA)){
+                        newDevice = device;
+                    }
+                }
+
+                // request routing change if necessary
+                setOutputDevice(mHardwareOutput, newDevice);
+            }
+            break;
+        // handle output device disconnection
+        case AudioSystem::DEVICE_STATE_UNAVAILABLE: {
+            if (!(mAvailableOutputDevices & device)) {
+                LOGW("setDeviceConnectionState() device not connected: %x", device);
+                return INVALID_OPERATION;
+            }
+
+            uint32_t newDevice = 0;
+            // get usage of disconnected device by all strategies
+            bool wasUsedForMedia = (getDeviceForStrategy(STRATEGY_MEDIA) & device) != 0;
+            bool wasUsedForSonification = (getDeviceForStrategy(STRATEGY_SONIFICATION) & device) != 0;
+            bool wasUsedforPhone = (getDeviceForStrategy(STRATEGY_PHONE) & device) != 0;
+            bool wasUsedforDtmf = (getDeviceForStrategy(STRATEGY_DTMF) & device) != 0;
+            LOGV("setDeviceConnectionState() disconnecting device %x used by media %d, sonification %d, phone %d",
+                    device, wasUsedForMedia, wasUsedForSonification, wasUsedforPhone);
+            // remove device from available output devices
+            mAvailableOutputDevices &= ~device;
+
+#ifdef WITH_A2DP
+            // handle A2DP device disconnection
+            if (AudioSystem::isA2dpDevice(device)) {
+                if (mA2dpOutput == 0 || mDuplicatedOutput == 0) {
+                    LOGW("setDeviceConnectionState() disconnecting A2DP and no A2DP output!");
+                    mAvailableOutputDevices |= device;
+                    return INVALID_OPERATION;
+                }
+
+                if (mA2dpDeviceAddress != device_address) {
+                    LOGW("setDeviceConnectionState() disconnecting unknow A2DP sink address %s", device_address);
+                    mAvailableOutputDevices |= device;
+                    return INVALID_OPERATION;
+                }
+
+                AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput);
+                AudioOutputDescriptor *a2dpOutputDesc = mOutputs.valueFor(mA2dpOutput);
+
+                // mute media during 2 seconds to avoid outputing sound on hardware output while music stream
+                // is switched from A2DP output and before music is paused by music application
+                setStrategyMute(STRATEGY_MEDIA, true, mHardwareOutput);
+                setStrategyMute(STRATEGY_MEDIA, false, mHardwareOutput, 2000);
+
+                // If the A2DP device was used by DTMF strategy, move all streams pertaining to DTMF strategy to
+                // hardware output
+                if (wasUsedforDtmf) {
+                    for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+                        if (getStrategy((AudioSystem::stream_type)i) == STRATEGY_DTMF) {
+                            mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, mHardwareOutput);
+                            hwOutputDesc->changeRefCount((AudioSystem::stream_type)i,
+                                    a2dpOutputDesc->mRefCount[i]);
+                        }
+                    }
+                    if (a2dpOutputDesc->isUsedByStrategy(STRATEGY_DTMF)) {
+                        newDevice = getDeviceForStrategy(STRATEGY_DTMF);
+                    }
+                }
+
+                // If the A2DP device was used by media strategy, move all streams pertaining to media strategy to
+                // hardware output
+                if (wasUsedForMedia) {
+                    for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+                        if (getStrategy((AudioSystem::stream_type)i) == STRATEGY_MEDIA) {
+                            mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, mHardwareOutput);
+                            hwOutputDesc->changeRefCount((AudioSystem::stream_type)i,
+                                    a2dpOutputDesc->mRefCount[i]);
+                        }
+                    }
+                    if (a2dpOutputDesc->isUsedByStrategy(STRATEGY_MEDIA)) {
+                        newDevice = getDeviceForStrategy(STRATEGY_MEDIA);
+                    }
+                }
+
+                // If the A2DP device was used by sonification strategy, move all streams pertaining to
+                // sonification strategy to hardware output.
+                // Note that newDevice is overwritten here giving sonification strategy a higher priority than
+                // media strategy.
+                if (wasUsedForSonification) {
+                    for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+                        if (getStrategy((AudioSystem::stream_type)i) == STRATEGY_SONIFICATION) {
+                            mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, mHardwareOutput);
+                        }
+                    }
+                    if (a2dpOutputDesc->isUsedByStrategy(STRATEGY_SONIFICATION)) {
+                        newDevice = getDeviceForStrategy(STRATEGY_SONIFICATION);
+                    }
+                }
+
+                // close A2DP and duplicated outputs
+                AudioParameter param;
+                param.add(String8("closing"), String8("true"));
+                mpClientInterface->setParameters(mA2dpOutput, param.toString());
+
+                LOGW("setDeviceConnectionState() closing A2DP and duplicated output!");
+                mpClientInterface->closeOutput(mDuplicatedOutput);
+                delete mOutputs.valueFor(mDuplicatedOutput);
+                mOutputs.removeItem(mDuplicatedOutput);
+                mDuplicatedOutput = 0;
+                mpClientInterface->closeOutput(mA2dpOutput);
+                delete mOutputs.valueFor(mA2dpOutput);
+                mOutputs.removeItem(mA2dpOutput);
+                mA2dpOutput = 0;
+            } else
+#endif
+            {
+                if (AudioSystem::isBluetoothScoDevice(device)) {
+                    // handle SCO device disconnection
+                    if (wasUsedforPhone &&
+                        mPhoneState == AudioSystem::MODE_IN_CALL) {
+                        // if in call, find new suitable device for phone strategy
+                        newDevice = getDeviceForStrategy(STRATEGY_PHONE);
+                    } else if (wasUsedforDtmf &&
+                               mOutputs.valueFor(mHardwareOutput)->isUsedByStrategy(STRATEGY_DTMF)) {
+                        newDevice = getDeviceForStrategy(STRATEGY_DTMF);
+                    }
+                    if ((mA2dpDeviceAddress == mScoDeviceAddress) &&
+                        (mPhoneState != AudioSystem::MODE_NORMAL)) {
+                        mpClientInterface->restoreOutput(mA2dpOutput);
+                    }
+                } else if (device == AudioSystem::DEVICE_OUT_WIRED_HEADSET ||
+                           device == AudioSystem::DEVICE_OUT_WIRED_HEADPHONE) {
+                    // if disconnecting a wired headset, we check the following by order of priority
+                    // to request a routing change if necessary:
+                    // 1: we are in call or the strategy phone is active on the hardware output:
+                    //      use device for strategy phone
+                    // 2: the strategy sonification is active on the hardware output:
+                    //      use device for strategy sonification
+                    // 3: the strategy media is active on the hardware output:
+                    //      use device for strategy media
+                    // 4: the strategy DTMF is active on the hardware output:
+                    //      use device for strategy DTMF
+                    if (wasUsedforPhone &&
+                        (mPhoneState == AudioSystem::MODE_IN_CALL ||
+                         mOutputs.valueFor(mHardwareOutput)->isUsedByStrategy(STRATEGY_PHONE))) {
+                        newDevice = getDeviceForStrategy(STRATEGY_PHONE);
+                    } else if (wasUsedForSonification &&
+                               mOutputs.valueFor(mHardwareOutput)->isUsedByStrategy(STRATEGY_SONIFICATION)){
+                        newDevice = getDeviceForStrategy(STRATEGY_SONIFICATION);
+                    } else if (wasUsedForMedia &&
+                               mOutputs.valueFor(mHardwareOutput)->isUsedByStrategy(STRATEGY_MEDIA)){
+                        newDevice = getDeviceForStrategy(STRATEGY_MEDIA);
+                    } else if (wasUsedforDtmf &&
+                               mOutputs.valueFor(mHardwareOutput)->isUsedByStrategy(STRATEGY_DTMF)){
+                        newDevice = getDeviceForStrategy(STRATEGY_DTMF);
+                    }
+                } else if (device == AudioSystem::DEVICE_OUT_TTY) {
+                    LOGV("setDeviceConnectionState() tty device");
+                    if (wasUsedforPhone &&
+                        (mPhoneState == AudioSystem::MODE_IN_CALL ||
+                         mOutputs.valueFor(mHardwareOutput)->isUsedByStrategy(STRATEGY_PHONE))) {
+                        newDevice = getDeviceForStrategy(STRATEGY_PHONE);
+                    }
+                } else if (device == AudioSystem::DEVICE_OUT_FM_SPEAKER ||
+                           device == AudioSystem::DEVICE_OUT_FM_HEADPHONE) {
+                    LOGV("setDeviceConnectionState() no mic headphone device");
+                    if (wasUsedForSonification &&
+                        mOutputs.valueFor(mHardwareOutput)->isUsedByStrategy(STRATEGY_SONIFICATION)){
+                        newDevice = getDeviceForStrategy(STRATEGY_SONIFICATION);
+                    } else if (wasUsedForMedia &&
+                               mOutputs.valueFor(mHardwareOutput)->isUsedByStrategy(STRATEGY_MEDIA)){
+                        newDevice = getDeviceForStrategy(STRATEGY_MEDIA);
+                    }
+                }
+            }
+            // request routing change if necessary
+            setOutputDevice(mHardwareOutput, newDevice);
+
+            // clear A2DP and SCO device address if necessary
+#ifdef WITH_A2DP
+            if (AudioSystem::isA2dpDevice(device)) {
+                mA2dpDeviceAddress = "";
+            }
+#endif
+            if (AudioSystem::isBluetoothScoDevice(device)) {
+                mScoDeviceAddress = "";
+            }
+            } break;
+
+        default:
+            LOGE("setDeviceConnectionState() invalid state: %x", state);
+            return BAD_VALUE;
+        }
+
+        if (device == AudioSystem::DEVICE_OUT_WIRED_HEADSET) {
+            device = AudioSystem::DEVICE_IN_WIRED_HEADSET;
+        } else if (device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO ||
+                   device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET ||
+                   device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT) {
+            device = AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+        } else {
+            return NO_ERROR;
+        }
+    }
+    // handle input devices
+    if (AudioSystem::isInputDevice(device)) {
+
+        switch (state)
+        {
+        // handle input device connection
+        case AudioSystem::DEVICE_STATE_AVAILABLE: {
+            if (mAvailableInputDevices & device) {
+                LOGW("setDeviceConnectionState() device already connected: %d", device);
+                return INVALID_OPERATION;
+            }
+            mAvailableInputDevices |= device;
+            }
+            break;
+
+        // handle input device disconnection
+        case AudioSystem::DEVICE_STATE_UNAVAILABLE: {
+            if (!(mAvailableInputDevices & device)) {
+                LOGW("setDeviceConnectionState() device not connected: %d", device);
+                return INVALID_OPERATION;
+            }
+            mAvailableInputDevices &= ~device;
+            } break;
+
+        default:
+            LOGE("setDeviceConnectionState() invalid state: %x", state);
+            return BAD_VALUE;
+        }
+
+        audio_io_handle_t activeInput = getActiveInput();
+        if (activeInput != 0) {
+            AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput);
+            uint32_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
+            if (newDevice != inputDesc->mDevice) {
+                LOGV("setDeviceConnectionState() changing device from %x to %x for input %d",
+                        inputDesc->mDevice, newDevice, activeInput);
+                inputDesc->mDevice = newDevice;
+                AudioParameter param = AudioParameter();
+                param.addInt(String8(AudioParameter::keyRouting), (int)newDevice);
+                mpClientInterface->setParameters(activeInput, param.toString());
+            }
+        }
+
+        return NO_ERROR;
+    }
+
+    LOGW("setDeviceConnectionState() invalid device: %x", device);
+    return BAD_VALUE;
+}
+
+AudioSystem::device_connection_state AudioPolicyManagerALSA::getDeviceConnectionState(AudioSystem::audio_devices device,
+                                                  const char *device_address)
+{
+    AudioSystem::device_connection_state state = AudioSystem::DEVICE_STATE_UNAVAILABLE;
+    String8 address = String8(device_address);
+    if (AudioSystem::isOutputDevice(device)) {
+        if (device & mAvailableOutputDevices) {
+#ifdef WITH_A2DP
+            if (AudioSystem::isA2dpDevice(device) &&
+                address != "" && mA2dpDeviceAddress != address) {
+                return state;
+            }
+#endif
+            if (AudioSystem::isBluetoothScoDevice(device) &&
+                address != "" && mScoDeviceAddress != address) {
+                return state;
+            }
+            state = AudioSystem::DEVICE_STATE_AVAILABLE;
+        }
+    } else if (AudioSystem::isInputDevice(device)) {
+        if (device & mAvailableInputDevices) {
+            state = AudioSystem::DEVICE_STATE_AVAILABLE;
+        }
+    }
+
+    return state;
+}
+
+void AudioPolicyManagerALSA::setPhoneState(int state)
+{
+    LOGV("setPhoneState() state %d", state);
+    uint32_t newDevice = 0;
+    if (state < 0 || state >= AudioSystem::NUM_MODES) {
+        LOGW("setPhoneState() invalid state %d", state);
+        return;
+    }
+
+    if (state == mPhoneState ) {
+        LOGW("setPhoneState() setting same state %d", state);
+        return;
+    }
+
+    // if leaving call state, handle special case of active streams
+    // pertaining to sonification strategy see handleIncallSonification()
+    if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+        LOGV("setPhoneState() in call state management: new state is %d", state);
+        for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+            handleIncallSonification(stream, false, true);
+        }
+    }
+
+    // store previous phone state for management of sonification strategy below
+    int oldState = mPhoneState;
+    uint32_t oldDtmfDevice = getDeviceForStrategy(STRATEGY_DTMF);
+    uint32_t oldSonificationDevice = getDeviceForStrategy(STRATEGY_SONIFICATION) & ~AudioSystem::DEVICE_OUT_SPEAKER;
