Primary audio HAL for FS1130

BUG: 26512598

Change-Id: I013302dbe24bb211324420e49b3a87b1fb3ddc74

5 files changed, 1332 insertions, 0 deletions

diff --git a/peripheral/audio/fs1130/Android.mk b/peripheral/audio/fs1130/Android.mk
new file mode 100644
index 0000000..fe1917d
--- /dev/null
+++ b/peripheral/audio/fs1130/Android.mk
@@ -0,0 +1,35 @@
+#
+# Copyright (C) 2016 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.
+
+ifneq ($(filter fs1130,$(TARGET_BOARD_PLATFORM)),)
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE := audio.primary.fs1130
+LOCAL_MODULE_RELATIVE_PATH := hw
+LOCAL_SRC_FILES := \
+    audio_hal.c
+LOCAL_C_INCLUDES += \
+    external/tinyalsa/include \
+    $(call include-path-for, audio-utils) \
+    $(call include-path-for, alsa-utils)
+LOCAL_SHARED_LIBRARIES := liblog libcutils libtinyalsa libaudioutils libalsautils
+LOCAL_MODULE_TAGS := optional
+LOCAL_CFLAGS := -Wno-unused-parameter
+
+include $(BUILD_SHARED_LIBRARY)
+endif
+
diff --git a/peripheral/audio/fs1130/audio_hal.c b/peripheral/audio/fs1130/audio_hal.c
new file mode 100644
index 0000000..e3b1223
--- /dev/null
+++ b/peripheral/audio/fs1130/audio_hal.c
@@ -0,0 +1,1105 @@
+/*
+ * Copyright 2016 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 "primary.audio"
+#define LOG_NDEBUG 0
+
+#include <errno.h>
+#include <inttypes.h>
+#include <pthread.h>
+#include <stdint.h>
+#include <stdlib.h>
+#include <sys/time.h>
+
+#include <log/log.h>
+#include <cutils/str_parms.h>
+#include <cutils/properties.h>
+
+#include <hardware/audio.h>
+#include <hardware/audio_alsaops.h>
+#include <hardware/hardware.h>
+
+#include <system/audio.h>
+
+#include <tinyalsa/asoundlib.h>
+
+#include <audio_utils/channels.h>
+
+#include "alsa_device_profile.h"
+#include "alsa_device_proxy.h"
+#include "alsa_logging.h"
+
+/* FOR TESTING:
+ * Set k_force_channels to force the number of channels to present to AudioFlinger.
+ *   0 disables (this is default: present the device channels to AudioFlinger).
+ *   2 forces to legacy stereo mode.
+ *
+ * Others values can be tried (up to 8).
+ * TODO: AudioFlinger cannot support more than 8 active output channels
+ * at this time, so limiting logic needs to be put here or communicated from above.
+ */
+static const unsigned k_force_channels = 0;
+
+#define DEFAULT_INPUT_BUFFER_SIZE_MS 20
+
+// stereo channel count
+#define FCC_2 2
+// fixed channel count of 8 limitation (for data processing in AudioFlinger)
+#define FCC_8 8
+
+struct audio_device {
+    struct audio_hw_device hw_device;
+
+    pthread_mutex_t lock; /* see note below on mutex acquisition order */
+
+    /* output */
+    alsa_device_profile out_profile;
+
+    /* input */
+    alsa_device_profile in_profile;
+
+    bool mic_muted;
+
+    bool standby;
+};
+
+struct stream_out {
+    struct audio_stream_out stream;
+
+    pthread_mutex_t lock;               /* see note below on mutex acquisition order */
+    pthread_mutex_t pre_lock;           /* acquire before lock to avoid DOS by playback thread */
+    bool standby;
+
+    struct audio_device *dev;           /* hardware information - only using this for the lock */
+
+    alsa_device_profile * profile;      /* Points to the alsa_device_profile in the audio_device */
+    alsa_device_proxy proxy;            /* state of the stream */
+
+    unsigned hal_channel_count;         /* channel count exposed to AudioFlinger.
+                                         * This may differ from the device channel count when
+                                         * the device is not compatible with AudioFlinger
+                                         * capabilities, e.g. exposes too many channels or
+                                         * too few channels. */
+    audio_channel_mask_t hal_channel_mask;   /* channel mask exposed to AudioFlinger. */
+
+    void * conversion_buffer;           /* any conversions are put into here
+                                         * they could come from here too if
+                                         * there was a previous conversion */
+    size_t conversion_buffer_size;      /* in bytes */
+};
+
+struct stream_in {
+    struct audio_stream_in stream;
+
+    pthread_mutex_t lock;               /* see note below on mutex acquisition order */
+    pthread_mutex_t pre_lock;           /* acquire before lock to avoid DOS by capture thread */
+    bool standby;
+
+    struct audio_device *dev;           /* hardware information - only using this for the lock */
+
+    alsa_device_profile * profile;      /* Points to the alsa_device_profile in the audio_device */
+    alsa_device_proxy proxy;            /* state of the stream */
+
+    unsigned hal_channel_count;         /* channel count exposed to AudioFlinger.
