hardware_legacy: provide HAL helpers for legacy audio users

This doesn't actually create a HAL, but rather a set of helper static
libraries that device specific libraries (i.e. the old libaudio pieces)
can link against to create a proper audio HAL module.

We provide an audio_policy static wrapper and audio hardware interface
static wrapper.

Change-Id: Ie56195447ad24b83888f752dca24674b0afd8a76
Signed-off-by: Dima Zavin <dima@android.com>

17 files changed, 1593 insertions, 61 deletions

diff --git a/audio/A2dpAudioInterface.cpp b/audio/A2dpAudioInterface.cpp
index d926cb1..2d78858 100644
--- a/audio/A2dpAudioInterface.cpp
+++ b/audio/A2dpAudioInterface.cpp
@@ -25,7 +25,8 @@
 #include "audio/liba2dp.h"
 #include <hardware_legacy/power.h>
 
-namespace android {
+
+namespace android_audio_legacy {
 
 static const char *sA2dpWakeLock = "A2dpOutputStream";
 #define MAX_WRITE_RETRIES  5
diff --git a/audio/A2dpAudioInterface.h b/audio/A2dpAudioInterface.h
index dbe2c6a..8fe9745 100644
--- a/audio/A2dpAudioInterface.h
+++ b/audio/A2dpAudioInterface.h
@@ -25,7 +25,8 @@
 #include <hardware_legacy/AudioHardwareBase.h>
 
 
-namespace android {
+namespace android_audio_legacy {
+    using android::Mutex;
 
 class A2dpAudioInterface : public AudioHardwareBase
 {
diff --git a/audio/AudioHardwareGeneric.cpp b/audio/AudioHardwareGeneric.cpp
index d63c031..61286e4 100644
--- a/audio/AudioHardwareGeneric.cpp
+++ b/audio/AudioHardwareGeneric.cpp
@@ -32,7 +32,9 @@
 #include "AudioHardwareGeneric.h"
 #include <media/AudioRecord.h>
 
-namespace android {
+#include <hardware_legacy/AudioSystemLegacy.h>
+
+namespace android_audio_legacy {
 
 // ----------------------------------------------------------------------------
 
diff --git a/audio/AudioHardwareGeneric.h b/audio/AudioHardwareGeneric.h
index aa4e78d..7b41e95 100644
--- a/audio/AudioHardwareGeneric.h
+++ b/audio/AudioHardwareGeneric.h
@@ -23,9 +23,12 @@
 
 #include <utils/threads.h>
 
+#include <hardware_legacy/AudioSystemLegacy.h>
 #include <hardware_legacy/AudioHardwareBase.h>
 
-namespace android {
+namespace android_audio_legacy {
+    using android::Mutex;
+    using android::AutoMutex;
 
 // ----------------------------------------------------------------------------
 
diff --git a/audio/AudioHardwareInterface.cpp b/audio/AudioHardwareInterface.cpp
index f58e4c0..9cec267 100644
--- a/audio/AudioHardwareInterface.cpp
+++ b/audio/AudioHardwareInterface.cpp
@@ -38,7 +38,7 @@
 // change to 1 to log routing calls
 #define LOG_ROUTING_CALLS 1
 
-namespace android {
+namespace android_audio_legacy {
 
 #if LOG_ROUTING_CALLS
 static const char* routingModeStrings[] =
@@ -66,46 +66,7 @@ static const char* displayMode(int mode)
 
 AudioHardwareInterface* AudioHardwareInterface::create()
 {
-    /*
-     * FIXME: This code needs to instantiate the correct audio device
-     * interface. For now - we use compile-time switches.
-     */
-    AudioHardwareInterface* hw = 0;
-    char value[PROPERTY_VALUE_MAX];
-
-#ifdef GENERIC_AUDIO
-    hw = new AudioHardwareGeneric();
-#else
-    // if running in emulation - use the emulator driver
-    if (property_get("ro.kernel.qemu", value, 0)) {
-        LOGD("Running in emulation - using generic audio driver");
-        hw = new AudioHardwareGeneric();
-    }
-    else {
-        LOGV("Creating Vendor Specific AudioHardware");
-        hw = createAudioHardware();
-    }
-#endif
-    if (hw->initCheck() != NO_ERROR) {
-        LOGW("Using stubbed audio hardware. No sound will be produced.");
-        delete hw;
-        hw = new AudioHardwareStub();
-    }
-    
-#ifdef WITH_A2DP
-    hw = new A2dpAudioInterface(hw);
-#endif
-
-#ifdef ENABLE_AUDIO_DUMP
-    // This code adds a record of buffers in a file to write calls made by AudioFlinger.
-    // It replaces the current AudioHardwareInterface object by an intermediate one which
-    // will record buffers in a file (after sending them to hardware) for testing purpose.
-    // This feature is enabled by defining symbol ENABLE_AUDIO_DUMP.
-    // The output file is set with setParameters("test_cmd_file_name=<name>"). Pause are not recorded in the file.
-    LOGV("opening PCM dump interface");
-    hw = new AudioDumpInterface(hw);    // replace interface
-#endif
-    return hw;
+    return NULL;
 }
 
 AudioStreamOut::~AudioStreamOut()
diff --git a/audio/AudioHardwareStub.cpp b/audio/AudioHardwareStub.cpp
index d481150..70a8309 100644
--- a/audio/AudioHardwareStub.cpp
+++ b/audio/AudioHardwareStub.cpp
@@ -25,7 +25,7 @@
 #include "AudioHardwareStub.h"
 #include <media/AudioRecord.h>
 
-namespace android {
+namespace android_audio_legacy {
 
 // ----------------------------------------------------------------------------
 
diff --git a/audio/AudioHardwareStub.h b/audio/AudioHardwareStub.h
index 06a29de..0858f37 100644
--- a/audio/AudioHardwareStub.h
+++ b/audio/AudioHardwareStub.h
@@ -23,7 +23,7 @@
 