+    mPhoneState = state;
+    bool force = false;
+    // check if a routing change is required for hardware output in the following
+    // order of priority:
+    // 1: a stream pertaining to sonification strategy is active
+    // 2: new state is incall
+    // 3: a stream pertaining to media strategy is active
+    // 4: a stream pertaining to DTMF strategy is active
+    if (mOutputs.valueFor(mHardwareOutput)->isUsedByStrategy(STRATEGY_SONIFICATION)) {
+        newDevice = getDeviceForStrategy(STRATEGY_SONIFICATION);
+    } else if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+        newDevice = getDeviceForStrategy(STRATEGY_PHONE);
+        // force routing command to audio hardware when starting call
+        // even if no device change is needed
+        force = true;
+    } else if (mOutputs.valueFor(mHardwareOutput)->isUsedByStrategy(STRATEGY_MEDIA)) {
+        newDevice = getDeviceForStrategy(STRATEGY_MEDIA);
+    } else if (mOutputs.valueFor(mHardwareOutput)->isUsedByStrategy(STRATEGY_DTMF)) {
+        newDevice = getDeviceForStrategy(STRATEGY_DTMF);
+    }
+
+
+    if (mA2dpOutput != 0) {
+        // If entering or exiting in call state, switch DTMF streams to/from A2DP output
+        // if necessary
+        uint32_t newDtmfDevice = getDeviceForStrategy(STRATEGY_DTMF);
+        uint32_t newSonificationDevice = getDeviceForStrategy(STRATEGY_SONIFICATION) & ~AudioSystem::DEVICE_OUT_SPEAKER;
+        if (state == AudioSystem::MODE_IN_CALL) {  // entering in call mode
+            // move DTMF streams from A2DP output to hardware output if necessary
+            if (AudioSystem::isA2dpDevice((AudioSystem::audio_devices)oldDtmfDevice) &&
+                !AudioSystem::isA2dpDevice((AudioSystem::audio_devices)newDtmfDevice)) {
+                for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+                    if (getStrategy((AudioSystem::stream_type)i) == STRATEGY_DTMF) {
+                        mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, mHardwareOutput);
+                        int refCount = mOutputs.valueFor(mA2dpOutput)->mRefCount[i];
+                        mOutputs.valueFor(mHardwareOutput)->changeRefCount((AudioSystem::stream_type)i,
+                                refCount);
+                        mOutputs.valueFor(mA2dpOutput)->changeRefCount((AudioSystem::stream_type)i,-refCount);
+                    }
+                }
+                if (newDevice == 0 && mOutputs.valueFor(mA2dpOutput)->isUsedByStrategy(STRATEGY_DTMF)) {
+                    newDevice = newDtmfDevice;
+                }
+            }
+            // move SONIFICATION streams from duplicated output to hardware output if necessary
+            if (AudioSystem::isA2dpDevice((AudioSystem::audio_devices)oldSonificationDevice) &&
+                !AudioSystem::isA2dpDevice((AudioSystem::audio_devices)newSonificationDevice)) {
+                for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+                    if (getStrategy((AudioSystem::stream_type)i) == STRATEGY_SONIFICATION) {
+                        mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, mHardwareOutput);
+                        int refCount = mOutputs.valueFor(mDuplicatedOutput)->mRefCount[i];
+                        mOutputs.valueFor(mHardwareOutput)->changeRefCount((AudioSystem::stream_type)i,
+                                refCount);
+                        mOutputs.valueFor(mDuplicatedOutput)->changeRefCount((AudioSystem::stream_type)i,-refCount);
+                    }
+                }
+            }
+        } else {  // exiting in call mode
+            // move DTMF streams from hardware output to A2DP output if necessary
+            if (!AudioSystem::isA2dpDevice((AudioSystem::audio_devices)oldDtmfDevice) &&
+                AudioSystem::isA2dpDevice((AudioSystem::audio_devices)newDtmfDevice)) {
+                for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+                    if (getStrategy((AudioSystem::stream_type)i) == STRATEGY_DTMF) {
+                        mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, mA2dpOutput);
+                        int refCount = mOutputs.valueFor(mHardwareOutput)->mRefCount[i];
+                        mOutputs.valueFor(mA2dpOutput)->changeRefCount((AudioSystem::stream_type)i, refCount);
+                        mOutputs.valueFor(mHardwareOutput)->changeRefCount((AudioSystem::stream_type)i, -refCount);
+                    }
+                }
+            }
+            // move SONIFICATION streams from hardware output to A2DP output if necessary
+            if (!AudioSystem::isA2dpDevice((AudioSystem::audio_devices)oldSonificationDevice) &&
+                AudioSystem::isA2dpDevice((AudioSystem::audio_devices)newSonificationDevice)) {
+                for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+                    if (getStrategy((AudioSystem::stream_type)i) == STRATEGY_SONIFICATION) {
+                        mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, mDuplicatedOutput);
+                        int refCount = mOutputs.valueFor(mHardwareOutput)->mRefCount[i];
+                        mOutputs.valueFor(mDuplicatedOutput)->changeRefCount((AudioSystem::stream_type)i, refCount);
+                        mOutputs.valueFor(mHardwareOutput)->changeRefCount((AudioSystem::stream_type)i, -refCount);
+                    }
+                }
+            }
+        }
+        // suspend A2DP output if SCO device address is the same as A2DP device address.
+        // no need to check that a SCO device is actually connected as mScoDeviceAddress == ""
+        // if none is connected and the test below will fail.
+        if (mA2dpDeviceAddress == mScoDeviceAddress) {
+            if (oldState == AudioSystem::MODE_NORMAL) {
+                mpClientInterface->suspendOutput(mA2dpOutput);
+            } else if (state == AudioSystem::MODE_NORMAL) {
+                mpClientInterface->restoreOutput(mA2dpOutput);
+            }
+        }
+    }
+
+    // force routing command to audio hardware when ending call
+    // even if no device change is needed
+    if (oldState == AudioSystem::MODE_IN_CALL) {
+        if (newDevice == 0) {
+            newDevice = mOutputs.valueFor(mHardwareOutput)->device();
+        }
+        force = true;
+    }
+    // change routing is necessary
+    setOutputDevice(mHardwareOutput, newDevice, force);
+
+    // if entering in call state, handle special case of active streams
+    // pertaining to sonification strategy see handleIncallSonification()
+    if (state == AudioSystem::MODE_IN_CALL) {
+        LOGV("setPhoneState() in call state management: new state is %d", state);
+        for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+            handleIncallSonification(stream, true, true);
+        }
+    }
+}
+
+void AudioPolicyManagerALSA::setRingerMode(uint32_t mode, uint32_t mask)
+{
+    LOGV("setRingerMode() mode %x, mask %x", mode, mask);
+
+    mRingerMode = mode;
+}
+
+void AudioPolicyManagerALSA::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config)
+{
+    LOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState);
+
+    switch(usage) {
+    case AudioSystem::FOR_COMMUNICATION:
+        if (config != AudioSystem::FORCE_SPEAKER && config != AudioSystem::FORCE_BT_SCO &&
+            config != AudioSystem::FORCE_NONE) {
+            LOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config);
+            return;
+        }
+        mForceUse[usage] = config;
+        // update hardware output routing immediately if in call, or if there is an active
+        // VOICE_CALL stream, as would be the case with an application that uses this stream
+        // for it to behave like in a telephony app (e.g. voicemail app that plays audio files
+        // streamed or downloaded to the device)
+        if ((mPhoneState == AudioSystem::MODE_IN_CALL) ||
+                (mOutputs.valueFor(mHardwareOutput)->isUsedByStream(AudioSystem::VOICE_CALL))) {
+            uint32_t device = getDeviceForStrategy(STRATEGY_PHONE);
+            setOutputDevice(mHardwareOutput, device);
+        }
+        break;
+    case AudioSystem::FOR_MEDIA:
+        if (config != AudioSystem::FORCE_HEADPHONES && config != AudioSystem::FORCE_BT_A2DP &&
+            config != AudioSystem::FORCE_WIRED_ACCESSORY && config != AudioSystem::FORCE_NONE) {
+            LOGW("setForceUse() invalid config %d for FOR_MEDIA", config);
+            return;
+        }
+        mForceUse[usage] = config;
+        break;
+    case AudioSystem::FOR_RECORD:
+        if (config != AudioSystem::FORCE_BT_SCO && config != AudioSystem::FORCE_WIRED_ACCESSORY &&
+            config != AudioSystem::FORCE_NONE) {
+            LOGW("setForceUse() invalid config %d for FOR_RECORD", config);
+            return;
+        }
+        mForceUse[usage] = config;
+        break;
+    default:
+        LOGW("setForceUse() invalid usage %d", usage);
+        break;
+    }
+}
+
+AudioSystem::forced_config AudioPolicyManagerALSA::getForceUse(AudioSystem::force_use usage)
+{
+    return mForceUse[usage];
+}
+
+void AudioPolicyManagerALSA::setSystemProperty(const char* property, const char* value)
+{
+    LOGV("setSystemProperty() property %s, value %s", property, value);
+    if (strcmp(property, "ro.camera.sound.forced") == 0) {
+        if (atoi(value)) {
+            LOGV("ENFORCED_AUDIBLE cannot be muted");
+            mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = false;
+        } else {
+            LOGV("ENFORCED_AUDIBLE can be muted");
+            mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = true;
+        }
+    }
+}
+
+audio_io_handle_t AudioPolicyManagerALSA::getOutput(AudioSystem::stream_type stream,
+                                    uint32_t samplingRate,
+                                    uint32_t format,
+                                    uint32_t channels,
+                                    AudioSystem::output_flags flags)
+{
+    audio_io_handle_t output = 0;
+    uint32_t latency = 0;
+    routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
+    uint32_t device = getDeviceForStrategy(strategy);
+    LOGV("getOutput() stream %d, samplingRate %d, format %d, channels %x, flags %x", stream, samplingRate, format, channels, flags);
+
+
+    // open a direct output if:
+    // 1 a direct output is explicitely requested
+    // 2 the audio format is compressed
+    if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
+         (format !=0 && !AudioSystem::isLinearPCM(format))) {
+
+        LOGV("getOutput() opening direct output device %x", device);
+        AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
+        outputDesc->mDevice = device;
+        outputDesc->mSamplingRate = samplingRate;
+        outputDesc->mFormat = format;
+        outputDesc->mChannels = channels;
+        outputDesc->mLatency = 0;
+        outputDesc->mFlags = (AudioSystem::output_flags)(flags | AudioSystem::OUTPUT_FLAG_DIRECT);
+        outputDesc->mRefCount[stream] = 1;
+        output = mpClientInterface->openOutput(&outputDesc->mDevice,
+                                        &outputDesc->mSamplingRate,
+                                        &outputDesc->mFormat,
+                                        &outputDesc->mChannels,
+                                        &outputDesc->mLatency,
+                                        outputDesc->mFlags);
+
+        // only accept an output with the requeted parameters
+        if ((samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) ||
+            (format != 0 && format != outputDesc->mFormat) ||
+            (channels != 0 && channels != outputDesc->mChannels)) {
+            LOGV("getOutput() failed opening direct output: samplingRate %d, format %d, channels %d",
+                    samplingRate, format, channels);
+            mpClientInterface->closeOutput(output);
+            delete outputDesc;
+            return 0;
+        }
+        mOutputs.add(output, outputDesc);
+        return output;
+    }
+
+    if (channels != 0 && channels != AudioSystem::CHANNEL_OUT_MONO &&
+        channels != AudioSystem::CHANNEL_OUT_STEREO) {
+        return 0;
+    }
+    // open a non direct output
+
+    // get which output is suitable for the specified stream. The actual routing change will happen
+    // when startOutput() will be called
+    uint32_t device2 = device & ~AudioSystem::DEVICE_OUT_SPEAKER;
+    if (AudioSystem::popCount((AudioSystem::audio_devices)device) == 2) {
+#ifdef WITH_A2DP
+        if (AudioSystem::isA2dpDevice((AudioSystem::audio_devices)device2)) {
+            // if playing on 2 devices among which one is A2DP, use duplicated output
+            LOGV("getOutput() using duplicated output");
+            LOGW_IF((mA2dpOutput == 0), "getOutput() A2DP device in multiple %x selected but A2DP output not opened", device);
+            output = mDuplicatedOutput;
+        } else
+#endif
+        {
+            // if playing on 2 devices among which none is A2DP, use hardware output
+            output = mHardwareOutput;
+        }
+        LOGV("getOutput() using output %d for 2 devices %x", output, device);
+    } else {
+#ifdef WITH_A2DP
+        if (AudioSystem::isA2dpDevice((AudioSystem::audio_devices)device2)) {
+            // if playing on A2DP device, use a2dp output
+            LOGW_IF((mA2dpOutput == 0), "getOutput() A2DP device %x selected but A2DP output not opened", device);
+            output = mA2dpOutput;
+        } else
+#endif
+        {
+            // if playing on not A2DP device, use hardware output
+            output = mHardwareOutput;
+        }
+    }
+
+
+    LOGW_IF((output ==0), "getOutput() could not find output for stream %d, samplingRate %d, format %d, channels %x, flags %x",
+                stream, samplingRate, format, channels, flags);
+
+    return output;
+}
+
+status_t AudioPolicyManagerALSA::startOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
+{
+    LOGV("startOutput() output %d, stream %d", output, stream);
+    ssize_t index = mOutputs.indexOfKey(output);
+    if (index < 0) {
+        LOGW("startOutput() unknow output %d", output);
+        return BAD_VALUE;
+    }
+
+    AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+    routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
+    uint32_t device = getDeviceForStrategy(strategy);
+
+    if (!outputDesc->isUsedByStrategy(strategy)) {
+        // if the stream started is the first active stream in its strategy, check if routing change
+        // must be done on hardware output
+        uint32_t newDevice = 0;
+        if (AudioSystem::popCount((AudioSystem::audio_devices)device) == 2) {
+#ifdef WITH_A2DP
+            uint32_t device2 = device & ~AudioSystem::DEVICE_OUT_SPEAKER;
+            if (AudioSystem::isA2dpDevice((AudioSystem::audio_devices)device2)) {
+                // if one device is A2DP, selected the second device for hardware output
+                device &= ~device2;
+            } else
+#endif
+            {
+                // we only support speaker + headset and speaker + headphone combinations on hardware output.
