Revert "kylin: Add audio, base on intel's"

This reverts commit 1b4e8fb06a88e9ffd4473a6923574bb20ba2bdb5.

Which does not compile on AOSP.

Change-Id: I170ac5e69985cbb69cffefbb9462de0538899df9

5 files changed, 0 insertions, 1500 deletions

diff --git a/peripheral/audio/generic/Android.mk b/peripheral/audio/generic/Android.mk
deleted file mode 100644
index f211787..0000000
--- a/peripheral/audio/generic/Android.mk
+++ /dev/null
@@ -1,49 +0,0 @@
-# Copyright (C) 2012 The Android Open Source Project
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#      http://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE_RELATIVE_PATH := hw
-LOCAL_SRC_FILES := \
-  audio_hal.c
-LOCAL_C_INCLUDES += \
-  external/tinyalsa/include \
-  $(call include-path-for, audio-utils) \
-  $(call include-path-for, alsa-utils)
-LOCAL_SHARED_LIBRARIES := liblog libcutils libtinyalsa libaudioutils libalsautils
-LOCAL_MODULE_TAGS := optional
-# setting to build for primary audio or usb audio
-# set -DTARGET_AUDIO_PRIMARY to 1 for Primary (audio jack)
-# set -DTARGET_AUDIO_PRIMARY to 0 for USB audio
-LOCAL_CFLAGS := -Wno-unused-parameter -DTARGET_AUDIO_PRIMARY=1
-LOCAL_MODULE := audio.primary.$(TARGET_BOARD_PLATFORM)
-include $(BUILD_SHARED_LIBRARY)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE_RELATIVE_PATH := hw
-LOCAL_SRC_FILES := \
-  audio_hal.c
-LOCAL_C_INCLUDES += \
-  external/tinyalsa/include \
-  $(call include-path-for, audio-utils) \
-  $(call include-path-for, alsa-utils)
-LOCAL_SHARED_LIBRARIES := liblog libcutils libtinyalsa libaudioutils libalsautils
-LOCAL_MODULE_TAGS := optional
-# setting to build for primary audio or usb audio
-# set -DTARGET_AUDIO_PRIMARY to 1 for Primary (audio jack)
-# set -DTARGET_AUDIO_PRIMARY to 0 for USB audio
-LOCAL_CFLAGS := -Wno-unused-parameter -DTARGET_AUDIO_PRIMARY=0
-LOCAL_MODULE := audio.usb.$(TARGET_BOARD_PLATFORM)
-include $(BUILD_SHARED_LIBRARY)
diff --git a/peripheral/audio/generic/audio_hal.c b/peripheral/audio/generic/audio_hal.c
deleted file mode 100644
index 8cc1639..0000000
--- a/peripheral/audio/generic/audio_hal.c
+++ /dev/null
@@ -1,1272 +0,0 @@
-/*
- * 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 "modules.audio.audio_hal"
-/*#define LOG_NDEBUG 0*/
-
-#include <errno.h>
-#include <inttypes.h>
-#include <math.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 <dirent.h>
-#include <sys/ioctl.h>
-#include <fcntl.h>
-#include <sound/asound.h>
-
-
-#define PCM_DEV_STR "pcm"
-#if TARGET_AUDIO_PRIMARY
-#define AUDIO_STR "rt5616"
-#else
-#define AUDIO_STR "USB Audio"
-#endif
-#define MAX_PATH_LEN 30
-
-#define NBR_RETRIES 5
-#define RETRY_WAIT_USEC 20000
-
-/* 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;
-
-#include "alsa_device_profile.h"
-#include "alsa_device_proxy.h"
-#include "alsa_logging.h"
-
-#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;
-#if TARGET_AUDIO_PRIMARY
-    unsigned int master_volume;
-#endif
-};
-
-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.
- */
-
-
-static int in_stream_card_number = -1, out_stream_card_number = -1;
-
-
-/*
- * Examines a pcm-device file to see if its a Audio device and
- * returns its card-number. If no match, returns -1.
- */
-static int first_valid_sound_card(char *pcm_name, bool is_out_stream)
-{
-    int fd;
-    char pcm_dev_path[MAX_PATH_LEN];
-    struct snd_pcm_info info;
-    char type;
-    int pcm_name_length;
-
-    ALOGV("%s enter",__func__);
-
-    pcm_name_length = strlen(pcm_name);
-    if (pcm_name_length < 2) {
-        return -1;
-    }
-    type = is_out_stream ? 'p' : 'c';
-    /* If pcm out then filename must end with 0p/0c */
-    if ((pcm_name[pcm_name_length -2] != '0') && (pcm_name[pcm_name_length - 1] != type)) {
-        ALOGV("%s exit",__func__);
-        return -1;
-    }
-
-    snprintf(pcm_dev_path, sizeof(pcm_dev_path), "/dev/snd/%s", pcm_name);
-    fd = open(pcm_dev_path, O_RDONLY);
-
-    if (fd != -1) {
-        if (!(ioctl(fd, SNDRV_PCM_IOCTL_INFO, &info))) {
-            if (strstr(info.id, AUDIO_STR)) {
-                close(fd);
-                ALOGV("%s exit",__func__);
-                return info.card;