 #include <hardware_legacy/AudioHardwareBase.h>
 
-namespace android {
+namespace android_audio_legacy {
 
 // ----------------------------------------------------------------------------
 
diff --git a/audio/AudioPolicyCompatClient.cpp b/audio/AudioPolicyCompatClient.cpp
new file mode 100644
index 0000000..a685594
--- /dev/null
+++ b/audio/AudioPolicyCompatClient.cpp
@@ -0,0 +1,142 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AudioPolicyCompatClient"
+//#define LOG_NDEBUG 0
+
+#include <stdint.h>
+
+#include <hardware/hardware.h>
+#include <hardware/audio.h>
+#include <hardware/audio_policy.h>
+#include <hardware/audio_policy_hal.h>
+
+#include <hardware_legacy/AudioSystemLegacy.h>
+
+#include "AudioPolicyCompatClient.h"
+
+namespace android_audio_legacy {
+
+audio_io_handle_t AudioPolicyCompatClient::openOutput(uint32_t *pDevices,
+                                uint32_t *pSamplingRate,
+                                uint32_t *pFormat,
+                                uint32_t *pChannels,
+                                uint32_t *pLatencyMs,
+                                AudioSystem::output_flags flags)
+{
+    return mServiceOps->open_output(mService, pDevices, pSamplingRate, pFormat,
+                                    pChannels, pLatencyMs,
+                                    (audio_policy_output_flags_t)flags);
+}
+
+audio_io_handle_t AudioPolicyCompatClient::openDuplicateOutput(audio_io_handle_t output1,
+                                                          audio_io_handle_t output2)
+{
+    return mServiceOps->open_duplicate_output(mService, output1, output2);
+}
+
+status_t AudioPolicyCompatClient::closeOutput(audio_io_handle_t output)
+{
+    return mServiceOps->close_output(mService, output);
+}
+
+status_t AudioPolicyCompatClient::suspendOutput(audio_io_handle_t output)
+{
+    return mServiceOps->suspend_output(mService, output);
+}
+
+status_t AudioPolicyCompatClient::restoreOutput(audio_io_handle_t output)
+{
+    return mServiceOps->restore_output(mService, output);
+}
+
+audio_io_handle_t AudioPolicyCompatClient::openInput(uint32_t *pDevices,
+                                uint32_t *pSamplingRate,
+                                uint32_t *pFormat,
+                                uint32_t *pChannels,
+                                uint32_t acoustics)
+{
+    return mServiceOps->open_input(mService, pDevices, pSamplingRate, pFormat,
+                                   pChannels, acoustics);
+}
+
+status_t AudioPolicyCompatClient::closeInput(audio_io_handle_t input)
+{
+    return mServiceOps->close_input(mService, input);
+}
+
+status_t AudioPolicyCompatClient::setStreamOutput(AudioSystem::stream_type stream,
+                                             audio_io_handle_t output)
+{
+    return mServiceOps->set_stream_output(mService, (audio_stream_type_t)stream,
+                                          output);
+}
+
+status_t AudioPolicyCompatClient::moveEffects(int session, audio_io_handle_t srcOutput,
+                                               audio_io_handle_t dstOutput)
+{
+    return mServiceOps->move_effects(mService, session, srcOutput, dstOutput);
+}
+
+String8 AudioPolicyCompatClient::getParameters(audio_io_handle_t ioHandle, const String8& keys)
+{
+    char *str;
+    String8 out_str8;
+
+    str = mServiceOps->get_parameters(mService, ioHandle, keys.string());
+    out_str8 = String8(str);
+    free(str);
+
+    return out_str8;
+}
+
+void AudioPolicyCompatClient::setParameters(audio_io_handle_t ioHandle,
+                                            const String8& keyValuePairs,
+                                            int delayMs)
+{
+    mServiceOps->set_parameters(mService, ioHandle, keyValuePairs.string(),
+                           delayMs);
+}
+
+status_t AudioPolicyCompatClient::setStreamVolume(
+                                             AudioSystem::stream_type stream,
+                                             float volume,
+                                             audio_io_handle_t output,
+                                             int delayMs)
+{
+    return mServiceOps->set_stream_volume(mService, (audio_stream_type_t)stream,
+                                          volume, output, delayMs);
+}
+
+status_t AudioPolicyCompatClient::startTone(ToneGenerator::tone_type tone,
+                                       AudioSystem::stream_type stream)
+{
+    return mServiceOps->start_tone(mService,
+                                   AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION,
+                                   (audio_stream_type_t)stream);
+}
+
+status_t AudioPolicyCompatClient::stopTone()
+{
+    return mServiceOps->stop_tone(mService);
+}
+
+status_t AudioPolicyCompatClient::setVoiceVolume(float volume, int delayMs)
+{
+    return mServiceOps->set_voice_volume(mService, volume, delayMs);
+}
+
+}; // namespace android_audio_legacy
diff --git a/audio/AudioPolicyCompatClient.h b/audio/AudioPolicyCompatClient.h
new file mode 100644
index 0000000..073d379
--- /dev/null
+++ b/audio/AudioPolicyCompatClient.h
@@ -0,0 +1,79 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_AUDIOPOLICYCLIENTLEGACY_H
+#define ANDROID_AUDIOPOLICYCLIENTLEGACY_H
+
+#include <hardware/audio.h>
+#include <hardware/audio_policy.h>
+#include <hardware/audio_policy_hal.h>
+
+#include <hardware_legacy/AudioSystemLegacy.h>
+#include <hardware_legacy/AudioPolicyInterface.h>
+
+/************************************/
+/* FOR BACKWARDS COMPATIBILITY ONLY */
+/************************************/
+namespace android_audio_legacy {
+
+class AudioPolicyCompatClient : public AudioPolicyClientInterface {
+public:
+    AudioPolicyCompatClient(struct audio_policy_service_ops *serviceOps,
+                            void *service) :
+            mServiceOps(serviceOps) , mService(service) {}
+
+    virtual audio_io_handle_t openOutput(uint32_t *pDevices,
+                                         uint32_t *pSamplingRate,
+                                         uint32_t *pFormat,
+                                         uint32_t *pChannels,
+                                         uint32_t *pLatencyMs,
+                                         AudioSystem::output_flags flags);
+    virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
+                                                  audio_io_handle_t output2);
+    virtual status_t closeOutput(audio_io_handle_t output);
+    virtual status_t suspendOutput(audio_io_handle_t output);
+    virtual status_t restoreOutput(audio_io_handle_t output);
+    virtual audio_io_handle_t openInput(uint32_t *pDevices,
+                                        uint32_t *pSamplingRate,
+                                        uint32_t *pFormat,
+                                        uint32_t *pChannels,
+                                        uint32_t acoustics);
+    virtual status_t closeInput(audio_io_handle_t input);
+    virtual status_t setStreamOutput(AudioSystem::stream_type stream, audio_io_handle_t output);
+    virtual status_t moveEffects(int session,
+                                 audio_io_handle_t srcOutput,
+                                 audio_io_handle_t dstOutput);
+
+    virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys);
+    virtual void setParameters(audio_io_handle_t ioHandle,
+                               const String8& keyValuePairs,
+                               int delayMs = 0);
+    virtual status_t setStreamVolume(AudioSystem::stream_type stream,
+                                     float volume,
+                                     audio_io_handle_t output,
+                                     int delayMs = 0);
+    virtual status_t startTone(ToneGenerator::tone_type tone, AudioSystem::stream_type stream);
+    virtual status_t stopTone();
+    virtual status_t setVoiceVolume(float volume, int delayMs = 0);
+
+private:
+    struct audio_policy_service_ops* mServiceOps;
+    void*                            mService;
+};
+
+}; // namespace android_audio_legacy
+
+#endif // ANDROID_AUDIOPOLICYCLIENTLEGACY_H
diff --git a/audio/AudioPolicyManagerBase.cpp b/audio/AudioPolicyManagerBase.cpp
index 32d92dc..6cdcec4 100644
--- a/audio/AudioPolicyManagerBase.cpp
+++ b/audio/AudioPolicyManagerBase.cpp
@@ -21,8 +21,7 @@
 #include <media/mediarecorder.h>
 #include <math.h>
 
-namespace android {
-
+namespace android_audio_legacy {
 
 // ----------------------------------------------------------------------------
 // AudioPolicyInterface implementation
@@ -542,7 +541,7 @@ audio_io_handle_t AudioPolicyManagerBase::getOutput(AudioSystem::stream_type str
     }
 
 
-    LOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d, format %d, channels %x, flags %x",
+    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;
@@ -2114,7 +2113,7 @@ bool AudioPolicyManagerBase::needsDirectOuput(AudioSystem::stream_type stream,
                                     uint32_t device)
 {
    return ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
-          (format != 0 && !AudioSystem::isLinearPCM(format)));
+          (format !=0 && !AudioSystem::isLinearPCM(format)));
 }
 
 uint32_t AudioPolicyManagerBase::getMaxEffectsCpuLoad()
@@ -2166,7 +2165,7 @@ void AudioPolicyManagerBase::AudioOutputDescriptor::changeRefCount(AudioSystem::
         return;
     }
     mRefCount[stream] += delta;
-    LOGV("changeRefCount() delta %d, stream %d, refCount %d", delta, stream, mRefCount[stream]);
+    LOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
 }
 
 uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::refCount()
@@ -2222,8 +2221,7 @@ status_t AudioPolicyManagerBase::AudioOutputDescriptor::dump(int fd)
 
 AudioPolicyManagerBase::AudioInputDescriptor::AudioInputDescriptor()
     : mSamplingRate(0), mFormat(0), mChannels(0),
-      mAcoustics((AudioSystem::audio_in_acoustics)0), mDevice(0), mRefCount(0),
-      mInputSource(0)
+     mAcoustics((AudioSystem::audio_in_acoustics)0), mDevice(0), mRefCount(0)
 {
 }
 