+                // other combinations will leave device = 0 and no routing will happen.
+                if (device != (AudioSystem::DEVICE_OUT_SPEAKER | AudioSystem::DEVICE_OUT_WIRED_HEADSET) &&
+                    device != (AudioSystem::DEVICE_OUT_SPEAKER | AudioSystem::DEVICE_OUT_WIRED_HEADPHONE)) {
+                    device = AudioSystem::DEVICE_OUT_SPEAKER;
+                }
+            }
+        }
+
+        // By order of priority
+        // 1 apply routing for phone strategy in any case
+        // 2 apply routing for notification strategy if no stream pertaining to
+        //   phone strategies is playing
+        // 3 apply routing for media strategy is not incall and neither phone nor sonification
+        //   strategies is active.
+        // 4 apply routing for DTMF strategy if no stream pertaining to
+        //   neither phone, sonification nor media strategy is playing
+        if (strategy == STRATEGY_PHONE) {
+            newDevice = device;
+        } else if (!mOutputs.valueFor(mHardwareOutput)->isUsedByStrategy(STRATEGY_PHONE)) {
+            if (strategy == STRATEGY_SONIFICATION) {
+                newDevice = device;
+            } else if (!mOutputs.valueFor(mHardwareOutput)->isUsedByStrategy(STRATEGY_SONIFICATION)) {
+                if (strategy == STRATEGY_MEDIA) {
+                    newDevice = device;
+                } else if (!mOutputs.valueFor(mHardwareOutput)->isUsedByStrategy(STRATEGY_MEDIA)) {
+                    // strategy == STRATEGY_DTMF
+                    newDevice = device;
+                }
+            }
+        }
+
+        // TODO: maybe mute stream is selected device was refused
+        setOutputDevice(mHardwareOutput, newDevice);
+    }
+
+    // incremenent usage count for this stream on the requested output:
+    // NOTE that the usage count is the same for duplicated output and hardware output which is
+    // necassary for a correct control of hardware output routing by startOutput() and stopOutput()
+    outputDesc->changeRefCount(stream, 1);
+
+    // handle special case for sonification while in call
+    if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+        handleIncallSonification(stream, true, false);
+    }
+
+    // apply volume rules for current stream and device if necessary
+    checkAndSetVolume(stream, mStreams[stream].mIndexCur, output, outputDesc->device());
+
+    return NO_ERROR;
+}
+
+status_t AudioPolicyManagerALSA::stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
+{
+    LOGV("stopOutput() output %d, stream %d", output, stream);
+    ssize_t index = mOutputs.indexOfKey(output);
+    if (index < 0) {
+        LOGW("stopOutput() unknow output %d", output);
+        return BAD_VALUE;
+    }
+
+    AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+    routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
+
+    // handle special case for sonification while in call
+    if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+        handleIncallSonification(stream, false, false);
+    }
+
+    if (outputDesc->isUsedByStrategy(strategy)) {
+        // decrement usage count of this stream on the output
+        outputDesc->changeRefCount(stream, -1);
+        if (!outputDesc->isUsedByStrategy(strategy)) {
+            // if the stream is the last of its strategy to use this output, change routing
+            // in the following order or priority:
+            // PHONE > SONIFICATION > MEDIA > DTMF
+            uint32_t newDevice = 0;
+            if (outputDesc->isUsedByStrategy(STRATEGY_PHONE)) {
+                newDevice = getDeviceForStrategy(STRATEGY_PHONE);
+            } else if (outputDesc->isUsedByStrategy(STRATEGY_SONIFICATION)) {
+                newDevice = getDeviceForStrategy(STRATEGY_SONIFICATION);
+            } else if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+                newDevice = getDeviceForStrategy(STRATEGY_PHONE);
+            } else if (outputDesc->isUsedByStrategy(STRATEGY_MEDIA)) {
+                newDevice = getDeviceForStrategy(STRATEGY_MEDIA);
+            } else if (outputDesc->isUsedByStrategy(STRATEGY_DTMF)) {
+                newDevice = getDeviceForStrategy(STRATEGY_DTMF);
+            }
+
+            // apply routing change if necessary.
+            // insert a delay of 2 times the audio hardware latency to ensure PCM
+            // buffers in audio flinger and audio hardware are emptied before the
+            // routing change is executed.
+            setOutputDevice(mHardwareOutput, newDevice, false, mOutputs.valueFor(mHardwareOutput)->mLatency*2);
+        }
+        // store time at which the last music track was stopped - see computeVolume()
+        if (stream == AudioSystem::MUSIC) {
+            mMusicStopTime = systemTime();
+        }
+        return NO_ERROR;
+    } else {
+        LOGW("stopOutput() refcount is already 0 for output %d", output);
+        return INVALID_OPERATION;
+    }
+}
+
+void AudioPolicyManagerALSA::releaseOutput(audio_io_handle_t output)
+{
+    LOGV("releaseOutput() %d", output);
+    ssize_t index = mOutputs.indexOfKey(output);
+    if (index < 0) {
+        LOGW("releaseOutput() releasing unknown output %d", output);
+        return;
+    }
+    if (mOutputs.valueAt(index)->mFlags & AudioSystem::OUTPUT_FLAG_DIRECT) {
+        mpClientInterface->closeOutput(output);
+        delete mOutputs.valueAt(index);
+        mOutputs.removeItem(output);
+    }
+}
+
+audio_io_handle_t AudioPolicyManagerALSA::getInput(int inputSource,
+                                    uint32_t samplingRate,
+                                    uint32_t format,
+                                    uint32_t channels,
+                                    AudioSystem::audio_in_acoustics acoustics)
+{
+    audio_io_handle_t input = 0;
+    uint32_t device = getDeviceForInputSource(inputSource);
+
+    LOGV("getInput() inputSource %d, samplingRate %d, format %d, channels %x, acoustics %x", inputSource, samplingRate, format, channels, acoustics);
+
+    if (device == 0) {
+        return 0;
+    }
+
+    // adapt channel selection to input source
+    switch(inputSource) {
+    case AUDIO_SOURCE_VOICE_UPLINK:
+        channels = AudioSystem::CHANNEL_IN_VOICE_UPLINK;
+        break;
+    case AUDIO_SOURCE_VOICE_DOWNLINK:
+        channels = AudioSystem::CHANNEL_IN_VOICE_DNLINK;
+        break;
+    case AUDIO_SOURCE_VOICE_CALL:
+        channels = (AudioSystem::CHANNEL_IN_VOICE_UPLINK | AudioSystem::CHANNEL_IN_VOICE_DNLINK);
+        break;
+    default:
+        break;
+    }
+
+    AudioInputDescriptor *inputDesc = new AudioInputDescriptor();
+
+    inputDesc->mInputSource = inputSource;
+    inputDesc->mDevice = device;
+    inputDesc->mSamplingRate = samplingRate;
+    inputDesc->mFormat = format;
+    inputDesc->mChannels = channels;
+    inputDesc->mAcoustics = acoustics;
+    inputDesc->mRefCount = 0;
+    input = mpClientInterface->openInput(&inputDesc->mDevice,
+                                    &inputDesc->mSamplingRate,
+                                    &inputDesc->mFormat,
+                                    &inputDesc->mChannels,
+                                    inputDesc->mAcoustics);
+
+    // only accept input with the exact requested set of parameters
+    if ((samplingRate != inputDesc->mSamplingRate) ||
+        (format != inputDesc->mFormat) ||
+        (channels != inputDesc->mChannels)) {
+        LOGV("getOutput() failed opening input: samplingRate %d, format %d, channels %d",
+                samplingRate, format, channels);
+        mpClientInterface->closeInput(input);
+        delete inputDesc;
+        return 0;
+    }
+    mInputs.add(input, inputDesc);
+    return input;
+}
+
+status_t AudioPolicyManagerALSA::startInput(audio_io_handle_t input)
+{
+    LOGV("startInput() input %d", input);
+    ssize_t index = mInputs.indexOfKey(input);
+    if (index < 0) {
+        LOGW("startInput() unknow input %d", input);
+        return BAD_VALUE;
+    }
+    AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
+
+    // refuse 2 active AudioRecord clients at the same time
+    if (getActiveInput() != 0) {
+        LOGW("startInput() input %d failed: other input already started", input);
+        return INVALID_OPERATION;
+    }
+
+    AudioParameter param = AudioParameter();
+    param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice);
+    mpClientInterface->setParameters(input, param.toString());
+
+    inputDesc->mRefCount = 1;
+    return NO_ERROR;
+}
+
+status_t AudioPolicyManagerALSA::stopInput(audio_io_handle_t input)
+{
+    LOGV("stopInput() input %d", input);
+    ssize_t index = mInputs.indexOfKey(input);
+    if (index < 0) {
+        LOGW("stopInput() unknow input %d", input);
+        return BAD_VALUE;
+    }
+    AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
+
+    if (inputDesc->mRefCount == 0) {
+        LOGW("stopInput() input %d already stopped", input);
+        return INVALID_OPERATION;
+    } else {
+        AudioParameter param = AudioParameter();
+        param.addInt(String8(AudioParameter::keyRouting), 0);
+        mpClientInterface->setParameters(input, param.toString());
+        inputDesc->mRefCount = 0;
+        return NO_ERROR;
+    }
+}
+
+void AudioPolicyManagerALSA::releaseInput(audio_io_handle_t input)
+{
+    LOGV("releaseInput() %d", input);
+    ssize_t index = mInputs.indexOfKey(input);
+    if (index < 0) {
+        LOGW("releaseInput() releasing unknown input %d", input);
+        return;
+    }
+    mpClientInterface->closeInput(input);
+    delete mInputs.valueAt(index);
+    mInputs.removeItem(input);
+    LOGV("releaseInput() exit");
+}
+
+
+
+void AudioPolicyManagerALSA::initStreamVolume(AudioSystem::stream_type stream,
+                                            int indexMin,
+                                            int indexMax)
+{
+    LOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
+    if (indexMin < 0 || indexMin >= indexMax) {
+        LOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d", stream , indexMin, indexMax);
+        return;
+    }
+    mStreams[stream].mIndexMin = indexMin;
+    mStreams[stream].mIndexMax = indexMax;
+}
+
+status_t AudioPolicyManagerALSA::setStreamVolumeIndex(AudioSystem::stream_type stream, int index)
+{
+
+    if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) {
+        return BAD_VALUE;
+    }
+
+    LOGV("setStreamVolumeIndex() stream %d, index %d", stream, index);
+    mStreams[stream].mIndexCur = index;
+
+    // compute and apply stream volume on all outputs according to connected device
+    status_t status = NO_ERROR;
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), mOutputs.valueAt(i)->device());
+        if (volStatus != NO_ERROR) {
+            status = volStatus;
+        }
+    }
+    return status;
+}
+
+status_t AudioPolicyManagerALSA::getStreamVolumeIndex(AudioSystem::stream_type stream, int *index)
+{
+    if (index == 0) {
+        return BAD_VALUE;
+    }
+    LOGV("getStreamVolumeIndex() stream %d", stream);
+    *index =  mStreams[stream].mIndexCur;
+    return NO_ERROR;
+}
+
+status_t AudioPolicyManagerALSA::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Hardware Output: %d\n", mHardwareOutput);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " A2DP Output: %d\n", mA2dpOutput);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Duplicated Output: %d\n", mDuplicatedOutput);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Output devices: %08x\n", mAvailableOutputDevices);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Input devices: %08x\n", mAvailableInputDevices);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " A2DP device address: %s\n", mA2dpDeviceAddress.string());
+    result.append(buffer);
+    snprintf(buffer, SIZE, " SCO device address: %s\n", mScoDeviceAddress.string());
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Ringer mode: %d\n", mRingerMode);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Force use for communications %d\n", mForceUse[AudioSystem::FOR_COMMUNICATION]);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AudioSystem::FOR_MEDIA]);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AudioSystem::FOR_RECORD]);
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+
+    snprintf(buffer, SIZE, "\nOutputs dump:\n");
+    write(fd, buffer, strlen(buffer));
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        snprintf(buffer, SIZE, "- Output %d dump:\n", mOutputs.keyAt(i));
+        write(fd, buffer, strlen(buffer));
+        mOutputs.valueAt(i)->dump(fd);
+    }
+
+    snprintf(buffer, SIZE, "\nInputs dump:\n");
+    write(fd, buffer, strlen(buffer));
+    for (size_t i = 0; i < mInputs.size(); i++) {
+        snprintf(buffer, SIZE, "- Input %d dump:\n", mInputs.keyAt(i));
+        write(fd, buffer, strlen(buffer));
+        mInputs.valueAt(i)->dump(fd);
+    }
+
+    snprintf(buffer, SIZE, "\nStreams dump:\n");
+    write(fd, buffer, strlen(buffer));
+    snprintf(buffer, SIZE, " Stream  Index Min  Index Max  Index Cur  Mute Count  Can be muted\n");
+    write(fd, buffer, strlen(buffer));
+    for (size_t i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
+        snprintf(buffer, SIZE, " %02d", i);
+        mStreams[i].dump(buffer + 3, SIZE);
+        write(fd, buffer, strlen(buffer));
+    }
+
+    return NO_ERROR;
+}
+
+// ----------------------------------------------------------------------------
+// AudioPolicyManagerALSA
+// ----------------------------------------------------------------------------
+
+// ---  class factory
+
+
+extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface)
+{
+    return new AudioPolicyManagerALSA(clientInterface);
+}
+
+extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
+{
+    delete interface;
+}
+
+AudioPolicyManagerALSA::AudioPolicyManagerALSA(AudioPolicyClientInterface *clientInterface)
+: mPhoneState(AudioSystem::MODE_NORMAL), mRingerMode(0), mMusicStopTime(0)
+{
+    mpClientInterface = clientInterface;
+
+    for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) {
+        mForceUse[i] = AudioSystem::FORCE_NONE;
+    }
+
+    // devices available by default are speaker, ear piece and microphone
+    mAvailableOutputDevices = AudioSystem::DEVICE_OUT_EARPIECE |
+                        AudioSystem::DEVICE_OUT_SPEAKER;
+    mAvailableInputDevices = AudioSystem::DEVICE_IN_BUILTIN_MIC;