-            }
-        } else {
-            ALOGE("ioctl failed for file: %s", pcm_dev_path);
-        }
-
-        close(fd);
-    }
-
-    ALOGV("%s exit",__func__);
-    return -1;
-}
-
-/*
- * Returns the number of the first valid Audio card
- * If none is found, returns -1.
- */
-static int get_first_sound_card(bool is_out_stream)
-{
-    DIR *dir;
-    struct dirent *de = NULL;
-    int card_nr;
-
-    ALOGV("%s enter",__func__);
-
-    dir = opendir("/dev/snd");
-    if (dir == NULL) {
-        ALOGE("Could not open directory /dev/snd");
-        ALOGV("%s exit",__func__);
-        return -1;
-    }
-
-    while ((de = readdir(dir))) {
-        if (strncmp(de->d_name, PCM_DEV_STR, sizeof(PCM_DEV_STR) - 1) == 0) {
-            if ((card_nr = first_valid_sound_card(de->d_name, is_out_stream)) != -1) {
-                closedir(dir);
-                ALOGV("%s exit",__func__);
-                return card_nr;
-            }
-        }
-    }
-
-    closedir(dir);
-    ALOGW("No card found in /dev/snd");
-    ALOGV("%s exit",__func__);
-    return -1;
-}
-
-static bool parse_card_device_params(bool is_out_stream, int *card, int *device)
-{
-    int try_time;
-    int found_card = -1;
-
-    if (is_out_stream) {
-        if (out_stream_card_number != -1) {
-            *card = out_stream_card_number;
-            *device = 0;
-            return true;
-        }
-    } else {
-        if (in_stream_card_number != -1) {
-            *card = in_stream_card_number;
-            *device = 0;
-            return true;
-        }
-    }
-
-    for (try_time = 0; try_time < NBR_RETRIES; try_time++) {
-        found_card = get_first_sound_card(is_out_stream);
-        if (found_card == -1)
-            usleep(RETRY_WAIT_USEC);
-        else
-            break;
-    }
-
-    if (found_card == -1) {
-        *card = -1;
-        *device = -1;
-        return false;
-    }
-
-    if (is_out_stream) {
-        out_stream_card_number = found_card;
-    } else {
-        in_stream_card_number = found_card;
-    }
-
-    *card = found_card;
-    *device = 0;
-
-    return true;
-}
-
-static char * device_get_parameters(alsa_device_profile * profile, const char * keys)
-{
-    if (profile->card < 0 || profile->device < 0) {
-        return strdup("");
-    }
-
-    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 = %s", 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 adjust_volume(const uint32_t volume)
-{
-    /*
-     * map [0, 100] to [0, 25]
-     */
-    return (int)(sqrt(volume) * 2.5f);
-}
-
-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(true, &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);
-
-    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() status:%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(true, &(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 {
-        ALOGE("%s: The requested sample rate (%d) is not valid", __func__, config->sample_rate);
-        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 {
-            ALOGE("%s: The requested format (0x%x) is not valid", __func__, config->format);
-            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(false, &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 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(false, &(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 {
-        ALOGE("%s: The requested sample rate (%d) is not valid", __func__, config->sample_rate);
-        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 {
-            ALOGE("%s: The requested format (0x%x) is not valid", __func__, config->format);
-            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)
-{
-#if TARGET_AUDIO_PRIMARY
-    struct mixer *mixer;
-    struct mixer_ctl *ctl;
-    struct audio_device * adev = (struct audio_device *)dev;
-
-    if ((0 > volume) || (1 < volume) || (NULL == adev))
-      return -EINVAL;
-
-    pthread_mutex_lock(&adev->lock);
-    adev->master_volume = (int)(volume*100);
-
-    if (!(mixer = mixer_open(0))) {
-      pthread_mutex_unlock(&adev->lock);
-      return -ENOSYS;
-    }
-
