HACK: Add dummy audio

At boot, it complained that it could not get passthrough implementation
for the following:
- android.hardware.audio@4.0::IDevicesFactory/default.
- android.hardware.audio.effect@4.0::IEffectsFactory/default.
- android.hardware.bluetooth.a2dp@1.0::IBluetoothAudioOffload/default.
- android.hardware.soundtrigger@2.1::ISoundTriggerHw/default.

With that makes android.hardware.audio start and makes the
boot goes furthurer:
  # lshal
  All binderized services (registered services through hwservicemanager)
  R Interface                                                                 Thread Use Server Clients
  [...]
  Y android.hardware.audio.effect@4.0::IEffectsFactory/default                0/4        202    210 175
  Y android.hardware.audio@4.0::IDevicesFactory/default                       0/4        202    210 175

The null linux audio driver was also compiled in to make sure
that the device is not broken due to bad audio policies, and
before that no sound card were present on the target device.

Signed-off-by: Denis 'GNUtoo' Carikli <GNUtoo@cyberdimension.org>

10 files changed, 2733 insertions, 10 deletions

diff --git a/BoardConfig.mk b/BoardConfig.mk
index d1b2613..a2eeb7b 100644
--- a/BoardConfig.mk
+++ b/BoardConfig.mk
@@ -63,4 +63,6 @@ BOARD_GPU_DRIVERS := swrast
 TARGET_USES_64_BIT_BINDER := true
 BOARD_BUILD_SYSTEM_ROOT_IMAGE := true
 
+BOARD_USES_GENERIC_AUDIO := true
+
 DEVICE_MANIFEST_FILE := device/samsung/i9305/manifest.xml
diff --git a/audio/Android.mk b/audio/Android.mk
new file mode 100644
index 0000000..7db7659
--- /dev/null
+++ b/audio/Android.mk
@@ -0,0 +1,58 @@
+#
+# 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_VENDOR_MODULE := true
+LOCAL_MODULE := audio.primary.i9305
+LOCAL_MODULE_RELATIVE_PATH := hw
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_SHARED_LIBRARIES := libcutils liblog
+
+LOCAL_SRC_FILES := audio_hw.c
+
+LOCAL_C_INCLUDES += \
+			external/tinyalsa/include \
+
+LOCAL_SHARED_LIBRARIES += \
+			libdl \
+			libtinyalsa
+
+LOCAL_CFLAGS := -Wno-unused-parameter
+LOCAL_HEADER_LIBRARIES := libhardware_headers
+
+include $(BUILD_SHARED_LIBRARY)
+
+include $(CLEAR_VARS)
+
+LOCAL_VENDOR_MODULE := true
+LOCAL_MODULE := audio.primary.i9305_legacy
+LOCAL_MODULE_RELATIVE_PATH := hw
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_SHARED_LIBRARIES := libcutils liblog
+
+LOCAL_SRC_FILES := audio_hw_legacy.c
+
+LOCAL_SHARED_LIBRARIES += \
+			libdl
+
+LOCAL_CFLAGS := -Wno-unused-parameter
+LOCAL_HEADER_LIBRARIES := libhardware_headers
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/audio/MODULE_LICENSE_APACHE2 b/audio/MODULE_LICENSE_APACHE2
new file mode 100644
index 0000000..e69de29
--- /dev/null
+++ b/audio/MODULE_LICENSE_APACHE2
diff --git a/audio/NOTICE b/audio/NOTICE
new file mode 100644
index 0000000..3237da6
--- /dev/null
+++ b/audio/NOTICE
@@ -0,0 +1,190 @@
+
+   Copyright (c) 2008-2009, The Android Open Source Project
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+
+   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.
+
+
+                                 Apache License
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+   TERMS AND CONDITIONS FOR USE, REPRODUCTION, AND DISTRIBUTION
+
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+
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+      of this License, Derivative Works shall not include works that remain
+      separable from, or merely link (or bind by name) to the interfaces of,
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+      "Contribution" shall mean any work of authorship, including
+      the original version of the Work and any modifications or additions
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+      "Contributor" shall mean Licensor and any individual or Legal Entity
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+   4. Redistribution. You may reproduce and distribute copies of the
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+          Derivative Works a copy of this License; and
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+          stating that You changed the files; and
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+      the terms of any separate license agreement you may have executed
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+   6. Trademarks. This License does not grant permission to use the trade
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diff --git a/audio/audio_hw.c b/audio/audio_hw.c
new file mode 100644
index 0000000..62a2daa
--- /dev/null
+++ b/audio/audio_hw.c
@@ -0,0 +1,1674 @@
+/*
+ * 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 "audio_hw_generic"
+
+#include <assert.h>
+#include <errno.h>
+#include <inttypes.h>
+#include <pthread.h>
+#include <stdint.h>
+#include <stdlib.h>
+#include <sys/time.h>
+#include <dlfcn.h>
+#include <fcntl.h>
+#include <unistd.h>
+
+#include <log/log.h>
+#include <cutils/str_parms.h>
+
+#include <hardware/hardware.h>
+#include <system/audio.h>
+#include <hardware/audio.h>
+#include <tinyalsa/asoundlib.h>
+
+#define PCM_CARD 0
+#define PCM_DEVICE 0
+
+
+#define OUT_PERIOD_MS 15
+#define OUT_PERIOD_COUNT 4
+
+#define IN_PERIOD_MS 15
+#define IN_PERIOD_COUNT 4
+
+struct generic_audio_device {
+    struct audio_hw_device device; // Constant after init
+    pthread_mutex_t lock;
+    bool mic_mute;                 // Proteced by this->lock
+    struct mixer* mixer;           // Proteced by this->lock
+};
+
+/* If not NULL, this is a pointer to the fallback module.
+ * This really is the original default audio device /dev/eac which we will use
+ * if no alsa devices are detected.
+ */
+static struct audio_module*  sFallback;
+static pthread_once_t sFallbackOnce = PTHREAD_ONCE_INIT;
+static void fallback_init(void);
+static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state);
+static int adev_get_microphones(const audio_hw_device_t *dev,
+                                struct audio_microphone_characteristic_t *mic_array,
+                                size_t *mic_count);
+
+
+typedef struct audio_vbuffer {
+    pthread_mutex_t lock;
+    uint8_t *  data;
+    size_t     frame_size;
+    size_t     frame_count;
+    size_t     head;
+    size_t     tail;
+    size_t     live;
+} audio_vbuffer_t;
+
+static int audio_vbuffer_init (audio_vbuffer_t * audio_vbuffer, size_t frame_count,
+                              size_t frame_size) {
+    if (!audio_vbuffer) {
+        return -EINVAL;
+    }
+    audio_vbuffer->frame_size = frame_size;
+    audio_vbuffer->frame_count = frame_count;
+    size_t bytes = frame_count * frame_size;
+    audio_vbuffer->data = calloc(bytes, 1);
+    if (!audio_vbuffer->data) {
+        return -ENOMEM;
+    }
+    audio_vbuffer->head = 0;
+    audio_vbuffer->tail = 0;
+    audio_vbuffer->live = 0;
+    pthread_mutex_init (&audio_vbuffer->lock, (const pthread_mutexattr_t *) NULL);
+    return 0;
+}
+
+static int audio_vbuffer_destroy (audio_vbuffer_t * audio_vbuffer) {
+    if (!audio_vbuffer) {
+        return -EINVAL;
+    }
+    free(audio_vbuffer->data);
+    pthread_mutex_destroy(&audio_vbuffer->lock);
+    return 0;
+}
+
+static int audio_vbuffer_live (audio_vbuffer_t * audio_vbuffer) {
+    if (!audio_vbuffer) {
+        return -EINVAL;
+    }
+    pthread_mutex_lock (&audio_vbuffer->lock);
+    int live = audio_vbuffer->live;
+    pthread_mutex_unlock (&audio_vbuffer->lock);
+    return live;
+}
+
+#define MIN(a,b) (((a)<(b))?(a):(b))
+static size_t audio_vbuffer_write (audio_vbuffer_t * audio_vbuffer, const void * buffer, size_t frame_count) {
+    size_t frames_written = 0;
+    pthread_mutex_lock (&audio_vbuffer->lock);
+
+    while (frame_count != 0) {
+        int frames = 0;
+        if (audio_vbuffer->live == 0 || audio_vbuffer->head > audio_vbuffer->tail) {
+            frames = MIN(frame_count, audio_vbuffer->frame_count - audio_vbuffer->head);
+        } else if (audio_vbuffer->head < audio_vbuffer->tail) {
+            frames = MIN(frame_count, audio_vbuffer->tail - (audio_vbuffer->head));
+        } else {
+            // Full
+            break;
+        }
+        memcpy(&audio_vbuffer->data[audio_vbuffer->head*audio_vbuffer->frame_size],
+               &((uint8_t*)buffer)[frames_written*audio_vbuffer->frame_size],
+               frames*audio_vbuffer->frame_size);
+        audio_vbuffer->live += frames;
+        frames_written += frames;
+        frame_count -= frames;
+        audio_vbuffer->head = (audio_vbuffer->head + frames) % audio_vbuffer->frame_count;
+    }
+
+    pthread_mutex_unlock (&audio_vbuffer->lock);
+    return frames_written;
+}
+
+static size_t audio_vbuffer_read (audio_vbuffer_t * audio_vbuffer, void * buffer, size_t frame_count) {
+    size_t frames_read = 0;
+    pthread_mutex_lock (&audio_vbuffer->lock);
+
+    while (frame_count != 0) {
+        int frames = 0;
+        if (audio_vbuffer->live == audio_vbuffer->frame_count ||
+            audio_vbuffer->tail > audio_vbuffer->head) {
+            frames = MIN(frame_count, audio_vbuffer->frame_count - audio_vbuffer->tail);
+        } else if (audio_vbuffer->tail < audio_vbuffer->head) {
+            frames = MIN(frame_count, audio_vbuffer->head - audio_vbuffer->tail);
+        } else {
+            break;
+        }
+        memcpy(&((uint8_t*)buffer)[frames_read*audio_vbuffer->frame_size],
+               &audio_vbuffer->data[audio_vbuffer->tail*audio_vbuffer->frame_size],
+               frames*audio_vbuffer->frame_size);
+        audio_vbuffer->live -= frames;
+        frames_read += frames;
+        frame_count -= frames;
+        audio_vbuffer->tail = (audio_vbuffer->tail + frames) % audio_vbuffer->frame_count;
+    }
+
+    pthread_mutex_unlock (&audio_vbuffer->lock);
+    return frames_read;
+}
+
+struct generic_stream_out {
+    struct audio_stream_out stream;   // Constant after init
+    pthread_mutex_t lock;
+    struct generic_audio_device *dev; // Constant after init
+    audio_devices_t device;           // Protected by this->lock
+    struct audio_config req_config;   // Constant after init
+    struct pcm_config pcm_config;     // Constant after init
+    audio_vbuffer_t buffer;           // Constant after init
+
+    // Time & Position Keeping
+    bool standby;                      // Protected by this->lock
+    uint64_t underrun_position;        // Protected by this->lock
+    struct timespec underrun_time;     // Protected by this->lock
+    uint64_t last_write_time_us;       // Protected by this->lock
+    uint64_t frames_total_buffered;    // Protected by this->lock
+    uint64_t frames_written;           // Protected by this->lock
+    uint64_t frames_rendered;          // Protected by this->lock
+
+    // Worker
+    pthread_t worker_thread;          // Constant after init