+
+    mA2dpDeviceAddress = String8("");
+    mScoDeviceAddress = String8("");
+
+    // open hardware output
+    AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
+    outputDesc->mDevice = (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER;
+    mHardwareOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
+                                    &outputDesc->mSamplingRate,
+                                    &outputDesc->mFormat,
+                                    &outputDesc->mChannels,
+                                    &outputDesc->mLatency,
+                                    outputDesc->mFlags);
+
+    if (mHardwareOutput == 0) {
+        LOGE("Failed to initialize hardware output stream, samplingRate: %d, format %d, channels %d",
+                outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannels);
+    } else {
+        mOutputs.add(mHardwareOutput, outputDesc);
+        setOutputDevice(mHardwareOutput, (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER, true);
+    }
+
+    mA2dpOutput = 0;
+    mDuplicatedOutput = 0;
+}
+
+AudioPolicyManagerALSA::~AudioPolicyManagerALSA()
+{
+   for (size_t i = 0; i < mOutputs.size(); i++) {
+        mpClientInterface->closeOutput(mOutputs.keyAt(i));
+        delete mOutputs.valueAt(i);
+   }
+   mOutputs.clear();
+   for (size_t i = 0; i < mInputs.size(); i++) {
+        mpClientInterface->closeInput(mInputs.keyAt(i));
+        delete mInputs.valueAt(i);
+   }
+   mInputs.clear();
+}
+
+// ---
+
+audio_io_handle_t AudioPolicyManagerALSA::getOutputForDevice(uint32_t device)
+{
+    audio_io_handle_t output = 0;
+    uint32_t lDevice;
+
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        lDevice = mOutputs.valueAt(i)->device();
+        LOGV("getOutputForDevice() output %d devices %x", mOutputs.keyAt(i), lDevice);
+
+        // We are only considering outputs connected to a mixer here => exclude direct outputs
+        if ((lDevice == device) &&
+           !(mOutputs.valueAt(i)->mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) {
+            output = mOutputs.keyAt(i);
+            LOGV("getOutputForDevice() found output %d for device %x", output, device);
+            break;
+        }
+    }
+    return output;
+}
+
+AudioPolicyManagerALSA::routing_strategy AudioPolicyManagerALSA::getStrategy(AudioSystem::stream_type stream)
+{
+    // stream to strategy mapping
+    switch (stream) {
+    case AudioSystem::VOICE_CALL:
+    case AudioSystem::BLUETOOTH_SCO:
+        return STRATEGY_PHONE;
+    case AudioSystem::RING:
+    case AudioSystem::NOTIFICATION:
+    case AudioSystem::ALARM:
+    case AudioSystem::ENFORCED_AUDIBLE:
+        return STRATEGY_SONIFICATION;
+    case AudioSystem::DTMF:
+        return STRATEGY_DTMF;
+    default:
+        LOGE("unknown stream type");
+    case AudioSystem::SYSTEM:
+        // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs
+        // while key clicks are played produces a poor result
+    case AudioSystem::TTS:
+    case AudioSystem::MUSIC:
+        return STRATEGY_MEDIA;
+    }
+}
+
+uint32_t AudioPolicyManagerALSA::getDeviceForStrategy(routing_strategy strategy)
+{
+    uint32_t device = 0;
+
+    switch (strategy) {
+    case STRATEGY_DTMF:
+        if (mPhoneState != AudioSystem::MODE_IN_CALL) {
+            // when off call, DTMF strategy follows the same rules as MEDIA strategy
+            device = getDeviceForStrategy(STRATEGY_MEDIA);
+            break;
+        }
+        // when in call, DTMF and PHONE strategies follow the same rules
+        // FALL THROUGH
+
+    case STRATEGY_PHONE:
+        // for phone strategy, we first consider the forced use and then the available devices by order
+        // of priority
+        switch (mForceUse[AudioSystem::FOR_COMMUNICATION]) {
+        case AudioSystem::FORCE_BT_SCO:
+            if (mPhoneState != AudioSystem::MODE_IN_CALL || strategy != STRATEGY_DTMF) {
+                device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
+                if (device) break;
+            }
+            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
+            if (device) break;
+            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO;
+            if (device) break;
+            // if SCO device is requested but no SCO device is available, fall back to default case
+            // FALL THROUGH
+
+        default:    // FORCE_NONE
+            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_TTY;
+            if (device) break;
+            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADPHONE;
+            if (device) break;
+            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET;
+            if (device) break;
+            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_EARPIECE;
+            if (device == 0) {
+                LOGE("getDeviceForStrategy() earpiece device not found");
+            }
+            break;
+
+        case AudioSystem::FORCE_SPEAKER:
+            if (mPhoneState != AudioSystem::MODE_IN_CALL || strategy != STRATEGY_DTMF) {
+                device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
+                if (device) break;
+            }
+            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_FM_SPEAKER;
+            if (device) break;
+            device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER;
+            if (device == 0) {
+                LOGE("getDeviceForStrategy() speaker device not found");
+            }
+            break;
+        }
+    break;
+
+    case STRATEGY_SONIFICATION:
+
+        // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
+        // handleIncallSonification().
+        if (mPhoneState == AudioSystem::MODE_IN_CALL) {
+            device = getDeviceForStrategy(STRATEGY_PHONE);
+            break;
+        }
+        device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER;
+        if (device == 0) {
+            LOGE("getDeviceForStrategy() speaker device not found");
+        }
+        // The second device used for sonification is the same as the device used by media strategy
+        // FALL THROUGH
+
+    case STRATEGY_MEDIA: {
+        uint32_t device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_AUX_DIGITAL;
+        if (device2 == 0) {
+            device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP;
+            if (device2 == 0) {
+                device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
+                if (device2 == 0) {
+                    device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
+                    if (device2 == 0) {
+                        device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_FM_HEADPHONE;
+                        if (device2 == 0) {
+                            device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_FM_SPEAKER;
+                            if (device2 == 0) {
+                                device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADPHONE;
+                                if (device2 == 0) {
+                                    device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET;
+                                    if (device2 == 0) {
+                                        device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER;
+                                        if (device == 0) {
+                                            LOGE("getDeviceForStrategy() speaker device not found");
+                                        }
+                                    }
+                                }
+                            }
+                        }
+                    }
+                }
+            }
+        }
+        // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION, 0 otherwise
+        device |= device2;
+        // Do not play media stream if in call and the requested device would change the hardware
+        // output routing
+        if (mPhoneState == AudioSystem::MODE_IN_CALL &&
+            !AudioSystem::isA2dpDevice((AudioSystem::audio_devices)device) &&
+            device != getDeviceForStrategy(STRATEGY_PHONE)) {
+            device = 0;
+            LOGV("getDeviceForStrategy() incompatible media and phone devices");
+        }
+        } break;
+
+    default:
+        LOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
+        break;
+    }
+
+    LOGV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
+    return device;
+}
+
+void AudioPolicyManagerALSA::setOutputDevice(audio_io_handle_t output, uint32_t device, bool force, int delayMs)
+{
+    LOGV("setOutputDevice() output %d device %x delayMs %d", output, device, delayMs);
+    if (mOutputs.indexOfKey(output) < 0) {
+        LOGW("setOutputDevice() unknown output %d", output);
+        return;
+    }
+#ifdef WITH_A2DP
+    if (output == mHardwareOutput) {
+        // clear A2DP devices from device bit field here so that the caller does not have to
+        // do it in case of multiple device selections
+        uint32_t device2 = device & ~AudioSystem::DEVICE_OUT_SPEAKER;
+        if (AudioSystem::isA2dpDevice((AudioSystem::audio_devices)device2)) {
+            LOGV("setOutputDevice() removing A2DP device");
+            device &= ~device2;
+        }
+    } else if (output == mA2dpOutput) {
+        // clear hardware devices from device bit field here so that the caller does not have to
+        // do it in case of multiple device selections (the second device is always DEVICE_OUT_SPEAKER)
+        // in this case
+        device &= ~AudioSystem::DEVICE_OUT_SPEAKER;
+    }
+#endif
+
+    // doing this check here allows the caller to call setOutputDevice() without conditions
+    if (device == 0) return;
+
+    uint32_t oldDevice = (uint32_t)mOutputs.valueFor(output)->device();
+    // Do not change the routing if the requested device is the same as current device. Doing this check
+    // here allows the caller to call setOutputDevice() without conditions
+    if (device == oldDevice && !force) {
+        LOGV("setOutputDevice() setting same device %x for output %d", device, output);
+        return;
+    }
+
+    mOutputs.valueFor(output)->mDevice = device;
+    // mute media streams if both speaker and headset are selected
+    if (device == (AudioSystem::DEVICE_OUT_SPEAKER | AudioSystem::DEVICE_OUT_WIRED_HEADSET) ||
+        device == (AudioSystem::DEVICE_OUT_SPEAKER | AudioSystem::DEVICE_OUT_WIRED_HEADPHONE)) {
+        setStrategyMute(STRATEGY_MEDIA, true, output);
+        // wait for the PCM output buffers to empty before proceeding with the rest of the command
+        usleep(mOutputs.valueFor(output)->mLatency*2*1000);
+    }
+    // suspend A2D output if SCO device is selected
+    if (AudioSystem::isBluetoothScoDevice((AudioSystem::audio_devices)device)) {
+         if (mA2dpOutput && mScoDeviceAddress == mA2dpDeviceAddress) {
+             mpClientInterface->suspendOutput(mA2dpOutput);
+         }
+    }
+    // do the routing
+    AudioParameter param = AudioParameter();
+    param.addInt(String8(AudioParameter::keyRouting), (int)device);
+    mpClientInterface->setParameters(mHardwareOutput, param.toString(), delayMs);
+    // update stream volumes according to new device
+    applyStreamVolumes(output, device, delayMs);
+
+    // if disconnecting SCO device, restore A2DP output
+    if (AudioSystem::isBluetoothScoDevice((AudioSystem::audio_devices)oldDevice)) {
+         if (mA2dpOutput && mScoDeviceAddress == mA2dpDeviceAddress) {
+             LOGV("restore A2DP output");
+             mpClientInterface->restoreOutput(mA2dpOutput);
+         }
+    }
+    // if changing from a combined headset + speaker route, unmute media streams
+    if (oldDevice == (AudioSystem::DEVICE_OUT_SPEAKER | AudioSystem::DEVICE_OUT_WIRED_HEADSET) ||
+        oldDevice == (AudioSystem::DEVICE_OUT_SPEAKER | AudioSystem::DEVICE_OUT_WIRED_HEADPHONE)) {
+        setStrategyMute(STRATEGY_MEDIA, false, output, delayMs);
+    }
+}
+
+uint32_t AudioPolicyManagerALSA::getDeviceForInputSource(int inputSource)
+{
+    uint32_t device;
+
+    switch(inputSource) {
+    case AUDIO_SOURCE_DEFAULT:
+    case AUDIO_SOURCE_MIC:
+    case AUDIO_SOURCE_VOICE_RECOGNITION:
+        if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO &&
+            mAvailableInputDevices & AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
+            device = AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+        } else if (mAvailableInputDevices & AudioSystem::DEVICE_IN_WIRED_HEADSET) {
+            device = AudioSystem::DEVICE_IN_WIRED_HEADSET;
+        } else {
+            device = AudioSystem::DEVICE_IN_BUILTIN_MIC;
+        }
+        break;
+    case AUDIO_SOURCE_CAMCORDER:
+        device = AudioSystem::DEVICE_IN_BUILTIN_MIC;
+        break;
+    case AUDIO_SOURCE_VOICE_UPLINK:
+    case AUDIO_SOURCE_VOICE_DOWNLINK:
+    case AUDIO_SOURCE_VOICE_CALL:
+        device = AudioSystem::DEVICE_IN_VOICE_CALL;
+        break;
+    default:
+        LOGW("getInput() invalid input source %d", inputSource);
+        device = 0;
+        break;
+    }
+    return device;
+}
+
+audio_io_handle_t AudioPolicyManagerALSA::getActiveInput()
+{
+    for (size_t i = 0; i < mInputs.size(); i++) {
+        if (mInputs.valueAt(i)->mRefCount > 0) {
+            return mInputs.keyAt(i);
+        }
+    }
+    return 0;
+}
+
+float AudioPolicyManagerALSA::computeVolume(int stream, int index, audio_io_handle_t output, uint32_t device)
+{
+    float volume = 1.0;
+    AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+    StreamDescriptor &streamDesc = mStreams[stream];
+
+    // Force max volume if stream cannot be muted
+    if (!streamDesc.mCanBeMuted) index = streamDesc.mIndexMax;
+
+    if (device == 0) {
+        device = outputDesc->device();
+    }
+
+    int volInt = (100 * (index - streamDesc.mIndexMin)) / (streamDesc.mIndexMax - streamDesc.mIndexMin);
+    volume = AudioSystem::linearToLog(volInt);
+
+    // if a heaset is connected, apply the following rules to ring tones and notifications
+    // to avoid sound level bursts in user's ears:
+    // - always attenuate ring tones and notifications volume by 6dB
+    // - if music is playing, always limit the volume to current music volume,
+    // with a minimum threshold at -36dB so that notification is always perceived.