diff --git a/audio/audio_hw_hal.cpp b/audio/audio_hw_hal.cpp
new file mode 100644
index 0000000..deb943d
--- /dev/null
+++ b/audio/audio_hw_hal.cpp
@@ -0,0 +1,577 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "legacy_audio_hw_hal"
+//#define LOG_NDEBUG 0
+
+#include <stdint.h>
+
+#include <hardware/hardware.h>
+#include <hardware/audio.h>
+#include <hardware/audio_hal.h>
+
+#include <hardware_legacy/AudioHardwareInterface.h>
+#include <hardware_legacy/AudioSystemLegacy.h>
+
+namespace android_audio_legacy {
+
+extern "C" {
+
+struct legacy_audio_module {
+    struct audio_module module;
+};
+
+struct legacy_audio_device {
+    struct audio_hw_device device;
+
+    struct AudioHardwareInterface *hwif;
+};
+
+struct legacy_stream_out {
+    struct audio_stream_out stream;
+
+    AudioStreamOut *legacy_out;
+};
+
+struct legacy_stream_in {
+    struct audio_stream_in stream;
+
+    AudioStreamIn *legacy_in;
+};
+
+/** audio_stream_out implementation **/
+static uint32_t out_get_sample_rate(const struct audio_stream *stream)
+{
+    const struct legacy_stream_out *out =
+        reinterpret_cast<const struct legacy_stream_out *>(stream);
+    return out->legacy_out->sampleRate();
+}
+
+static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
+{
+    struct legacy_stream_out *out =
+        reinterpret_cast<struct legacy_stream_out *>(stream);
+
+    LOGE("(%s:%d) %s: Implement me!", __FILE__, __LINE__, __func__);
+    /* TODO: implement this */
+    return 0;
+}
+
+static size_t out_get_buffer_size(const struct audio_stream *stream)
+{
+    const struct legacy_stream_out *out =
+        reinterpret_cast<const struct legacy_stream_out *>(stream);
+    return out->legacy_out->bufferSize();
+}
+
+static uint32_t out_get_channels(const struct audio_stream *stream)
+{
+    const struct legacy_stream_out *out =
+        reinterpret_cast<const struct legacy_stream_out *>(stream);
+    return out->legacy_out->channels();
+}
+
+static int out_get_format(const struct audio_stream *stream)
+{
+    const struct legacy_stream_out *out =
+        reinterpret_cast<const struct legacy_stream_out *>(stream);
+    return out->legacy_out->format();
+}
+
+static int out_set_format(struct audio_stream *stream, int format)
+{
+    struct legacy_stream_out *out =
+        reinterpret_cast<struct legacy_stream_out *>(stream);
+    LOGE("(%s:%d) %s: Implement me!", __FILE__, __LINE__, __func__);
+    /* TODO: implement me */
+    return 0;
+}
+
+static int out_standby(struct audio_stream *stream)
+{
+    struct legacy_stream_out *out =
+        reinterpret_cast<struct legacy_stream_out *>(stream);
+    return out->legacy_out->standby();
+}
+
+static int out_dump(const struct audio_stream *stream, int fd)
+{
+    const struct legacy_stream_out *out =
+        reinterpret_cast<const struct legacy_stream_out *>(stream);
+    Vector<String16> args;
+    return out->legacy_out->dump(fd, args);
+}
+
+static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+    struct legacy_stream_out *out =
+        reinterpret_cast<struct legacy_stream_out *>(stream);
+    return out->legacy_out->setParameters(String8(kvpairs));
+}
+
+static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
+{
+    const struct legacy_stream_out *out =
+        reinterpret_cast<const struct legacy_stream_out *>(stream);
+    String8 s8;
+    s8 = out->legacy_out->getParameters(String8(keys));
+    return strdup(s8.string());
+}
+
+static uint32_t out_get_latency(const struct audio_stream_out *stream)
+{
+    const struct legacy_stream_out *out =
+        reinterpret_cast<const struct legacy_stream_out *>(stream);
+    return out->legacy_out->latency();
+}
+
+static int out_set_volume(struct audio_stream_out *stream, float left,
+                          float right)
+{
+    struct legacy_stream_out *out =
+        reinterpret_cast<struct legacy_stream_out *>(stream);
+    return out->legacy_out->setVolume(left, right);
+}
+
+static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
+                         size_t bytes)
+{
+    struct legacy_stream_out *out =
+        reinterpret_cast<struct legacy_stream_out *>(stream);
+    return out->legacy_out->write(buffer, bytes);
+}
+
+static int out_get_render_position(const struct audio_stream_out *stream,
+                                   uint32_t *dsp_frames)
+{
+    const struct legacy_stream_out *out =
+        reinterpret_cast<const struct legacy_stream_out *>(stream);
+    return out->legacy_out->getRenderPosition(dsp_frames);
+}
+
+/** audio_stream_in implementation **/
+static uint32_t in_get_sample_rate(const struct audio_stream *stream)
+{
+    const struct legacy_stream_in *in =
+        reinterpret_cast<const struct legacy_stream_in *>(stream);
+    return in->legacy_in->sampleRate();
+}
+
+static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
+{
+    struct legacy_stream_in *in =
+        reinterpret_cast<struct legacy_stream_in *>(stream);
+
+    LOGE("(%s:%d) %s: Implement me!", __FILE__, __LINE__, __func__);
+    /* TODO: implement this */
+    return 0;
+}
+
+static size_t in_get_buffer_size(const struct audio_stream *stream)
+{
+    const struct legacy_stream_in *in =
+        reinterpret_cast<const struct legacy_stream_in *>(stream);
+    return in->legacy_in->bufferSize();
+}
+
+static uint32_t in_get_channels(const struct audio_stream *stream)
+{
+    const struct legacy_stream_in *in =
+        reinterpret_cast<const struct legacy_stream_in *>(stream);
+    return in->legacy_in->channels();
+}
+
+static int in_get_format(const struct audio_stream *stream)
+{
+    const struct legacy_stream_in *in =
+        reinterpret_cast<const struct legacy_stream_in *>(stream);
+    return in->legacy_in->format();
+}
+
+static int in_set_format(struct audio_stream *stream, int format)
+{
+    struct legacy_stream_in *in =
+        reinterpret_cast<struct legacy_stream_in *>(stream);
+    LOGE("(%s:%d) %s: Implement me!", __FILE__, __LINE__, __func__);
+    /* TODO: implement me */
+    return 0;
+}
+
+static int in_standby(struct audio_stream *stream)
+{
+    struct legacy_stream_in *in = reinterpret_cast<struct legacy_stream_in *>(stream);
+    return in->legacy_in->standby();
+}
+
+static int in_dump(const struct audio_stream *stream, int fd)
+{
+    const struct legacy_stream_in *in =
+        reinterpret_cast<const struct legacy_stream_in *>(stream);
+    Vector<String16> args;
+    return in->legacy_in->dump(fd, args);
+}
+
+static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+    struct legacy_stream_in *in =
+        reinterpret_cast<struct legacy_stream_in *>(stream);
+    return in->legacy_in->setParameters(String8(kvpairs));
+}
+
+static char * in_get_parameters(const struct audio_stream *stream,
+                                const char *keys)
+{
+    const struct legacy_stream_in *in =
+        reinterpret_cast<const struct legacy_stream_in *>(stream);
+    String8 s8;
+    s8 = in->legacy_in->getParameters(String8(keys));
+    return strdup(s8.string());
+}
+
+static int in_set_gain(struct audio_stream_in *stream, float gain)
+{
+    struct legacy_stream_in *in =
+        reinterpret_cast<struct legacy_stream_in *>(stream);
+    return in->legacy_in->setGain(gain);
+}
+
+static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
+                       size_t bytes)
+{
+    struct legacy_stream_in *in =
+        reinterpret_cast<struct legacy_stream_in *>(stream);
+    return in->legacy_in->read(buffer, bytes);
+}
+
+static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
+{
+    struct legacy_stream_in *in =
+        reinterpret_cast<struct legacy_stream_in *>(stream);
+    return in->legacy_in->getInputFramesLost();
+}
+
+/** audio_hw_device implementation **/
+static inline struct legacy_audio_device * to_ladev(struct audio_hw_device *dev)
+{
+    return reinterpret_cast<struct legacy_audio_device *>(dev);
+}
+
+static inline const struct legacy_audio_device * to_cladev(const struct audio_hw_device *dev)
+{
+    return reinterpret_cast<const struct legacy_audio_device *>(dev);
+}
+
+static uint32_t adev_get_supported_devices(const struct audio_hw_device *dev)
+{
+    /* XXX: The old AudioHardwareInterface interface is not smart enough to