+    pthread_cond_t worker_wake;       // Protected by this->lock
+    bool worker_standby;              // Protected by this->lock
+    bool worker_exit;                 // Protected by this->lock
+};
+
+struct generic_stream_in {
+    struct audio_stream_in stream;    // Constant after init
+    pthread_mutex_t lock;
+    struct generic_audio_device *dev; // Constant after init
+    audio_devices_t device;           // Protected by this->lock
+    struct audio_config req_config;   // Constant after init
+    struct pcm *pcm;                  // Protected by this->lock
+    struct pcm_config pcm_config;     // Constant after init
+    int16_t *stereo_to_mono_buf;      // Protected by this->lock
+    size_t stereo_to_mono_buf_size;   // Protected by this->lock
+    audio_vbuffer_t buffer;           // Protected by this->lock
+
+    // Time & Position Keeping
+    bool standby;                     // Protected by this->lock
+    int64_t standby_position;         // Protected by this->lock
+    struct timespec standby_exit_time;// Protected by this->lock
+    int64_t standby_frames_read;      // Protected by this->lock
+
+    // Worker
+    pthread_t worker_thread;          // Constant after init
+    pthread_cond_t worker_wake;       // Protected by this->lock
+    bool worker_standby;              // Protected by this->lock
+    bool worker_exit;                 // Protected by this->lock
+};
+
+static struct pcm_config pcm_config_out = {
+    .channels = 2,
+    .rate = 0,
+    .period_size = 0,
+    .period_count = OUT_PERIOD_COUNT,
+    .format = PCM_FORMAT_S16_LE,
+    .start_threshold = 0,
+};
+
+static struct pcm_config pcm_config_in = {
+    .channels = 2,
+    .rate = 0,
+    .period_size = 0,
+    .period_count = IN_PERIOD_COUNT,
+    .format = PCM_FORMAT_S16_LE,
+    .start_threshold = 0,
+    .stop_threshold = INT_MAX,
+};
+
+static pthread_mutex_t adev_init_lock = PTHREAD_MUTEX_INITIALIZER;
+static unsigned int audio_device_ref_count = 0;
+
+static uint32_t out_get_sample_rate(const struct audio_stream *stream)
+{
+    struct generic_stream_out *out = (struct generic_stream_out *)stream;
+    return out->req_config.sample_rate;
+}
+
+static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
+{
+    return -ENOSYS;
+}
+
+static size_t out_get_buffer_size(const struct audio_stream *stream)
+{
+    struct generic_stream_out *out = (struct generic_stream_out *)stream;
+    int size = out->pcm_config.period_size *
+                audio_stream_out_frame_size(&out->stream);
+
+    return size;
+}
+
+static audio_channel_mask_t out_get_channels(const struct audio_stream *stream)
+{
+    struct generic_stream_out *out = (struct generic_stream_out *)stream;
+    return out->req_config.channel_mask;
+}
+
+static audio_format_t out_get_format(const struct audio_stream *stream)
+{
+    struct generic_stream_out *out = (struct generic_stream_out *)stream;
+
+    return out->req_config.format;
+}
+
+static int out_set_format(struct audio_stream *stream, audio_format_t format)
+{
+    return -ENOSYS;
+}
+
+static int out_dump(const struct audio_stream *stream, int fd)
+{
+    struct generic_stream_out *out = (struct generic_stream_out *)stream;
+    pthread_mutex_lock(&out->lock);
+    dprintf(fd, "\tout_dump:\n"
+                "\t\tsample rate: %u\n"
+                "\t\tbuffer size: %zu\n"
+                "\t\tchannel mask: %08x\n"
+                "\t\tformat: %d\n"
+                "\t\tdevice: %08x\n"
+                "\t\taudio dev: %p\n\n",
+                out_get_sample_rate(stream),
+                out_get_buffer_size(stream),
+                out_get_channels(stream),
+                out_get_format(stream),
+                out->device,
+                out->dev);
+    pthread_mutex_unlock(&out->lock);
+    return 0;
+}
+
+static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+    struct generic_stream_out *out = (struct generic_stream_out *)stream;
+    struct str_parms *parms;
+    char value[32];
+    int ret = -ENOSYS;
+    int success;
+    long val;
+    char *end;
+
+    if (kvpairs == NULL || kvpairs[0] == 0) {
+        return 0;
+    }
+    pthread_mutex_lock(&out->lock);
+    if (out->standby) {
+        parms = str_parms_create_str(kvpairs);
+        success = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING,
+                                value, sizeof(value));
+        if (success >= 0) {
+            errno = 0;
+            val = strtol(value, &end, 10);
+            if (errno == 0 && (end != NULL) && (*end == '\0') && ((int)val == val)) {
+                out->device = (int)val;
+                ret = 0;
+            }
+        }
+
+        // NO op for AUDIO_PARAMETER_DEVICE_CONNECT and AUDIO_PARAMETER_DEVICE_DISCONNECT
+        success = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_CONNECT,
+                                value, sizeof(value));
+        if (success >= 0) {
+            ret = 0;
+        }
+        success = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_DISCONNECT,
+                                value, sizeof(value));
+        if (success >= 0) {
+            ret = 0;
+        }
+
+        if (ret != 0) {
+            ALOGD("%s Unsupported parameter %s", __FUNCTION__, kvpairs);
+        }
+
+        str_parms_destroy(parms);
+    }
+    pthread_mutex_unlock(&out->lock);
+    return ret;
+}
+
+static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
+{
+    struct generic_stream_out *out = (struct generic_stream_out *)stream;
+    struct str_parms *query = str_parms_create_str(keys);
+    char *str = NULL;
+    char value[256];
+    struct str_parms *reply = str_parms_create();
+    int ret;
+    bool get = false;
+
+    ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
+    if (ret >= 0) {
+        pthread_mutex_lock(&out->lock);
+        str_parms_add_int(reply, AUDIO_PARAMETER_STREAM_ROUTING, out->device);
+        pthread_mutex_unlock(&out->lock);
+        get = true;
+    }
+
+    if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS)) {
+        value[0] = 0;
+        strcat(value, "AUDIO_FORMAT_PCM_16_BIT");
+        str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_FORMATS, value);
+        get = true;
+    }
+
+    if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_FORMAT)) {
+        value[0] = 0;
+        strcat(value, "AUDIO_FORMAT_PCM_16_BIT");
+        str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_FORMAT, value);
+        get = true;
+    }
+
+    if (get) {
+        str = strdup(str_parms_to_str(reply));
+    }
+    else {
+        ALOGD("%s Unsupported paramter: %s", __FUNCTION__, keys);
+    }
+
+    str_parms_destroy(query);
+    str_parms_destroy(reply);
+    return str;
+}
+
+static uint32_t out_get_latency(const struct audio_stream_out *stream)
+{
+    struct generic_stream_out *out = (struct generic_stream_out *)stream;
+    return (out->pcm_config.period_size * 1000) / out->pcm_config.rate;
+}
+
+static int out_set_volume(struct audio_stream_out *stream, float left,
+                          float right)
+{
+    return -ENOSYS;
+}
+
+static void *out_write_worker(void * args)
+{
+    struct generic_stream_out *out = (struct generic_stream_out *)args;
+    struct pcm *pcm = NULL;
+    uint8_t *buffer = NULL;
+    int buffer_frames;
+    int buffer_size;
+    bool restart = false;
+    bool shutdown = false;
+    while (true) {
+        pthread_mutex_lock(&out->lock);
+        while (out->worker_standby || restart) {
+            restart = false;
+            if (pcm) {
+                pcm_close(pcm); // Frees pcm
+                pcm = NULL;
+                free(buffer);
+                buffer=NULL;
+            }
+            if (out->worker_exit) {
+                break;
+            }
+            pthread_cond_wait(&out->worker_wake, &out->lock);
+        }
+
+        if (out->worker_exit) {
+            if (!out->worker_standby) {
+                ALOGE("Out worker not in standby before exiting");
+            }
+            shutdown = true;
+        }
+
+        while (!shutdown && audio_vbuffer_live(&out->buffer) == 0) {
+            pthread_cond_wait(&out->worker_wake, &out->lock);
+        }
+
+        if (shutdown) {
+            pthread_mutex_unlock(&out->lock);
+            break;
+        }
+
+        if (!pcm) {
+            pcm = pcm_open(PCM_CARD, PCM_DEVICE,
+                          PCM_OUT | PCM_MONOTONIC, &out->pcm_config);
+            if (!pcm_is_ready(pcm)) {
+                ALOGE("pcm_open(out) failed: %s: channels %d format %d rate %d",
+                  pcm_get_error(pcm),
+                  out->pcm_config.channels,
+                  out->pcm_config.format,
+                  out->pcm_config.rate
+                   );
+                pthread_mutex_unlock(&out->lock);
+                break;
+            }
+            buffer_frames = out->pcm_config.period_size;
+            buffer_size = pcm_frames_to_bytes(pcm, buffer_frames);
+            buffer = malloc(buffer_size);
+            if (!buffer) {
+                ALOGE("could not allocate write buffer");
+                pthread_mutex_unlock(&out->lock);
+                break;
+            }
+        }
+        int frames = audio_vbuffer_read(&out->buffer, buffer, buffer_frames);
+        pthread_mutex_unlock(&out->lock);
+        int ret = pcm_write(pcm, buffer, pcm_frames_to_bytes(pcm, frames));
+        if (ret != 0) {
+            ALOGE("pcm_write failed %s", pcm_get_error(pcm));
+            restart = true;
+        }
+    }
+    if (buffer) {
+        free(buffer);
+    }
+
+    return NULL;
+}
+
+// Call with in->lock held
+static void get_current_output_position(struct generic_stream_out *out,
+                                       uint64_t * position,
+                                       struct timespec * timestamp) {
+    struct timespec curtime = { .tv_sec = 0, .tv_nsec = 0 };
+    clock_gettime(CLOCK_MONOTONIC, &curtime);
+    const int64_t now_us = (curtime.tv_sec * 1000000000LL + curtime.tv_nsec) / 1000;
+    if (timestamp) {
+        *timestamp = curtime;
+    }
+    int64_t position_since_underrun;
+    if (out->standby) {
+        position_since_underrun = 0;
+    } else {
+        const int64_t first_us = (out->underrun_time.tv_sec * 1000000000LL +
+                                  out->underrun_time.tv_nsec) / 1000;
+        position_since_underrun = (now_us - first_us) *
+                out_get_sample_rate(&out->stream.common) /
+                1000000;
+        if (position_since_underrun < 0) {
+            position_since_underrun = 0;
+        }
+    }
+    *position = out->underrun_position + position_since_underrun;
+
+    // The device will reuse the same output stream leading to periods of
+    // underrun.
