13 files changed, 1049 insertions, 59 deletions

diff --git a/modules/Android.mk b/modules/Android.mk
index 871b984..faa8bb3 100644
--- a/modules/Android.mk
+++ b/modules/Android.mk
@@ -1,2 +1,2 @@
-hardware_modules := gralloc hwcomposer audio nfc local_time power usbaudio
+hardware_modules := gralloc hwcomposer audio nfc nfc-nci local_time power usbaudio audio_remote_submix
 include $(call all-named-subdir-makefiles,$(hardware_modules))
diff --git a/modules/audio/audio_hw.c b/modules/audio/audio_hw.c
index 78bed03..b051cf9 100644
--- a/modules/audio/audio_hw.c
+++ b/modules/audio/audio_hw.c
@@ -55,7 +55,7 @@ static size_t out_get_buffer_size(const struct audio_stream *stream)
     return 4096;
 }
 
-static uint32_t out_get_channels(const struct audio_stream *stream)
+static audio_channel_mask_t out_get_channels(const struct audio_stream *stream)
 {
     return AUDIO_CHANNEL_OUT_STEREO;
 }
@@ -150,7 +150,7 @@ static size_t in_get_buffer_size(const struct audio_stream *stream)
     return 320;
 }
 
-static uint32_t in_get_channels(const struct audio_stream *stream)
+static audio_channel_mask_t in_get_channels(const struct audio_stream *stream)
 {
     return AUDIO_CHANNEL_IN_MONO;
 }
@@ -292,8 +292,17 @@ static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
 }
 
 #ifndef ICS_AUDIO_BLOB
-static int adev_get_master_volume(struct audio_hw_device *dev,
-                                  float *volume)
+static int adev_get_master_volume(struct audio_hw_device *dev, float *volume)
+{
+    return -ENOSYS;
+}
+
+static int adev_set_master_mute(struct audio_hw_device *dev, bool muted)
+{
+    return -ENOSYS;
+}
+
+static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted)
 {
     return -ENOSYS;
 }
@@ -376,29 +385,6 @@ static int adev_close(hw_device_t *device)
     return 0;
 }
 
-static uint32_t adev_get_supported_devices(const struct audio_hw_device *dev)
-{
-    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_open(const hw_module_t* module, const char* name,
                      hw_device_t** device)
 {
@@ -413,16 +399,17 @@ static int adev_open(const hw_module_t* module, const char* name,
         return -ENOMEM;
 
     adev->device.common.tag = HARDWARE_DEVICE_TAG;
-    adev->device.common.version = AUDIO_DEVICE_API_VERSION_1_0;
+    adev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
     adev->device.common.module = (struct hw_module_t *) module;
     adev->device.common.close = adev_close;
 
-    adev->device.get_supported_devices = adev_get_supported_devices;
     adev->device.init_check = adev_init_check;
     adev->device.set_voice_volume = adev_set_voice_volume;
     adev->device.set_master_volume = adev_set_master_volume;
 #ifndef ICS_AUDIO_BLOB
     adev->device.get_master_volume = adev_get_master_volume;
+    adev->device.set_master_mute = adev_set_master_mute;
+    adev->device.get_master_mute = adev_get_master_mute;
 #endif
     adev->device.set_mode = adev_set_mode;
     adev->device.set_mic_mute = adev_set_mic_mute;
diff --git a/modules/audio/audio_policy.c b/modules/audio/audio_policy.c
index ce74ff8..82aa449 100644
--- a/modules/audio/audio_policy.c
+++ b/modules/audio/audio_policy.c
@@ -98,7 +98,7 @@ static audio_io_handle_t ap_get_output(struct audio_policy *pol,
                                        audio_stream_type_t stream,
                                        uint32_t sampling_rate,
                                        audio_format_t format,
-                                       uint32_t channels,
+                                       audio_channel_mask_t channelMask,
                                        audio_output_flags_t flags)
 {
     return 0;
@@ -124,7 +124,7 @@ static void ap_release_output(struct audio_policy *pol,
 static audio_io_handle_t ap_get_input(struct audio_policy *pol, audio_source_t inputSource,
                                       uint32_t sampling_rate,
                                       audio_format_t format,
-                                      uint32_t channels,
+                                      audio_channel_mask_t channelMask,
                                       audio_in_acoustics_t acoustics)
 {
     return 0;
@@ -195,13 +195,13 @@ static audio_devices_t ap_get_devices_for_stream(const struct audio_policy *pol,
 }
 
 static audio_io_handle_t ap_get_output_for_effect(struct audio_policy *pol,
-                                            struct effect_descriptor_s *desc)
+                                            const struct effect_descriptor_s *desc)
 {
     return 0;
 }
 