+     * tell us this, so we'll lie and basically tell AF that we support the
+     * below input/output devices and cross our fingers. To do things properly,
+     * audio hardware interfaces that need advanced features (like this) should
+     * convert to the new HAL interface and not use this wrapper. */
+
+    return (/* OUT */
+            AUDIO_DEVICE_OUT_EARPIECE |
+            AUDIO_DEVICE_OUT_SPEAKER |
+            AUDIO_DEVICE_OUT_WIRED_HEADSET |
+            AUDIO_DEVICE_OUT_WIRED_HEADPHONE |
+            AUDIO_DEVICE_OUT_AUX_DIGITAL |
+            AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET |
+            AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET |
+            AUDIO_DEVICE_OUT_ALL_SCO |
+            AUDIO_DEVICE_OUT_DEFAULT |
+            /* IN */
+            AUDIO_DEVICE_IN_COMMUNICATION |
+            AUDIO_DEVICE_IN_AMBIENT |
+            AUDIO_DEVICE_IN_BUILTIN_MIC |
+            AUDIO_DEVICE_IN_WIRED_HEADSET |
+            AUDIO_DEVICE_IN_AUX_DIGITAL |
+            AUDIO_DEVICE_IN_BACK_MIC |
+            AUDIO_DEVICE_IN_ALL_SCO |
+            AUDIO_DEVICE_IN_DEFAULT);
+}
+
+static int adev_init_check(const struct audio_hw_device *dev)
+{
+    const struct legacy_audio_device *ladev = to_cladev(dev);
+
+    return ladev->hwif->initCheck();
+}
+
+static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
+{
+    struct legacy_audio_device *ladev = to_ladev(dev);
+    return ladev->hwif->setVoiceVolume(volume);
+}
+
+static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
+{
+    struct legacy_audio_device *ladev = to_ladev(dev);
+    return ladev->hwif->setMasterVolume(volume);
+}
+
+static int adev_set_mode(struct audio_hw_device *dev, int mode)
+{
+    struct legacy_audio_device *ladev = to_ladev(dev);
+    return ladev->hwif->setMode(mode);
+}
+
+static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
+{
+    struct legacy_audio_device *ladev = to_ladev(dev);
+    return ladev->hwif->setMicMute(state);
+}
+
+static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
+{
+    const struct legacy_audio_device *ladev = to_cladev(dev);
+    return ladev->hwif->getMicMute(state);
+}
+
+static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
+{
+    struct legacy_audio_device *ladev = to_ladev(dev);
+    return ladev->hwif->setParameters(String8(kvpairs));
+}
+
+static char * adev_get_parameters(const struct audio_hw_device *dev,
+                                  const char *keys)
+{
+    const struct legacy_audio_device *ladev = to_cladev(dev);
+    String8 s8;
+
+    s8 = ladev->hwif->getParameters(String8(keys));
+    return strdup(s8.string());
+}
+
+static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
+                                         uint32_t sample_rate, int format,
+                                         int channel_count)
+{
+    const struct legacy_audio_device *ladev = to_cladev(dev);
+    return ladev->hwif->getInputBufferSize(sample_rate, format, channel_count);
+}
+
+static int adev_open_output_stream(struct audio_hw_device *dev,
+                                   uint32_t devices,
+                                   int *format,
+                                   uint32_t *channels,
+                                   uint32_t *sample_rate,
+                                   struct audio_stream_out **stream_out)
+{
+    struct legacy_audio_device *ladev = to_ladev(dev);
+    status_t status;
+    struct legacy_stream_out *out;
+    int ret;
+
+    out = (struct legacy_stream_out *)calloc(1, sizeof(*out));
+    if (!out)
+        return -ENOMEM;
+
+    out->legacy_out = ladev->hwif->openOutputStream(devices, format, channels,
+                                                    sample_rate, &status);
+    if (!out->legacy_out) {
+        ret = status;
+        goto err_open;
+    }
+
+    out->stream.common.get_sample_rate = out_get_sample_rate;
+    out->stream.common.set_sample_rate = out_set_sample_rate;
+    out->stream.common.get_buffer_size = out_get_buffer_size;
+    out->stream.common.get_channels = out_get_channels;
+    out->stream.common.get_format = out_get_format;
+    out->stream.common.set_format = out_set_format;
+    out->stream.common.standby = out_standby;
+    out->stream.common.dump = out_dump;
+    out->stream.common.set_parameters = out_set_parameters;
+    out->stream.common.get_parameters = out_get_parameters;
+    out->stream.get_latency = out_get_latency;
+    out->stream.set_volume = out_set_volume;
+    out->stream.write = out_write;
+    out->stream.get_render_position = out_get_render_position;
+
+    *stream_out = &out->stream;
+    return 0;
+
+err_open:
+    free(out);
+    *stream_out = NULL;
+    return ret;
+}
+
+static void adev_close_output_stream(struct audio_hw_device *dev,
+                                     struct audio_stream_out* stream)
+{
+    struct legacy_audio_device *ladev = to_ladev(dev);
+    struct legacy_stream_out *out = reinterpret_cast<struct legacy_stream_out *>(stream);
+
+    ladev->hwif->closeOutputStream(out->legacy_out);
+    free(out);
+}
+
+/** This method creates and opens the audio hardware input stream */
+static int adev_open_input_stream(struct audio_hw_device *dev,
+                                  uint32_t devices, int *format,
+                                  uint32_t *channels, uint32_t *sample_rate,
+                                  audio_in_acoustics_t acoustics,
+                                  struct audio_stream_in **stream_in)
+{
+    struct legacy_audio_device *ladev = to_ladev(dev);
+    status_t status;
+    struct legacy_stream_in *in;
+    int ret;
+
+    in = (struct legacy_stream_in *)calloc(1, sizeof(*in));
+    if (!in)
+        return -ENOMEM;
+
+    in->legacy_in = ladev->hwif->openInputStream(devices, format, channels,
+                                    sample_rate, &status,
+                                    (AudioSystem::audio_in_acoustics)acoustics);
+    if (!in->legacy_in) {
+        ret = status;
+        goto err_open;
+    }
+
+    in->stream.common.get_sample_rate = in_get_sample_rate;
+    in->stream.common.set_sample_rate = in_set_sample_rate;
+    in->stream.common.get_buffer_size = in_get_buffer_size;
+    in->stream.common.get_channels = in_get_channels;
+    in->stream.common.get_format = in_get_format;
+    in->stream.common.set_format = in_set_format;
+    in->stream.common.standby = in_standby;
+    in->stream.common.dump = in_dump;
+    in->stream.common.set_parameters = in_set_parameters;
+    in->stream.common.get_parameters = in_get_parameters;
+    in->stream.set_gain = in_set_gain;
+    in->stream.read = in_read;
+    in->stream.get_input_frames_lost = in_get_input_frames_lost;
+
+    *stream_in = &in->stream;
+    return 0;
+
+err_open:
+    free(in);
+    *stream_in = NULL;
+    return ret;
+}
+
+static void adev_close_input_stream(struct audio_hw_device *dev,
+                               struct audio_stream_in *stream)
+{
+    struct legacy_audio_device *ladev = to_ladev(dev);
+    struct legacy_stream_in *in =
+        reinterpret_cast<struct legacy_stream_in *>(stream);
+
+    ladev->hwif->closeInputStream(in->legacy_in);
+    free(in);
+}
+
+static int adev_dump(const struct audio_hw_device *dev, int fd)
+{
+    const struct legacy_audio_device *ladev = to_cladev(dev);
+    Vector<String16> args;
+
+    return ladev->hwif->dumpState(fd, args);
+}
+
+static int legacy_adev_close(hw_device_t* device)
+{
+    struct audio_hw_device *hwdev =
+                        reinterpret_cast<struct audio_hw_device *>(device);
+    struct legacy_audio_device *ladev = to_ladev(hwdev);
+
+    if (!ladev)
+        return 0;
+
+    if (ladev->hwif)
+        delete ladev->hwif;
+
+    free(ladev);
+    return 0;
+}
+
+static int legacy_adev_open(const hw_module_t* module, const char* name,
+                            hw_device_t** device)
+{
+    struct legacy_audio_device *ladev;
+    int ret;
+
+    if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
+        return -EINVAL;
+
+    ladev = (struct legacy_audio_device *)calloc(1, sizeof(*ladev));
+    if (!ladev)
+        return -ENOMEM;
+
+    ladev->device.common.tag = HARDWARE_DEVICE_TAG;
+    ladev->device.common.version = 0;