+                                         * This may differ from the device channel count when
+                                         * the device is not compatible with AudioFlinger
+                                         * capabilities, e.g. exposes too many channels or
+                                         * too few channels. */
+    audio_channel_mask_t hal_channel_mask;   /* channel mask exposed to AudioFlinger. */
+
+    /* We may need to read more data from the device in order to data reduce to 16bit, 4chan */
+    void * conversion_buffer;           /* any conversions are put into here
+                                         * they could come from here too if
+                                         * there was a previous conversion */
+    size_t conversion_buffer_size;      /* in bytes */
+};
+
+/*
+ * NOTE: when multiple mutexes have to be acquired, always take the
+ * stream_in or stream_out mutex first, followed by the audio_device mutex.
+ * stream pre_lock is always acquired before stream lock to prevent starvation of control thread by
+ * higher priority playback or capture thread.
+ */
+
+/*
+ * Extract the card and device numbers from the supplied key/value pairs.
+ *   kvpairs    A null-terminated string containing the key/value pairs or card and device.
+ *              i.e. "card=1;device=42"
+ *   card   A pointer to a variable to receive the parsed-out card number.
+ *   device A pointer to a variable to receive the parsed-out device number.
+ * NOTE: The variables pointed to by card and device return -1 (undefined) if the
+ *  associated key/value pair is not found in the provided string.
+ *  Return true if the kvpairs string contain a card/device spec, false otherwise.
+ */
+static bool parse_card_device_params(const char *kvpairs, int *card, int *device)
+{
+    struct str_parms * parms = str_parms_create_str(kvpairs);
+    char value[32];
+    int param_val;
+
+    // initialize to "undefined" state.
+    *card = -1;
+    *device = -1;
+
+    param_val = str_parms_get_str(parms, "card", value, sizeof(value));
+    if (param_val >= 0) {
+        *card = atoi(value);
+    }
+
+    param_val = str_parms_get_str(parms, "device", value, sizeof(value));
+    if (param_val >= 0) {
+        *device = atoi(value);
+    }
+
+    // FIXME: hardwire device if it is not found automatically
+    if (*card < 0 || *device < 0) {
+        if (strstr(kvpairs, "output_stream")) {
+            *card = 0;
+            *device = 2;
+        } else if (strstr(kvpairs, "input_stream")) {
+            *card = 0;
+            *device = 4;        /* ?? */
+        } else {
+            *card = 0;
+            *device = 2;
+        }
+    }
+
+    str_parms_destroy(parms);
+
+    ALOGV("parse_card_device_params kvpairs:\"%s\" => card:%d device:%d", kvpairs, *card, *device);
+    return *card >= 0 && *device >= 0;
+}
+
+static char * device_get_parameters(alsa_device_profile * profile, const char * keys)
+{
+    if (profile->card < 0 || profile->device < 0) {
+        char* no_result = strdup("");
+        ALOGV("device_get_parameters keys:\"%s\" => no_result:\"%s\"", keys, no_result);
+        return no_result;
+    }
+
+    struct str_parms *query = str_parms_create_str(keys);
+    struct str_parms *result = str_parms_create();
+
+    /* These keys are from hardware/libhardware/include/audio.h */
+    /* supported sample rates */
+    if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)) {
+        char* rates_list = profile_get_sample_rate_strs(profile);
+        str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES,
+                          rates_list);
+        free(rates_list);
+    }
+
+    /* supported channel counts */
+    if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS)) {
+        char* channels_list = profile_get_channel_count_strs(profile);
+        str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_CHANNELS,
+                          channels_list);
+        free(channels_list);
+    }
+
+    /* supported sample formats */
+    if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS)) {
+        char * format_params = profile_get_format_strs(profile);
+        str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_FORMATS,
+                          format_params);
+        free(format_params);
+    }
+    str_parms_destroy(query);
+
+    char* result_str = str_parms_to_str(result);
+    str_parms_destroy(result);
+
+    ALOGV("device_get_parameters keys:\"%s\" => \"%s\"", keys, result_str);
+
+    return result_str;
+}
+
+void lock_input_stream(struct stream_in *in)
+{
+    pthread_mutex_lock(&in->pre_lock);
+    pthread_mutex_lock(&in->lock);
+    pthread_mutex_unlock(&in->pre_lock);
+}
+
+void lock_output_stream(struct stream_out *out)
+{
+    pthread_mutex_lock(&out->pre_lock);
+    pthread_mutex_lock(&out->lock);
+    pthread_mutex_unlock(&out->pre_lock);
+}
+
+/*
+ * HAl Functions
+ */
+/**
+ * NOTE: when multiple mutexes have to be acquired, always respect the
+ * following order: hw device > out stream
+ */
+
+/*
+ * OUT functions
+ */
+static uint32_t out_get_sample_rate(const struct audio_stream *stream)
+{
+    uint32_t rate = proxy_get_sample_rate(&((struct stream_out*)stream)->proxy);
+    ALOGV("out_get_sample_rate => %d", rate);
+    return rate;
+}
+
+static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
+{
+    return 0;
+}
+
+static size_t out_get_buffer_size(const struct audio_stream *stream)
+{
+    const struct stream_out* out = (const struct stream_out*)stream;
+    size_t buffer_size =
+        proxy_get_period_size(&out->proxy) * audio_stream_out_frame_size(&(out->stream));
+    return buffer_size;
+}
+
+static uint32_t out_get_channels(const struct audio_stream *stream)
+{
+    const struct stream_out *out = (const struct stream_out*)stream;
+    return out->hal_channel_mask;
+}
+
+static audio_format_t out_get_format(const struct audio_stream *stream)
+{
+    /* Note: The HAL doesn't do any FORMAT conversion at this time. It
+     * Relies on the framework to provide data in the specified format.