 static int ap_register_effect(struct audio_policy *pol,
-                              struct effect_descriptor_s *desc,
+                              const struct effect_descriptor_s *desc,
                               audio_io_handle_t output,
                               uint32_t strategy,
                               int session,
diff --git a/modules/audio_remote_submix/Android.mk b/modules/audio_remote_submix/Android.mk
new file mode 100644
index 0000000..5f54902
--- /dev/null
+++ b/modules/audio_remote_submix/Android.mk
@@ -0,0 +1,30 @@
+# Copyright (C) 2012 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.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE := audio.r_submix.default
+LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/hw
+LOCAL_SRC_FILES := \
+	audio_hw.cpp
+LOCAL_C_INCLUDES += \
+	frameworks/av/include/ \
+	frameworks/native/include/
+LOCAL_SHARED_LIBRARIES := liblog libcutils libutils libnbaio
+LOCAL_STATIC_LIBRARIES := libmedia_helper
+LOCAL_MODULE_TAGS := optional
+include $(BUILD_SHARED_LIBRARY)
+
diff --git a/modules/audio_remote_submix/audio_hw.cpp b/modules/audio_remote_submix/audio_hw.cpp
new file mode 100755
index 0000000..3756274
--- /dev/null
+++ b/modules/audio_remote_submix/audio_hw.cpp
@@ -0,0 +1,839 @@
+/*
+ * Copyright (C) 2012 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 "r_submix"
+//#define LOG_NDEBUG 0
+
+#include <errno.h>
+#include <pthread.h>
+#include <stdint.h>
+#include <sys/time.h>
+#include <stdlib.h>
+
+#include <cutils/log.h>
+#include <cutils/str_parms.h>
+#include <cutils/properties.h>
+
+#include <hardware/hardware.h>
+#include <system/audio.h>
+#include <hardware/audio.h>
+
+#include <media/nbaio/MonoPipe.h>
+#include <media/nbaio/MonoPipeReader.h>
+#include <media/AudioBufferProvider.h>
+
+#include <utils/String8.h>
+#include <media/AudioParameter.h>
+
+extern "C" {
+
+namespace android {
+
+#define MAX_PIPE_DEPTH_IN_FRAMES     (1024*8)
+// The duration of MAX_READ_ATTEMPTS * READ_ATTEMPT_SLEEP_MS must be stricly inferior to
+//   the duration of a record buffer at the current record sample rate (of the device, not of
+//   the recording itself). Here we have:
+//      3 * 5ms = 15ms < 1024 frames * 1000 / 48000 = 21.333ms
+#define MAX_READ_ATTEMPTS            3
+#define READ_ATTEMPT_SLEEP_MS        5 // 5ms between two read attempts when pipe is empty
+#define DEFAULT_RATE_HZ              48000 // default sample rate
+
+struct submix_config {
+    audio_format_t format;
+    audio_channel_mask_t channel_mask;
+    unsigned int rate; // sample rate for the device
+    unsigned int period_size; // size of the audio pipe is period_size * period_count in frames
+    unsigned int period_count;
+};
+
+struct submix_audio_device {
+    struct audio_hw_device device;
+    bool output_standby;
+    bool input_standby;
+    submix_config config;
+    // Pipe variables: they handle the ring buffer that "pipes" audio:
+    //  - from the submix virtual audio output == what needs to be played
+    //    remotely, seen as an output for AudioFlinger
+    //  - to the virtual audio source == what is captured by the component
+    //    which "records" the submix / virtual audio source, and handles it as needed.
+    // A usecase example is one where the component capturing the audio is then sending it over
+    // Wifi for presentation on a remote Wifi Display device (e.g. a dongle attached to a TV, or a
+    // TV with Wifi Display capabilities), or to a wireless audio player.
+    sp<MonoPipe>       rsxSink;
+    sp<MonoPipeReader> rsxSource;
+
+    // device lock, also used to protect access to the audio pipe
+    pthread_mutex_t lock;
+};
+
+struct submix_stream_out {
+    struct audio_stream_out stream;
+    struct submix_audio_device *dev;
+};
+
+struct submix_stream_in {
+    struct audio_stream_in stream;
+    struct submix_audio_device *dev;
+    bool output_standby; // output standby state as seen from record thread
+
+    // wall clock when recording starts
+    struct timespec record_start_time;
+    // how many frames have been requested to be read
+    int64_t read_counter_frames;
+};
+
+
+/* audio HAL functions */
+
+static uint32_t out_get_sample_rate(const struct audio_stream *stream)
+{
+    const struct submix_stream_out *out =
+            reinterpret_cast<const struct submix_stream_out *>(stream);
+    uint32_t out_rate = out->dev->config.rate;
+    //ALOGV("out_get_sample_rate() returns %u", out_rate);
+    return out_rate;
+}
+
+static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
+{
+    if ((rate != 44100) && (rate != 48000)) {
+        ALOGE("out_set_sample_rate(rate=%u) rate unsupported", rate);
+        return -ENOSYS;
+    }
+    struct submix_stream_out *out = reinterpret_cast<struct submix_stream_out *>(stream);
+    //ALOGV("out_set_sample_rate(rate=%u)", rate);
+    out->dev->config.rate = rate;
+    return 0;
+}
+
+static size_t out_get_buffer_size(const struct audio_stream *stream)
+{
+    const struct submix_stream_out *out =
+            reinterpret_cast<const struct submix_stream_out *>(stream);
+    const struct submix_config& config_out = out->dev->config;
+    size_t buffer_size = config_out.period_size * popcount(config_out.channel_mask)
+                            * sizeof(int16_t); // only PCM 16bit
+    //ALOGV("out_get_buffer_size() returns %u, period size=%u",
+    //        buffer_size, config_out.period_size);
+    return buffer_size;
+}
+
+static audio_channel_mask_t out_get_channels(const struct audio_stream *stream)
+{
+    const struct submix_stream_out *out =
+            reinterpret_cast<const struct submix_stream_out *>(stream);