+    ladev->device.common.module = const_cast<hw_module_t*>(module);
+    ladev->device.common.close = legacy_adev_close;
+
+    ladev->device.get_supported_devices = adev_get_supported_devices;
+    ladev->device.init_check = adev_init_check;
+    ladev->device.set_voice_volume = adev_set_voice_volume;
+    ladev->device.set_master_volume = adev_set_master_volume;
+    ladev->device.set_mode = adev_set_mode;
+    ladev->device.set_mic_mute = adev_set_mic_mute;
+    ladev->device.get_mic_mute = adev_get_mic_mute;
+    ladev->device.set_parameters = adev_set_parameters;
+    ladev->device.get_parameters = adev_get_parameters;
+    ladev->device.get_input_buffer_size = adev_get_input_buffer_size;
+    ladev->device.open_output_stream = adev_open_output_stream;
+    ladev->device.close_output_stream = adev_close_output_stream;
+    ladev->device.open_input_stream = adev_open_input_stream;
+    ladev->device.close_input_stream = adev_close_input_stream;
+    ladev->device.dump = adev_dump;
+
+    ladev->hwif = createAudioHardware();
+    if (!ladev->hwif) {
+        ret = -EIO;
+        goto err_create_audio_hw;
+    }
+
+    *device = &ladev->device.common;
+
+    return 0;
+
+err_create_audio_hw:
+    free(ladev);
+    return ret;
+}
+
+static struct hw_module_methods_t legacy_audio_module_methods = {
+        open: legacy_adev_open
+};
+
+struct legacy_audio_module HAL_MODULE_INFO_SYM = {
+    module: {
+        common: {
+            tag: HARDWARE_MODULE_TAG,
+            version_major: 1,
+            version_minor: 0,
+            id: AUDIO_HARDWARE_MODULE_ID,
+            name: "LEGACY Audio HW HAL",
+            author: "The Android Open Source Project",
+            methods: &legacy_audio_module_methods,
+            dso : NULL,
+            reserved : {0},
+        },
+    },
+};
+
+}; // extern "C"
+
+}; // namespace android_audio_legacy
diff --git a/audio/audio_policy_hal.cpp b/audio/audio_policy_hal.cpp
new file mode 100644
index 0000000..f115861
--- /dev/null
+++ b/audio/audio_policy_hal.cpp
@@ -0,0 +1,419 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "legacy_audio_policy_hal"
+//#define LOG_NDEBUG 0
+
+#include <stdint.h>
+
+#include <hardware/hardware.h>
+#include <hardware/audio.h>
+#include <hardware/audio_policy.h>
+#include <hardware/audio_policy_hal.h>
+
+#include <hardware_legacy/AudioPolicyInterface.h>
+#include <hardware_legacy/AudioSystemLegacy.h>
+
+#include "AudioPolicyCompatClient.h"
+
+namespace android_audio_legacy {
+
+extern "C" {
+
+struct legacy_ap_module {
+    struct audio_policy_module module;
+};
+
+struct legacy_ap_device {
+    struct audio_policy_device device;
+};
+
+struct legacy_audio_policy {
+    struct audio_policy policy;
+
+    void *service;
+    struct audio_policy_service_ops *aps_ops;
+    AudioPolicyCompatClient *service_client;
+    AudioPolicyInterface *apm;
+};
+
+static inline struct legacy_audio_policy * to_lap(struct audio_policy *pol)
+{
+    return reinterpret_cast<struct legacy_audio_policy *>(pol);
+}
+
+static inline const struct legacy_audio_policy * to_clap(const struct audio_policy *pol)
+{
+    return reinterpret_cast<const struct legacy_audio_policy *>(pol);
+}
+
+
+static int ap_set_device_connection_state(struct audio_policy *pol,
+                                          audio_devices_t device,
+                                          audio_policy_dev_state_t state,
+                                          const char *device_address)
+{
+    struct legacy_audio_policy *lap = to_lap(pol);
+    return lap->apm->setDeviceConnectionState(
+                    (AudioSystem::audio_devices)device,
+                    (AudioSystem::device_connection_state)state,
+                    device_address);
+}
+
+static audio_policy_dev_state_t ap_get_device_connection_state(
+                                            const struct audio_policy *pol,
+                                            audio_devices_t device,
+                                            const char *device_address)
+{
+    const struct legacy_audio_policy *lap = to_clap(pol);
+    return (audio_policy_dev_state_t)lap->apm->getDeviceConnectionState(
+                    (AudioSystem::audio_devices)device,
+                    device_address);
+}
+
+static void ap_set_phone_state(struct audio_policy *pol, int state)
+{
+    struct legacy_audio_policy *lap = to_lap(pol);
+    lap->apm->setPhoneState(state);
+}
+
+    /* indicate a change in ringer mode */
+static void ap_set_ringer_mode(struct audio_policy *pol, uint32_t mode,
+                               uint32_t mask)
+{
+    struct legacy_audio_policy *lap = to_lap(pol);
+    lap->apm->setRingerMode(mode, mask);
+}
+
+    /* force using a specific device category for the specified usage */
+static void ap_set_force_use(struct audio_policy *pol,
+                          audio_policy_force_use_t usage,
+                          audio_policy_forced_cfg_t config)
+{
+    struct legacy_audio_policy *lap = to_lap(pol);
+    lap->apm->setForceUse((AudioSystem::force_use)usage,
+                          (AudioSystem::forced_config)config);
+}
+
+    /* retreive current device category forced for a given usage */
+static audio_policy_forced_cfg_t ap_get_force_use(
+                                               const struct audio_policy *pol,
+                                               audio_policy_force_use_t usage)
+{
+    const struct legacy_audio_policy *lap = to_clap(pol);
+    return (audio_policy_forced_cfg_t)lap->apm->getForceUse(
+                          (AudioSystem::force_use)usage);
+}
+
+/* if can_mute is true, then audio streams that are marked ENFORCED_AUDIBLE
+ * can still be muted. */
+static void ap_set_can_mute_enforced_audible(struct audio_policy *pol,
+                                             bool can_mute)
+{
+    struct legacy_audio_policy *lap = to_lap(pol);
+    lap->apm->setSystemProperty("ro.camera.sound.forced", can_mute ? "0" : "1");
+}
+
+static int ap_init_check(const struct audio_policy *pol)
+{
+    const struct legacy_audio_policy *lap = to_clap(pol);
+    return lap->apm->initCheck();
+}
+
+static audio_io_handle_t ap_get_output(struct audio_policy *pol,
+                                       audio_stream_type_t stream,
+                                       uint32_t sampling_rate,
+                                       uint32_t format,
+                                       uint32_t channels,
+                                       audio_policy_output_flags_t flags)
+{
+    struct legacy_audio_policy *lap = to_lap(pol);
+
+    LOGV("%s: tid %d", __func__, gettid());
+    return lap->apm->getOutput((AudioSystem::stream_type)stream,
+                               sampling_rate, format, channels,
+                               (AudioSystem::output_flags)flags);
+}
+
+static int ap_start_output(struct audio_policy *pol, audio_io_handle_t output,
+                           audio_stream_type_t stream, int session)
+{
+    struct legacy_audio_policy *lap = to_lap(pol);
+    return lap->apm->startOutput(output, (AudioSystem::stream_type)stream,
+                                 session);
+}
+
+static int ap_stop_output(struct audio_policy *pol, audio_io_handle_t output,
+                          audio_stream_type_t stream, int session)
+{
+    struct legacy_audio_policy *lap = to_lap(pol);
+    return lap->apm->stopOutput(output, (AudioSystem::stream_type)stream,
+                                session);
+}
+
+static void ap_release_output(struct audio_policy *pol,
+                              audio_io_handle_t output)
+{
+    struct legacy_audio_policy *lap = to_lap(pol);
+    lap->apm->releaseOutput(output);
+}
+
+static audio_io_handle_t ap_get_input(struct audio_policy *pol, int inputSource,
+                                      uint32_t sampling_rate,
+                                      uint32_t format,
+                                      uint32_t channels,
+                                      audio_in_acoustics_t acoustics)
+{
+    struct legacy_audio_policy *lap = to_lap(pol);
+    return lap->apm->getInput(inputSource, sampling_rate, format, channels,
+                              (AudioSystem::audio_in_acoustics)acoustics);
+}
+