+     * This could change in the future.
+     */
+    alsa_device_proxy * proxy = &((struct stream_out*)stream)->proxy;
+    audio_format_t format = audio_format_from_pcm_format(proxy_get_format(proxy));
+    return format;
+}
+
+static int out_set_format(struct audio_stream *stream, audio_format_t format)
+{
+    return 0;
+}
+
+static int out_standby(struct audio_stream *stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+
+    lock_output_stream(out);
+    if (!out->standby) {
+        pthread_mutex_lock(&out->dev->lock);
+        proxy_close(&out->proxy);
+        pthread_mutex_unlock(&out->dev->lock);
+        out->standby = true;
+    }
+    pthread_mutex_unlock(&out->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)
+{
+    ALOGV("out_set_parameters keys:%s", kvpairs);
+
+    struct stream_out *out = (struct stream_out *)stream;
+
+    int routing = 0;
+    int ret_value = 0;
+    int card = -1;
+    int device = -1;
+
+    if (!parse_card_device_params(kvpairs, &card, &device)) {
+        // nothing to do
+        return ret_value;
+    }
+
+    lock_output_stream(out);
+    /* Lock the device because that is where the profile lives */
+    pthread_mutex_lock(&out->dev->lock);
+
+    if (!profile_is_cached_for(out->profile, card, device)) {
+        /* cannot read pcm device info if playback is active */
+        if (!out->standby)
+            ret_value = -ENOSYS;
+        else {
+            int saved_card = out->profile->card;
+            int saved_device = out->profile->device;
+            out->profile->card = card;
+            out->profile->device = device;
+            ret_value = profile_read_device_info(out->profile) ? 0 : -EINVAL;
+            if (ret_value != 0) {
+                out->profile->card = saved_card;
+                out->profile->device = saved_device;
+            }
+        }
+    }
+
+    pthread_mutex_unlock(&out->dev->lock);
+    pthread_mutex_unlock(&out->lock);
+
+    return ret_value;
+}
+
+static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    lock_output_stream(out);
+    pthread_mutex_lock(&out->dev->lock);
+
+    char * params_str =  device_get_parameters(out->profile, keys);
+
+    pthread_mutex_unlock(&out->lock);
+    pthread_mutex_unlock(&out->dev->lock);
+
+    ALOGV("out_get_parameters keys:\"%s\" => param_str:\"%s\"", keys, params_str);
+    return params_str;
+}
+
+static uint32_t out_get_latency(const struct audio_stream_out *stream)
+{
+    alsa_device_proxy * proxy = &((struct stream_out*)stream)->proxy;
+    return proxy_get_latency(proxy);
+}
+
+static int out_set_volume(struct audio_stream_out *stream, float left, float right)
+{
+    return -ENOSYS;
+}
+
+/* must be called with hw device and output stream mutexes locked */
+static int start_output_stream(struct stream_out *out)
+{
+    ALOGV("start_output_stream(card:%d device:%d)", out->profile->card, out->profile->device);
+
+    return proxy_open(&out->proxy);
+}
+
+static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, size_t bytes)
+{
+    int ret;
+    struct stream_out *out = (struct stream_out *)stream;
+
+    lock_output_stream(out);
+    if (out->standby) {
+        pthread_mutex_lock(&out->dev->lock);
+        ret = start_output_stream(out);
+        pthread_mutex_unlock(&out->dev->lock);
+        if (ret != 0) {
+            goto err;
+        }
+        out->standby = false;
+    }
+
+    alsa_device_proxy* proxy = &out->proxy;
+    const void * write_buff = buffer;
+    int num_write_buff_bytes = bytes;
+    const int num_device_channels = proxy_get_channel_count(proxy); /* what we told alsa */
+    const int num_req_channels = out->hal_channel_count; /* what we told AudioFlinger */
+    if (num_device_channels != num_req_channels) {
+        /* allocate buffer */
+        const size_t required_conversion_buffer_size =
+                 bytes * num_device_channels / num_req_channels;
+        if (required_conversion_buffer_size > out->conversion_buffer_size) {
+            out->conversion_buffer_size = required_conversion_buffer_size;
+            out->conversion_buffer = realloc(out->conversion_buffer,
+                                             out->conversion_buffer_size);
+        }
+        /* convert data */
+        const audio_format_t audio_format = out_get_format(&(out->stream.common));
+        const unsigned sample_size_in_bytes = audio_bytes_per_sample(audio_format);
+        num_write_buff_bytes =
+                adjust_channels(write_buff, num_req_channels,
+                                out->conversion_buffer, num_device_channels,