+    if (*position > out->frames_written) {
+        ALOGW("Not supplying enough data to HAL, expected position %" PRIu64 " , only wrote "
+              "%" PRIu64,
+              *position, out->frames_written);
+
+        *position = out->frames_written;
+        out->underrun_position = *position;
+        out->underrun_time = curtime;
+        out->frames_total_buffered = 0;
+    }
+}
+
+
+static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
+                         size_t bytes)
+{
+    struct generic_stream_out *out = (struct generic_stream_out *)stream;
+    const size_t frames =  bytes / audio_stream_out_frame_size(stream);
+
+    pthread_mutex_lock(&out->lock);
+
+    if (out->worker_standby) {
+        out->worker_standby = false;
+    }
+
+    uint64_t current_position;
+    struct timespec current_time;
+
+    get_current_output_position(out, &current_position, &current_time);
+    const uint64_t now_us = (current_time.tv_sec * 1000000000LL +
+                             current_time.tv_nsec) / 1000;
+    if (out->standby) {
+        out->standby = false;
+        out->underrun_time = current_time;
+        out->frames_rendered = 0;
+        out->frames_total_buffered = 0;
+    }
+
+    size_t frames_written = audio_vbuffer_write(&out->buffer, buffer, frames);
+    pthread_cond_signal(&out->worker_wake);
+
+    /* Implementation just consumes bytes if we start getting backed up */
+    out->frames_written += frames;
+    out->frames_rendered += frames;
+    out->frames_total_buffered += frames;
+
+    // We simulate the audio device blocking when it's write buffers become
+    // full.
+
+    // At the beginning or after an underrun, try to fill up the vbuffer.
+    // This will be throttled by the PlaybackThread
+    int frames_sleep = out->frames_total_buffered < out->buffer.frame_count ? 0 : frames;
+
+    uint64_t sleep_time_us = frames_sleep * 1000000LL /
+                            out_get_sample_rate(&stream->common);
+
+    // If the write calls are delayed, subtract time off of the sleep to
+    // compensate
+    uint64_t time_since_last_write_us = now_us - out->last_write_time_us;
+    if (time_since_last_write_us < sleep_time_us) {
+        sleep_time_us -= time_since_last_write_us;
+    } else {
+        sleep_time_us = 0;
+    }
+    out->last_write_time_us = now_us + sleep_time_us;
+
+    pthread_mutex_unlock(&out->lock);
+
+    if (sleep_time_us > 0) {
+        usleep(sleep_time_us);
+    }
+
+    if (frames_written < frames) {
+        ALOGW("Hardware backing HAL too slow, could only write %zu of %zu frames", frames_written, frames);
+    }
+
+    /* Always consume all bytes */
+    return bytes;
+}
+
+static int out_get_presentation_position(const struct audio_stream_out *stream,
+                                   uint64_t *frames, struct timespec *timestamp)
+
+{
+    if (stream == NULL || frames == NULL || timestamp == NULL) {
+        return -EINVAL;
+    }
+    struct generic_stream_out *out = (struct generic_stream_out *)stream;
+
+    pthread_mutex_lock(&out->lock);
+    get_current_output_position(out, frames, timestamp);
+    pthread_mutex_unlock(&out->lock);
+
+    return 0;
+}
+
+static int out_get_render_position(const struct audio_stream_out *stream,
+                                   uint32_t *dsp_frames)
+{
+    if (stream == NULL || dsp_frames == NULL) {
+        return -EINVAL;
+    }
+    struct generic_stream_out *out = (struct generic_stream_out *)stream;
+    pthread_mutex_lock(&out->lock);
+    *dsp_frames = out->frames_rendered;
+    pthread_mutex_unlock(&out->lock);
+    return 0;
+}
+
+// Must be called with out->lock held
+static void do_out_standby(struct generic_stream_out *out)
+{
+    int frames_sleep = 0;
+    uint64_t sleep_time_us = 0;
+    if (out->standby) {
+        return;
+    }
+    while (true) {
+        get_current_output_position(out, &out->underrun_position, NULL);
+        frames_sleep = out->frames_written - out->underrun_position;
+
+        if (frames_sleep == 0) {
+            break;
+        }
+
+        sleep_time_us = frames_sleep * 1000000LL /
+                        out_get_sample_rate(&out->stream.common);
+
+        pthread_mutex_unlock(&out->lock);
+        usleep(sleep_time_us);
+        pthread_mutex_lock(&out->lock);
+    }
+    out->worker_standby = true;
+    out->standby = true;
+}
+
+static int out_standby(struct audio_stream *stream)
+{
+    struct generic_stream_out *out = (struct generic_stream_out *)stream;
+    pthread_mutex_lock(&out->lock);
+    do_out_standby(out);
+    pthread_mutex_unlock(&out->lock);
+    return 0;
+}
+
+static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+    // out_add_audio_effect is a no op
+    return 0;
+}
+
+static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+    // out_remove_audio_effect is a no op
+    return 0;
+}
+
+static int out_get_next_write_timestamp(const struct audio_stream_out *stream,
+                                        int64_t *timestamp)
+{
+    return -ENOSYS;
+}
+
+static uint32_t in_get_sample_rate(const struct audio_stream *stream)
+{
+    struct generic_stream_in *in = (struct generic_stream_in *)stream;
+    return in->req_config.sample_rate;
+}
+
+static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
+{
+    return -ENOSYS;
+}
+
+static int refine_output_parameters(uint32_t *sample_rate, audio_format_t *format, audio_channel_mask_t *channel_mask)
+{
+    static const uint32_t sample_rates [] = {8000,11025,16000,22050,24000,32000,
+                                            44100,48000};
+    static const int sample_rates_count = sizeof(sample_rates)/sizeof(uint32_t);
+    bool inval = false;
+    if (*format != AUDIO_FORMAT_PCM_16_BIT) {
+        *format = AUDIO_FORMAT_PCM_16_BIT;
+        inval = true;
+    }
+
+    int channel_count = popcount(*channel_mask);
+    if (channel_count != 1 && channel_count != 2) {
+        *channel_mask = AUDIO_CHANNEL_IN_STEREO;
+        inval = true;
+    }
+
+    int i;
+    for (i = 0; i < sample_rates_count; i++) {
+        if (*sample_rate < sample_rates[i]) {
+            *sample_rate = sample_rates[i];
+            inval=true;
+            break;
+        }
+        else if (*sample_rate == sample_rates[i]) {
+            break;
+        }
+        else if (i == sample_rates_count-1) {
+            // Cap it to the highest rate we support
+            *sample_rate = sample_rates[i];
+            inval=true;
+        }
+    }
+
+    if (inval) {
+        return -EINVAL;
+    }
+    return 0;
+}
+
+static int refine_input_parameters(uint32_t *sample_rate, audio_format_t *format, audio_channel_mask_t *channel_mask)
+{
+    static const uint32_t sample_rates [] = {8000, 11025, 16000, 22050, 44100, 48000};
+    static const int sample_rates_count = sizeof(sample_rates)/sizeof(uint32_t);
+    bool inval = false;
+    // Only PCM_16_bit is supported. If this is changed, stereo to mono drop
+    // must be fixed in in_read
+    if (*format != AUDIO_FORMAT_PCM_16_BIT) {
+        *format = AUDIO_FORMAT_PCM_16_BIT;
+        inval = true;
+    }
+
+    int channel_count = popcount(*channel_mask);
+    if (channel_count != 1 && channel_count != 2) {
+        *channel_mask = AUDIO_CHANNEL_IN_STEREO;
+        inval = true;
+    }
+
+    int i;
+    for (i = 0; i < sample_rates_count; i++) {
+        if (*sample_rate < sample_rates[i]) {
+            *sample_rate = sample_rates[i];
+            inval=true;
+            break;
+        }
+        else if (*sample_rate == sample_rates[i]) {
+            break;
+        }
+        else if (i == sample_rates_count-1) {
+            // Cap it to the highest rate we support
+            *sample_rate = sample_rates[i];
+            inval=true;
+        }
+    }
+
+    if (inval) {
+        return -EINVAL;
+    }
+    return 0;
+}
+
+static int check_input_parameters(uint32_t sample_rate, audio_format_t format,
+                                  audio_channel_mask_t channel_mask)
+{
+    return refine_input_parameters(&sample_rate, &format, &channel_mask);
+}
+
+static size_t get_input_buffer_size(uint32_t sample_rate, audio_format_t format,
+                                    audio_channel_mask_t channel_mask)
+{
+    size_t size;
+    int channel_count = popcount(channel_mask);
+    if (check_input_parameters(sample_rate, format, channel_mask) != 0)
+        return 0;
+
+    size = sample_rate*IN_PERIOD_MS/1000;
+    // Audioflinger expects audio buffers to be multiple of 16 frames
+    size = ((size + 15) / 16) * 16;
+    size *= sizeof(short) * channel_count;
+
+    return size;
+}
+
+
+static size_t in_get_buffer_size(const struct audio_stream *stream)
+{
+    struct generic_stream_in *in = (struct generic_stream_in *)stream;
+    int size = get_input_buffer_size(in->req_config.sample_rate,
+                                 in->req_config.format,
+                                 in->req_config.channel_mask);
+
+    return size;
+}
+
+static audio_channel_mask_t in_get_channels(const struct audio_stream *stream)
+{
+    struct generic_stream_in *in = (struct generic_stream_in *)stream;
+    return in->req_config.channel_mask;
+}
+
+static audio_format_t in_get_format(const struct audio_stream *stream)
+{
+    struct generic_stream_in *in = (struct generic_stream_in *)stream;
+    return in->req_config.format;
+}
+
+static int in_set_format(struct audio_stream *stream, audio_format_t format)
+{
+    return -ENOSYS;
+}
+
+static int in_dump(const struct audio_stream *stream, int fd)
+{
+    struct generic_stream_in *in = (struct generic_stream_in *)stream;
+
+    pthread_mutex_lock(&in->lock);
+    dprintf(fd, "\tin_dump:\n"
+                "\t\tsample rate: %u\n"
+                "\t\tbuffer size: %zu\n"
+                "\t\tchannel mask: %08x\n"