+static int ap_start_input(struct audio_policy *pol, audio_io_handle_t input)
+{
+    struct legacy_audio_policy *lap = to_lap(pol);
+    return lap->apm->startInput(input);
+}
+
+static int ap_stop_input(struct audio_policy *pol, audio_io_handle_t input)
+{
+    struct legacy_audio_policy *lap = to_lap(pol);
+    return lap->apm->stopInput(input);
+}
+
+static void ap_release_input(struct audio_policy *pol, audio_io_handle_t input)
+{
+    struct legacy_audio_policy *lap = to_lap(pol);
+    lap->apm->releaseInput(input);
+}
+
+static void ap_init_stream_volume(struct audio_policy *pol,
+                                  audio_stream_type_t stream, int index_min,
+                                  int index_max)
+{
+    struct legacy_audio_policy *lap = to_lap(pol);
+    lap->apm->initStreamVolume((AudioSystem::stream_type)stream, index_min,
+                               index_max);
+}
+
+static int ap_set_stream_volume_index(struct audio_policy *pol,
+                                      audio_stream_type_t stream,
+                                      int index)
+{
+    struct legacy_audio_policy *lap = to_lap(pol);
+    return lap->apm->setStreamVolumeIndex((AudioSystem::stream_type)stream,
+                                          index);
+}
+
+static int ap_get_stream_volume_index(const struct audio_policy *pol,
+                                      audio_stream_type_t stream,
+                                      int *index)
+{
+    const struct legacy_audio_policy *lap = to_clap(pol);
+    return lap->apm->getStreamVolumeIndex((AudioSystem::stream_type)stream,
+                                          index);
+}
+
+static uint32_t ap_get_strategy_for_stream(const struct audio_policy *pol,
+                                           audio_stream_type_t stream)
+{
+    const struct legacy_audio_policy *lap = to_clap(pol);
+    return lap->apm->getStrategyForStream((AudioSystem::stream_type)stream);
+}
+
+static uint32_t ap_get_devices_for_stream(const struct audio_policy *pol,
+                                       audio_stream_type_t stream)
+{
+    const struct legacy_audio_policy *lap = to_clap(pol);
+    return lap->apm->getDevicesForStream((AudioSystem::stream_type)stream);
+}
+
+static audio_io_handle_t ap_get_output_for_effect(struct audio_policy *pol,
+                                            struct effect_descriptor_s *desc)
+{
+    struct legacy_audio_policy *lap = to_lap(pol);
+    return lap->apm->getOutputForEffect(desc);
+}
+
+static int ap_register_effect(struct audio_policy *pol,
+                              struct effect_descriptor_s *desc,
+                              audio_io_handle_t output,
+                              uint32_t strategy,
+                              int session,
+                              int id)
+{
+    struct legacy_audio_policy *lap = to_lap(pol);
+    return lap->apm->registerEffect(desc, output, strategy, session, id);
+}
+
+static int ap_unregister_effect(struct audio_policy *pol, int id)
+{
+    struct legacy_audio_policy *lap = to_lap(pol);
+    return lap->apm->unregisterEffect(id);
+}
+
+static bool ap_is_stream_active(const struct audio_policy *pol, int stream,
+                                uint32_t in_past_ms)
+{
+    const struct legacy_audio_policy *lap = to_clap(pol);
+    return lap->apm->isStreamActive(stream, in_past_ms);
+}
+
+static int ap_dump(const struct audio_policy *pol, int fd)
+{
+    const struct legacy_audio_policy *lap = to_clap(pol);
+    return lap->apm->dump(fd);
+}
+
+static int create_legacy_ap(const struct audio_policy_device *device,
+                            struct audio_policy_service_ops *aps_ops,
+                            void *service,
+                            struct audio_policy **ap)
+{
+    struct legacy_audio_policy *lap;
+    int ret;
+
+    if (!service || !aps_ops)
+        return -EINVAL;
+
+    lap = (struct legacy_audio_policy *)calloc(1, sizeof(*lap));
+    if (!lap)
+        return -ENOMEM;
+
+    lap->policy.set_device_connection_state = ap_set_device_connection_state;
+    lap->policy.get_device_connection_state = ap_get_device_connection_state;
+    lap->policy.set_phone_state = ap_set_phone_state;
+    lap->policy.set_ringer_mode = ap_set_ringer_mode;
+    lap->policy.set_force_use = ap_set_force_use;
+    lap->policy.get_force_use = ap_get_force_use;
+    lap->policy.set_can_mute_enforced_audible =
+        ap_set_can_mute_enforced_audible;
+    lap->policy.init_check = ap_init_check;
+    lap->policy.get_output = ap_get_output;
+    lap->policy.start_output = ap_start_output;
+    lap->policy.stop_output = ap_stop_output;
+    lap->policy.release_output = ap_release_output;
+    lap->policy.get_input = ap_get_input;
+    lap->policy.start_input = ap_start_input;
+    lap->policy.stop_input = ap_stop_input;
+    lap->policy.release_input = ap_release_input;
+    lap->policy.init_stream_volume = ap_init_stream_volume;
+    lap->policy.set_stream_volume_index = ap_set_stream_volume_index;
+    lap->policy.get_stream_volume_index = ap_get_stream_volume_index;
+    lap->policy.get_strategy_for_stream = ap_get_strategy_for_stream;
+    lap->policy.get_devices_for_stream = ap_get_devices_for_stream;
+    lap->policy.get_output_for_effect = ap_get_output_for_effect;
+    lap->policy.register_effect = ap_register_effect;
+    lap->policy.unregister_effect = ap_unregister_effect;
+    lap->policy.is_stream_active = ap_is_stream_active;
+    lap->policy.dump = ap_dump;
+
+    lap->service = service;
+    lap->aps_ops = aps_ops;
+    lap->service_client =
+        new AudioPolicyCompatClient(aps_ops, service);
+    if (!lap->service_client) {
+        ret = -ENOMEM;
+        goto err_new_compat_client;
+    }
+
+    lap->apm = createAudioPolicyManager(lap->service_client);
+    if (!lap->apm) {
+        ret = -ENOMEM;
+        goto err_create_apm;
+    }
+
+    *ap = &lap->policy;
+    return 0;
+
+err_create_apm:
+    delete lap->service_client;
+err_new_compat_client:
+    free(lap);
+    *ap = NULL;
+    return ret;
+}
+
+static int destroy_legacy_ap(const struct audio_policy_device *ap_dev,
+                             struct audio_policy *ap)
+{
+    struct legacy_audio_policy *lap = to_lap(ap);
+
+    if (!lap)
+        return 0;
+
+    if (lap->apm)
+        destroyAudioPolicyManager(lap->apm);
+    if (lap->service_client)
+        delete lap->service_client;
+    free(lap);
+    return 0;
+}
+
+static int legacy_ap_dev_close(hw_device_t* device)
+{
+    if (device)
+        free(device);
+    return 0;
+}
+
+static int legacy_ap_dev_open(const hw_module_t* module, const char* name,
+                                    hw_device_t** device)
+{
+    struct legacy_ap_device *dev;
+
+    if (strcmp(name, AUDIO_POLICY_INTERFACE) != 0)
+        return -EINVAL;
+
+    dev = (struct legacy_ap_device *)calloc(1, sizeof(*dev));
+    if (!dev)
+        return -ENOMEM;
+
+    dev->device.common.tag = HARDWARE_DEVICE_TAG;
+    dev->device.common.version = 0;
+    dev->device.common.module = const_cast<hw_module_t*>(module);
+    dev->device.common.close = legacy_ap_dev_close;
+    dev->device.create_audio_policy = create_legacy_ap;
+    dev->device.destroy_audio_policy = destroy_legacy_ap;
+
+    *device = &dev->device.common;
+
+    return 0;
+}
+
+static struct hw_module_methods_t legacy_ap_module_methods = {
+        open: legacy_ap_dev_open
+};
+
+struct legacy_ap_module HAL_MODULE_INFO_SYM = {
+    module: {
+        common: {
+            tag: HARDWARE_MODULE_TAG,
+            version_major: 1,
+            version_minor: 0,
+            id: AUDIO_POLICY_HARDWARE_MODULE_ID,
+            name: "LEGACY Audio Policy HAL",
+            author: "The Android Open Source Project",
+            methods: &legacy_ap_module_methods,
+            dso : NULL,
+            reserved : {0},
+        },
+    },
+};
+
+}; // extern "C"
+
+}; // namespace android_audio_legacy
diff --git a/include/hardware_legacy/AudioHardwareBase.h b/include/hardware_legacy/AudioHardwareBase.h
index c34135f..f8e5b8f 100644
--- a/include/hardware_legacy/AudioHardwareBase.h
+++ b/include/hardware_legacy/AudioHardwareBase.h
@@ -17,10 +17,11 @@
 #ifndef ANDROID_AUDIO_HARDWARE_BASE_H
 #define ANDROID_AUDIO_HARDWARE_BASE_H
 