+                                sample_size_in_bytes, num_write_buff_bytes);
+        write_buff = out->conversion_buffer;
+    }
+
+    if (write_buff != NULL && num_write_buff_bytes != 0) {
+        proxy_write(&out->proxy, write_buff, num_write_buff_bytes);
+    }
+
+    pthread_mutex_unlock(&out->lock);
+
+    return bytes;
+
+err:
+    pthread_mutex_unlock(&out->lock);
+    if (ret != 0) {
+        usleep(bytes * 1000000 / audio_stream_out_frame_size(stream) /
+               out_get_sample_rate(&stream->common));
+    }
+
+    return bytes;
+}
+
+static int out_get_render_position(const struct audio_stream_out *stream, uint32_t *dsp_frames)
+{
+    return -EINVAL;
+}
+
+static int out_get_presentation_position(const struct audio_stream_out *stream,
+                                         uint64_t *frames, struct timespec *timestamp)
+{
+    struct stream_out *out = (struct stream_out *)stream; // discard const qualifier
+    lock_output_stream(out);
+
+    const alsa_device_proxy *proxy = &out->proxy;
+    const int ret = proxy_get_presentation_position(proxy, frames, timestamp);
+
+    pthread_mutex_unlock(&out->lock);
+    // ALOGV("out_get_presentation_position => ret:%d frames:%llu", ret, (unsigned long long)*frames);
+    return ret;
+}
+
+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;
+}
+
+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*/)
+{
+    ALOGV("adev_open_output_stream() handle:0x%X, device:0x%X, flags:0x%X, addr:%s",
+          handle, devices, flags, address);
+
+    struct audio_device *adev = (struct audio_device *)dev;
+
+    struct stream_out *out;
+
+    out = (struct stream_out *)calloc(1, sizeof(struct stream_out));
+    if (!out)
+        return -ENOMEM;
+
+    /* setup function pointers */
+    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);
+    pthread_mutex_init(&out->pre_lock, (const pthread_mutexattr_t *) NULL);
+
+    out->dev = adev;
+    pthread_mutex_lock(&adev->lock);
+    out->profile = &adev->out_profile;
+
+    // build this to hand to the alsa_device_proxy
+    struct pcm_config proxy_config;
+    memset(&proxy_config, 0, sizeof(proxy_config));
+
+    /* Pull out the card/device pair */
+    parse_card_device_params(address, &(out->profile->card), &(out->profile->device));
+
+    profile_read_device_info(out->profile);
+
+    pthread_mutex_unlock(&adev->lock);
+
+    int ret = 0;
+
+    /* Rate */
+    if (config->sample_rate == 0) {
+        proxy_config.rate = config->sample_rate = profile_get_default_sample_rate(out->profile);
+    } else if (profile_is_sample_rate_valid(out->profile, config->sample_rate)) {
+        proxy_config.rate = config->sample_rate;
+    } else {
+        proxy_config.rate = config->sample_rate = profile_get_default_sample_rate(out->profile);
+        ret = -EINVAL;
+    }
+
+    /* Format */
+    if (config->format == AUDIO_FORMAT_DEFAULT) {
+        proxy_config.format = profile_get_default_format(out->profile);
+        config->format = audio_format_from_pcm_format(proxy_config.format);
+    } else {
+        enum pcm_format fmt = pcm_format_from_audio_format(config->format);
+        if (profile_is_format_valid(out->profile, fmt)) {
+            proxy_config.format = fmt;
+        } else {
+            proxy_config.format = profile_get_default_format(out->profile);
+            config->format = audio_format_from_pcm_format(proxy_config.format);
+            ret = -EINVAL;
+        }
+    }
+
+    /* Channels */
+    unsigned proposed_channel_count = 0;
+    if (k_force_channels) {
+        proposed_channel_count = k_force_channels;
+    } else if (config->channel_mask == AUDIO_CHANNEL_NONE) {
+        proposed_channel_count =  profile_get_default_channel_count(out->profile);
+    }
+    if (proposed_channel_count != 0) {
+        if (proposed_channel_count <= FCC_2) {
+            // use channel position mask for mono and stereo
+            config->channel_mask = audio_channel_out_mask_from_count(proposed_channel_count);
+        } else {
+            // use channel index mask for multichannel
+            config->channel_mask =
+                    audio_channel_mask_for_index_assignment_from_count(proposed_channel_count);
+        }
+        out->hal_channel_count = proposed_channel_count;
+    } else {
+        out->hal_channel_count = audio_channel_count_from_out_mask(config->channel_mask);
+    }
+    /* we can expose any channel mask, and emulate internally based on channel count. */
+    out->hal_channel_mask = config->channel_mask;
+
+    /* no validity checks are needed as proxy_prepare() forces channel_count to be valid.