+                "\t\tformat: %d\n"
+                "\t\tdevice: %08x\n"
+                "\t\taudio dev: %p\n\n",
+                in_get_sample_rate(stream),
+                in_get_buffer_size(stream),
+                in_get_channels(stream),
+                in_get_format(stream),
+                in->device,
+                in->dev);
+    pthread_mutex_unlock(&in->lock);
+    return 0;
+}
+
+static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+    struct generic_stream_in *in = (struct generic_stream_in *)stream;
+    struct str_parms *parms;
+    char value[32];
+    int ret = -ENOSYS;
+    int success;
+    long val;
+    char *end;
+
+    if (kvpairs == NULL || kvpairs[0] == 0) {
+        return 0;
+    }
+    pthread_mutex_lock(&in->lock);
+    if (in->standby) {
+        parms = str_parms_create_str(kvpairs);
+
+        success = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING,
+                                value, sizeof(value));
+        if (success >= 0) {
+            errno = 0;
+            val = strtol(value, &end, 10);
+            if ((errno == 0) && (end != NULL) && (*end == '\0') && ((int)val == val)) {
+                in->device = (int)val;
+                ret = 0;
+            }
+        }
+        // NO op for AUDIO_PARAMETER_DEVICE_CONNECT and AUDIO_PARAMETER_DEVICE_DISCONNECT
+        success = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_CONNECT,
+                                value, sizeof(value));
+        if (success >= 0) {
+            ret = 0;
+        }
+        success = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_DISCONNECT,
+                                value, sizeof(value));
+        if (success >= 0) {
+            ret = 0;
+        }
+
+        if (ret != 0) {
+            ALOGD("%s: Unsupported parameter %s", __FUNCTION__, kvpairs);
+        }
+
+        str_parms_destroy(parms);
+    }
+    pthread_mutex_unlock(&in->lock);
+    return ret;
+}
+
+static char * in_get_parameters(const struct audio_stream *stream,
+                                const char *keys)
+{
+    struct generic_stream_in *in = (struct generic_stream_in *)stream;
+    struct str_parms *query = str_parms_create_str(keys);
+    char *str = NULL;
+    char value[256];
+    struct str_parms *reply = str_parms_create();
+    int ret;
+    bool get = false;
+
+    ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
+    if (ret >= 0) {
+        str_parms_add_int(reply, AUDIO_PARAMETER_STREAM_ROUTING, in->device);
+        get = true;
+    }
+
+    if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS)) {
+        value[0] = 0;
+        strcat(value, "AUDIO_FORMAT_PCM_16_BIT");
+        str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_FORMATS, value);
+        get = true;
+    }
+
+    if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_FORMAT)) {
+        value[0] = 0;
+        strcat(value, "AUDIO_FORMAT_PCM_16_BIT");
+        str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_FORMAT, value);
+        get = true;
+    }
+
+    if (get) {
+        str = strdup(str_parms_to_str(reply));
+    }
+    else {
+        ALOGD("%s Unsupported paramter: %s", __FUNCTION__, keys);
+    }
+
+    str_parms_destroy(query);
+    str_parms_destroy(reply);
+    return str;
+}
+
+static int in_set_gain(struct audio_stream_in *stream, float gain)
+{
+    // in_set_gain is a no op
+    return 0;
+}
+
+// Call with in->lock held
+static void get_current_input_position(struct generic_stream_in *in,
+                                       int64_t * position,
+                                       struct timespec * timestamp) {
+    struct timespec t = { .tv_sec = 0, .tv_nsec = 0 };
+    clock_gettime(CLOCK_MONOTONIC, &t);
+    const int64_t now_us = (t.tv_sec * 1000000000LL + t.tv_nsec) / 1000;
+    if (timestamp) {
+        *timestamp = t;
+    }
+    int64_t position_since_standby;
+    if (in->standby) {
+        position_since_standby = 0;
+    } else {
+        const int64_t first_us = (in->standby_exit_time.tv_sec * 1000000000LL +
+                                  in->standby_exit_time.tv_nsec) / 1000;
+        position_since_standby = (now_us - first_us) *
+                in_get_sample_rate(&in->stream.common) /
+                1000000;
+        if (position_since_standby < 0) {
+            position_since_standby = 0;
+        }
+    }
+    *position = in->standby_position + position_since_standby;
+}
+
+// Must be called with in->lock held
+static void do_in_standby(struct generic_stream_in *in)
+{
+    if (in->standby) {
+        return;
+    }
+    in->worker_standby = true;
+    get_current_input_position(in, &in->standby_position, NULL);
+    in->standby = true;
+}
+
+static int in_standby(struct audio_stream *stream)
+{
+    struct generic_stream_in *in = (struct generic_stream_in *)stream;
+    pthread_mutex_lock(&in->lock);
+    do_in_standby(in);
+    pthread_mutex_unlock(&in->lock);
+    return 0;
+}
+
+static void *in_read_worker(void * args)
+{
+    struct generic_stream_in *in = (struct generic_stream_in *)args;
+    struct pcm *pcm = NULL;
+    uint8_t *buffer = NULL;
+    size_t buffer_frames;
+    int buffer_size;
+
+    bool restart = false;
+    bool shutdown = false;
+    while (true) {
+        pthread_mutex_lock(&in->lock);
+        while (in->worker_standby || restart) {
+            restart = false;
+            if (pcm) {
+                pcm_close(pcm); // Frees pcm
+                pcm = NULL;
+                free(buffer);
+                buffer=NULL;
+            }
+            if (in->worker_exit) {
+                break;
+            }
+            pthread_cond_wait(&in->worker_wake, &in->lock);
+        }
+
+        if (in->worker_exit) {
+            if (!in->worker_standby) {
+                ALOGE("In worker not in standby before exiting");
+            }
+            shutdown = true;
+        }
+        if (shutdown) {
+            pthread_mutex_unlock(&in->lock);
+            break;
+        }
+        if (!pcm) {
+            pcm = pcm_open(PCM_CARD, PCM_DEVICE,
+                          PCM_IN | PCM_MONOTONIC, &in->pcm_config);
+            if (!pcm_is_ready(pcm)) {
+                ALOGE("pcm_open(in) failed: %s: channels %d format %d rate %d",
+                  pcm_get_error(pcm),
+                  in->pcm_config.channels,
+                  in->pcm_config.format,
+                  in->pcm_config.rate
+                   );
+                pthread_mutex_unlock(&in->lock);
+                break;
+            }
+            buffer_frames = in->pcm_config.period_size;
+            buffer_size = pcm_frames_to_bytes(pcm, buffer_frames);
+            buffer = malloc(buffer_size);
+            if (!buffer) {
+                ALOGE("could not allocate worker read buffer");
+                pthread_mutex_unlock(&in->lock);
+                break;
+            }
+        }
+        pthread_mutex_unlock(&in->lock);
+        int ret = pcm_read(pcm, buffer, pcm_frames_to_bytes(pcm, buffer_frames));
+        if (ret != 0) {
+            ALOGW("pcm_read failed %s", pcm_get_error(pcm));
+            restart = true;
+            continue;
+        }
+
+        pthread_mutex_lock(&in->lock);
+        size_t frames_written = audio_vbuffer_write(&in->buffer, buffer, buffer_frames);
+        pthread_mutex_unlock(&in->lock);
+
+        if (frames_written != buffer_frames) {
+            ALOGW("in_read_worker only could write %zu / %zu frames", frames_written, buffer_frames);
+        }
+    }
+    if (buffer) {
+        free(buffer);
+    }
+    return NULL;
+}
+
+static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
+                       size_t bytes)
+{
+    struct generic_stream_in *in = (struct generic_stream_in *)stream;
+    struct generic_audio_device *adev = in->dev;
+    const size_t frames =  bytes / audio_stream_in_frame_size(stream);
+    bool mic_mute = false;
+    size_t read_bytes = 0;
+
+    adev_get_mic_mute(&adev->device, &mic_mute);
+    pthread_mutex_lock(&in->lock);
+
+    if (in->worker_standby) {
+        in->worker_standby = false;
+    }
+    pthread_cond_signal(&in->worker_wake);
+
+    int64_t current_position;
+    struct timespec current_time;
+
+    get_current_input_position(in, &current_position, &current_time);
+    if (in->standby) {
+        in->standby = false;
+        in->standby_exit_time = current_time;
+        in->standby_frames_read = 0;
+    }
+
+    const int64_t frames_available = current_position - in->standby_position - in->standby_frames_read;
+    assert(frames_available >= 0);
+
+    const size_t frames_wait = ((uint64_t)frames_available > frames) ? 0 : frames - frames_available;
+
+    int64_t sleep_time_us  = frames_wait * 1000000LL /
+                             in_get_sample_rate(&stream->common);
+
+    pthread_mutex_unlock(&in->lock);
+
+    if (sleep_time_us > 0) {
+        usleep(sleep_time_us);
+    }
+
+    pthread_mutex_lock(&in->lock);
+    int read_frames = 0;
+    if (in->standby) {
+        ALOGW("Input put to sleep while read in progress");
+        goto exit;
+    }
+    in->standby_frames_read += frames;
+
+    if (popcount(in->req_config.channel_mask) == 1 &&
+        in->pcm_config.channels == 2) {
+        // Need to resample to mono
+        if (in->stereo_to_mono_buf_size < bytes*2) {
+            in->stereo_to_mono_buf = realloc(in->stereo_to_mono_buf,
+                                             bytes*2);
+            if (!in->stereo_to_mono_buf) {
+                ALOGE("Failed to allocate stereo_to_mono_buff");
+                goto exit;
+            }
+        }
+
+        read_frames = audio_vbuffer_read(&in->buffer, in->stereo_to_mono_buf, frames);
+
+        // Currently only pcm 16 is supported.