-#include "hardware_legacy/AudioHardwareInterface.h"
+#include <hardware_legacy/AudioHardwareInterface.h>
 
+#include <hardware/audio.h>
 
-namespace android {
+namespace android_audio_legacy {
 
 // ----------------------------------------------------------------------------
 
diff --git a/include/hardware_legacy/AudioHardwareInterface.h b/include/hardware_legacy/AudioHardwareInterface.h
index ba65d9a..198cff3 100644
--- a/include/hardware_legacy/AudioHardwareInterface.h
+++ b/include/hardware_legacy/AudioHardwareInterface.h
@@ -26,10 +26,17 @@
 #include <utils/String8.h>
 
 #include <media/IAudioFlinger.h>
-#include "media/AudioSystem.h"
+#include <hardware_legacy/AudioSystemLegacy.h>
 
+#include <hardware/audio.h>
+#include <hardware/audio_hal.h>
 
-namespace android {
+#include <cutils/bitops.h>
+
+namespace android_audio_legacy {
+    using android::Vector;
+    using android::String16;
+    using android::String8;
 
 // ----------------------------------------------------------------------------
 
@@ -62,7 +69,7 @@ public:
     /**
      * return the frame size (number of bytes per sample).
      */
-    uint32_t    frameSize() const { return AudioSystem::popCount(channels())*((format()==AudioSystem::PCM_16_BIT)?sizeof(int16_t):sizeof(int8_t)); }
+    uint32_t    frameSize() const { return popcount(channels())*((format()==AUDIO_FORMAT_PCM_16_BIT)?sizeof(int16_t):sizeof(int8_t)); }
 
     /**
      * return the audio hardware driver latency in milli seconds.
diff --git a/include/hardware_legacy/AudioPolicyInterface.h b/include/hardware_legacy/AudioPolicyInterface.h
index 76f9c7a..7b9fb94 100644
--- a/include/hardware_legacy/AudioPolicyInterface.h
+++ b/include/hardware_legacy/AudioPolicyInterface.h
@@ -21,8 +21,12 @@
 #include <media/ToneGenerator.h>
 #include <utils/String8.h>
 
-namespace android {
+#include <hardware_legacy/AudioSystemLegacy.h>
 
+namespace android_audio_legacy {
+    using android::Vector;
+    using android::String8;
+    using android::ToneGenerator;
 
 // ----------------------------------------------------------------------------
 
diff --git a/include/hardware_legacy/AudioPolicyManagerBase.h b/include/hardware_legacy/AudioPolicyManagerBase.h
index 1b03267..2ad1710 100644
--- a/include/hardware_legacy/AudioPolicyManagerBase.h
+++ b/include/hardware_legacy/AudioPolicyManagerBase.h
@@ -23,7 +23,8 @@
 #include <hardware_legacy/AudioPolicyInterface.h>
 
 
-namespace android {
+namespace android_audio_legacy {
+    using android::KeyedVector;
 