+     * and we emulate any channel count discrepancies in out_write(). */
+    proxy_config.channels = proposed_channel_count;
+
+    proxy_prepare(&out->proxy, out->profile, &proxy_config);
+
+    /* TODO The retry mechanism isn't implemented in AudioPolicyManager/AudioFlinger. */
+    ret = 0;
+
+    out->conversion_buffer = NULL;
+    out->conversion_buffer_size = 0;
+
+    out->standby = true;
+
+    *stream_out = &out->stream;
+
+    return ret;
+
+err_open:
+    free(out);
+    *stream_out = NULL;
+    return -ENOSYS;
+}
+
+static void adev_close_output_stream(struct audio_hw_device *dev,
+                                     struct audio_stream_out *stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    ALOGV("adev_close_output_stream(c:%d d:%d)", out->profile->card, out->profile->device);
+
+    /* Close the pcm device */
+    out_standby(&stream->common);
+
+    free(out->conversion_buffer);
+
+    out->conversion_buffer = NULL;
+    out->conversion_buffer_size = 0;
+
+    free(stream);
+}
+
+static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
+                                         const struct audio_config *config)
+{
+    /* TODO This needs to be calculated based on format/channels/rate */
+    return 320;
+}
+
+/*
+ * IN functions
+ */
+static uint32_t in_get_sample_rate(const struct audio_stream *stream)
+{
+    uint32_t rate = proxy_get_sample_rate(&((const struct stream_in *)stream)->proxy);
+    ALOGV("in_get_sample_rate() = %d", rate);
+    return rate;
+}
+
+static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
+{
+    ALOGV("in_set_sample_rate(%d) - NOPE", rate);
+    return -ENOSYS;
+}
+
+static size_t in_get_buffer_size(const struct audio_stream *stream)
+{
+    const struct stream_in * in = ((const struct stream_in*)stream);
+    return proxy_get_period_size(&in->proxy) * audio_stream_in_frame_size(&(in->stream));
+}
+
+static uint32_t in_get_channels(const struct audio_stream *stream)
+{
+    const struct stream_in *in = (const struct stream_in*)stream;
+    return in->hal_channel_mask;
+}
+
+static audio_format_t in_get_format(const struct audio_stream *stream)
+{
+     alsa_device_proxy *proxy = &((struct stream_in*)stream)->proxy;
+     audio_format_t format = audio_format_from_pcm_format(proxy_get_format(proxy));
+     return format;
+}
+
+static int in_set_format(struct audio_stream *stream, audio_format_t format)
+{
+    ALOGV("in_set_format(%d) - NOPE", format);
+
+    return -ENOSYS;
+}
+
+static int in_standby(struct audio_stream *stream)
+{
+    struct stream_in *in = (struct stream_in *)stream;
+
+    lock_input_stream(in);
+    if (!in->standby) {
+        pthread_mutex_lock(&in->dev->lock);
+        proxy_close(&in->proxy);
+        pthread_mutex_unlock(&in->dev->lock);
+        in->standby = true;
+    }
+
+    pthread_mutex_unlock(&in->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)
+{
+    ALOGV("in_set_parameters() keys:%s", kvpairs);
+
+    struct stream_in *in = (struct stream_in *)stream;
+
+    char value[32];
+    int param_val;
+    int routing = 0;
+    int ret_value = 0;
+    int card = -1;
+    int device = -1;
+
+    if (!parse_card_device_params(kvpairs, &card, &device)) {
+        // nothing to do
+        return ret_value;
+    }
+
+    lock_input_stream(in);
+    pthread_mutex_lock(&in->dev->lock);
+
+    if (card >= 0 && device >= 0 && !profile_is_cached_for(in->profile, card, device)) {
+        /* cannot read pcm device info if playback is active */
+        if (!in->standby)
+            ret_value = -ENOSYS;
+        else {
+            int saved_card = in->profile->card;
+            int saved_device = in->profile->device;
+            in->profile->card = card;
+            in->profile->device = device;
+            ret_value = profile_read_device_info(in->profile) ? 0 : -EINVAL;
+            if (ret_value != 0) {
+                in->profile->card = saved_card;
+                in->profile->device = saved_device;
+            }
+        }
+    }
+
+    pthread_mutex_unlock(&in->dev->lock);
+    pthread_mutex_unlock(&in->lock);
+
+    return ret_value;
+}
+
+static char * in_get_parameters(const struct audio_stream *stream, const char *keys)
+{
+    struct stream_in *in = (struct stream_in *)stream;
+
+    lock_input_stream(in);
+    pthread_mutex_lock(&in->dev->lock);
+
+    char * params_str =  device_get_parameters(in->profile, keys);
+
+    pthread_mutex_unlock(&in->dev->lock);
+    pthread_mutex_unlock(&in->lock);
+
+    return params_str;
+}
+
+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 in_set_gain(struct audio_stream_in *stream, float gain)
+{
+    return 0;
+}
+
+/* must be called with hw device and output stream mutexes locked */
+static int start_input_stream(struct stream_in *in)
+{
+    ALOGV("ustart_input_stream(card:%d device:%d)", in->profile->card, in->profile->device);
+
+    return proxy_open(&in->proxy);
+}
+
+/* TODO mutex stuff here (see out_write) */
+static ssize_t in_read(struct audio_stream_in *stream, void* buffer, size_t bytes)
+{
+    size_t num_read_buff_bytes = 0;
+    void * read_buff = buffer;
+    void * out_buff = buffer;
+    int ret = 0;
+
+    struct stream_in * in = (struct stream_in *)stream;
+
+    lock_input_stream(in);
+    if (in->standby) {
+        pthread_mutex_lock(&in->dev->lock);
+        ret = start_input_stream(in);
+        pthread_mutex_unlock(&in->dev->lock);
+        if (ret != 0) {
+            goto err;
+        }
+        in->standby = false;
+    }
+
+    alsa_device_profile * profile = in->profile;
+
+    /*
+     * OK, we need to figure out how much data to read to be able to output the requested
+     * number of bytes in the HAL format (16-bit, stereo).