+    uint32_t channels = out->dev->config.channel_mask;
+    //ALOGV("out_get_channels() returns %08x", channels);
+    return channels;
+}
+
+static audio_format_t out_get_format(const struct audio_stream *stream)
+{
+    return AUDIO_FORMAT_PCM_16_BIT;
+}
+
+static int out_set_format(struct audio_stream *stream, audio_format_t format)
+{
+    if (format != AUDIO_FORMAT_PCM_16_BIT) {
+        return -ENOSYS;
+    } else {
+        return 0;
+    }
+}
+
+static int out_standby(struct audio_stream *stream)
+{
+    ALOGI("out_standby()");
+
+    const struct submix_stream_out *out = reinterpret_cast<const struct submix_stream_out *>(stream);
+
+    pthread_mutex_lock(&out->dev->lock);
+
+    out->dev->output_standby = true;
+
+    pthread_mutex_unlock(&out->dev->lock);
+
+    return 0;
+}
+
+static int out_dump(const struct audio_stream *stream, int fd)
+{
+    return 0;
+}
+
+static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+    int exiting = -1;
+    AudioParameter parms = AudioParameter(String8(kvpairs));
+    // FIXME this is using hard-coded strings but in the future, this functionality will be
+    //       converted to use audio HAL extensions required to support tunneling
+    if ((parms.getInt(String8("exiting"), exiting) == NO_ERROR) && (exiting > 0)) {
+        const struct submix_stream_out *out =
+                reinterpret_cast<const struct submix_stream_out *>(stream);
+
+        pthread_mutex_lock(&out->dev->lock);
+
+        MonoPipe* sink = out->dev->rsxSink.get();
+        if (sink != NULL) {
+            sink->incStrong(out);
+        } else {
+            pthread_mutex_unlock(&out->dev->lock);
+            return 0;
+        }
+
+        ALOGI("shutdown");
+        sink->shutdown(true);
+
+        sink->decStrong(out);
+
+        pthread_mutex_unlock(&out->dev->lock);
+    }
+
+    return 0;
+}
+
+static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
+{
+    return strdup("");
+}
+
+static uint32_t out_get_latency(const struct audio_stream_out *stream)
+{
+    const struct submix_stream_out *out =
+            reinterpret_cast<const struct submix_stream_out *>(stream);
+    const struct submix_config * config_out = &(out->dev->config);
+    uint32_t latency = (MAX_PIPE_DEPTH_IN_FRAMES * 1000) / config_out->rate;
+    ALOGV("out_get_latency() returns %u", latency);
+    return latency;
+}
+
+static int out_set_volume(struct audio_stream_out *stream, float left,
+                          float right)
+{
+    return -ENOSYS;
+}
+
+static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
+                         size_t bytes)
+{
+    //ALOGV("out_write(bytes=%d)", bytes);
+    ssize_t written_frames = 0;
+    struct submix_stream_out *out = reinterpret_cast<struct submix_stream_out *>(stream);
+
+    const size_t frame_size = audio_stream_frame_size(&stream->common);
+    const size_t frames = bytes / frame_size;
+
+    pthread_mutex_lock(&out->dev->lock);
+
+    out->dev->output_standby = false;
+
+    MonoPipe* sink = out->dev->rsxSink.get();
+    if (sink != NULL) {
+        if (sink->isShutdown()) {
+            pthread_mutex_unlock(&out->dev->lock);
+            // the pipe has already been shutdown, this buffer will be lost but we must
+            //   simulate timing so we don't drain the output faster than realtime
+            usleep(frames * 1000000 / out_get_sample_rate(&stream->common));
+            return bytes;
+        }
+        sink->incStrong(buffer);
+    } else {
+        pthread_mutex_unlock(&out->dev->lock);
+        ALOGE("out_write without a pipe!");
+        ALOG_ASSERT("out_write without a pipe!");
+        return 0;
+    }
+
+    pthread_mutex_unlock(&out->dev->lock);
+
+    written_frames = sink->write(buffer, frames);
+    if (written_frames < 0) {
+        if (written_frames == (ssize_t)NEGOTIATE) {
+            ALOGE("out_write() write to pipe returned NEGOTIATE");
+
+            pthread_mutex_lock(&out->dev->lock);
+            sink->decStrong(buffer);
+            pthread_mutex_unlock(&out->dev->lock);
+
+            written_frames = 0;
+            return 0;
+        } else {
+            // write() returned UNDERRUN or WOULD_BLOCK, retry
+            ALOGE("out_write() write to pipe returned unexpected %16lx", written_frames);
+            written_frames = sink->write(buffer, frames);
+        }
+    }
+
+    pthread_mutex_lock(&out->dev->lock);
+
+    sink->decStrong(buffer);
+
+    pthread_mutex_unlock(&out->dev->lock);
+
+    if (written_frames < 0) {
+        ALOGE("out_write() failed writing to pipe with %16lx", written_frames);
+        return 0;
+    } else {
+        ALOGV("out_write() wrote %lu bytes)", written_frames * frame_size);
+        return written_frames * frame_size;
+    }
+}
+
+static int out_get_render_position(const struct audio_stream_out *stream,
+                                   uint32_t *dsp_frames)
+{
+    return -EINVAL;
+}
+
+static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+    return 0;
+}
+
+static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+    return 0;
+}
+
+static int out_get_next_write_timestamp(const struct audio_stream_out *stream,
+                                        int64_t *timestamp)
+{
+    return -EINVAL;
+}
+
+/** audio_stream_in implementation **/
+static uint32_t in_get_sample_rate(const struct audio_stream *stream)
+{
+    const struct submix_stream_in *in = reinterpret_cast<const struct submix_stream_in *>(stream);
+    //ALOGV("in_get_sample_rate() returns %u", in->dev->config.rate);
+    return in->dev->config.rate;
+}
+
+static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
+{
+    return -ENOSYS;
+}
+