+        uint16_t *src = (uint16_t *)in->stereo_to_mono_buf;
+        uint16_t *dst = (uint16_t *)buffer;
+        size_t i;
+        // Resample stereo 16 to mono 16 by dropping one channel.
+        // The stereo stream is interleaved L-R-L-R
+        for (i = 0; i < frames; i++) {
+            *dst = *src;
+            src += 2;
+            dst += 1;
+        }
+    } else {
+        read_frames = audio_vbuffer_read(&in->buffer, buffer, frames);
+    }
+
+exit:
+    read_bytes = read_frames*audio_stream_in_frame_size(stream);
+
+    if (mic_mute) {
+        read_bytes = 0;
+    }
+
+    if (read_bytes < bytes) {
+        memset (&((uint8_t *)buffer)[read_bytes], 0, bytes-read_bytes);
+    }
+
+    pthread_mutex_unlock(&in->lock);
+
+    return bytes;
+}
+
+static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
+{
+    return 0;
+}
+
+static int in_get_capture_position(const struct audio_stream_in *stream,
+                                int64_t *frames, int64_t *time)
+{
+    struct generic_stream_in *in = (struct generic_stream_in *)stream;
+    pthread_mutex_lock(&in->lock);
+    struct timespec current_time;
+    get_current_input_position(in, frames, &current_time);
+    *time = (current_time.tv_sec * 1000000000LL + current_time.tv_nsec);
+    pthread_mutex_unlock(&in->lock);
+    return 0;
+}
+
+static int in_get_active_microphones(const struct audio_stream_in *stream,
+                                     struct audio_microphone_characteristic_t *mic_array,
+                                     size_t *mic_count)
+{
+    return adev_get_microphones(NULL, mic_array, mic_count);
+}
+
+static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+    // in_add_audio_effect is a no op
+    return 0;
+}
+
+static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+    // in_add_audio_effect is a no op
+    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,
+                                   const char *address __unused)
+{
+    struct generic_audio_device *adev = (struct generic_audio_device *)dev;
+    struct generic_stream_out *out;
+    int ret = 0;
+
+    if (refine_output_parameters(&config->sample_rate, &config->format, &config->channel_mask)) {
+        ALOGE("Error opening output stream format %d, channel_mask %04x, sample_rate %u",
+              config->format, config->channel_mask, config->sample_rate);
+        ret = -EINVAL;
+        goto error;
+    }
+
+    out = (struct generic_stream_out *)calloc(1, sizeof(struct generic_stream_out));
+
+    if (!out)
+        return -ENOMEM;
+
+    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_presentation_position = out_get_presentation_position;
+    out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
+
+    pthread_mutex_init(&out->lock, (const pthread_mutexattr_t *) NULL);
+    out->dev = adev;
+    out->device = devices;
+    memcpy(&out->req_config, config, sizeof(struct audio_config));
+    memcpy(&out->pcm_config, &pcm_config_out, sizeof(struct pcm_config));
+    out->pcm_config.rate = config->sample_rate;
+    out->pcm_config.period_size = out->pcm_config.rate*OUT_PERIOD_MS/1000;
+
+    out->standby = true;
+    out->underrun_position = 0;
+    out->underrun_time.tv_sec = 0;
+    out->underrun_time.tv_nsec = 0;
+    out->last_write_time_us = 0;
+    out->frames_total_buffered = 0;
+    out->frames_written = 0;
+    out->frames_rendered = 0;
+
+    ret = audio_vbuffer_init(&out->buffer,
+                      out->pcm_config.period_size*out->pcm_config.period_count,
+                      out->pcm_config.channels *
+                      pcm_format_to_bits(out->pcm_config.format) >> 3);
+    if (ret == 0) {
+        pthread_cond_init(&out->worker_wake, NULL);
+        out->worker_standby = true;
+        out->worker_exit = false;
+        pthread_create(&out->worker_thread, NULL, out_write_worker, out);
+
+    }
+    *stream_out = &out->stream;
+
+
+error:
+
+    return ret;
+}
+
+static void adev_close_output_stream(struct audio_hw_device *dev,
+                                     struct audio_stream_out *stream)
+{
+    struct generic_stream_out *out = (struct generic_stream_out *)stream;
+    pthread_mutex_lock(&out->lock);
+    do_out_standby(out);
+
+    out->worker_exit = true;
+    pthread_cond_signal(&out->worker_wake);
+    pthread_mutex_unlock(&out->lock);
+
+    pthread_join(out->worker_thread, NULL);
+    pthread_mutex_destroy(&out->lock);
+    audio_vbuffer_destroy(&out->buffer);
+    free(stream);
+}
+
+static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
+{
+    return 0;
+}
+
+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)
+{
+    return 0;
+}
+
+static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
+{
+    // adev_set_voice_volume is a no op (simulates phones)
+    return 0;
+}
+
+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)
+{
+    // adev_set_mode is a no op (simulates phones)
+    return 0;
+}
+
+static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
+{
+    struct generic_audio_device *adev = (struct generic_audio_device *)dev;
+    pthread_mutex_lock(&adev->lock);
+    adev->mic_mute = state;
+    pthread_mutex_unlock(&adev->lock);
+    return 0;
+}
+
+static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
+{
+    struct generic_audio_device *adev = (struct generic_audio_device *)dev;
+    pthread_mutex_lock(&adev->lock);
+    *state = adev->mic_mute;
+    pthread_mutex_unlock(&adev->lock);
+    return 0;
+}
+
+
+static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
+                                         const struct audio_config *config)
+{
+    return get_input_buffer_size(config->sample_rate, config->format, config->channel_mask);
+}
+
+
+static void adev_close_input_stream(struct audio_hw_device *dev,
+                                   struct audio_stream_in *stream)
+{
+    struct generic_stream_in *in = (struct generic_stream_in *)stream;
+    pthread_mutex_lock(&in->lock);
+    do_in_standby(in);
+
+    in->worker_exit = true;
+    pthread_cond_signal(&in->worker_wake);
+    pthread_mutex_unlock(&in->lock);
+    pthread_join(in->worker_thread, NULL);
+
+    if (in->stereo_to_mono_buf != NULL) {
+        free(in->stereo_to_mono_buf);
+        in->stereo_to_mono_buf_size = 0;
+    }
+
+    pthread_mutex_destroy(&in->lock);
+    audio_vbuffer_destroy(&in->buffer);
+    free(stream);
+}
+
+
+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,
+                                  audio_input_flags_t flags __unused,
+                                  const char *address __unused,
+                                  audio_source_t source __unused)
+{
+    struct generic_audio_device *adev = (struct generic_audio_device *)dev;
+    struct generic_stream_in *in;
+    int ret = 0;
+    if (refine_input_parameters(&config->sample_rate, &config->format, &config->channel_mask)) {
+        ALOGE("Error opening input stream format %d, channel_mask %04x, sample_rate %u",
+              config->format, config->channel_mask, config->sample_rate);
+        ret = -EINVAL;
+        goto error;
+    }
+
+    in = (struct generic_stream_in *)calloc(1, sizeof(struct generic_stream_in));
+    if (!in) {
+        ret = -ENOMEM;
+        goto error;
+    }
+
+    in->stream.common.get_sample_rate = in_get_sample_rate;
+    in->stream.common.set_sample_rate = in_set_sample_rate;         // no op
+    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;                   // no op
+    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;       // no op
+    in->stream.common.remove_audio_effect = in_remove_audio_effect; // no op
+    in->stream.set_gain = in_set_gain;                              // no op
+    in->stream.read = in_read;
+    in->stream.get_input_frames_lost = in_get_input_frames_lost;    // no op
+    in->stream.get_capture_position = in_get_capture_position;
+    in->stream.get_active_microphones = in_get_active_microphones;
+
+    pthread_mutex_init(&in->lock, (const pthread_mutexattr_t *) NULL);
+    in->dev = adev;
+    in->device = devices;
+    memcpy(&in->req_config, config, sizeof(struct audio_config));
+    memcpy(&in->pcm_config, &pcm_config_in, sizeof(struct pcm_config));
+    in->pcm_config.rate = config->sample_rate;
+    in->pcm_config.period_size = in->pcm_config.rate*IN_PERIOD_MS/1000;
+
+    in->stereo_to_mono_buf = NULL;
+    in->stereo_to_mono_buf_size = 0;
+
+    in->standby = true;
+    in->standby_position = 0;
+    in->standby_exit_time.tv_sec = 0;
+    in->standby_exit_time.tv_nsec = 0;
+    in->standby_frames_read = 0;
+
+    ret = audio_vbuffer_init(&in->buffer,
+                      in->pcm_config.period_size*in->pcm_config.period_count,
+                      in->pcm_config.channels *
+                      pcm_format_to_bits(in->pcm_config.format) >> 3);
+    if (ret == 0) {
+        pthread_cond_init(&in->worker_wake, NULL);
+        in->worker_standby = true;
+        in->worker_exit = false;
+        pthread_create(&in->worker_thread, NULL, in_read_worker, in);
+    }
+
+    *stream_in = &in->stream;
+
+error:
+    return ret;
+}
+
+
+static int adev_dump(const audio_hw_device_t *dev, int fd)
+{
+    return 0;
+}
+
+static int adev_get_microphones(const audio_hw_device_t *dev,
+                                struct audio_microphone_characteristic_t *mic_array,
+                                size_t *mic_count)
+{
+    if (mic_count == NULL) {
+        return -ENOSYS;
+    }
+
+    if (*mic_count == 0) {
+        *mic_count = 1;
+        return 0;
+    }
+
+    if (mic_array == NULL) {
+        return -ENOSYS;
+    }
+
+    strncpy(mic_array->device_id, "mic_default", AUDIO_MICROPHONE_ID_MAX_LEN - 1);