+     */
+    num_read_buff_bytes = bytes;
+    int num_device_channels = proxy_get_channel_count(&in->proxy); /* what we told Alsa */
+    int num_req_channels = in->hal_channel_count; /* what we told AudioFlinger */
+
+    if (num_device_channels != num_req_channels) {
+        num_read_buff_bytes = (num_device_channels * num_read_buff_bytes) / num_req_channels;
+    }
+
+    /* Setup/Realloc the conversion buffer (if necessary). */
+    if (num_read_buff_bytes != bytes) {
+        if (num_read_buff_bytes > in->conversion_buffer_size) {
+            /*TODO Remove this when AudioPolicyManger/AudioFlinger support arbitrary formats
+              (and do these conversions themselves) */
+            in->conversion_buffer_size = num_read_buff_bytes;
+            in->conversion_buffer = realloc(in->conversion_buffer, in->conversion_buffer_size);
+        }
+        read_buff = in->conversion_buffer;
+    }
+
+    ret = proxy_read(&in->proxy, read_buff, num_read_buff_bytes);
+    if (ret == 0) {
+        if (num_device_channels != num_req_channels) {
+            // ALOGV("chans dev:%d req:%d", num_device_channels, num_req_channels);
+
+            out_buff = buffer;
+            /* Num Channels conversion */
+            if (num_device_channels != num_req_channels) {
+                audio_format_t audio_format = in_get_format(&(in->stream.common));
+                unsigned sample_size_in_bytes = audio_bytes_per_sample(audio_format);
+
+                num_read_buff_bytes =
+                    adjust_channels(read_buff, num_device_channels,
+                                    out_buff, num_req_channels,
+                                    sample_size_in_bytes, num_read_buff_bytes);
+            }
+        }
+
+        /* no need to acquire in->dev->lock to read mic_muted here as we don't change its state */
+        if (num_read_buff_bytes > 0 && in->dev->mic_muted)
+            memset(buffer, 0, num_read_buff_bytes);
+    } else {
+        num_read_buff_bytes = 0; // reset the value after USB headset is unplugged
+    }
+
+err:
+    pthread_mutex_unlock(&in->lock);
+
+    return num_read_buff_bytes;
+}
+
+static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
+{
+    return 0;
+}
+
+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)
+{
+    ALOGV("in adev_open_input_stream() rate:%" PRIu32 ", chanMask:0x%" PRIX32 ", fmt:%" PRIu8,
+          config->sample_rate, config->channel_mask, config->format);
+
+    struct stream_in *in = (struct stream_in *)calloc(1, sizeof(struct stream_in));
+    int ret = 0;
+
+    if (in == NULL)
+        return -ENOMEM;
+
+    /* setup function pointers */
+    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;
+
+    pthread_mutex_init(&in->lock, (const pthread_mutexattr_t *) NULL);
+    pthread_mutex_init(&in->pre_lock, (const pthread_mutexattr_t *) NULL);
+
+    in->dev = (struct audio_device *)dev;
+    pthread_mutex_lock(&in->dev->lock);
+
+    in->profile = &in->dev->in_profile;
+
+    struct pcm_config proxy_config;
+    memset(&proxy_config, 0, sizeof(proxy_config));
+
+    /* Pull out the card/device pair */
+    parse_card_device_params(address, &(in->profile->card), &(in->profile->device));
+
+    profile_read_device_info(in->profile);
+    pthread_mutex_unlock(&in->dev->lock);
+
+    /* Rate */
+    if (config->sample_rate == 0) {
+        proxy_config.rate = config->sample_rate = profile_get_default_sample_rate(in->profile);
+    } else if (profile_is_sample_rate_valid(in->profile, config->sample_rate)) {
+        proxy_config.rate = config->sample_rate;
+    } else {
+        proxy_config.rate = config->sample_rate = profile_get_default_sample_rate(in->profile);
+        ret = -EINVAL;
+    }
+
+    /* Format */
+    if (config->format == AUDIO_FORMAT_DEFAULT) {
+        proxy_config.format = profile_get_default_format(in->profile);
+        config->format = audio_format_from_pcm_format(proxy_config.format);
+    } else {
+        enum pcm_format fmt = pcm_format_from_audio_format(config->format);
+        if (profile_is_format_valid(in->profile, fmt)) {
+            proxy_config.format = fmt;
+        } else {
+            proxy_config.format = profile_get_default_format(in->profile);
+            config->format = audio_format_from_pcm_format(proxy_config.format);
+            ret = -EINVAL;
+        }
+    }
+
+    /* Channels */
+    unsigned proposed_channel_count = 0;
+    if (k_force_channels) {
+        proposed_channel_count = k_force_channels;
+    } else if (config->channel_mask == AUDIO_CHANNEL_NONE) {
+        proposed_channel_count = profile_get_default_channel_count(in->profile);
+    }
+    if (proposed_channel_count != 0) {
+        config->channel_mask = audio_channel_in_mask_from_count(proposed_channel_count);
+        if (config->channel_mask == AUDIO_CHANNEL_INVALID)
+            config->channel_mask =
+                    audio_channel_mask_for_index_assignment_from_count(proposed_channel_count);
+        in->hal_channel_count = proposed_channel_count;
+    } else {
+        in->hal_channel_count = audio_channel_count_from_in_mask(config->channel_mask);
+    }
+    /* we can expose any channel mask, and emulate internally based on channel count. */