+static size_t in_get_buffer_size(const struct audio_stream *stream)
+{
+    const struct submix_stream_in *in = reinterpret_cast<const struct submix_stream_in *>(stream);
+    ALOGV("in_get_buffer_size() returns %u",
+            in->dev->config.period_size * audio_stream_frame_size(stream));
+    return in->dev->config.period_size * audio_stream_frame_size(stream);
+}
+
+static audio_channel_mask_t in_get_channels(const struct audio_stream *stream)
+{
+    return AUDIO_CHANNEL_IN_STEREO;
+}
+
+static audio_format_t in_get_format(const struct audio_stream *stream)
+{
+    return AUDIO_FORMAT_PCM_16_BIT;
+}
+
+static int in_set_format(struct audio_stream *stream, audio_format_t format)
+{
+    if (format != AUDIO_FORMAT_PCM_16_BIT) {
+        return -ENOSYS;
+    } else {
+        return 0;
+    }
+}
+
+static int in_standby(struct audio_stream *stream)
+{
+    ALOGI("in_standby()");
+    const struct submix_stream_in *in = reinterpret_cast<const struct submix_stream_in *>(stream);
+
+    pthread_mutex_lock(&in->dev->lock);
+
+    in->dev->input_standby = true;
+
+    pthread_mutex_unlock(&in->dev->lock);
+
+    return 0;
+}
+
+static int in_dump(const struct audio_stream *stream, int fd)
+{
+    return 0;
+}
+
+static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+    return 0;
+}
+
+static char * in_get_parameters(const struct audio_stream *stream,
+                                const char *keys)
+{
+    return strdup("");
+}
+
+static int in_set_gain(struct audio_stream_in *stream, float gain)
+{
+    return 0;
+}
+
+static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
+                       size_t bytes)
+{
+    //ALOGV("in_read bytes=%u", bytes);
+    ssize_t frames_read = -1977;
+    struct submix_stream_in *in = reinterpret_cast<struct submix_stream_in *>(stream);
+    const size_t frame_size = audio_stream_frame_size(&stream->common);
+    const size_t frames_to_read = bytes / frame_size;
+
+    pthread_mutex_lock(&in->dev->lock);
+
+    const bool output_standby_transition = (in->output_standby != in->dev->output_standby);
+    in->output_standby = in->dev->output_standby;
+
+    if (in->dev->input_standby || output_standby_transition) {
+        in->dev->input_standby = false;
+        // keep track of when we exit input standby (== first read == start "real recording")
+        // or when we start recording silence, and reset projected time
+        int rc = clock_gettime(CLOCK_MONOTONIC, &in->record_start_time);
+        if (rc == 0) {
+            in->read_counter_frames = 0;
+        }
+    }
+
+    in->read_counter_frames += frames_to_read;
+
+    MonoPipeReader* source = in->dev->rsxSource.get();
+    if (source != NULL) {
+        source->incStrong(buffer);
+    } else {
+        ALOGE("no audio pipe yet we're trying to read!");
+        pthread_mutex_unlock(&in->dev->lock);
+        usleep((bytes / frame_size) * 1000000 / in_get_sample_rate(&stream->common));
+        memset(buffer, 0, bytes);
+        return bytes;
+    }
+
+    pthread_mutex_unlock(&in->dev->lock);
+
+    // read the data from the pipe (it's non blocking)
+    size_t remaining_frames = frames_to_read;
+    int attempts = 0;
+    char* buff = (char*)buffer;
+    while ((remaining_frames > 0) && (attempts < MAX_READ_ATTEMPTS)) {
+        attempts++;
+        frames_read = source->read(buff, remaining_frames, AudioBufferProvider::kInvalidPTS);
+        if (frames_read > 0) {
+            remaining_frames -= frames_read;
+            buff += frames_read * frame_size;
+            //ALOGV("  in_read (att=%d) got %ld frames, remaining=%u",
+            //      attempts, frames_read, remaining_frames);
+        } else {
+            //ALOGE("  in_read read returned %ld", frames_read);
+            usleep(READ_ATTEMPT_SLEEP_MS * 1000);
+        }
+    }
+
+    // done using the source
+    pthread_mutex_lock(&in->dev->lock);
+
+    source->decStrong(buffer);
+
+    pthread_mutex_unlock(&in->dev->lock);
+
+    if (remaining_frames > 0) {
+        ALOGV("  remaining_frames = %d", remaining_frames);
+        memset(((char*)buffer)+ bytes - (remaining_frames * frame_size), 0,
+                remaining_frames * frame_size);
+    }
+
+    // compute how much we need to sleep after reading the data by comparing the wall clock with
+    //   the projected time at which we should return.
+    struct timespec time_after_read;// wall clock after reading from the pipe
+    struct timespec record_duration;// observed record duration
+    int rc = clock_gettime(CLOCK_MONOTONIC, &time_after_read);
+    const uint32_t sample_rate = in_get_sample_rate(&stream->common);
+    if (rc == 0) {
+        // for how long have we been recording?
+        record_duration.tv_sec  = time_after_read.tv_sec - in->record_start_time.tv_sec;
+        record_duration.tv_nsec = time_after_read.tv_nsec - in->record_start_time.tv_nsec;
+        if (record_duration.tv_nsec < 0) {
+            record_duration.tv_sec--;
+            record_duration.tv_nsec += 1000000000;
+        }
+
+        // read_counter_frames contains the number of frames that have been read since the beginning
+        // of recording (including this call): it's converted to usec and compared to how long we've
+        // been recording for, which gives us how long we must wait to sync the projected recording
+        // time, and the observed recording time
+        long projected_vs_observed_offset_us =
+                ((int64_t)(in->read_counter_frames
+                            - (record_duration.tv_sec*sample_rate)))
+                        * 1000000 / sample_rate
+                - (record_duration.tv_nsec / 1000);
+