+    if ((device &
+        (AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP |
+        AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
+        AudioSystem::DEVICE_OUT_WIRED_HEADSET |
+        AudioSystem::DEVICE_OUT_WIRED_HEADPHONE)) &&
+        (getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION)) {
+        volume *= SONIFICATION_HEADSET_VOLUME_FACTOR;
+        // when the phone is ringing we must consider that music could have been paused just before
+        // by the music application and behave as if music was active if the last music track was
+        // just stopped
+        if (outputDesc->isUsedByStream(AudioSystem::MUSIC) ||
+            ((mPhoneState == AudioSystem::MODE_RINGTONE) &&
+             (systemTime() - mMusicStopTime < seconds(SONIFICATION_HEADSET_MUSIC_DELAY)))) {
+            float musicVol = computeVolume(AudioSystem::MUSIC, mStreams[AudioSystem::MUSIC].mIndexCur, output, device);
+            float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ? musicVol : SONIFICATION_HEADSET_VOLUME_MIN;
+            if (volume > minVol) {
+                volume = minVol;
+                LOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol);
+            }
+        }
+    }
+
+    return volume;
+}
+
+status_t AudioPolicyManagerALSA::checkAndSetVolume(int stream, int index, audio_io_handle_t output, uint32_t device, int delayMs, bool force)
+{
+
+    // do not change actual stream volume if the stream is muted
+    if (mStreams[stream].mMuteCount != 0) {
+        LOGV("checkAndSetVolume() stream %d muted count %d", stream, mStreams[stream].mMuteCount);
+        return NO_ERROR;
+    }
+
+    // do not change in call volume if bluetooth is connected and vice versa
+    if ((stream == AudioSystem::VOICE_CALL && mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) ||
+        (stream == AudioSystem::BLUETOOTH_SCO && mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO)) {
+        LOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
+             stream, mForceUse[AudioSystem::FOR_COMMUNICATION]);
+        return INVALID_OPERATION;
+    }
+
+    float volume = computeVolume(stream, index, output, device || force);
+    // do not set volume if the float value did not change
+    if (volume != mOutputs.valueFor(output)->mCurVolume[stream]) {
+        mOutputs.valueFor(output)->mCurVolume[stream] = volume;
+        LOGV("setStreamVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs);
+        if (stream == AudioSystem::VOICE_CALL ||
+            stream == AudioSystem::DTMF ||
+            stream == AudioSystem::BLUETOOTH_SCO) {
+            float voiceVolume = -1.0;
+            // offset value to reflect actual hardware volume that never reaches 0
+            // 1% corresponds roughly to first step in VOICE_CALL stream volume setting (see AudioService.java)
+            volume = 0.01 + 0.99 * volume;
+            if (stream == AudioSystem::VOICE_CALL) {
+                voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
+            } else if (stream == AudioSystem::BLUETOOTH_SCO) {
+                voiceVolume = 1.0;
+            }
+            if (voiceVolume >= 0 && output == mHardwareOutput) {
+                mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
+            }
+        }
+        mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output, delayMs);
+    }
+
+    return NO_ERROR;
+}
+
+void AudioPolicyManagerALSA::applyStreamVolumes(audio_io_handle_t output, uint32_t device, int delayMs)
+{
+    LOGV("applyStreamVolumes() for output %d and device %x", output, device);
+
+    for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+        checkAndSetVolume(stream, mStreams[stream].mIndexCur, output, device, delayMs);
+    }
+}
+
+void AudioPolicyManagerALSA::setStrategyMute(routing_strategy strategy, bool on, audio_io_handle_t output, int delayMs)
+{
+    LOGV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output);
+    for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+        if (getStrategy((AudioSystem::stream_type)stream) == strategy) {
+            setStreamMute(stream, on, output, delayMs);
+        }
+    }
+}
+
+void AudioPolicyManagerALSA::setStreamMute(int stream, bool on, audio_io_handle_t output, int delayMs)
+{
+    StreamDescriptor &streamDesc = mStreams[stream];
+    uint32_t device = mOutputs.valueFor(output)->mDevice;
+
+    LOGV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d", stream, on, output, streamDesc.mMuteCount);
+
+    if (on) {
+        if (streamDesc.mMuteCount == 0) {
+            if (streamDesc.mCanBeMuted) {
+                checkAndSetVolume(stream, 0, output, device, delayMs);
+            }
+        }
+        // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored
+        streamDesc.mMuteCount++;
+    } else {
+        if (streamDesc.mMuteCount == 0) {
+            LOGW("setStreamMute() unmuting non muted stream!");
+            return;
+        }
+        if (--streamDesc.mMuteCount == 0) {
+            checkAndSetVolume(stream, streamDesc.mIndexCur, output, device, delayMs);
+        }
+    }
+}
+
+void AudioPolicyManagerALSA::handleIncallSonification(int stream, bool starting, bool stateChange)
+{
+    // if the stream pertains to sonification strategy and we are in call we must
+    // mute the stream if it is low visibility. If it is high visibility, we must play a tone
+    // in the device used for phone strategy and play the tone if the selected device does not
+    // interfere with the device used for phone strategy
+    // if stateChange is true, we are called from setPhoneState() and we must mute or unmute as
+    // many times as there are active tracks on the output
+
+    if (getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) {
+        AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mHardwareOutput);
+        LOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
+                stream, starting, outputDesc->mDevice, stateChange);
+        if (outputDesc->isUsedByStream((AudioSystem::stream_type)stream)) {
+            int muteCount = 1;
+            if (stateChange) {
+                muteCount = outputDesc->mRefCount[stream];
+            }
+            if (AudioSystem::isLowVisibility((AudioSystem::stream_type)stream)) {
+                LOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
+                for (int i = 0; i < muteCount; i++) {
+                    setStreamMute(stream, starting, mHardwareOutput);
+                }
+            } else {
+                LOGV("handleIncallSonification() high visibility ");
+                if (outputDesc->mDevice & getDeviceForStrategy(STRATEGY_PHONE)) {
+                    LOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
+                    for (int i = 0; i < muteCount; i++) {
+                        setStreamMute(stream, starting, mHardwareOutput);
+                    }
+                }
+                if (starting) {
+                    mpClientInterface->startTone(ToneGenerator::TONE_SUP_CALL_WAITING, AudioSystem::VOICE_CALL);
+                } else {
+                    mpClientInterface->stopTone();
+                }
+            }
+        }
+    }
+}
+
+
+// --- AudioOutputDescriptor class implementation
+
+AudioPolicyManagerALSA::AudioOutputDescriptor::AudioOutputDescriptor()
+    : mSamplingRate(0), mFormat(0), mChannels(0), mLatency(0),
+    mFlags((AudioSystem::output_flags)0), mDevice(0), mOutput1(0), mOutput2(0)
+{
+    // clear usage count for all stream types
+    for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
+        mRefCount[i] = 0;
+        mCurVolume[i] = -1.0;
+    }
+}
+
+uint32_t AudioPolicyManagerALSA::AudioOutputDescriptor::device()
+{
+    uint32_t device = 0;
+    if (isDuplicated()) {
+        device = mOutput1->mDevice | mOutput2->mDevice;
+    } else {
+        device = mDevice;
+    }
+    return device;
+}
+
+void AudioPolicyManagerALSA::AudioOutputDescriptor::changeRefCount(AudioSystem::stream_type stream, int delta)
+{
+    // forward usage count change to attached outputs
+    if (isDuplicated()) {
+        mOutput1->changeRefCount(stream, delta);
+        mOutput2->changeRefCount(stream, delta);
+    }
+    if ((delta + (int)mRefCount[stream]) < 0) {
+        LOGW("changeRefCount() invalid delta %d for stream %d, refCount %d", delta, stream, mRefCount[stream]);
+        mRefCount[stream] = 0;
+        return;
+    }
+    mRefCount[stream] += delta;
+    LOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
+}
+
+bool AudioPolicyManagerALSA::AudioOutputDescriptor::isUsedByStrategy(routing_strategy strategy)
+{
+    for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+        if (AudioPolicyManagerALSA::getStrategy((AudioSystem::stream_type)i) == strategy &&
+            isUsedByStream((AudioSystem::stream_type)i)) {
+            return true;
+        }
+    }
+    return false;
+}
+
+status_t AudioPolicyManagerALSA::AudioOutputDescriptor::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Format: %d\n", mFormat);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Channels: %08x\n", mChannels);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Stream volume refCount\n");
+    result.append(buffer);
+    for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
+        snprintf(buffer, SIZE, " %02d     %.03f  %d\n", i, mCurVolume[i], mRefCount[i]);
+        result.append(buffer);
+    }
+    write(fd, result.string(), result.size());
+
+    return NO_ERROR;
+}
+
+// --- AudioInputDescriptor class implementation
+
+AudioPolicyManagerALSA::AudioInputDescriptor::AudioInputDescriptor()
+    : mSamplingRate(0), mFormat(0), mChannels(0),
+     mAcoustics((AudioSystem::audio_in_acoustics)0), mDevice(0), mRefCount(0)
+{
+}
+
+status_t AudioPolicyManagerALSA::AudioInputDescriptor::dump(int fd)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Format: %d\n", mFormat);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Channels: %08x\n", mChannels);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Acoustics %08x\n", mAcoustics);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
+    result.append(buffer);
+    snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
+    result.append(buffer);
+    write(fd, result.string(), result.size());
+
+    return NO_ERROR;
+}
+
+// --- StreamDescriptor class implementation
+
+void AudioPolicyManagerALSA::StreamDescriptor::dump(char* buffer, size_t size)
+{
+    snprintf(buffer, size, "      %02d         %02d         %02d         %02d          %d\n",
+            mIndexMin,
+            mIndexMax,
+            mIndexCur,
+            mMuteCount,
+            mCanBeMuted);
+}
+
+
+}; // namespace android
diff --git a/AudioPolicyManagerALSA.h b/AudioPolicyManagerALSA.h
new file mode 100644
index 0000000..b31747c
--- /dev/null
+++ b/AudioPolicyManagerALSA.h
@@ -0,0 +1,201 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+
+#include <stdint.h>
+#include <sys/types.h>
+#include <utils/Timers.h>
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
+#include <hardware_legacy/AudioPolicyInterface.h>
+
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+#define MAX_DEVICE_ADDRESS_LEN 20
+// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB
+#define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5
+// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB
+#define SONIFICATION_HEADSET_VOLUME_MIN  0.016
+// Time in seconds during which we consider that music is still active after a music
+// track was stopped - see computeVolume()
+#define SONIFICATION_HEADSET_MUSIC_DELAY  5
+class AudioPolicyManagerALSA: public AudioPolicyInterface
+{
+
+public:
+                AudioPolicyManagerALSA(AudioPolicyClientInterface *clientInterface);
+        virtual ~AudioPolicyManagerALSA();
+
+        // AudioPolicyInterface
+        virtual status_t setDeviceConnectionState(AudioSystem::audio_devices device,
+                                                          AudioSystem::device_connection_state state,
+                                                          const char *device_address);
+        virtual AudioSystem::device_connection_state getDeviceConnectionState(AudioSystem::audio_devices device,
+                                                                              const char *device_address);
+        virtual void setPhoneState(int state);
+        virtual void setRingerMode(uint32_t mode, uint32_t mask);
+        virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config);
+        virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage);
+        virtual void setSystemProperty(const char* property, const char* value);
+        virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream,
+                                            uint32_t samplingRate,
+                                            uint32_t format,
+                                            uint32_t channels,
+                                            AudioSystem::output_flags flags);
+        virtual status_t startOutput(audio_io_handle_t output, AudioSystem::stream_type stream);
+        virtual status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream);
+        virtual void releaseOutput(audio_io_handle_t output);
+        virtual audio_io_handle_t getInput(int inputSource,
+                                            uint32_t samplingRate,
+                                            uint32_t format,
+                                            uint32_t channels,
+                                            AudioSystem::audio_in_acoustics acoustics);
+        // indicates to the audio policy manager that the input starts being used.
+        virtual status_t startInput(audio_io_handle_t input);
+        // indicates to the audio policy manager that the input stops being used.