+    in->hal_channel_mask = config->channel_mask;
+
+    proxy_config.channels = profile_get_default_channel_count(in->profile);
+    proxy_prepare(&in->proxy, in->profile, &proxy_config);
+
+    in->standby = true;
+
+    in->conversion_buffer = NULL;
+    in->conversion_buffer_size = 0;
+
+    *stream_in = &in->stream;
+
+    return ret;
+}
+
+static void adev_close_input_stream(struct audio_hw_device *dev, struct audio_stream_in *stream)
+{
+    struct stream_in *in = (struct stream_in *)stream;
+
+    /* Close the pcm device */
+    in_standby(&stream->common);
+
+    free(in->conversion_buffer);
+
+    free(stream);
+}
+
+/*
+ * ADEV Functions
+ */
+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)
+{
+    return -ENOSYS;
+}
+
+static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
+{
+    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)
+{
+    struct audio_device * adev = (struct audio_device *)dev;
+    pthread_mutex_lock(&adev->lock);
+    adev->mic_muted = state;
+    pthread_mutex_unlock(&adev->lock);
+    return -ENOSYS;
+}
+
+static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
+{
+    return -ENOSYS;
+}
+
+static int adev_dump(const audio_hw_device_t *device, int fd)
+{
+    return 0;
+}
+
+static int adev_close(hw_device_t *device)
+{
+    struct audio_device *adev = (struct audio_device *)device;
+    free(device);
+
+    return 0;
+}
+
+static int adev_open(const hw_module_t* module, const char* name, hw_device_t** device)
+{
+    if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
+        return -EINVAL;
+
+    struct audio_device *adev = calloc(1, sizeof(struct audio_device));
+    if (!adev)
+        return -ENOMEM;
+
+    profile_init(&adev->out_profile, PCM_OUT);
+    profile_init(&adev->in_profile, PCM_IN);
+
+    adev->hw_device.common.tag = HARDWARE_DEVICE_TAG;
+    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.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;
+    adev->hw_device.set_mode = adev_set_mode;
+    adev->hw_device.set_mic_mute = adev_set_mic_mute;
+    adev->hw_device.get_mic_mute = adev_get_mic_mute;
+    adev->hw_device.set_parameters = adev_set_parameters;
+    adev->hw_device.get_parameters = adev_get_parameters;
+    adev->hw_device.get_input_buffer_size = adev_get_input_buffer_size;
+    adev->hw_device.open_output_stream = adev_open_output_stream;
+    adev->hw_device.close_output_stream = adev_close_output_stream;
+    adev->hw_device.open_input_stream = adev_open_input_stream;
+    adev->hw_device.close_input_stream = adev_close_input_stream;
+    adev->hw_device.dump = adev_dump;
+
+    *device = &adev->hw_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 = "audio HW HAL",
+        .author = "The Android Open Source Project",
+        .methods = &hal_module_methods,
+    },
+};
diff --git a/peripheral/audio/fs1130/audio_policy.conf b/peripheral/audio/fs1130/audio_policy.conf
new file mode 100644
index 0000000..0c95ce5
--- /dev/null
+++ b/peripheral/audio/fs1130/audio_policy.conf
@@ -0,0 +1,80 @@
+#
+# Copyright 2016 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.
+#
+
+# 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|AUDIO_DEVICE_OUT_SPDIF
+  default_output_device AUDIO_DEVICE_OUT_SPEAKER
+  attached_input_devices AUDIO_DEVICE_IN_SPDIF
+}
+
+audio_hw_modules {
+  primary {
+    outputs {
+      primary {
+        sampling_rates 44100|48000
+        channel_masks AUDIO_CHANNEL_OUT_STEREO
+        formats AUDIO_FORMAT_PCM_16_BIT
+        devices AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_SPDIF
+        flags AUDIO_OUTPUT_FLAG_PRIMARY
+      }
+    }
+    inputs {
+      primary {
+        sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000
+        channel_masks AUDIO_CHANNEL_IN_MONO|AUDIO_CHANNEL_IN_STEREO
+        formats AUDIO_FORMAT_PCM_16_BIT|AUDIO_FORMAT_AMR_NB|AUDIO_FORMAT_AMR_WB|AUDIO_FORMAT_QCELP|AUDIO_FORMAT_EVRC|AUDIO_FORMAT_EVRCB|AUDIO_FORMAT_EVRCWB|AUDIO_FORMAT_EVRCNW
+        devices AUDIO_DEVICE_IN_SPDIF
+      }
+    }
+  }
+  a2dp {
+    outputs {
+      a2dp {
+        sampling_rates 44100
+        channel_masks AUDIO_CHANNEL_OUT_STEREO
+        formats AUDIO_FORMAT_PCM_16_BIT
+        devices AUDIO_DEVICE_OUT_ALL_A2DP
+      }
+    }
+  }
+  usb {
+    global_configuration {
+      attached_output_devices AUDIO_DEVICE_OUT_USB_DEVICE
+      attached_input_devices AUDIO_DEVICE_IN_USB_DEVICE
+    }
+    outputs {
+      usb_device {
+        sampling_rates dynamic
+        channel_masks dynamic
+        formats dynamic
+        devices AUDIO_DEVICE_OUT_USB_DEVICE
+        flags AUDIO_OUTPUT_FLAG_PRIMARY
+      }
+    }
+    inputs {
+      usb_device {
+        sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000
+        channel_masks AUDIO_CHANNEL_IN_MONO|AUDIO_CHANNEL_IN_STEREO
+        formats AUDIO_FORMAT_PCM_16_BIT
+        devices AUDIO_DEVICE_IN_USB_DEVICE
+      }
+    }
+  }
+}
diff --git a/peripheral/audio/fs1130/media_codecs.xml b/peripheral/audio/fs1130/media_codecs.xml
new file mode 100644
index 0000000..ded3cb3
--- /dev/null
+++ b/peripheral/audio/fs1130/media_codecs.xml
@@ -0,0 +1,82 @@
+<?xml version="1.0" encoding="utf-8" ?>
+<!-- Copyright 2016 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.