+    mic_array->device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+    strncpy(mic_array->address, AUDIO_BOTTOM_MICROPHONE_ADDRESS,
+            AUDIO_DEVICE_MAX_ADDRESS_LEN - 1);
+    memset(mic_array->channel_mapping, AUDIO_MICROPHONE_CHANNEL_MAPPING_UNUSED,
+           sizeof(mic_array->channel_mapping));
+    mic_array->location = AUDIO_MICROPHONE_LOCATION_UNKNOWN;
+    mic_array->group = 0;
+    mic_array->index_in_the_group = 0;
+    mic_array->sensitivity = AUDIO_MICROPHONE_SENSITIVITY_UNKNOWN;
+    mic_array->max_spl = AUDIO_MICROPHONE_SPL_UNKNOWN;
+    mic_array->min_spl = AUDIO_MICROPHONE_SPL_UNKNOWN;
+    mic_array->directionality = AUDIO_MICROPHONE_DIRECTIONALITY_UNKNOWN;
+    mic_array->num_frequency_responses = 0;
+    mic_array->geometric_location.x = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
+    mic_array->geometric_location.y = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
+    mic_array->geometric_location.z = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
+    mic_array->orientation.x = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
+    mic_array->orientation.y = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
+    mic_array->orientation.z = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
+
+    *mic_count = 1;
+    return 0;
+}
+
+static int adev_close(hw_device_t *dev)
+{
+    struct generic_audio_device *adev = (struct generic_audio_device *)dev;
+    int ret = 0;
+    if (!adev)
+        return 0;
+
+    pthread_mutex_lock(&adev_init_lock);
+
+    if (audio_device_ref_count == 0) {
+        ALOGE("adev_close called when ref_count 0");
+        ret = -EINVAL;
+        goto error;
+    }
+
+    if ((--audio_device_ref_count) == 0) {
+        if (adev->mixer) {
+            mixer_close(adev->mixer);
+        }
+        free(adev);
+    }
+
+error:
+    pthread_mutex_unlock(&adev_init_lock);
+    return ret;
+}
+
+static int adev_open(const hw_module_t* module, const char* name,
+                     hw_device_t** device)
+{
+    static struct generic_audio_device *adev;
+
+    if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
+        return -EINVAL;
+
+    pthread_once(&sFallbackOnce, fallback_init);
+    if (sFallback != NULL) {
+        return sFallback->common.methods->open(&sFallback->common, name, device);
+    }
+
+    pthread_mutex_lock(&adev_init_lock);
+    if (audio_device_ref_count != 0) {
+        *device = &adev->device.common;
+        audio_device_ref_count++;
+        ALOGV("%s: returning existing instance of adev", __func__);
+        ALOGV("%s: exit", __func__);
+        goto unlock;
+    }
+    adev = calloc(1, sizeof(struct generic_audio_device));
+
+    pthread_mutex_init(&adev->lock, (const pthread_mutexattr_t *) NULL);
+
+    adev->device.common.tag = HARDWARE_DEVICE_TAG;
+    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.init_check = adev_init_check;               // no op
+    adev->device.set_voice_volume = adev_set_voice_volume;   // no op
+    adev->device.set_master_volume = adev_set_master_volume; // no op
+    adev->device.get_master_volume = adev_get_master_volume; // no op
+    adev->device.set_master_mute = adev_set_master_mute;     // no op
+    adev->device.get_master_mute = adev_get_master_mute;     // no op
+    adev->device.set_mode = adev_set_mode;                   // no op
+    adev->device.set_mic_mute = adev_set_mic_mute;
+    adev->device.get_mic_mute = adev_get_mic_mute;
+    adev->device.set_parameters = adev_set_parameters;       // no op
+    adev->device.get_parameters = adev_get_parameters;       // no op
+    adev->device.get_input_buffer_size = adev_get_input_buffer_size;
+    adev->device.open_output_stream = adev_open_output_stream;
+    adev->device.close_output_stream = adev_close_output_stream;
+    adev->device.open_input_stream = adev_open_input_stream;
+    adev->device.close_input_stream = adev_close_input_stream;
+    adev->device.dump = adev_dump;
+    adev->device.get_microphones = adev_get_microphones;
+
+    *device = &adev->device.common;
+
+    adev->mixer = mixer_open(PCM_CARD);
+    struct mixer_ctl *ctl;
+
+    // Set default mixer ctls
+    // Enable channels and set volume
+    for (int i = 0; i < (int)mixer_get_num_ctls(adev->mixer); i++) {
+        ctl = mixer_get_ctl(adev->mixer, i);
+        ALOGD("mixer %d name %s", i, mixer_ctl_get_name(ctl));
+        if (!strcmp(mixer_ctl_get_name(ctl), "Master Playback Volume") ||
+            !strcmp(mixer_ctl_get_name(ctl), "Capture Volume")) {
+            for (int z = 0; z < (int)mixer_ctl_get_num_values(ctl); z++) {
+                ALOGD("set ctl %d to %d", z, 100);
+                mixer_ctl_set_percent(ctl, z, 100);
+            }
+            continue;
+        }
+        if (!strcmp(mixer_ctl_get_name(ctl), "Master Playback Switch") ||
+            !strcmp(mixer_ctl_get_name(ctl), "Capture Switch")) {
+            for (int z = 0; z < (int)mixer_ctl_get_num_values(ctl); z++) {
+                ALOGD("set ctl %d to %d", z, 1);
+                mixer_ctl_set_value(ctl, z, 1);
+            }
+            continue;
+        }
+    }
+
+    audio_device_ref_count++;
+
+unlock:
+    pthread_mutex_unlock(&adev_init_lock);
+    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 = "Generic audio HW HAL",
+        .author = "The Android Open Source Project",
+        .methods = &hal_module_methods,
+    },
+};
+
+/* This function detects whether or not we should be using an alsa audio device
+ * or fall back to the legacy default_audio driver.
+ */
+static void
+fallback_init(void)
+{
+    void* module;
+
+    FILE *fptr = fopen ("/proc/asound/pcm", "r");
+    if (fptr != NULL) {
+      // asound/pcm is empty if there are no devices
+      int c = fgetc(fptr);
+      fclose(fptr);
+      if (c != EOF) {
+          ALOGD("Emulator host-side ALSA audio emulation detected.");
+          return;
+      }
+    }
+
+    ALOGD("Emulator without host-side ALSA audio emulation detected.");
+#if __LP64__
+    module = dlopen("/vendor/lib64/hw/audio.primary.i9305_legacy.so",
+                    RTLD_LAZY|RTLD_LOCAL);
+#else
+    module = dlopen("/vendor/lib/hw/audio.primary.i9305_legacy.so",
+                    RTLD_LAZY|RTLD_LOCAL);
+#endif
+    if (module != NULL) {
+        sFallback = (struct audio_module *)(dlsym(module, HAL_MODULE_INFO_SYM_AS_STR));
+        if (sFallback == NULL) {
+            dlclose(module);
+        }
+    }
+    if (sFallback == NULL) {
+        ALOGE("Could not find legacy fallback module!?");
+    }
+}
diff --git a/audio/audio_hw_legacy.c b/audio/audio_hw_legacy.c
new file mode 100644
index 0000000..3d6a253
--- /dev/null
+++ b/audio/audio_hw_legacy.c
@@ -0,0 +1,714 @@
+/*
+ * 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 "audio_hw_generic"
+/*#define LOG_NDEBUG 0*/
+
+#include <errno.h>
+#include <pthread.h>
+#include <stdint.h>
+#include <stdlib.h>
+#include <sys/time.h>
+#include <fcntl.h>
+#include <unistd.h>
+
+#include <log/log.h>
+#include <cutils/str_parms.h>
+
+#include <hardware/hardware.h>
+#include <system/audio.h>
+#include <hardware/audio.h>
+
+
+#define AUDIO_DEVICE_NAME "/dev/eac"
+#define OUT_BUFFER_SIZE 4096
+#define OUT_LATENCY_MS 20
+#define IN_SAMPLING_RATE 8000
+#define IN_BUFFER_SIZE 320
+
+
+struct generic_audio_device {
+    struct audio_hw_device device;
+    pthread_mutex_t lock;
+    struct audio_stream_out *output;
+    struct audio_stream_in *input;
+    int fd;
+    bool mic_mute;
+};
+
+
+struct generic_stream_out {
+    struct audio_stream_out stream;
+    struct generic_audio_device *dev;
+    audio_devices_t device;
+    uint32_t sample_rate;
+};
+
+struct generic_stream_in {
+    struct audio_stream_in stream;
+    struct generic_audio_device *dev;
+    audio_devices_t device;
+};
+
+
+static uint32_t out_get_sample_rate(const struct audio_stream *stream)
+{
+    struct generic_stream_out *out = (struct generic_stream_out *)stream;
+    return out->sample_rate;
+}
+
+static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
+{
+    return -ENOSYS;
+}
+
+static size_t out_get_buffer_size(const struct audio_stream *stream)
+{
+    return OUT_BUFFER_SIZE;
+}
+
+static audio_channel_mask_t out_get_channels(const struct audio_stream *stream)
+{
+    return AUDIO_CHANNEL_OUT_STEREO;
+}
+
+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)
+{
+    return -ENOSYS;
+}
+
+static int out_standby(struct audio_stream *stream)
+{
+    // out_standby is a no op
+    return 0;
+}
+
+static int out_dump(const struct audio_stream *stream, int fd)
+{
+    struct generic_stream_out *out = (struct generic_stream_out *)stream;
+
+    dprintf(fd, "\tout_dump:\n"
+                "\t\tsample rate: %u\n"
+                "\t\tbuffer size: %zu\n"
+                "\t\tchannel mask: %08x\n"
+                "\t\tformat: %d\n"
+                "\t\tdevice: %08x\n"
+                "\t\taudio dev: %p\n\n",
+                out_get_sample_rate(stream),
+                out_get_buffer_size(stream),
+                out_get_channels(stream),
+                out_get_format(stream),
+                out->device,
+                out->dev);
+
+    return 0;
+}
+
+static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+    struct generic_stream_out *out = (struct generic_stream_out *)stream;
+    struct str_parms *parms;
+    char value[32];
+    int ret;
+    long val;
+    char *end;
+
+    parms = str_parms_create_str(kvpairs);
+
+    ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING,
+                            value, sizeof(value));
+    if (ret >= 0) {
+        errno = 0;
+        val = strtol(value, &end, 10);
+        if (errno == 0 && (end != NULL) && (*end == '\0') && ((int)val == val)) {