+        virtual status_t stopInput(audio_io_handle_t input);
+        virtual void releaseInput(audio_io_handle_t input);
+        virtual void initStreamVolume(AudioSystem::stream_type stream,
+                                                    int indexMin,
+                                                    int indexMax);
+        virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream, int index);
+        virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, int *index);
+
+        virtual status_t dump(int fd);
+
+private:
+
+        enum routing_strategy {
+            STRATEGY_MEDIA,
+            STRATEGY_PHONE,
+            STRATEGY_SONIFICATION,
+            STRATEGY_DTMF,
+            NUM_STRATEGIES
+        };
+
+        // descriptor for audio outputs. Used to maintain current configuration of each opened audio output
+        // and keep track of the usage of this output by each audio stream type.
+        class AudioOutputDescriptor
+        {
+        public:
+            AudioOutputDescriptor();
+
+            status_t    dump(int fd);
+
+            uint32_t device();
+            void changeRefCount(AudioSystem::stream_type, int delta);
+            bool isUsedByStrategy(routing_strategy strategy);
+            bool isUsedByStream(AudioSystem::stream_type stream) { return mRefCount[stream] > 0 ? true : false; }
+            bool isDuplicated() { return (mDevice == 0); } // by convention mDevice is 0 for duplicated outputs
+
+            uint32_t mSamplingRate;             //
+            uint32_t mFormat;                   //
+            uint32_t mChannels;                 // output configuration
+            uint32_t mLatency;                  //
+            AudioSystem::output_flags mFlags;   //
+            uint32_t mDevice;                   // current device this output is routed to
+            uint32_t mRefCount[AudioSystem::NUM_STREAM_TYPES]; // number of streams of each type using this output
+            AudioOutputDescriptor *mOutput1;    // used by duplicated outputs: first output
+            AudioOutputDescriptor *mOutput2;    // used by duplicated outputs: second output
+            float mCurVolume[AudioSystem::NUM_STREAM_TYPES];   // current stream volume
+        };
+
+        // descriptor for audio inputs. Used to maintain current configuration of each opened audio input
+        // and keep track of the usage of this input.
+        class AudioInputDescriptor
+        {
+        public:
+            AudioInputDescriptor();
+
+            status_t    dump(int fd);
+
+            uint32_t mSamplingRate;                     //
+            uint32_t mFormat;                           // input configuration
+            uint32_t mChannels;                         //
+            AudioSystem::audio_in_acoustics mAcoustics; //
+            uint32_t mDevice;                           // current device this input is routed to
+            uint32_t mRefCount;                         // number of AudioRecord clients using this output
+            int      mInputSource;                     // input source selected by application (mediarecorder.h)
+        };
+
+        // stream descriptor used for volume control
+        class StreamDescriptor
+        {
+        public:
+            StreamDescriptor()
+            :   mIndexMin(0), mIndexMax(1), mIndexCur(1), mMuteCount(0), mCanBeMuted(true) {}
+
+            void dump(char* buffer, size_t size);
+
+            int mIndexMin;      // min volume index
+            int mIndexMax;      // max volume index
+            int mIndexCur;      // current volume index
+            int mMuteCount;     // mute request counter
+            bool mCanBeMuted;   // true is the stream can be muted
+        };
+
+        // return the strategy corresponding to a given stream type
+        static routing_strategy getStrategy(AudioSystem::stream_type stream);
+        // return the output handle of an output routed to the specified device, 0 if no output
+        // is routed to the device
+        audio_io_handle_t getOutputForDevice(uint32_t device);
+        // return appropriate device for streams handled by the specified strategy according to current
+        // phone state, connected devices...
+        uint32_t getDeviceForStrategy(routing_strategy strategy);
+        // change the route of the specified output
+        void setOutputDevice(audio_io_handle_t output, uint32_t device, bool force = false, int delayMs = 0);
+        // select input device corresponding to requested audio source
+        uint32_t getDeviceForInputSource(int inputSource);
+        // return io handle of active input or 0 if no input is active
+        audio_io_handle_t getActiveInput();
+        // compute the actual volume for a given stream according to the requested index and a particular
+        // device
+        float computeVolume(int stream, int index, audio_io_handle_t output, uint32_t device);
+        // check that volume change is permitted, compute and send new volume to audio hardware
+        status_t checkAndSetVolume(int stream, int index, audio_io_handle_t output, uint32_t device, int delayMs = 0, bool force = false);
+        // apply all stream volumes to the specified output and device
+        void applyStreamVolumes(audio_io_handle_t output, uint32_t device, int delayMs = 0);
+        // Mute or unmute all streams handled by the specified strategy on the specified output
+        void setStrategyMute(routing_strategy strategy, bool on, audio_io_handle_t output, int delayMs = 0);
+        // Mute or unmute the stream on the specified output
+        void setStreamMute(int stream, bool on, audio_io_handle_t output, int delayMs = 0);
+        // handle special cases for sonification strategy while in call: mute streams or replace by
+        // a special tone in the device used for communication
+        void handleIncallSonification(int stream, bool starting, bool stateChange);
+
+        AudioPolicyClientInterface *mpClientInterface;  // audio policy client interface
+        audio_io_handle_t mHardwareOutput;              // hardware output handler
+        audio_io_handle_t mA2dpOutput;                  // A2DP output handler
+        audio_io_handle_t mDuplicatedOutput;            // duplicated output handler: outputs to hardware and A2DP.
+
+        KeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mOutputs;   // list of output descriptors
+        KeyedVector<audio_io_handle_t, AudioInputDescriptor *> mInputs;     // list of input descriptors
+        uint32_t mAvailableOutputDevices;                                   // bit field of all available output devices
+        uint32_t mAvailableInputDevices;                                    // bit field of all available input devices
+        int mPhoneState;                                                    // current phone state
+        uint32_t                 mRingerMode;                               // current ringer mode
+        AudioSystem::forced_config mForceUse[AudioSystem::NUM_FORCE_USE];   // current forced use configuration
+
+        StreamDescriptor mStreams[AudioSystem::NUM_STREAM_TYPES];           // stream descriptors for volume control
+        String8 mA2dpDeviceAddress;                                         // A2DP device MAC address
+        String8 mScoDeviceAddress;                                          // SCO device MAC address
+        nsecs_t mMusicStopTime;                                             // time when last music stream was stopped
+};
+
+};
diff --git a/AudioStreamInALSA.cpp b/AudioStreamInALSA.cpp
new file mode 100644
index 0000000..f645357
--- /dev/null
+++ b/AudioStreamInALSA.cpp
@@ -0,0 +1,155 @@
+/* AudioStreamInALSA.cpp
+ **
+ ** Copyright 2008-2009 Wind River Systems
+ **
+ ** Licensed under the Apache License, Version 2.0 (the "License");
+ ** you may not use this file except in compliance with the License.
+ ** You may obtain a copy of the License at
+ **
+ **     http://www.apache.org/licenses/LICENSE-2.0
+ **
+ ** Unless required by applicable law or agreed to in writing, software
+ ** distributed under the License is distributed on an "AS IS" BASIS,
+ ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ ** See the License for the specific language governing permissions and
+ ** limitations under the License.
+ */
+
+#include <errno.h>
+#include <stdarg.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <dlfcn.h>
+
+#define LOG_TAG "AudioHardwareALSA"
+#include <utils/Log.h>
+#include <utils/String8.h>
+
+#include <cutils/properties.h>
+#include <media/AudioRecord.h>
+#include <hardware_legacy/power.h>
+
+#include "AudioHardwareALSA.h"
+
+namespace android
+{
+
+AudioStreamInALSA::AudioStreamInALSA(AudioHardwareALSA *parent,
+        alsa_handle_t *handle,
+        AudioSystem::audio_in_acoustics audio_acoustics) :
+    ALSAStreamOps(parent, handle),
+    mAcoustics(audio_acoustics)
+{
+    acoustic_device_t *aDev = acoustics();
+
+    if (aDev) aDev->set_params(aDev, mAcoustics, NULL);
+}
+
+AudioStreamInALSA::~AudioStreamInALSA()
+{
+    close();
+}
+
+status_t AudioStreamInALSA::setGain(float gain)
+{
+    return mixer() ? mixer()->setMasterGain(gain) : (status_t)NO_INIT;
+}
+
+ssize_t AudioStreamInALSA::read(void *buffer, ssize_t bytes)
+{
+    AutoMutex lock(mLock);
+
+    if (!mPowerLock) {
+        acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioInLock");
+        mPowerLock = true;
+    }
+
+    acoustic_device_t *aDev = acoustics();
+
+    // If there is an acoustics module read method, then it overrides this
+    // implementation (unlike AudioStreamOutALSA write).
+    if (aDev && aDev->read)
+        return aDev->read(aDev, buffer, bytes);
+
+    snd_pcm_sframes_t n, frames = snd_pcm_bytes_to_frames(mHandle->handle, bytes);
+    status_t          err;
+
+    do {
+        n = snd_pcm_readi(mHandle->handle, buffer, frames);
+        if (n < frames) {
+            if (mHandle->handle) {
+                if (n < 0) {
+                    n = snd_pcm_recover(mHandle->handle, n, 0);
+
+                    if (aDev && aDev->recover) aDev->recover(aDev, n);
+                } else
+                    n = snd_pcm_prepare(mHandle->handle);
+            }
+            return static_cast<ssize_t>(n);
+        }
+    } while (n == -EAGAIN);
+
+    return static_cast<ssize_t>(snd_pcm_frames_to_bytes(mHandle->handle, n));
+}
+
+status_t AudioStreamInALSA::dump(int fd, const Vector<String16>& args)
+{
+    return NO_ERROR;
+}
+
+status_t AudioStreamInALSA::open(int mode)
+{
+    AutoMutex lock(mLock);
+
+    status_t status = ALSAStreamOps::open(mode);
+
+    acoustic_device_t *aDev = acoustics();
+
+    if (status == NO_ERROR && aDev)
+        status = aDev->use_handle(aDev, mHandle);
+
+    return status;
+}
+
+status_t AudioStreamInALSA::close()
+{
+    AutoMutex lock(mLock);
+
+    acoustic_device_t *aDev = acoustics();
+
+    if (mHandle && aDev) aDev->cleanup(aDev);
+
+    close();
+
+    if (mPowerLock) {
+        release_wake_lock ("AudioInLock");
+        mPowerLock = false;
+    }
+
+    return NO_ERROR;
+}
+
+status_t AudioStreamInALSA::standby()
+{
+    AutoMutex lock(mLock);
+
+    if (mPowerLock) {
+        release_wake_lock ("AudioInLock");
+        mPowerLock = false;
+    }
+
+    return NO_ERROR;
+}
+
+status_t AudioStreamInALSA::setAcousticParams(void *params)
+{
+    AutoMutex lock(mLock);
+
+    acoustic_device_t *aDev = acoustics();
+
+    return aDev ? aDev->set_params(aDev, mAcoustics, params) : (status_t)NO_ERROR;
+}
+
+}       // namespace android
diff --git a/AudioStreamOutALSA.cpp b/AudioStreamOutALSA.cpp
new file mode 100644
index 0000000..b042bfb
--- /dev/null
+++ b/AudioStreamOutALSA.cpp
@@ -0,0 +1,169 @@
+/* AudioStreamOutALSA.cpp
+ **
+ ** Copyright 2008-2009 Wind River Systems
+ **
+ ** Licensed under the Apache License, Version 2.0 (the "License");
+ ** you may not use this file except in compliance with the License.
+ ** You may obtain a copy of the License at
+ **
+ **     http://www.apache.org/licenses/LICENSE-2.0
+ **
+ ** Unless required by applicable law or agreed to in writing, software
+ ** distributed under the License is distributed on an "AS IS" BASIS,
+ ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ ** See the License for the specific language governing permissions and
+ ** limitations under the License.
+ */
+
+#include <errno.h>
+#include <stdarg.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <dlfcn.h>
+
+#define LOG_TAG "AudioHardwareALSA"
+#include <utils/Log.h>
+#include <utils/String8.h>
+
+#include <cutils/properties.h>
+#include <media/AudioRecord.h>
+#include <hardware_legacy/power.h>
+
+#include "AudioHardwareALSA.h"
+
+#ifndef ALSA_DEFAULT_SAMPLE_RATE
+#define ALSA_DEFAULT_SAMPLE_RATE 44100 // in Hz
+#endif
+
+namespace android
+{
+
+// ----------------------------------------------------------------------------
+
+static const int DEFAULT_SAMPLE_RATE = ALSA_DEFAULT_SAMPLE_RATE;
+
+// ----------------------------------------------------------------------------
+
+AudioStreamOutALSA::AudioStreamOutALSA(AudioHardwareALSA *parent, alsa_handle_t *handle) :
+    ALSAStreamOps(parent, handle)
+{
+}
+
+AudioStreamOutALSA::~AudioStreamOutALSA()
+{
+    close();
+}
+
+uint32_t AudioStreamOutALSA::channels() const
+{
+    int c = ALSAStreamOps::channels();
+    return c;
+}
+
+status_t AudioStreamOutALSA::setVolume(float left, float right)
+{
+    return mixer()->setVolume (mHandle->curDev, left, right);
+}
+
+ssize_t AudioStreamOutALSA::write(const void *buffer, size_t bytes)
+{
+    AutoMutex lock(mLock);
+
+    if (!mPowerLock) {
+        acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioOutLock");
+        mPowerLock = true;
+    }
+
+    acoustic_device_t *aDev = acoustics();
+
+    // For output, we will pass the data on to the acoustics module, but the actual
+    // data is expected to be sent to the audio device directly as well.