+-->
+
+<!--
+<!DOCTYPE MediaCodecs [
+<!ELEMENT Include EMPTY>
+<!ATTLIST Include href CDATA #REQUIRED>
+<!ELEMENT MediaCodecs (Decoders|Encoders|Include)*>
+<!ELEMENT Decoders (MediaCodec|Include)*>
+<!ELEMENT Encoders (MediaCodec|Include)*>
+<!ELEMENT MediaCodec (Type|Quirk|Include)*>
+<!ATTLIST MediaCodec name CDATA #REQUIRED>
+<!ATTLIST MediaCodec type CDATA>
+<!ELEMENT Type EMPTY>
+<!ATTLIST Type name CDATA #REQUIRED>
+<!ELEMENT Quirk EMPTY>
+<!ATTLIST Quirk name CDATA #REQUIRED>
+]>
+
+There's a simple and a complex syntax to declare the availability of a
+media codec:
+
+A codec that properly follows the OpenMax spec and therefore doesn't have any
+quirks and that only supports a single content type can be declared like so:
+
+    <MediaCodec name="OMX.foo.bar" type="something/interesting" />
+
+If a codec has quirks OR supports multiple content types, the following syntax
+can be used:
+
+    <MediaCodec name="OMX.foo.bar" >
+        <Type name="something/interesting" />
+        <Type name="something/else" />
+        ...
+        <Quirk name="requires-allocate-on-input-ports" />
+        <Quirk name="requires-allocate-on-output-ports" />
+        <Quirk name="output-buffers-are-unreadable" />
+    </MediaCodec>
+
+Only the three quirks included above are recognized at this point:
+
+"requires-allocate-on-input-ports"
+    must be advertised if the component does not properly support specification
+    of input buffers using the OMX_UseBuffer(...) API but instead requires
+    OMX_AllocateBuffer to be used.
+
+"requires-allocate-on-output-ports"
+    must be advertised if the component does not properly support specification
+    of output buffers using the OMX_UseBuffer(...) API but instead requires
+    OMX_AllocateBuffer to be used.
+
+"output-buffers-are-unreadable"
+    must be advertised if the emitted output buffers of a decoder component
+    are not readable, i.e. use a custom format even though abusing one of
+    the official OMX colorspace constants.
+    Clients of such decoders will not be able to access the decoded data,
+    naturally making the component much less useful. The only use for
+    a component with this quirk is to render the output to the screen.
+    Audio decoders MUST NOT advertise this quirk.
+    Video decoders that advertise this quirk must be accompanied by a
+    corresponding color space converter for thumbnail extraction,
+    matching surfaceflinger support that can render the custom format to
+    a texture and possibly other code, so just DON'T USE THIS QUIRK.
+
+-->
+
+<MediaCodecs>
+    <Include href="media_codecs_google_audio.xml" />
+</MediaCodecs>
diff --git a/peripheral/audio/fs1130/peripheral.mk b/peripheral/audio/fs1130/peripheral.mk
new file mode 100644
index 0000000..2459f87
--- /dev/null
+++ b/peripheral/audio/fs1130/peripheral.mk
@@ -0,0 +1,30 @@
+#
+# Copyright 2016 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.
+#
+
+AUDIO_SRC := hardware/bsp/imagination/peripheral/audio/fs1130
+
+# Audio feature flags.
+BOARD_USES_ALSA_AUDIO := true
+BOARD_USES_TINY_ALSA_AUDIO := true
+
+# Audio configuration files.
+PRODUCT_COPY_FILES += \
+  frameworks/av/media/libstagefright/data/media_codecs_google_audio.xml:system/etc/media_codecs_google_audio.xml \
+  $(AUDIO_SRC)/audio_policy.conf:system/etc/audio_policy.conf \
+  $(AUDIO_SRC)/media_codecs.xml:system/etc/media_codecs.xml
+
+DEVICE_PACKAGES += \
+  audio.primary.fs1130