+            out->device = (int)val;
+        } else {
+            ret = -EINVAL;
+        }
+    }
+
+    str_parms_destroy(parms);
+    return ret;
+}
+
+static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
+{
+    struct generic_stream_out *out = (struct generic_stream_out *)stream;
+    struct str_parms *query = str_parms_create_str(keys);
+    char *str;
+    char value[256];
+    struct str_parms *reply = str_parms_create();
+    int ret;
+
+    ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
+    if (ret >= 0) {
+        str_parms_add_int(reply, AUDIO_PARAMETER_STREAM_ROUTING, out->device);
+        str = strdup(str_parms_to_str(reply));
+    } else {
+        str = strdup(keys);
+    }
+
+    str_parms_destroy(query);
+    str_parms_destroy(reply);
+    return str;
+}
+
+static uint32_t out_get_latency(const struct audio_stream_out *stream)
+{
+    return OUT_LATENCY_MS;
+}
+
+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)
+{
+    struct generic_stream_out *out = (struct generic_stream_out *)stream;
+    struct generic_audio_device *adev = out->dev;
+
+    pthread_mutex_lock(&adev->lock);
+    if (adev->fd >= 0)
+        bytes = write(adev->fd, buffer, bytes);
+    pthread_mutex_unlock(&adev->lock);
+
+    return bytes;
+}
+
+static int out_get_render_position(const struct audio_stream_out *stream,
+                                   uint32_t *dsp_frames)
+{
+    return -ENOSYS;
+}
+
+static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+    // out_add_audio_effect is a no op
+    return 0;
+}
+
+static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+    // out_remove_audio_effect is a no op
+    return 0;
+}
+
+static int out_get_next_write_timestamp(const struct audio_stream_out *stream,
+                                        int64_t *timestamp)
+{
+    return -ENOSYS;
+}
+
+/** audio_stream_in implementation **/
+static uint32_t in_get_sample_rate(const struct audio_stream *stream)
+{
+    return IN_SAMPLING_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)
+{
+    return IN_BUFFER_SIZE;
+}
+
+static audio_channel_mask_t in_get_channels(const struct audio_stream *stream)
+{
+    return AUDIO_CHANNEL_IN_MONO;
+}
+
+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)
+{
+    return -ENOSYS;
+}
+
+static int in_standby(struct audio_stream *stream)
+{
+    // in_standby is a no op
+    return 0;
+}
+
+static int in_dump(const struct audio_stream *stream, int fd)
+{
+    struct generic_stream_in *in = (struct generic_stream_in *)stream;
+
+    dprintf(fd, "\tin_dump:\n"
+                "\t\tsample rate: %u\n"
+                "\t\tbuffer size: %zu\n"
+                "\t\tchannel mask: %08x\n"
+                "\t\tformat: %d\n"
+                "\t\tdevice: %08x\n"
+                "\t\taudio dev: %p\n\n",
+                in_get_sample_rate(stream),
+                in_get_buffer_size(stream),
+                in_get_channels(stream),
+                in_get_format(stream),
+                in->device,
+                in->dev);
+
+    return 0;
+}
+
+static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+    struct generic_stream_in *in = (struct generic_stream_in *)stream;
+    struct str_parms *parms;
+    char value[32];
+    int ret;
+    long val;
+    char *end;
+
+    parms = str_parms_create_str(kvpairs);
+
+    ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING,
+                            value, sizeof(value));
+    if (ret >= 0) {
+        errno = 0;
+        val = strtol(value, &end, 10);
+        if ((errno == 0) && (end != NULL) && (*end == '\0') && ((int)val == val)) {
+            in->device = (int)val;
+        } else {
+            ret = -EINVAL;
+        }
+    }
+
+    str_parms_destroy(parms);
+    return ret;
+}
+
+static char * in_get_parameters(const struct audio_stream *stream,
+                                const char *keys)
+{
+    struct generic_stream_in *in = (struct generic_stream_in *)stream;
+    struct str_parms *query = str_parms_create_str(keys);
+    char *str;
+    char value[256];
+    struct str_parms *reply = str_parms_create();
+    int ret;
+
+    ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
+    if (ret >= 0) {
+        str_parms_add_int(reply, AUDIO_PARAMETER_STREAM_ROUTING, in->device);
+        str = strdup(str_parms_to_str(reply));
+    } else {
+        str = strdup(keys);
+    }
+
+    str_parms_destroy(query);
+    str_parms_destroy(reply);
+    return str;
+}
+
+static int in_set_gain(struct audio_stream_in *stream, float gain)
+{
+    // in_set_gain is a no op
+    return 0;
+}
+
+static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
+                       size_t bytes)
+{
+    struct generic_stream_in *in = (struct generic_stream_in *)stream;
+    struct generic_audio_device *adev = in->dev;
+
+    pthread_mutex_lock(&adev->lock);
+    if (adev->fd >= 0)
+        bytes = read(adev->fd, buffer, bytes);
+    if (adev->mic_mute && (bytes > 0)) {
+        memset(buffer, 0, bytes);
+    }
+    pthread_mutex_unlock(&adev->lock);
+
+    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)
+{
+    // in_add_audio_effect is a no op
+    return 0;
+}
+
+static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+    // in_add_audio_effect is a no op
+    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,
+                                   const char *address __unused)
+{
+    struct generic_audio_device *adev = (struct generic_audio_device *)dev;
+    struct generic_stream_out *out;
+    static const uint32_t sample_rates [] = { 44100, 48000 };
+    static const int sample_rates_count = sizeof(sample_rates)/sizeof(sample_rates[0]);
+    int ret = 0;
+
+    pthread_mutex_lock(&adev->lock);
+    if (adev->output != NULL) {
+        ret = -ENOSYS;
+        goto error;
+    }
+
+    if ((config->format != AUDIO_FORMAT_PCM_16_BIT) ||
+        (config->channel_mask != AUDIO_CHANNEL_OUT_STEREO) ) {
+        ALOGE("Error opening output stream, format %d, channel_mask %04x",
+              config->format, config->channel_mask);
+        config->format = AUDIO_FORMAT_PCM_16_BIT;
+        config->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+        ret = -EINVAL;
+    }
+
+    for (int idx = 0; idx < sample_rates_count; idx++) {
+        if (config->sample_rate < sample_rates[idx]) {
+            config->sample_rate = sample_rates[idx];
+            ALOGE("Error opening output stream, sample_rate %u", config->sample_rate);
+            ret = -EINVAL;
+            break;
+        } else if (config->sample_rate == sample_rates[idx]) {
+            break;
+        } else if (idx == sample_rates_count-1) {
+            // Cap it to the highest rate we support
+            config->sample_rate = sample_rates[idx];
+            ALOGE("Error opening output stream, sample_rate %u", config->sample_rate);
+            ret = -EINVAL;
+        }
+    }
+
+    if (ret != 0) goto error;
+
+    out = (struct generic_stream_out *)calloc(1, sizeof(struct generic_stream_out));
+
+    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;
+    out->sample_rate = config->sample_rate;
+
+    out->dev = adev;
+    out->device = devices;
+    adev->output = (struct audio_stream_out *)out;
+    *stream_out = &out->stream;
+
+error:
+    pthread_mutex_unlock(&adev->lock);
+
+    return ret;
+}
+
+static void adev_close_output_stream(struct audio_hw_device *dev,
+                                     struct audio_stream_out *stream)
+{
+    struct generic_audio_device *adev = (struct generic_audio_device *)dev;
+
+    pthread_mutex_lock(&adev->lock);
+    if (stream == adev->output) {
+        free(stream);
+        adev->output = NULL;
+    }
+    pthread_mutex_unlock(&adev->lock);
+}
+
+static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
+{
+    return 0;
+}
+
+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)
+{
+    struct generic_audio_device *adev = (struct generic_audio_device *)dev;
+
+    if (adev->fd >= 0)
+        return 0;
+
+    return -ENODEV;
+}
+
+static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
+{
+    // adev_set_voice_volume is a no op (simulates phones)
+    return 0;
+}
+
+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)
+{
+    // adev_set_mode is a no op (simulates phones)
+    return 0;
+}
+
+static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
+{
+    struct generic_audio_device *adev = (struct generic_audio_device *)dev;
+
+    pthread_mutex_lock(&adev->lock);
+    adev->mic_mute = state;
+    pthread_mutex_unlock(&adev->lock);
+    return 0;
+}
+
+static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
+{
+    struct generic_audio_device *adev = (struct generic_audio_device *)dev;
+
+    pthread_mutex_lock(&adev->lock);
+    *state = adev->mic_mute;
+    pthread_mutex_unlock(&adev->lock);
+
+    return 0;
+}
+
+static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
+                                         const struct audio_config *config)
+{
+    return IN_BUFFER_SIZE;
+}
+
+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,
+                                  audio_input_flags_t flags __unused,
+                                  const char *address __unused,
+                                  audio_source_t source __unused)
+{
+    struct generic_audio_device *adev = (struct generic_audio_device *)dev;
+    struct generic_stream_in *in;
+    int ret = 0;
+
+    pthread_mutex_lock(&adev->lock);
+    if (adev->input != NULL) {
+        ret = -ENOSYS;
+        goto error;
+    }
+
+    if ((config->format != AUDIO_FORMAT_PCM_16_BIT) ||
+        (config->channel_mask != AUDIO_CHANNEL_IN_MONO) ||
+        (config->sample_rate != IN_SAMPLING_RATE)) {
