diff --git a/libs/audioflinger/AudioHardwareInterface.cpp b/libs/audioflinger/AudioHardwareInterface.cpp
new file mode 100644
index 0000000..ac76a19
--- /dev/null
+++ b/libs/audioflinger/AudioHardwareInterface.cpp
@@ -0,0 +1,247 @@
+/*
+**
+** Copyright 2007, 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 <cutils/properties.h>
+#include <string.h>
+#include <unistd.h>
+
+#define LOG_TAG "AudioHardwareInterface"
+#include <utils/Log.h>
+#include <utils/String8.h>
+
+#include "AudioHardwareStub.h"
+#include "AudioHardwareGeneric.h"
+
+//#define DUMP_FLINGER_OUT        // if defined allows recording samples in a file
+#ifdef DUMP_FLINGER_OUT
+#include "AudioDumpInterface.h"
+#endif
+
+
+// change to 1 to log routing calls
+#define LOG_ROUTING_CALLS 0
+
+namespace android {
+
+#if LOG_ROUTING_CALLS
+static const char* routingModeStrings[] =
+{
+    "OUT OF RANGE",
+    "INVALID",
+    "CURRENT",
+    "NORMAL",
+    "RINGTONE",
+    "IN_CALL"
+};
+
+static const char* routeStrings[] =
+{
+    "EARPIECE ",
+    "SPEAKER ",
+    "BLUETOOTH ",
+    "HEADSET "
+    "BLUETOOTH_A2DP "
+};
+static const char* routeNone = "NONE";
+
+static const char* displayMode(int mode)
+{
+    if ((mode < -2) || (mode > 2))
+        return routingModeStrings[0];
+    return routingModeStrings[mode+3];
+}
+
+static const char* displayRoutes(uint32_t routes)
+{
+    static char routeStr[80];
+    if (routes == 0)
+        return routeNone;
+    routeStr[0] = 0;
+    int bitMask = 1;
+    for (int i = 0; i < 4; ++i, bitMask <<= 1) {
+        if (routes & bitMask) {
+            strcat(routeStr, routeStrings[i]);
+        }
+    }
+    routeStr[strlen(routeStr)-1] = 0;
+    return routeStr;
+}
+#endif
+
+// ----------------------------------------------------------------------------
+
+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 DUMP_FLINGER_OUT
+    // 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 DUMP_FLINGER_OUT.
+    // The output file is FLINGER_DUMP_NAME. Pause are not recorded in the file.
+    
+    hw = new AudioDumpInterface(hw);    // replace interface
+#endif
+    return hw;
+}
+
+AudioStreamOut::~AudioStreamOut()
+{
+}
+
+AudioStreamIn::~AudioStreamIn() {}
+
+AudioHardwareBase::AudioHardwareBase()
+{
+    // force a routing update on initialization
+    memset(&mRoutes, 0, sizeof(mRoutes));
+    mMode = 0;
+}
+
+// generics for audio routing - the real work is done in doRouting
+status_t AudioHardwareBase::setRouting(int mode, uint32_t routes)
+{
+#if LOG_ROUTING_CALLS
+    LOGD("setRouting: mode=%s, routes=[%s]", displayMode(mode), displayRoutes(routes));
+#endif
+    if (mode == AudioSystem::MODE_CURRENT)
+        mode = mMode;
+    if ((mode < 0) || (mode >= AudioSystem::NUM_MODES))
+        return BAD_VALUE;
+    uint32_t old = mRoutes[mode];
+    mRoutes[mode] = routes;
+    if ((mode != mMode) || (old == routes))
+        return NO_ERROR;
+#if LOG_ROUTING_CALLS
+    const char* oldRouteStr = strdup(displayRoutes(old));
+    LOGD("doRouting: mode=%s, old route=[%s], new route=[%s]",
+           displayMode(mode), oldRouteStr, displayRoutes(routes));
+    delete oldRouteStr;
+#endif
+    return doRouting();
+}
+
+status_t AudioHardwareBase::getRouting(int mode, uint32_t* routes)
+{
+    if (mode == AudioSystem::MODE_CURRENT)
+        mode = mMode;
+    if ((mode < 0) || (mode >= AudioSystem::NUM_MODES))
+        return BAD_VALUE;
+    *routes = mRoutes[mode];
+#if LOG_ROUTING_CALLS
+    LOGD("getRouting: mode=%s, routes=[%s]",
+           displayMode(mode), displayRoutes(*routes));
+#endif
+    return NO_ERROR;
+}
+
+status_t AudioHardwareBase::setMode(int mode)
+{
+#if LOG_ROUTING_CALLS
+    LOGD("setMode(%s)", displayMode(mode));
+#endif
+    if ((mode < 0) || (mode >= AudioSystem::NUM_MODES))
+        return BAD_VALUE;
+    if (mMode == mode)
+        return NO_ERROR;
+#if LOG_ROUTING_CALLS
+    LOGD("doRouting: old mode=%s, new mode=%s route=[%s]",
+            displayMode(mMode), displayMode(mode), displayRoutes(mRoutes[mode]));
+#endif
+    mMode = mode;
+    return doRouting();
+}
+
+status_t AudioHardwareBase::getMode(int* mode)
+{
+    // Implement: set audio routing
+    *mode = mMode;
+    return NO_ERROR;
+}
+
+status_t AudioHardwareBase::setParameter(const char* key, const char* value)
+{
+    // default implementation is to ignore
+    return NO_ERROR;
+}
+
+
+// default implementation
+size_t AudioHardwareBase::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
+{
+    if (sampleRate != 8000) {
+        LOGW("getInputBufferSize bad sampling rate: %d", sampleRate);
+        return 0;
+    }
+    if (format != AudioSystem::PCM_16_BIT) {
+        LOGW("getInputBufferSize bad format: %d", format);
+        return 0;
+    }
+    if (channelCount != 1) {
+        LOGW("getInputBufferSize bad channel count: %d", channelCount);
+        return 0;
+    }
+
+    return 320;
+}
+
+status_t AudioHardwareBase::dumpState(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+    snprintf(buffer, SIZE, "AudioHardwareBase::dumpState\n");
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmMode: %d\n", mMode);
+    result.append(buffer);
+    for (int i = 0, n = AudioSystem::NUM_MODES; i < n; ++i) {
+        snprintf(buffer, SIZE, "\tmRoutes[%d]: %d\n", i, mRoutes[i]);
+        result.append(buffer);
+    }
+    ::write(fd, result.string(), result.size());
+    dump(fd, args);  // Dump the state of the concrete child.
+    return NO_ERROR;
+}
+
+// ----------------------------------------------------------------------------
+
+}; // namespace android