+        ALOGV("  record duration %5lds %3ldms, will wait: %7ldus",
+                record_duration.tv_sec, record_duration.tv_nsec/1000000,
+                projected_vs_observed_offset_us);
+        if (projected_vs_observed_offset_us > 0) {
+            usleep(projected_vs_observed_offset_us);
+        }
+    }
+
+
+    ALOGV("in_read returns %d", bytes);
+    return bytes;
+
+}
+
+static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
+{
+    return 0;
+}
+
+static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+    return 0;
+}
+
+static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+    return 0;
+}
+
+static int adev_open_output_stream(struct audio_hw_device *dev,
+                                   audio_io_handle_t handle,
+                                   audio_devices_t devices,
+                                   audio_output_flags_t flags,
+                                   struct audio_config *config,
+                                   struct audio_stream_out **stream_out)
+{
+    ALOGV("adev_open_output_stream()");
+    struct submix_audio_device *rsxadev = (struct submix_audio_device *)dev;
+    struct submix_stream_out *out;
+    int ret;
+
+    out = (struct submix_stream_out *)calloc(1, sizeof(struct submix_stream_out));
+    if (!out) {
+        ret = -ENOMEM;
+        goto err_open;
+    }
+
+    pthread_mutex_lock(&rsxadev->lock);
+
+    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.common.add_audio_effect = out_add_audio_effect;
+    out->stream.common.remove_audio_effect = out_remove_audio_effect;
+    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;
+    out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
+
+    config->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+    rsxadev->config.channel_mask = config->channel_mask;
+
+    if ((config->sample_rate != 48000) || (config->sample_rate != 44100)) {
+        config->sample_rate = DEFAULT_RATE_HZ;
+    }
+    rsxadev->config.rate = config->sample_rate;
+
+    config->format = AUDIO_FORMAT_PCM_16_BIT;
+    rsxadev->config.format = config->format;
+
+    rsxadev->config.period_size = 1024;
+    rsxadev->config.period_count = 4;
+    out->dev = rsxadev;
+
+    *stream_out = &out->stream;
+
+    // initialize pipe
+    {
+        ALOGV("  initializing pipe");
+        const NBAIO_Format format =
+                config->sample_rate == 48000 ? Format_SR48_C2_I16 : Format_SR44_1_C2_I16;
+        const NBAIO_Format offers[1] = {format};
+        size_t numCounterOffers = 0;
+        // creating a MonoPipe with optional blocking set to true.
+        MonoPipe* sink = new MonoPipe(MAX_PIPE_DEPTH_IN_FRAMES, format, true/*writeCanBlock*/);
+        ssize_t index = sink->negotiate(offers, 1, NULL, numCounterOffers);
+        ALOG_ASSERT(index == 0);
+        MonoPipeReader* source = new MonoPipeReader(sink);
+        numCounterOffers = 0;
+        index = source->negotiate(offers, 1, NULL, numCounterOffers);
+        ALOG_ASSERT(index == 0);
+        rsxadev->rsxSink = sink;
+        rsxadev->rsxSource = source;
+    }
+
+    pthread_mutex_unlock(&rsxadev->lock);
+
+    return 0;
+
+err_open:
+    *stream_out = NULL;
+    return ret;
+}
+
+static void adev_close_output_stream(struct audio_hw_device *dev,
+                                     struct audio_stream_out *stream)
+{
+    ALOGV("adev_close_output_stream()");
+    struct submix_audio_device *rsxadev = (struct submix_audio_device *)dev;
+
+    pthread_mutex_lock(&rsxadev->lock);
+
+    rsxadev->rsxSink.clear();
+    rsxadev->rsxSource.clear();
+    free(stream);
+
+    pthread_mutex_unlock(&rsxadev->lock);
+}
+
+static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
+{
+    return -ENOSYS;
+}
+
+static char * adev_get_parameters(const struct audio_hw_device *dev,
+                                  const char *keys)
+{
+    return strdup("");;
+}
+
+static int adev_init_check(const struct audio_hw_device *dev)
+{
+    ALOGI("adev_init_check()");
+    return 0;
+}
+
+static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
+{
+    return -ENOSYS;
+}
+
+static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
+{
+    return -ENOSYS;
+}
+
+static int adev_get_master_volume(struct audio_hw_device *dev, float *volume)
+{
+    return -ENOSYS;
+}
+
+static int adev_set_master_mute(struct audio_hw_device *dev, bool muted)
+{
+    return -ENOSYS;
+}
+
+static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted)
+{
+    return -ENOSYS;
+}
+
+static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
+{
+    return 0;
+}
+
+static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
+{
+    return -ENOSYS;
+}
+
+static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
+{
+    return -ENOSYS;
+}
+
+static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
+                                         const struct audio_config *config)
+{
+    //### TODO correlate this with pipe parameters
+    return 4096;
+}
+
+static int adev_open_input_stream(struct audio_hw_device *dev,
+                                  audio_io_handle_t handle,
+                                  audio_devices_t devices,
+                                  struct audio_config *config,
+                                  struct audio_stream_in **stream_in)
+{
+    ALOGI("adev_open_input_stream()");
+
+    struct submix_audio_device *rsxadev = (struct submix_audio_device *)dev;
+    struct submix_stream_in *in;
+    int ret;
+
+    in = (struct submix_stream_in *)calloc(1, sizeof(struct submix_stream_in));
+    if (!in) {
+        ret = -ENOMEM;
+        goto err_open;
+    }
+
+    pthread_mutex_lock(&rsxadev->lock);
+