-    ctl = mixer_get_ctl_by_name(mixer, "HP Playback Volume");
-    mixer_ctl_set_value(ctl,0,adjust_volume(adev->master_volume));
-    mixer_ctl_set_value(ctl,1,adjust_volume(adev->master_volume));
-
-    mixer_close(mixer);
-    pthread_mutex_unlock(&adev->lock);
-    return 0;
-#else
-    return -ENOSYS;
-#endif
-}
-
-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 TARGET_AUDIO_PRIMARY
-    struct mixer *mixer;
-    struct mixer_ctl *ctl;
-#endif
-    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;
-#if TARGET_AUDIO_PRIMARY
-    mixer = mixer_open(0);
-
-    if (mixer) {
-        /* setting master volume to value 50 */
-        adev->master_volume = 50;
-
-	int ret = 0;
-        ctl = mixer_get_ctl_by_name(mixer, "HP Playback Switch");
-        ret = mixer_ctl_set_value(ctl,0,1);
-        ret = mixer_ctl_set_value(ctl,1,1);
-        ctl = mixer_get_ctl_by_name(mixer, "HP Playback Volume");
-        mixer_ctl_set_value(ctl,0,adjust_volume(adev->master_volume));
-        mixer_ctl_set_value(ctl,1,adjust_volume(adev->master_volume));
-        ctl = mixer_get_ctl_by_name(mixer, "HPO MIX DAC1 Switch");
-        mixer_ctl_set_value(ctl,0,1);
-        ctl = mixer_get_ctl_by_name(mixer, "HPO MIX DAC1 Switch");
-        mixer_ctl_set_value(ctl,0,1);
-        ctl = mixer_get_ctl_by_name(mixer, "OUT MIXR DAC R1 Switch");
-        mixer_ctl_set_value(ctl,0,1);
-        ctl = mixer_get_ctl_by_name(mixer, "OUT MIXL DAC L1 Switch");
-        mixer_ctl_set_value(ctl,0,1);
-        ctl = mixer_get_ctl_by_name(mixer, "Stereo DAC MIXR DAC R1 Switch");
-        mixer_ctl_set_value(ctl,0,1);
-        ctl = mixer_get_ctl_by_name(mixer, "Stereo DAC MIXL DAC L1 Switch");
-        mixer_ctl_set_value(ctl,0,1);
-
-        mixer_close(mixer);
-    }
-#endif
-    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/generic/audio_policy.conf b/peripheral/audio/generic/audio_policy.conf
deleted file mode 100644
index ec3624f..0000000
--- a/peripheral/audio/generic/audio_policy.conf
+++ /dev/null
@@ -1,69 +0,0 @@
-#
-# Copyright (C) 2015 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
-  default_output_device AUDIO_DEVICE_OUT_SPEAKER
-  attached_input_devices AUDIO_DEVICE_IN_BUILTIN_MIC|AUDIO_DEVICE_IN_REMOTE_SUBMIX
-}
-audio_hw_modules {
-  primary {
-    outputs {
-      primary {
-        sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|88200|96000|176400|192000|
-        channel_masks AUDIO_CHANNEL_OUT_STEREO
-        formats AUDIO_FORMAT_PCM_16_BIT
-        devices AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET|AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_USB_DEVICE
-        flags AUDIO_OUTPUT_FLAG_PRIMARY
-      }
-    }
-    inputs {
-      primary {
-        sampling_rates 8000|11025|16000|22050|32000|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_BLUETOOTH_SCO_HEADSET|AUDIO_DEVICE_IN_WIRED_HEADSET|AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET|AUDIO_DEVICE_IN_USB_DEVICE
-      }
-    }
-  }
-  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/generic/media_codecs.xml b/peripheral/audio/generic/media_codecs.xml
deleted file mode 100644
index 934619c..0000000
--- a/peripheral/audio/generic/media_codecs.xml
+++ /dev/null
@@ -1,82 +0,0 @@
-<?xml version="1.0" encoding="utf-8" ?>
-<!-- Copyright (C) 2015 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/generic/peripheral.mk b/peripheral/audio/generic/peripheral.mk
deleted file mode 100644
index 1d17efb..0000000
--- a/peripheral/audio/generic/peripheral.mk
+++ /dev/null
@@ -1,28 +0,0 @@
-#
-# Copyright 2015 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 configuration files
-PRODUCT_COPY_FILES += \
-  frameworks/av/media/libstagefright/data/media_codecs_google_audio.xml:system/etc/media_codecs_google_audio.xml
-
-PRODUCT_COPY_FILES += \
-  hardware/bsp/rockchip/peripheral/audio/generic/media_codecs.xml:system/etc/media_codecs.xml \
-  hardware/bsp/rockchip/peripheral/audio/generic/audio_policy.conf:system/etc/audio_policy.conf
-
-# Primary audio HAL
-DEVICE_PACKAGES += \
-    audio.primary.$(TARGET_BOARD_PLATFORM) \
-    audio.usb.$(TARGET_BOARD_PLATFORM)