+        ALOGE("Error opening input stream format %d, channel_mask %04x, sample_rate %u",
+              config->format, config->channel_mask, config->sample_rate);
+        config->format = AUDIO_FORMAT_PCM_16_BIT;
+        config->channel_mask = AUDIO_CHANNEL_IN_MONO;
+        config->sample_rate = IN_SAMPLING_RATE;
+        ret = -EINVAL;
+        goto error;
+    }
+
+    in = (struct generic_stream_in *)calloc(1, sizeof(struct generic_stream_in));
+
+    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;
+
+    in->dev = adev;
+    in->device = devices;
+    adev->input = (struct audio_stream_in *)in;
+    *stream_in = &in->stream;
+
+error:
+    pthread_mutex_unlock(&adev->lock);
+
+    return ret;
+}
+
+static void adev_close_input_stream(struct audio_hw_device *dev,
+                                   struct audio_stream_in *stream)
+{
+    struct generic_audio_device *adev = (struct generic_audio_device *)dev;
+
+    pthread_mutex_lock(&adev->lock);
+    if (stream == adev->input) {
+        free(stream);
+        adev->input = NULL;
+    }
+    pthread_mutex_unlock(&adev->lock);
+}
+
+static int adev_dump(const audio_hw_device_t *dev, int fd)
+{
+    struct generic_audio_device *adev = (struct generic_audio_device *)dev;
+
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+
+    dprintf(fd, "\nadev_dump:\n"
+                "\tfd: %d\n"
+                "\tmic_mute: %s\n"
+                "\toutput: %p\n"
+                "\tinput: %p\n\n",
+                adev->fd,
+                adev->mic_mute ? "true": "false",
+                adev->output,
+                adev->input);
+
+    if (adev->output != NULL)
+        out_dump((const struct audio_stream *)adev->output, fd);
+    if (adev->input != NULL)
+        in_dump((const struct audio_stream *)adev->input, fd);
+
+    return 0;
+}
+
+static int adev_close(hw_device_t *dev)
+{
+    struct generic_audio_device *adev = (struct generic_audio_device *)dev;
+
+    adev_close_output_stream((struct audio_hw_device *)dev, adev->output);
+    adev_close_input_stream((struct audio_hw_device *)dev, adev->input);
+
+    if (adev->fd >= 0)
+        close(adev->fd);
+
+    free(dev);
+    return 0;
+}
+
+static int adev_open(const hw_module_t* module, const char* name,
+                     hw_device_t** device)
+{
+    struct generic_audio_device *adev;
+    int fd;
+
+    if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
+        return -EINVAL;
+
+    fd = open(AUDIO_DEVICE_NAME, O_RDWR);
+    if (fd < 0)
+        return -ENOSYS;
+
+    adev = calloc(1, sizeof(struct generic_audio_device));
+
+    adev->fd = fd;
+
+    adev->device.common.tag = HARDWARE_DEVICE_TAG;
+    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.init_check = adev_init_check;
+    adev->device.set_voice_volume = adev_set_voice_volume;
+    adev->device.set_master_volume = adev_set_master_volume;
+    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;
+    adev->device.set_mode = adev_set_mode;
+    adev->device.set_mic_mute = adev_set_mic_mute;
+    adev->device.get_mic_mute = adev_get_mic_mute;
+    adev->device.set_parameters = adev_set_parameters;
+    adev->device.get_parameters = adev_get_parameters;
+    adev->device.get_input_buffer_size = adev_get_input_buffer_size;
+    adev->device.open_output_stream = adev_open_output_stream;
+    adev->device.close_output_stream = adev_close_output_stream;
+    adev->device.open_input_stream = adev_open_input_stream;
+    adev->device.close_input_stream = adev_close_input_stream;
+    adev->device.dump = adev_dump;
+
+    *device = &adev->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 = "Generic audio HW HAL",
+        .author = "The Android Open Source Project",
+        .methods = &hal_module_methods,
+    },
+};
diff --git a/audio_policy.conf b/audio_policy.conf
new file mode 100644
index 0000000..0945c25
--- /dev/null
+++ b/audio_policy.conf
@@ -0,0 +1,64 @@
+#
+# Audio policy configuration for generic device builds (goldfish audio HAL - emulator)
+#
+
+# Global configuration section: lists input and output devices always present on the device
+# as well as the output device selected by default.
+# Devices are designated by a string that corresponds to the enum in audio.h
+
+global_configuration {
+  attached_output_devices AUDIO_DEVICE_OUT_SPEAKER
+  default_output_device AUDIO_DEVICE_OUT_SPEAKER
+  attached_input_devices AUDIO_DEVICE_IN_BUILTIN_MIC|AUDIO_DEVICE_IN_REMOTE_SUBMIX
+}
+
+# audio hardware module section: contains descriptors for all audio hw modules present on the
+# device. Each hw module node is named after the corresponding hw module library base name.
+# For instance, "primary" corresponds to audio.primary.<device>.so.
+# The "primary" module is mandatory and must include at least one output with
+# AUDIO_OUTPUT_FLAG_PRIMARY flag.
+# Each module descriptor contains one or more output profile descriptors and zero or more
+# input profile descriptors. Each profile lists all the parameters supported by a given output
+# or input stream category.
+# The "channel_masks", "formats", "devices" and "flags" are specified using strings corresponding
+# to enums in audio.h and audio_policy.h. They are concatenated by use of "|" without space or "\n".
+
+audio_hw_modules {
+  primary {
+    outputs {
+      primary {
+        sampling_rates 8000|11025|16000|22050|24000|44100|48000
+        channel_masks AUDIO_CHANNEL_OUT_MONO|AUDIO_CHANNEL_OUT_STEREO
+        formats AUDIO_FORMAT_PCM_16_BIT
+        devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_WIRED_HEADSET
+        flags AUDIO_OUTPUT_FLAG_PRIMARY
+      }
+    }
+    inputs {
+      primary {
+        sampling_rates 8000|11025|16000|22050|44100|48000
+        channel_masks AUDIO_CHANNEL_IN_MONO|AUDIO_CHANNEL_IN_STEREO
+        formats AUDIO_FORMAT_PCM_16_BIT
+        devices AUDIO_DEVICE_IN_BUILTIN_MIC|AUDIO_DEVICE_IN_WIRED_HEADSET
+      }
+    }
+  }
+  r_submix {
+    outputs {
+      submix {
+        sampling_rates 48000
+        channel_masks AUDIO_CHANNEL_OUT_STEREO
+        formats AUDIO_FORMAT_PCM_16_BIT
+        devices AUDIO_DEVICE_OUT_REMOTE_SUBMIX
+      }
+    }
+    inputs {
+      submix {
+        sampling_rates 48000
+        channel_masks AUDIO_CHANNEL_IN_STEREO
+        formats AUDIO_FORMAT_PCM_16_BIT
+        devices AUDIO_DEVICE_IN_REMOTE_SUBMIX
+      }
+    }
+  }
+}
diff --git a/device.mk b/device.mk
index 028854c..574cf21 100644
--- a/device.mk
+++ b/device.mk
@@ -30,15 +30,33 @@ PRODUCT_PACKAGES += \
     android.hardware.graphics.composer@2.1-service \
 
 # Audio
+PRODUCT_PACKAGES += audio.primary.i9305
+PRODUCT_PACKAGES += audio.primary.i9305_legacy
+PRODUCT_PACKAGES += android.hardware.audio@4.0-impl
+PRODUCT_PACKAGES += android.hardware.audio@4.0-service
+PRODUCT_PACKAGES += android.hardware.audio.effect@4.0-impl
+PRODUCT_PACKAGES += android.hardware.audio.effect@4.0-service
+
+# A2DP
 PRODUCT_PACKAGES += \
-    audio.primary.default \
+  audio.a2dp.default \
+  android.hardware.bluetooth.a2dp@1.0-impl \
+  android.hardware.bluetooth.a2dp@1.0-service \
 
-# Audio HAL packages
+# Sound trigger
 PRODUCT_PACKAGES += \
-    android.hardware.audio@2.0-impl \
-    android.hardware.audio@2.0-service \
-    android.hardware.audio.effect@2.0-impl \
-    android.hardware.soundtrigger@2.0-impl \
+    sound_trigger.stub.default \
+    android.hardware.soundtrigger@2.1-impl \
+    android.hardware.soundtrigger@2.1-service \
+
+PRODUCT_COPY_FILES += \
+    frameworks/av/media/libeffects/data/audio_effects.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_effects.xml \
+    device/samsung/i9305/audio_policy.conf:$(TARGET_COPY_OUT_VENDOR)/etc/audio_policy.conf \
+    frameworks/av/services/audiopolicy/config/audio_policy_configuration_generic.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_policy_configuration.xml \
+    frameworks/av/services/audiopolicy/config/primary_audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/primary_audio_policy_configuration.xml \
+    frameworks/av/services/audiopolicy/config/r_submix_audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/r_submix_audio_policy_configuration.xml \
+    frameworks/av/services/audiopolicy/config/audio_policy_volumes.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_policy_volumes.xml \
+    frameworks/av/services/audiopolicy/config/default_volume_tables.xml:$(TARGET_COPY_OUT_VENDOR)/etc/default_volume_tables.xml
 
 # DRM HAL packages
 PRODUCT_PACKAGES += \
diff --git a/init.smdk4x12.rc b/init.smdk4x12.rc
index 9fb48ea..2584a24 100644
--- a/init.smdk4x12.rc
+++ b/init.smdk4x12.rc
@@ -15,6 +15,9 @@
 #
 
 on boot
+    # Audio support
+    setprop ro.hardware.audio.primary i9305
+    # adb support
     mkdir /dev/usb-ffs 0770 shell shell
     mkdir /dev/usb-ffs/adb 0770 shell shell
     mount functionfs adb /dev/usb-ffs/adb uid=2000,gid=2000
diff --git a/manifest.xml b/manifest.xml
index bc12422..6998ba5 100644
--- a/manifest.xml
+++ b/manifest.xml
@@ -43,19 +43,19 @@
             <instance>default</instance>
         </interface>
     </hal>
-    <hal>
+    <hal format="hidl">
         <name>android.hardware.audio.effect</name>
         <transport>hwbinder</transport>
-        <version>2.0</version>
+        <version>4.0</version>
         <interface>
             <name>IEffectsFactory</name>
             <instance>default</instance>
         </interface>
     </hal>
-    <hal>
+    <hal format="hidl">
         <name>android.hardware.audio</name>
         <transport>hwbinder</transport>
-        <version>2.0</version>
+        <version>4.0</version>
         <interface>
             <name>IDevicesFactory</name>
             <instance>default</instance>