+    if (aDev && aDev->write)
+        aDev->write(aDev, buffer, bytes);
+
+    snd_pcm_sframes_t n;
+    size_t            sent = 0;
+    status_t          err;
+
+    do {
+        n = snd_pcm_writei(mHandle->handle,
+                           (char *)buffer + sent,
+                           snd_pcm_bytes_to_frames(mHandle->handle, bytes));
+        if (n == -EBADFD) {
+            // Somehow the stream is in a bad state. The driver probably
+            // has a bug and snd_pcm_recover() doesn't seem to handle this.
+            mHandle->module->open(mHandle, mHandle->curDev, mHandle->curMode);
+
+            if (aDev && aDev->recover) aDev->recover(aDev, n);
+        }
+        else if (n < 0) {
+            if (mHandle->handle) {
+                // snd_pcm_recover() will return 0 if successful in recovering from
+                // an error, or -errno if the error was unrecoverable.
+                n = snd_pcm_recover(mHandle->handle, n, 1);
+
+                if (aDev && aDev->recover) aDev->recover(aDev, n);
+
+                if (n) return static_cast<ssize_t>(n);
+            }
+        }
+        else
+            sent += static_cast<ssize_t>(snd_pcm_frames_to_bytes(mHandle->handle, n));
+
+    } while (mHandle->handle && sent < bytes);
+
+    return sent;
+}
+
+status_t AudioStreamOutALSA::dump(int fd, const Vector<String16>& args)
+{
+    return NO_ERROR;
+}
+
+status_t AudioStreamOutALSA::open(int mode)
+{
+    AutoMutex lock(mLock);
+
+    return ALSAStreamOps::open(mode);
+}
+
+status_t AudioStreamOutALSA::close()
+{
+    AutoMutex lock(mLock);
+
+    snd_pcm_drain (mHandle->handle);
+    close();
+
+    if (mPowerLock) {
+        release_wake_lock ("AudioOutLock");
+        mPowerLock = false;
+    }
+
+    return NO_ERROR;
+}
+
+status_t AudioStreamOutALSA::standby()
+{
+    AutoMutex lock(mLock);
+
+    snd_pcm_drain (mHandle->handle);
+
+    if (mPowerLock) {
+        release_wake_lock ("AudioOutLock");
+        mPowerLock = false;
+    }
+
+    return NO_ERROR;
+}
+
+#define USEC_TO_MSEC(x) ((x + 999) / 1000)
+
+uint32_t AudioStreamOutALSA::latency() const
+{
+    // Android wants latency in milliseconds.
+    return USEC_TO_MSEC (mHandle->latency);
+}
+
+}       // namespace android
diff --git a/acoustics_default.cpp b/acoustics_default.cpp
index 91a169d..6280979 100644
--- a/acoustics_default.cpp
+++ b/acoustics_default.cpp
@@ -15,71 +15,85 @@
  ** limitations under the License.
  */
 
-
-#define LOG_TAG "AudioHardwareALSA"
+#define LOG_TAG "AcousticsModule"
 #include <utils/Log.h>
 
 #include "AudioHardwareALSA.h"
 
 namespace android
 {
-    static int s_device_open(const hw_module_t*, const char*, hw_device_t**);
-    static int s_device_close (hw_device_t*);
-    static status_t s_set_acoustics (snd_pcm_t *, AudioSystem::audio_in_acoustics);
-    static ssize_t s_filter (snd_pcm_t *, void *, ssize_t);
-
-    static hw_module_methods_t s_module_methods = {
-        open: s_device_open
-    };
-
-    extern "C" const hw_module_t HAL_MODULE_INFO_SYM = {
-            tag: HARDWARE_MODULE_TAG,
-            version_major: 1,
-            version_minor: 0,
-            id: ACOUSTICS_HARDWARE_MODULE_ID,
-            name: "ALSA acoustics module",
-            author: "Wind River",
-            methods: &s_module_methods,
-            reserved: {}
-    };
-
-    static int s_device_open(const hw_module_t* module,
-                             const char* name,
-                             hw_device_t** device)
-    {
-        acoustic_device_t *dev;
-        dev = (acoustic_device_t *)malloc(sizeof(*dev));
-        if (! dev) return -ENOMEM;
-
-        memset(dev, 0, sizeof(*dev));
-
-        /* initialize the procs */
-        dev->common.tag = HARDWARE_DEVICE_TAG;
-        dev->common.version = 0;
-        dev->common.module = (hw_module_t *)module;
-        dev->common.close = s_device_close;
-        dev->set_acoustics = s_set_acoustics;
-        dev->filter = s_filter;
-
-        *device = &dev->common;
-        return 0;
-    }
-
-    static int s_device_close (hw_device_t* device)
-    {
-        free(device);
-        return 0;
-    }
-
-    static status_t s_set_acoustics (snd_pcm_t *handle, AudioSystem::audio_in_acoustics acoustics)
-    {
-        LOGD("Acoustics set_acoustics stub called with %d.", (int)acoustics);
-        return NO_ERROR;
-    }
-
-    static ssize_t s_filter (snd_pcm_t *handle, void *buffer, ssize_t frames)
-    {
-        // Default acoustics doesn't apply any filtering
-        return frames;
-    }
+
+static int s_device_open(const hw_module_t*, const char*, hw_device_t**);
+static int s_device_close(hw_device_t*);
+
+static status_t s_use_handle(acoustic_device_t *, alsa_handle_t *);
+static status_t s_cleanup(acoustic_device_t *);
+static status_t s_set_params(acoustic_device_t *,
+        AudioSystem::audio_in_acoustics, void *params);
+
+static hw_module_methods_t s_module_methods = {
+    open            : s_device_open
+};
+
+extern "C" const hw_module_t HAL_MODULE_INFO_SYM = {
+    tag             : HARDWARE_MODULE_TAG,
+    version_major   : 1,
+    version_minor   : 0,
+    id              : ACOUSTICS_HARDWARE_MODULE_ID,
+    name            : "ALSA acoustics module",
+    author          : "Wind River",
+    methods         : &s_module_methods,
+    dso             : 0,
+    reserved        : { 0, },
+};
+
+static int s_device_open(const hw_module_t* module, const char* name,
+        hw_device_t** device)
+{
+    acoustic_device_t *dev;
+    dev = (acoustic_device_t *) malloc(sizeof(*dev));
+    if (!dev) return -ENOMEM;
+
+    memset(dev, 0, sizeof(*dev));
+
+    /* initialize the procs */
+    dev->common.tag = HARDWARE_DEVICE_TAG;
+    dev->common.version = 0;
+    dev->common.module = (hw_module_t *) module;
+    dev->common.close = s_device_close;
+
+    // Required methods...
+    dev->use_handle = s_use_handle;
+    dev->cleanup = s_cleanup;
+    dev->set_params = s_set_params;
+
+    // read, write, and recover are optional methods...
+
+    *device = &dev->common;
+    return 0;
+}
+
+static int s_device_close(hw_device_t* device)
+{
+    free(device);
+    return 0;
+}
+
+static status_t s_use_handle(acoustic_device_t *dev, alsa_handle_t *h)
+{
+    return NO_ERROR;
+}
+
+static status_t s_cleanup(acoustic_device_t *dev)
+{
+    LOGD("Acoustics close stub called.");
+    return NO_ERROR;
+}
+
+static status_t s_set_params(acoustic_device_t *dev,
+        AudioSystem::audio_in_acoustics acoustics, void *params)
+{
+    LOGD("Acoustics set_params stub called with %d.", (int)acoustics);
+    return NO_ERROR;
+}
 }
diff --git a/alsa_default.cpp b/alsa_default.cpp
new file mode 100644
index 0000000..0e8e71e
--- /dev/null
+++ b/alsa_default.cpp
@@ -0,0 +1,535 @@
+/* alsa_default.cpp
+ **
+ ** Copyright 2009 Wind River Systems
+ **
+ ** Licensed under the Apache License, Version 2.0 (the "License");
+ ** you may not use this file except in compliance with the License.
+ ** You may obtain a copy of the License at
+ **
+ **     http://www.apache.org/licenses/LICENSE-2.0
+ **
+ ** Unless required by applicable law or agreed to in writing, software
+ ** distributed under the License is distributed on an "AS IS" BASIS,
+ ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ ** See the License for the specific language governing permissions and
+ ** limitations under the License.
+ */
+
+#define LOG_TAG "ALSAModule"
+#include <utils/Log.h>
+
+#include "AudioHardwareALSA.h"
+#include <media/AudioRecord.h>
+
+#undef DISABLE_HARWARE_RESAMPLING
+
+#define ALSA_NAME_MAX 128
+
+#define ALSA_STRCAT(x,y) \
+    if (strlen(x) + strlen(y) < ALSA_NAME_MAX) \
+        strcat(x, y);
+
+#ifndef ALSA_DEFAULT_SAMPLE_RATE
+#define ALSA_DEFAULT_SAMPLE_RATE 44100 // in Hz
+#endif
+
+namespace android
+{
+
+static int s_device_open(const hw_module_t*, const char*, hw_device_t**);
+static int s_device_close(hw_device_t*);
+static status_t s_init(alsa_device_t *, ALSAHandleList &);
+static status_t s_open(alsa_handle_t *, uint32_t, int);
+static status_t s_close(alsa_handle_t *);
+static status_t s_route(alsa_handle_t *, uint32_t, int);
+
+static hw_module_methods_t s_module_methods = {
+    open            : s_device_open
+};
+
+extern "C" const hw_module_t HAL_MODULE_INFO_SYM = {
+    tag             : HARDWARE_MODULE_TAG,
+    version_major   : 1,
+    version_minor   : 0,
+    id              : ALSA_HARDWARE_MODULE_ID,
+    name            : "ALSA module",
+    author          : "Wind River",
+    methods         : &s_module_methods,
+    dso             : 0,
+    reserved        : { 0, },
+};
+
+static int s_device_open(const hw_module_t* module, const char* name,
+        hw_device_t** device)
+{
+    alsa_device_t *dev;
+    dev = (alsa_device_t *) malloc(sizeof(*dev));
+    if (!dev) return -ENOMEM;
+
+    memset(dev, 0, sizeof(*dev));
+
+    /* initialize the procs */
+    dev->common.tag = HARDWARE_DEVICE_TAG;
+    dev->common.version = 0;
+    dev->common.module = (hw_module_t *) module;
+    dev->common.close = s_device_close;
+    dev->init = s_init;
+    dev->open = s_open;
+    dev->close = s_close;
+    dev->route = s_route;
+
+    *device = &dev->common;
+    return 0;
+}
+
+static int s_device_close(hw_device_t* device)
+{
+    free(device);
+    return 0;
+}
+
+// ----------------------------------------------------------------------------
+
+static const int DEFAULT_SAMPLE_RATE = ALSA_DEFAULT_SAMPLE_RATE;
+
+static const char *devicePrefix[SND_PCM_STREAM_LAST + 1] = {
+        /* SND_PCM_STREAM_PLAYBACK : */"AndroidOut",
+        /* SND_PCM_STREAM_CAPTURE  : */"AndroidIn",
+};
+
+static alsa_handle_t _defaultsOut = {
+    module      : 0,
+    devices     : AudioSystem::DEVICE_OUT_ALL,
+    curDev      : 0,
+    curMode     : 0,
+    handle      : 0,
+    format      : SND_PCM_FORMAT_S16_LE, // AudioSystem::PCM_16_BIT
+    channels    : 2,
+    sampleRate  : DEFAULT_SAMPLE_RATE,
+    latency     : 200000, // Desired Delay in usec
+    bufferSize  : DEFAULT_SAMPLE_RATE / 5, // Desired Number of samples
+    modPrivate  : 0,
+};
+
+static alsa_handle_t _defaultsIn = {
+    module      : 0,
+    devices     : AudioSystem::DEVICE_IN_ALL,
+    curDev      : 0,
+    curMode     : 0,
+    handle      : 0,
+    format      : SND_PCM_FORMAT_S16_LE, // AudioSystem::PCM_16_BIT
+    channels    : 1,
+    sampleRate  : AudioRecord::DEFAULT_SAMPLE_RATE,
+    latency     : 250000, // Desired Delay in usec
+    bufferSize  : 2048, // Desired Number of samples
+    modPrivate  : 0,
+};
+
+struct device_suffix_t {
+    const AudioSystem::audio_devices device;
+    const char *suffix;
+};
+
+/* The following table(s) need to match in order of the route bits
+ */
+static const device_suffix_t deviceSuffix[] = {
+        {AudioSystem::DEVICE_OUT_EARPIECE,       "_Earpiece"},
+        {AudioSystem::DEVICE_OUT_SPEAKER,        "_Speaker"},
+        {AudioSystem::DEVICE_OUT_BLUETOOTH_SCO,  "_Bluetooth"},
+        {AudioSystem::DEVICE_OUT_WIRED_HEADSET,  "_Headset"},
+        {AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP, "_Bluetooth-A2DP"},
+        {AudioSystem::DEVICE_OUT_FM_HEADPHONE,   "_FM"},
+};
+
+static const int deviceSuffixLen = (sizeof(deviceSuffix)
+        / sizeof(device_suffix_t));
+
+// ----------------------------------------------------------------------------
+
+const char *deviceName(alsa_handle_t *handle, uint32_t device, int mode)
+{
+    static char devString[ALSA_NAME_MAX];
+    int hasDevExt = 0;
+
+    strcpy(devString, devicePrefix[0]);
+
+    for (int dev = 0; device && dev < deviceSuffixLen; dev++)
+        if (device & deviceSuffix[dev].device) {
+            ALSA_STRCAT (devString, deviceSuffix[dev].suffix);
+            device &= ~deviceSuffix[dev].device;
+            hasDevExt = 1;
+        }
+
+    if (hasDevExt) switch (mode) {
+    case AudioSystem::MODE_NORMAL:
+        ALSA_STRCAT (devString, "_normal")
+        ;
+        break;
+    case AudioSystem::MODE_RINGTONE:
+        ALSA_STRCAT (devString, "_ringtone")
+        ;
+        break;
+    case AudioSystem::MODE_IN_CALL:
+        ALSA_STRCAT (devString, "_incall")
+        ;
+        break;
+    };
+
+    return devString;
+}
+
+snd_pcm_stream_t direction(alsa_handle_t *handle)
+{
+    return (handle->devices & AudioSystem::DEVICE_OUT_ALL) ? SND_PCM_STREAM_PLAYBACK
+            : SND_PCM_STREAM_CAPTURE;
+}
+
+const char *streamName(alsa_handle_t *handle)
+{
+    return snd_pcm_stream_name(direction(handle));
+}
+
+status_t setHardwareParams(alsa_handle_t *handle)
+{
+    snd_pcm_hw_params_t *hardwareParams;
+    status_t err;
+
+    snd_pcm_uframes_t bufferSize = handle->bufferSize;
+    unsigned int requestedRate = handle->sampleRate;
+    unsigned int latency = handle->latency;
+
+    // snd_pcm_format_description() and snd_pcm_format_name() do not perform
+    // proper bounds checking.