 // ----------------------------------------------------------------------------
 
diff --git a/include/hardware_legacy/AudioSystemLegacy.h b/include/hardware_legacy/AudioSystemLegacy.h
new file mode 100644
index 0000000..e72bb93
--- /dev/null
+++ b/include/hardware_legacy/AudioSystemLegacy.h
@@ -0,0 +1,336 @@
+/*
+ * Copyright (C) 2008 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_AUDIOSYSTEM_LEGACY_H_
+#define ANDROID_AUDIOSYSTEM_LEGACY_H_
+
+#include <utils/Errors.h>
+#include <media/AudioParameter.h>
+
+#include <hardware/audio.h>
+#include <hardware/audio_policy.h>
+
+namespace android_audio_legacy {
+
+using android::status_t;
+using android::AudioParameter;
+
+enum {
+    OK                  = android::OK,
+    NO_ERROR            = android::NO_ERROR,
+
+    UNKNOWN_ERROR       = android::UNKNOWN_ERROR,
+
+    NO_MEMORY           = android::NO_MEMORY,
+    INVALID_OPERATION   = android::INVALID_OPERATION,
+    BAD_VALUE           = android::BAD_VALUE,
+    BAD_TYPE            = android::BAD_TYPE,
+    NAME_NOT_FOUND      = android::NAME_NOT_FOUND,
+    PERMISSION_DENIED   = android::PERMISSION_DENIED,
+    NO_INIT             = android::NO_INIT,
+    ALREADY_EXISTS      = android::ALREADY_EXISTS,
+    DEAD_OBJECT         = android::DEAD_OBJECT,
+    FAILED_TRANSACTION  = android::FAILED_TRANSACTION,
+    JPARKS_BROKE_IT     = android::JPARKS_BROKE_IT,
+    BAD_INDEX           = android::BAD_INDEX,
+    NOT_ENOUGH_DATA     = android::NOT_ENOUGH_DATA,
+    WOULD_BLOCK         = android::WOULD_BLOCK,
+    TIMED_OUT           = android::TIMED_OUT,
+    UNKNOWN_TRANSACTION = android::UNKNOWN_TRANSACTION,
+};
+
+enum audio_source {
+    AUDIO_SOURCE_DEFAULT = 0,
+    AUDIO_SOURCE_MIC = 1,
+    AUDIO_SOURCE_VOICE_UPLINK = 2,
+    AUDIO_SOURCE_VOICE_DOWNLINK = 3,
+    AUDIO_SOURCE_VOICE_CALL = 4,
+    AUDIO_SOURCE_CAMCORDER = 5,
+    AUDIO_SOURCE_VOICE_RECOGNITION = 6,
+    AUDIO_SOURCE_VOICE_COMMUNICATION = 7,
+    AUDIO_SOURCE_MAX = AUDIO_SOURCE_VOICE_COMMUNICATION,
+
+    AUDIO_SOURCE_LIST_END  // must be last - used to validate audio source type
+};
+
+class AudioSystem {
+public:
+#if 1
+    enum stream_type {
+        DEFAULT          =-1,
+        VOICE_CALL       = 0,
+        SYSTEM           = 1,
+        RING             = 2,
+        MUSIC            = 3,
+        ALARM            = 4,
+        NOTIFICATION     = 5,
+        BLUETOOTH_SCO    = 6,
+        ENFORCED_AUDIBLE = 7, // Sounds that cannot be muted by user and must be routed to speaker
+        DTMF             = 8,
+        TTS              = 9,
+        NUM_STREAM_TYPES
+    };
+
+    // Audio sub formats (see AudioSystem::audio_format).
+    enum pcm_sub_format {
+        PCM_SUB_16_BIT          = 0x1, // must be 1 for backward compatibility
+        PCM_SUB_8_BIT           = 0x2, // must be 2 for backward compatibility
+    };
+
+    enum audio_sessions {
+        SESSION_OUTPUT_STAGE = AUDIO_SESSION_OUTPUT_STAGE,
+        SESSION_OUTPUT_MIX = AUDIO_SESSION_OUTPUT_MIX,
+    };
+
+    // MP3 sub format field definition : can use 11 LSBs in the same way as MP3 frame header to specify
+    // bit rate, stereo mode, version...
+    enum mp3_sub_format {
+        //TODO
+    };
+
+    // AMR NB/WB sub format field definition: specify frame block interleaving, bandwidth efficient or octet aligned,
+    // encoding mode for recording...
+    enum amr_sub_format {
+        //TODO
+    };
+
+    // AAC sub format field definition: specify profile or bitrate for recording...
+    enum aac_sub_format {
+        //TODO
+    };
+
+    // VORBIS sub format field definition: specify quality for recording...
+    enum vorbis_sub_format {
+        //TODO
+    };
+
+    // Audio format consists in a main format field (upper 8 bits) and a sub format field (lower 24 bits).
+    // The main format indicates the main codec type. The sub format field indicates options and parameters
+    // for each format. The sub format is mainly used for record to indicate for instance the requested bitrate
+    // or profile. It can also be used for certain formats to give informations not present in the encoded
+    // audio stream (e.g. octet alignement for AMR).
+    enum audio_format {
+        INVALID_FORMAT      = -1,
+        FORMAT_DEFAULT      = 0,
+        PCM                 = 0x00000000, // must be 0 for backward compatibility
+        MP3                 = 0x01000000,
+        AMR_NB              = 0x02000000,
+        AMR_WB              = 0x03000000,
+        AAC                 = 0x04000000,
+        HE_AAC_V1           = 0x05000000,
+        HE_AAC_V2           = 0x06000000,
+        VORBIS              = 0x07000000,
+        MAIN_FORMAT_MASK    = 0xFF000000,
+        SUB_FORMAT_MASK     = 0x00FFFFFF,
+        // Aliases
+        PCM_16_BIT          = (PCM|PCM_SUB_16_BIT),
+        PCM_8_BIT          = (PCM|PCM_SUB_8_BIT)
+    };
+
+
+    // Channel mask definitions must be kept in sync with JAVA values in /media/java/android/media/AudioFormat.java
+    enum audio_channels {
+        // output channels
+        CHANNEL_OUT_FRONT_LEFT = 0x4,
+        CHANNEL_OUT_FRONT_RIGHT = 0x8,
+        CHANNEL_OUT_FRONT_CENTER = 0x10,
+        CHANNEL_OUT_LOW_FREQUENCY = 0x20,
+        CHANNEL_OUT_BACK_LEFT = 0x40,
+        CHANNEL_OUT_BACK_RIGHT = 0x80,
+        CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x100,
+        CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x200,
+        CHANNEL_OUT_BACK_CENTER = 0x400,
+        CHANNEL_OUT_MONO = CHANNEL_OUT_FRONT_LEFT,
+        CHANNEL_OUT_STEREO = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT),
+        CHANNEL_OUT_QUAD = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
+                CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT),
+        CHANNEL_OUT_SURROUND = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
+                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_BACK_CENTER),
+        CHANNEL_OUT_5POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
+                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT),
+        CHANNEL_OUT_7POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
+                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT |
+                CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER),
+        CHANNEL_OUT_ALL = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
+                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT |
+                CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER | CHANNEL_OUT_BACK_CENTER),
+
+        // input channels
+        CHANNEL_IN_LEFT = 0x4,
+        CHANNEL_IN_RIGHT = 0x8,
+        CHANNEL_IN_FRONT = 0x10,
+        CHANNEL_IN_BACK = 0x20,
+        CHANNEL_IN_LEFT_PROCESSED = 0x40,
+        CHANNEL_IN_RIGHT_PROCESSED = 0x80,
+        CHANNEL_IN_FRONT_PROCESSED = 0x100,
+        CHANNEL_IN_BACK_PROCESSED = 0x200,
+        CHANNEL_IN_PRESSURE = 0x400,
+        CHANNEL_IN_X_AXIS = 0x800,
+        CHANNEL_IN_Y_AXIS = 0x1000,
+        CHANNEL_IN_Z_AXIS = 0x2000,
+        CHANNEL_IN_VOICE_UPLINK = 0x4000,
+        CHANNEL_IN_VOICE_DNLINK = 0x8000,
+        CHANNEL_IN_MONO = CHANNEL_IN_FRONT,
+        CHANNEL_IN_STEREO = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT),
+        CHANNEL_IN_ALL = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT | CHANNEL_IN_FRONT | CHANNEL_IN_BACK|
+                CHANNEL_IN_LEFT_PROCESSED | CHANNEL_IN_RIGHT_PROCESSED | CHANNEL_IN_FRONT_PROCESSED | CHANNEL_IN_BACK_PROCESSED|
+                CHANNEL_IN_PRESSURE | CHANNEL_IN_X_AXIS | CHANNEL_IN_Y_AXIS | CHANNEL_IN_Z_AXIS |
+                CHANNEL_IN_VOICE_UPLINK | CHANNEL_IN_VOICE_DNLINK)
+    };
+
+    enum audio_mode {
+        MODE_INVALID = -2,
+        MODE_CURRENT = -1,
+        MODE_NORMAL = 0,
+        MODE_RINGTONE,
+        MODE_IN_CALL,
+        MODE_IN_COMMUNICATION,
+        NUM_MODES  // not a valid entry, denotes end-of-list
+    };
+
+    enum audio_in_acoustics {
+        AGC_ENABLE    = 0x0001,
+        AGC_DISABLE   = 0,
+        NS_ENABLE     = 0x0002,
+        NS_DISABLE    = 0,
+        TX_IIR_ENABLE = 0x0004,
+        TX_DISABLE    = 0
+    };
+
+    enum audio_devices {
+        // output devices
+        DEVICE_OUT_EARPIECE = 0x1,
+        DEVICE_OUT_SPEAKER = 0x2,
+        DEVICE_OUT_WIRED_HEADSET = 0x4,
+        DEVICE_OUT_WIRED_HEADPHONE = 0x8,
+        DEVICE_OUT_BLUETOOTH_SCO = 0x10,
+        DEVICE_OUT_BLUETOOTH_SCO_HEADSET = 0x20,
+        DEVICE_OUT_BLUETOOTH_SCO_CARKIT = 0x40,
+        DEVICE_OUT_BLUETOOTH_A2DP = 0x80,
+        DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100,
+        DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = 0x200,
+        DEVICE_OUT_AUX_DIGITAL = 0x400,
+        DEVICE_OUT_ANLG_DOCK_HEADSET = 0x800,
+        DEVICE_OUT_DGTL_DOCK_HEADSET = 0x1000,
+        DEVICE_OUT_DEFAULT = 0x8000,
+        DEVICE_OUT_ALL = (DEVICE_OUT_EARPIECE | DEVICE_OUT_SPEAKER | DEVICE_OUT_WIRED_HEADSET |
+                DEVICE_OUT_WIRED_HEADPHONE | DEVICE_OUT_BLUETOOTH_SCO | DEVICE_OUT_BLUETOOTH_SCO_HEADSET |
+                DEVICE_OUT_BLUETOOTH_SCO_CARKIT | DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
+                DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER | DEVICE_OUT_AUX_DIGITAL |
+                DEVICE_OUT_ANLG_DOCK_HEADSET | DEVICE_OUT_DGTL_DOCK_HEADSET |
+                DEVICE_OUT_DEFAULT),
+        DEVICE_OUT_ALL_A2DP = (DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
+                DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
+
+        // input devices
+        DEVICE_IN_COMMUNICATION = 0x10000,
+        DEVICE_IN_AMBIENT = 0x20000,
+        DEVICE_IN_BUILTIN_MIC = 0x40000,
+        DEVICE_IN_BLUETOOTH_SCO_HEADSET = 0x80000,
+        DEVICE_IN_WIRED_HEADSET = 0x100000,
+        DEVICE_IN_AUX_DIGITAL = 0x200000,
+        DEVICE_IN_VOICE_CALL = 0x400000,
+        DEVICE_IN_BACK_MIC = 0x800000,
+        DEVICE_IN_DEFAULT = 0x80000000,
+
+        DEVICE_IN_ALL = (DEVICE_IN_COMMUNICATION | DEVICE_IN_AMBIENT | DEVICE_IN_BUILTIN_MIC |
+                DEVICE_IN_BLUETOOTH_SCO_HEADSET | DEVICE_IN_WIRED_HEADSET | DEVICE_IN_AUX_DIGITAL |
+                DEVICE_IN_VOICE_CALL | DEVICE_IN_BACK_MIC | DEVICE_IN_DEFAULT)
+    };
+
+    // request to open a direct output with getOutput() (by opposition to sharing an output with other AudioTracks)
+    enum output_flags {
+        OUTPUT_FLAG_INDIRECT = 0x0,
+        OUTPUT_FLAG_DIRECT = 0x1
+    };
+
+    // device categories used for setForceUse()
+    enum forced_config {
+        FORCE_NONE,
+        FORCE_SPEAKER,
+        FORCE_HEADPHONES,
+        FORCE_BT_SCO,
+        FORCE_BT_A2DP,
+        FORCE_WIRED_ACCESSORY,
+        FORCE_BT_CAR_DOCK,
+        FORCE_BT_DESK_DOCK,
+        FORCE_ANALOG_DOCK,
+        FORCE_DIGITAL_DOCK,
+        NUM_FORCE_CONFIG,
+        FORCE_DEFAULT = FORCE_NONE
+    };
+
+    // usages used for setForceUse()
+    enum force_use {
+        FOR_COMMUNICATION,
+        FOR_MEDIA,
+        FOR_RECORD,
+        FOR_DOCK,
+        NUM_FORCE_USE
+    };
+
+    //
+    // AudioPolicyService interface
+    //
+
+    // device connection states used for setDeviceConnectionState()
+    enum device_connection_state {
+        DEVICE_STATE_UNAVAILABLE,
+        DEVICE_STATE_AVAILABLE,
+        NUM_DEVICE_STATES
+    };
+
+#endif
+
+    static uint32_t popCount(uint32_t u) {
+        return popcount(u);
+    }
+
+#if 1
+    static bool isOutputDevice(audio_devices device) {
+        return audio_is_output_device((audio_devices_t)device);
+    }
+    static bool isInputDevice(audio_devices device) {
+        return audio_is_input_device((audio_devices_t)device);
+    }
+    static bool isA2dpDevice(audio_devices device) {
+        return audio_is_a2dp_device((audio_devices_t)device);
+    }
+    static bool isBluetoothScoDevice(audio_devices device) {
+        return audio_is_bluetooth_sco_device((audio_devices_t)device);
+    }
+    static bool isLowVisibility(stream_type stream) {
+        return audio_is_low_visibility((audio_stream_type_t)stream);
+    }
+    static bool isValidFormat(uint32_t format) {
+        return audio_is_valid_format(format);
+    }
+    static bool isLinearPCM(uint32_t format) {
+        return audio_is_linear_pcm(format);
+    }
+    static bool isOutputChannel(uint32_t channel) {
+        return audio_is_output_channel(channel);
+    }
+    static bool isInputChannel(uint32_t channel) {
+        return audio_is_input_channel(channel);
+    }
+
+#endif
+};
+
+};  // namespace android
+
+#endif // ANDROID_AUDIOSYSTEM_LEGACY_H_