hal/audio_hw.c

/*
 * Copyright (c) 2013, The Linux Foundation. All rights reserved.
 * Not a Contribution.
 *
 * Copyright (C) 2013 The Android Open Source Project
 *
 * Licensed under the Apache License, Version 2.0 (the "License");
 * you may not use this file except in compliance with the License.
 * You may obtain a copy of the License at
 *
 *      http://www.apache.org/licenses/LICENSE-2.0
 *
 * Unless required by applicable law or agreed to in writing, software
 * distributed under the License is distributed on an "AS IS" BASIS,
 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
 * See the License for the specific language governing permissions and
 * limitations under the License.
 */

#define LOG_TAG "audio_hw_primary"
/*#define LOG_NDEBUG 0*/
#define LOG_NDDEBUG 0

#include <errno.h>
#include <pthread.h>
#include <stdint.h>
#include <sys/time.h>
#include <stdlib.h>
#include <math.h>

#include <cutils/log.h>
#include <cutils/str_parms.h>
#include <cutils/properties.h>

#include <hardware/audio_effect.h>
#include <audio_effects/effect_aec.h>
#include <audio_effects/effect_ns.h>
#include "audio_hw.h"
#include "platform_api.h"
#include <platform.h>
#include "audio_extn.h"

struct pcm_config pcm_config_deep_buffer = {
    .channels = 2,
    .rate = DEFAULT_OUTPUT_SAMPLING_RATE,
    .period_size = DEEP_BUFFER_OUTPUT_PERIOD_SIZE,
    .period_count = DEEP_BUFFER_OUTPUT_PERIOD_COUNT,
    .format = PCM_FORMAT_S16_LE,
    .start_threshold = DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4,
    .stop_threshold = INT_MAX,
    .avail_min = DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4,
};

struct pcm_config pcm_config_low_latency = {
    .channels = 2,
    .rate = DEFAULT_OUTPUT_SAMPLING_RATE,
    .period_size = LOW_LATENCY_OUTPUT_PERIOD_SIZE,
    .period_count = LOW_LATENCY_OUTPUT_PERIOD_COUNT,
    .format = PCM_FORMAT_S16_LE,
    .start_threshold = LOW_LATENCY_OUTPUT_PERIOD_SIZE / 4,
    .stop_threshold = INT_MAX,
    .avail_min = LOW_LATENCY_OUTPUT_PERIOD_SIZE / 4,
};

struct pcm_config pcm_config_hdmi_multi = {
    .channels = HDMI_MULTI_DEFAULT_CHANNEL_COUNT, /* changed when the stream is opened */
    .rate = DEFAULT_OUTPUT_SAMPLING_RATE, /* changed when the stream is opened */
    .period_size = HDMI_MULTI_PERIOD_SIZE,
    .period_count = HDMI_MULTI_PERIOD_COUNT,
    .format = PCM_FORMAT_S16_LE,
    .start_threshold = 0,
    .stop_threshold = INT_MAX,
    .avail_min = 0,
};

struct pcm_config pcm_config_audio_capture = {
    .channels = 2,
    .period_count = AUDIO_CAPTURE_PERIOD_COUNT,
    .format = PCM_FORMAT_S16_LE,
};

static const char * const use_case_table[AUDIO_USECASE_MAX] = {
    [USECASE_AUDIO_PLAYBACK_DEEP_BUFFER] = "deep-buffer-playback",
    [USECASE_AUDIO_PLAYBACK_LOW_LATENCY] = "low-latency-playback",
    [USECASE_AUDIO_PLAYBACK_MULTI_CH] = "multi-channel-playback",
    [USECASE_AUDIO_RECORD] = "audio-record",
    [USECASE_AUDIO_RECORD_LOW_LATENCY] = "low-latency-record",
    [USECASE_AUDIO_PLAYBACK_FM] = "play-fm",
    [USECASE_VOICE_CALL] = "voice-call",
    [USECASE_VOICE2_CALL] = "voice2-call",
    [USECASE_VOLTE_CALL] = "volte-call",
    [USECASE_QCHAT_CALL] = "qchat-call",
};


#define STRING_TO_ENUM(string) { #string, string }

struct string_to_enum {
    const char *name;
    uint32_t value;
};

static const struct string_to_enum out_channels_name_to_enum_table[] = {
    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
};

static struct audio_device *adev = NULL;
static pthread_mutex_t adev_init_lock;
static bool is_adev_initialised = false;

static int enable_audio_route(struct audio_device *adev,
                              struct audio_usecase *usecase,
                              bool update_mixer)
{
    snd_device_t snd_device;
    char mixer_path[MIXER_PATH_MAX_LENGTH];

    if (usecase == NULL)
        return -EINVAL;