+    bool validFormat = (static_cast<int> (handle->format)
+            > SND_PCM_FORMAT_UNKNOWN) && (static_cast<int> (handle->format)
+            <= SND_PCM_FORMAT_LAST);
+    const char *formatDesc = validFormat ? snd_pcm_format_description(
+            handle->format) : "Invalid Format";
+    const char *formatName = validFormat ? snd_pcm_format_name(handle->format)
+            : "UNKNOWN";
+
+    if (snd_pcm_hw_params_malloc(&hardwareParams) < 0) {
+        LOG_ALWAYS_FATAL("Failed to allocate ALSA hardware parameters!");
+        return NO_INIT;
+    }
+
+    err = snd_pcm_hw_params_any(handle->handle, hardwareParams);
+    if (err < 0) {
+        LOGE("Unable to configure hardware: %s", snd_strerror(err));
+        goto done;
+    }
+
+    // Set the interleaved read and write format.
+    err = snd_pcm_hw_params_set_access(handle->handle, hardwareParams,
+            SND_PCM_ACCESS_RW_INTERLEAVED);
+    if (err < 0) {
+        LOGE("Unable to configure PCM read/write format: %s",
+                snd_strerror(err));
+        goto done;
+    }
+
+    err = snd_pcm_hw_params_set_format(handle->handle, hardwareParams,
+            handle->format);
+    if (err < 0) {
+        LOGE("Unable to configure PCM format %s (%s): %s",
+                formatName, formatDesc, snd_strerror(err));
+        goto done;
+    }
+
+    LOGV("Set %s PCM format to %s (%s)", streamName(), formatName, formatDesc);
+
+    err = snd_pcm_hw_params_set_channels(handle->handle, hardwareParams,
+            handle->channels);
+    if (err < 0) {
+        LOGE("Unable to set channel count to %i: %s",
+                handle->channels, snd_strerror(err));
+        goto done;
+    }
+
+    LOGV("Using %i %s for %s.", handle->channels,
+            handle->channels == 1 ? "channel" : "channels", streamName());
+
+    err = snd_pcm_hw_params_set_rate_near(handle->handle, hardwareParams,
+            &requestedRate, 0);
+
+    if (err < 0)
+        LOGE("Unable to set %s sample rate to %u: %s",
+                streamName(handle), handle->sampleRate, snd_strerror(err));
+    else if (requestedRate != handle->sampleRate)
+        // Some devices have a fixed sample rate, and can not be changed.
+        // This may cause resampling problems; i.e. PCM playback will be too
+        // slow or fast.
+        LOGW("Requested rate (%u HZ) does not match actual rate (%u HZ)",
+                handle->sampleRate, requestedRate);
+    else
+        LOGV("Set %s sample rate to %u HZ", stream, requestedRate);
+
+#ifdef DISABLE_HARWARE_RESAMPLING
+    // Disable hardware re-sampling.
+    err = snd_pcm_hw_params_set_rate_resample(handle->handle,
+            hardwareParams,
+            static_cast<int>(resample));
+    if (err < 0) {
+        LOGE("Unable to %s hardware resampling: %s",
+                resample ? "enable" : "disable",
+                snd_strerror(err));
+        goto done;
+    }
+#endif
+
+    // Make sure we have at least the size we originally wanted
+    err = snd_pcm_hw_params_set_buffer_size(handle->handle, hardwareParams,
+            bufferSize);
+    if (err < 0) {
+        LOGE("Unable to set buffer size to %d:  %s",
+                (int)bufferSize, snd_strerror(err));
+        goto done;
+    }
+
+    // Setup buffers for latency
+    err = snd_pcm_hw_params_set_buffer_time_near(handle->handle,
+            hardwareParams, &latency, NULL);
+    if (err < 0) {
+        /* That didn't work, set the period instead */
+        unsigned int periodTime = latency / 4;
+        err = snd_pcm_hw_params_set_period_time_near(handle->handle,
+                hardwareParams, &periodTime, NULL);
+        if (err < 0) {
+            LOGE("Unable to set the period time for latency: %s", snd_strerror(err));
+            goto done;
+        }
+        snd_pcm_uframes_t periodSize;
+        err = snd_pcm_hw_params_get_period_size(hardwareParams, &periodSize,
+                NULL);
+        if (err < 0) {
+            LOGE("Unable to get the period size for latency: %s", snd_strerror(err));
+            goto done;
+        }
+        bufferSize = periodSize * 4;
+        if (bufferSize < handle->bufferSize) bufferSize = handle->bufferSize;
+        err = snd_pcm_hw_params_set_buffer_size_near(handle->handle,
+                hardwareParams, &bufferSize);
+        if (err < 0) {
+            LOGE("Unable to set the buffer size for latency: %s", snd_strerror(err));
+            goto done;
+        }
+    } else {
+        // OK, we got buffer time near what we expect. See what that did for bufferSize.
+        err = snd_pcm_hw_params_get_buffer_size(hardwareParams, &bufferSize);
+        if (err < 0) {
+            LOGE("Unable to get the buffer size for latency: %s", snd_strerror(err));
+            goto done;
+        }
+        // Does set_buffer_time_near change the passed value? It should.
+        err = snd_pcm_hw_params_get_buffer_time(hardwareParams, &latency, NULL);
+        if (err < 0) {
+            LOGE("Unable to get the buffer time for latency: %s", snd_strerror(err));
+            goto done;
+        }
+        unsigned int periodTime = latency / 4;
+        err = snd_pcm_hw_params_set_period_time_near(handle->handle,
+                hardwareParams, &periodTime, NULL);
+        if (err < 0) {
+            LOGE("Unable to set the period time for latency: %s", snd_strerror(err));
+            goto done;
+        }
+    }
+
+    LOGV("Buffer size: %d", (int)bufferSize);
+    LOGV("Latency: %d", (int)latency);
+
+    handle->bufferSize = bufferSize;
+    handle->latency = latency;
+
+    // Commit the hardware parameters back to the device.
+    err = snd_pcm_hw_params(handle->handle, hardwareParams);
+    if (err < 0) LOGE("Unable to set hardware parameters: %s", snd_strerror(err));
+
+    done:
+    snd_pcm_hw_params_free(hardwareParams);
+
+    return err;
+}
+
+status_t setSoftwareParams(alsa_handle_t *handle)
+{
+    snd_pcm_sw_params_t * softwareParams;
+    int err;
+
+    snd_pcm_uframes_t bufferSize = 0;
+    snd_pcm_uframes_t periodSize = 0;
+    snd_pcm_uframes_t startThreshold, stopThreshold;
+
+    if (snd_pcm_sw_params_malloc(&softwareParams) < 0) {
+        LOG_ALWAYS_FATAL("Failed to allocate ALSA software parameters!");
+        return NO_INIT;
+    }
+
+    // Get the current software parameters
+    err = snd_pcm_sw_params_current(handle->handle, softwareParams);
+    if (err < 0) {
+        LOGE("Unable to get software parameters: %s", snd_strerror(err));
+        goto done;
+    }
+
+    // Configure ALSA to start the transfer when the buffer is almost full.
+    snd_pcm_get_params(handle->handle, &bufferSize, &periodSize);
+
+    if (handle->devices & AudioSystem::DEVICE_OUT_ALL) {
+        // For playback, configure ALSA to start the transfer when the
+        // buffer is full.
+        startThreshold = bufferSize - 1;
+        stopThreshold = bufferSize;
+    } else {
+        // For recording, configure ALSA to start the transfer on the
+        // first frame.
+        startThreshold = 1;
+        stopThreshold = bufferSize;
+    }
+
+    err = snd_pcm_sw_params_set_start_threshold(handle->handle, softwareParams,
+            startThreshold);
+    if (err < 0) {
+        LOGE("Unable to set start threshold to %lu frames: %s",
+                startThreshold, snd_strerror(err));
+        goto done;
+    }
+
+    err = snd_pcm_sw_params_set_stop_threshold(handle->handle, softwareParams,
+            stopThreshold);
+    if (err < 0) {
+        LOGE("Unable to set stop threshold to %lu frames: %s",
+                stopThreshold, snd_strerror(err));
+        goto done;
+    }
+
+    // Allow the transfer to start when at least periodSize samples can be
+    // processed.
+    err = snd_pcm_sw_params_set_avail_min(handle->handle, softwareParams,
+            periodSize);
+    if (err < 0) {
+        LOGE("Unable to configure available minimum to %lu: %s",
+                periodSize, snd_strerror(err));
+        goto done;
+    }
+
+    // Commit the software parameters back to the device.
+    err = snd_pcm_sw_params(handle->handle, softwareParams);
+    if (err < 0) LOGE("Unable to configure software parameters: %s",
+            snd_strerror(err));
+
+    done:
+    snd_pcm_sw_params_free(softwareParams);
+
+    return err;
+}
+
+// ----------------------------------------------------------------------------
+
+static status_t s_init(alsa_device_t *module, ALSAHandleList &list)
+{
+    list.clear();
+
+    snd_pcm_uframes_t bufferSize = _defaultsOut.bufferSize;
+
+    for (size_t i = 1; (bufferSize & ~i) != 0; i <<= 1)
+        bufferSize &= ~i;
+
+    _defaultsOut.module = module;
+    _defaultsOut.bufferSize = bufferSize;
+
+    list.push_back(_defaultsOut);
+
+    bufferSize = _defaultsIn.bufferSize;
+
+    for (size_t i = 1; (bufferSize & ~i) != 0; i <<= 1)
+        bufferSize &= ~i;
+
+    _defaultsIn.module = module;
+    _defaultsIn.bufferSize = bufferSize;
+
+    list.push_back(_defaultsIn);
+
+    return NO_ERROR;
+}
+
+static status_t s_open(alsa_handle_t *handle, uint32_t devices, int mode)
+{
+    // Close off previously opened device.
+    // It would be nice to determine if the underlying device actually
+    // changes, but we might be recovering from an error or manipulating
+    // mixer settings (see asound.conf).
+    //
+    s_close(handle);
+
+    LOGD("open called for devices %08x in mode %d...", devices, mode);
+
+    const char *stream = streamName(handle);
+    const char *devName = deviceName(handle, devices, mode);
+
+    int err;
+
+    for (;;) {
+        // The PCM stream is opened in blocking mode, per ALSA defaults.  The
+        // AudioFlinger seems to assume blocking mode too, so asynchronous mode
+        // should not be used.
+        err = snd_pcm_open(&handle->handle, devName, direction(handle),
+                SND_PCM_ASYNC);
+        if (err == 0) break;
+
+        // See if there is a less specific name we can try.
+        // Note: We are changing the contents of a const char * here.
+        char *tail = strrchr(devName, '_');
+        if (!tail) break;
+        *tail = 0;
+    }
+
+    if (err < 0) {
+        // None of the Android defined audio devices exist. Open a generic one.
+        devName = "default";
+        err = snd_pcm_open(&handle->handle, devName, direction(handle), 0);
+    }
+
+    if (err < 0) {
+        LOGE("Failed to Initialize any ALSA %s device: %s",
+                stream, strerror(err));
+        return NO_INIT;
+    }
+
+    err = setHardwareParams(handle);
+
+    if (err == NO_ERROR) err = setSoftwareParams(handle);
+
+    LOGI("Initialized ALSA %s device %s", stream, devName);
+
+    handle->curDev = devices;
+    handle->curMode = mode;
+
+    return err;
+}
+
+static status_t s_close(alsa_handle_t *handle)
+{
+    status_t err = NO_ERROR;
+    snd_pcm_t *h = handle->handle;
+    handle->handle = 0;
+    handle->curDev = 0;
+    handle->curMode = 0;
+    if (h) {
+        snd_pcm_drain(h);
+        err = snd_pcm_close(h);
+    }
+
+    return err;
+}
+
+static status_t s_route(alsa_handle_t *handle, uint32_t devices, int mode)
+{
+    LOGD("route called for devices %08x in mode %d...", devices, mode);
+
+    if (handle->handle && handle->curDev == devices && handle->curMode == mode) return NO_ERROR;
+
+    return s_open(handle, devices, mode);
+}
+
+}