+    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.common.add_audio_effect = in_add_audio_effect;
+    in->stream.common.remove_audio_effect = in_remove_audio_effect;
+    in->stream.set_gain = in_set_gain;
+    in->stream.read = in_read;
+    in->stream.get_input_frames_lost = in_get_input_frames_lost;
+
+    config->channel_mask = AUDIO_CHANNEL_IN_STEREO;
+    rsxadev->config.channel_mask = config->channel_mask;
+
+    if ((config->sample_rate != 48000) || (config->sample_rate != 44100)) {
+        config->sample_rate = DEFAULT_RATE_HZ;
+    }
+    rsxadev->config.rate = config->sample_rate;
+
+    config->format = AUDIO_FORMAT_PCM_16_BIT;
+    rsxadev->config.format = config->format;
+
+    rsxadev->config.period_size = 1024;
+    rsxadev->config.period_count = 4;
+
+    *stream_in = &in->stream;
+
+    in->dev = rsxadev;
+
+    in->read_counter_frames = 0;
+    in->output_standby = rsxadev->output_standby;
+
+    pthread_mutex_unlock(&rsxadev->lock);
+
+    return 0;
+
+err_open:
+    *stream_in = NULL;
+    return ret;
+}
+
+static void adev_close_input_stream(struct audio_hw_device *dev,
+                                   struct audio_stream_in *stream)
+{
+    ALOGV("adev_close_input_stream()");
+    struct submix_audio_device *rsxadev = (struct submix_audio_device *)dev;
+
+    pthread_mutex_lock(&rsxadev->lock);
+
+    MonoPipe* sink = rsxadev->rsxSink.get();
+    if (sink != NULL) {
+        ALOGI("shutdown");
+        sink->shutdown(true);
+    }
+
+    free(stream);
+
+    pthread_mutex_unlock(&rsxadev->lock);
+}
+
+static int adev_dump(const audio_hw_device_t *device, int fd)
+{
+    return 0;
+}
+
+static int adev_close(hw_device_t *device)
+{
+    ALOGI("adev_close()");
+    free(device);
+    return 0;
+}
+
+static int adev_open(const hw_module_t* module, const char* name,
+                     hw_device_t** device)
+{
+    ALOGI("adev_open(name=%s)", name);
+    struct submix_audio_device *rsxadev;
+
+    if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
+        return -EINVAL;
+
+    rsxadev = (submix_audio_device*) calloc(1, sizeof(struct submix_audio_device));
+    if (!rsxadev)
+        return -ENOMEM;
+
+    rsxadev->device.common.tag = HARDWARE_DEVICE_TAG;
+    rsxadev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
+    rsxadev->device.common.module = (struct hw_module_t *) module;
+    rsxadev->device.common.close = adev_close;
+
+    rsxadev->device.init_check = adev_init_check;
+    rsxadev->device.set_voice_volume = adev_set_voice_volume;
+    rsxadev->device.set_master_volume = adev_set_master_volume;
+    rsxadev->device.get_master_volume = adev_get_master_volume;
+    rsxadev->device.set_master_mute = adev_set_master_mute;
+    rsxadev->device.get_master_mute = adev_get_master_mute;
+    rsxadev->device.set_mode = adev_set_mode;
+    rsxadev->device.set_mic_mute = adev_set_mic_mute;
+    rsxadev->device.get_mic_mute = adev_get_mic_mute;
+    rsxadev->device.set_parameters = adev_set_parameters;
+    rsxadev->device.get_parameters = adev_get_parameters;
+    rsxadev->device.get_input_buffer_size = adev_get_input_buffer_size;
+    rsxadev->device.open_output_stream = adev_open_output_stream;
+    rsxadev->device.close_output_stream = adev_close_output_stream;
+    rsxadev->device.open_input_stream = adev_open_input_stream;
+    rsxadev->device.close_input_stream = adev_close_input_stream;
+    rsxadev->device.dump = adev_dump;
+
+    rsxadev->input_standby = true;
+    rsxadev->output_standby = true;
+
+    *device = &rsxadev->device.common;
+
+    return 0;
+}
+
+static struct hw_module_methods_t hal_module_methods = {
+    /* open */ adev_open,
+};
+
+struct audio_module HAL_MODULE_INFO_SYM = {
+    /* common */ {
+        /* tag */                HARDWARE_MODULE_TAG,
+        /* module_api_version */ AUDIO_MODULE_API_VERSION_0_1,
+        /* hal_api_version */    HARDWARE_HAL_API_VERSION,
+        /* id */                 AUDIO_HARDWARE_MODULE_ID,
+        /* name */               "Wifi Display audio HAL",
+        /* author */             "The Android Open Source Project",
+        /* methods */            &hal_module_methods,
+        /* dso */                NULL,
+        /* reserved */           { 0 },
+    },
+};
+
+} //namespace android
+
+} //extern "C"
diff --git a/modules/gralloc/framebuffer.cpp b/modules/gralloc/framebuffer.cpp
index a37e2c0..326f2ae 100644
--- a/modules/gralloc/framebuffer.cpp
+++ b/modules/gralloc/framebuffer.cpp
@@ -312,11 +312,6 @@ int fb_device_open(hw_module_t const* module, const char* name,
 {
     int status = -EINVAL;
     if (!strcmp(name, GRALLOC_HARDWARE_FB0)) {
-        alloc_device_t* gralloc_device;
-        status = gralloc_open(module, &gralloc_device);
-        if (status < 0)
-            return status;
-
         /* initialize our state here */
         fb_context_t *dev = (fb_context_t*)malloc(sizeof(*dev));
         memset(dev, 0, sizeof(*dev));
diff --git a/modules/gralloc/gralloc.cpp b/modules/gralloc/gralloc.cpp
index a6b4edd..99aeb01 100644
--- a/modules/gralloc/gralloc.cpp
+++ b/modules/gralloc/gralloc.cpp
@@ -219,6 +219,7 @@ static int gralloc_alloc(alloc_device_t* dev,
         case HAL_PIXEL_FORMAT_RGB_565:
         case HAL_PIXEL_FORMAT_RGBA_5551:
         case HAL_PIXEL_FORMAT_RGBA_4444:
+        case HAL_PIXEL_FORMAT_RAW_SENSOR:
             bpp = 2;
             break;
         default:
diff --git a/modules/hwcomposer/hwcomposer.cpp b/modules/hwcomposer/hwcomposer.cpp
index 0e04cac..f0a5512 100644
--- a/modules/hwcomposer/hwcomposer.cpp
+++ b/modules/hwcomposer/hwcomposer.cpp
@@ -29,7 +29,7 @@
 /*****************************************************************************/
 
 struct hwc_context_t {
-    hwc_composer_device_t device;
+    hwc_composer_device_1_t device;
     /* our private state goes below here */
 };
 
@@ -54,7 +54,7 @@ hwc_module_t HAL_MODULE_INFO_SYM = {
 
 /*****************************************************************************/
 
-static void dump_layer(hwc_layer_t const* l) {
+static void dump_layer(hwc_layer_1_t const* l) {
     ALOGD("\ttype=%d, flags=%08x, handle=%p, tr=%02x, blend=%04x, {%d,%d,%d,%d}, {%d,%d,%d,%d}",
             l->compositionType, l->flags, l->handle, l->transform, l->blending,
             l->sourceCrop.left,
@@ -67,26 +67,26 @@ static void dump_layer(hwc_layer_t const* l) {
             l->displayFrame.bottom);
 }
 
-static int hwc_prepare(hwc_composer_device_t *dev, hwc_layer_list_t* list) {
-    if (list && (list->flags & HWC_GEOMETRY_CHANGED)) {
-        for (size_t i=0 ; i<list->numHwLayers ; i++) {
+static int hwc_prepare(hwc_composer_device_1_t *dev,
+        size_t numDisplays, hwc_display_contents_1_t** displays) {
+    if (displays && (displays[0]->flags & HWC_GEOMETRY_CHANGED)) {
+        for (size_t i=0 ; i<displays[0]->numHwLayers ; i++) {
             //dump_layer(&list->hwLayers[i]);
-            list->hwLayers[i].compositionType = HWC_FRAMEBUFFER;
+            displays[0]->hwLayers[i].compositionType = HWC_FRAMEBUFFER;
         }
     }
     return 0;
 }
 
-static int hwc_set(hwc_composer_device_t *dev,
-        hwc_display_t dpy,
-        hwc_surface_t sur,
-        hwc_layer_list_t* list)
+static int hwc_set(hwc_composer_device_1_t *dev,
+        size_t numDisplays, hwc_display_contents_1_t** displays)
 {
     //for (size_t i=0 ; i<list->numHwLayers ; i++) {
     //    dump_layer(&list->hwLayers[i]);
     //}
 
-    EGLBoolean sucess = eglSwapBuffers((EGLDisplay)dpy, (EGLSurface)sur);
+    EGLBoolean sucess = eglSwapBuffers((EGLDisplay)displays[0]->dpy,
+            (EGLSurface)displays[0]->sur);
     if (!sucess) {
         return HWC_EGL_ERROR;
     }
@@ -117,7 +117,7 @@ static int hwc_device_open(const struct hw_module_t* module, const char* name,
 
         /* initialize the procs */
         dev->device.common.tag = HARDWARE_DEVICE_TAG;
-        dev->device.common.version = 0;
+        dev->device.common.version = HWC_DEVICE_API_VERSION_1_0;
         dev->device.common.module = const_cast<hw_module_t*>(module);
         dev->device.common.close = hwc_device_close;
 
diff --git a/modules/nfc-nci/Android.mk b/modules/nfc-nci/Android.mk
new file mode 100644
index 0000000..97262ef
--- /dev/null
+++ b/modules/nfc-nci/Android.mk
@@ -0,0 +1,25 @@
+# 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.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE := nfc_nci.default
+LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/hw
+LOCAL_SRC_FILES := nfc_nci_example.c
+LOCAL_SHARED_LIBRARIES := liblog libcutils
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/modules/nfc-nci/nfc_nci_example.c b/modules/nfc-nci/nfc_nci_example.c
new file mode 100644
index 0000000..2514225
--- /dev/null
+++ b/modules/nfc-nci/nfc_nci_example.c
@@ -0,0 +1,119 @@
+/*
+ * Copyright (C) 2012 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 <errno.h>
+#include <string.h>
+
+#include <cutils/log.h>
+#include <hardware/hardware.h>
+#include <hardware/nfc.h>
+
+
+/*
+ * NCI HAL method implementations. These must be overriden
+ */
+static int hal_open(const struct nfc_nci_device *dev,
+        nfc_stack_callback_t *p_cback, nfc_stack_data_callback_t *p_data_cback) {
+    ALOGE("NFC-NCI HAL: %s", __FUNCTION__);
+    return 0;
+}
+
+static int hal_write(const struct nfc_nci_device *dev,
+        uint16_t data_len, const uint8_t *p_data) {
+    ALOGE("NFC-NCI HAL: %s", __FUNCTION__);
+    return 0;
+}
+
+static int hal_core_initialized(const struct nfc_nci_device *dev,
+        uint8_t* p_core_init_rsp_params) {
+    ALOGE("NFC-NCI HAL: %s", __FUNCTION__);
+    return 0;
+}
+
+static int hal_pre_discover(const struct nfc_nci_device *dev) {
+    ALOGE("NFC-NCI HAL: %s", __FUNCTION__);
+    return 0;
+}
+
+static int hal_close(const struct nfc_nci_device *dev) {
+    ALOGE("NFC-NCI HAL: %s", __FUNCTION__);
+    return 0;
+}
+
+static int hal_control_granted (const struct nfc_nci_device *p_dev)
+{
+    ALOGE("NFC-NCI HAL: %s", __FUNCTION__);
+    return 0;
+}
+
+
+static int hal_power_cycle (const struct nfc_nci_device *p_dev)
+{
+    ALOGE("NFC-NCI HAL: %s", __FUNCTION__);
+    return 0;
+}
+
+/*
+ * Generic device handling below - can generally be left unchanged.
+ */
+/* Close an opened nfc device instance */
+static int nfc_close(hw_device_t *dev) {
+    free(dev);
+    return 0;
+}
+
+static int nfc_open(const hw_module_t* module, const char* name,
+        hw_device_t** device) {
+    if (strcmp(name, NFC_NCI_CONTROLLER) == 0) {
+        nfc_nci_device_t *dev = calloc(1, sizeof(nfc_nci_device_t));
+
+        dev->common.tag = HARDWARE_DEVICE_TAG;
+        dev->common.version = 0x00010000; // [31:16] major, [15:0] minor
+        dev->common.module = (struct hw_module_t*) module;
+        dev->common.close = nfc_close;
+
+        // NCI HAL method pointers
+        dev->open = hal_open;
+        dev->write = hal_write;
+        dev->core_initialized = hal_core_initialized;
+        dev->pre_discover = hal_pre_discover;
+        dev->close = hal_close;
+        dev->control_granted = hal_control_granted;
+        dev->power_cycle = hal_power_cycle;
+
+        *device = (hw_device_t*) dev;
+
+        return 0;
+    } else {
+        return -EINVAL;
+    }
+}
+
+
+static struct hw_module_methods_t nfc_module_methods = {
+    .open = nfc_open,
+};
+
+struct nfc_nci_module_t HAL_MODULE_INFO_SYM = {
+    .common = {
+        .tag = HARDWARE_MODULE_TAG,
+        .module_api_version = 0x0100, // [15:8] major, [7:0] minor (1.0)
+        .hal_api_version = 0x00, // 0 is only valid value
+        .id = NFC_NCI_HARDWARE_MODULE_ID,
+        .name = "Default NFC NCI HW HAL",
+        .author = "The Android Open Source Project",
+        .methods = &nfc_module_methods,
+    },
+};
diff --git a/modules/nfc/Android.mk b/modules/nfc/Android.mk
index d541b21..429fb43 100644
--- a/modules/nfc/Android.mk
+++ b/modules/nfc/Android.mk
@@ -18,7 +18,7 @@ include $(CLEAR_VARS)
 
 LOCAL_MODULE := nfc.default
 LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/hw
-LOCAL_SRC_FILES := nfc_hw_example.c
+LOCAL_SRC_FILES := nfc_pn544_example.c
 LOCAL_SHARED_LIBRARIES := liblog libcutils
 LOCAL_MODULE_TAGS := optional
 
diff --git a/modules/nfc/nfc_hw_example.c b/modules/nfc/nfc_pn544_example.c
index 54c9c56..54c9c56 100644
--- a/modules/nfc/nfc_hw_example.c
+++ b/modules/nfc/nfc_pn544_example.c
diff --git a/modules/usbaudio/audio_hw.c b/modules/usbaudio/audio_hw.c
index 9283016..f33c343 100644
--- a/modules/usbaudio/audio_hw.c
+++ b/modules/usbaudio/audio_hw.c
@@ -379,11 +379,6 @@ static int adev_close(hw_device_t *device)
     return 0;
 }
 
-static uint32_t adev_get_supported_devices(const struct audio_hw_device *dev)
-{
-    return AUDIO_DEVICE_OUT_ALL_USB;
-}
-
 static int adev_open(const hw_module_t* module, const char* name,
                      hw_device_t** device)
 {
@@ -398,11 +393,10 @@ static int adev_open(const hw_module_t* module, const char* name,
         return -ENOMEM;
 
     adev->hw_device.common.tag = HARDWARE_DEVICE_TAG;
-    adev->hw_device.common.version = AUDIO_DEVICE_API_VERSION_1_0;
+    adev->hw_device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
     adev->hw_device.common.module = (struct hw_module_t *) module;
     adev->hw_device.common.close = adev_close;
 
-    adev->hw_device.get_supported_devices = adev_get_supported_devices;
     adev->hw_device.init_check = adev_init_check;
     adev->hw_device.set_voice_volume = adev_set_voice_volume;
     adev->hw_device.set_master_volume = adev_set_master_volume;