    ALOGV("%s: enter: usecase(%d)", __func__, usecase->id);

    if (usecase->type == PCM_CAPTURE)
        snd_device = usecase->in_snd_device;
    else
        snd_device = usecase->out_snd_device;

    strcpy(mixer_path, use_case_table[usecase->id]);
    platform_add_backend_name(mixer_path, snd_device);
    ALOGV("%s: apply mixer path: %s", __func__, mixer_path);
    audio_route_apply_path(adev->audio_route, mixer_path);
    if (update_mixer)
        audio_route_update_mixer(adev->audio_route);

    ALOGV("%s: exit", __func__);
    return 0;
}

int disable_audio_route(struct audio_device *adev,
                        struct audio_usecase *usecase,
                        bool update_mixer)
{
    snd_device_t snd_device;
    char mixer_path[MIXER_PATH_MAX_LENGTH];

    if (usecase == NULL)
        return -EINVAL;

    ALOGV("%s: enter: usecase(%d)", __func__, usecase->id);
    if (usecase->type == PCM_CAPTURE)
        snd_device = usecase->in_snd_device;
    else
        snd_device = usecase->out_snd_device;
    strcpy(mixer_path, use_case_table[usecase->id]);
    platform_add_backend_name(mixer_path, snd_device);
    ALOGV("%s: reset mixer path: %s", __func__, mixer_path);
    audio_route_reset_path(adev->audio_route, mixer_path);
    if (update_mixer)
        audio_route_update_mixer(adev->audio_route);

    ALOGV("%s: exit", __func__);
    return 0;
}

static int enable_snd_device(struct audio_device *adev,
                             snd_device_t snd_device,
                             bool update_mixer)
{
    if (snd_device < SND_DEVICE_MIN ||
        snd_device >= SND_DEVICE_MAX) {
        ALOGE("%s: Invalid sound device %d", __func__, snd_device);
        return -EINVAL;
    }

    adev->snd_dev_ref_cnt[snd_device]++;
    if (adev->snd_dev_ref_cnt[snd_device] > 1) {
        ALOGV("%s: snd_device(%d: %s) is already active",
              __func__, snd_device, platform_get_snd_device_name(snd_device));
        return 0;
    }

    if (platform_send_audio_calibration(adev->platform, snd_device) < 0) {
        adev->snd_dev_ref_cnt[snd_device]--;
        return -EINVAL;
    }

    ALOGV("%s: snd_device(%d: %s)", __func__,
          snd_device, platform_get_snd_device_name(snd_device));
    audio_route_apply_path(adev->audio_route, platform_get_snd_device_name(snd_device));
    if (update_mixer)
        audio_route_update_mixer(adev->audio_route);

    return 0;
}

int disable_snd_device(struct audio_device *adev,
                       snd_device_t snd_device,
                       bool update_mixer)
{
    if (snd_device < SND_DEVICE_MIN ||
        snd_device >= SND_DEVICE_MAX) {
        ALOGE("%s: Invalid sound device %d", __func__, snd_device);
        return -EINVAL;
    }
    if (adev->snd_dev_ref_cnt[snd_device] <= 0) {
        ALOGE("%s: device ref cnt is already 0", __func__);
        return -EINVAL;
    }
    adev->snd_dev_ref_cnt[snd_device]--;
    if (adev->snd_dev_ref_cnt[snd_device] == 0) {
        ALOGV("%s: snd_device(%d: %s)", __func__,
              snd_device, platform_get_snd_device_name(snd_device));
        audio_route_reset_path(adev->audio_route, platform_get_snd_device_name(snd_device));
        if (update_mixer)
            audio_route_update_mixer(adev->audio_route);
    }
    return 0;
}

static void check_usecases_codec_backend(struct audio_device *adev,
                                          struct audio_usecase *uc_info,
                                          snd_device_t snd_device)
{
    struct listnode *node;
    struct audio_usecase *usecase;
    bool switch_device[AUDIO_USECASE_MAX];
    int i, num_uc_to_switch = 0;

    /*
     * This function is to make sure that all the usecases that are active on
     * the hardware codec backend are always routed to any one device that is
     * handled by the hardware codec.
     * For example, if low-latency and deep-buffer usecases are currently active
     * on speaker and out_set_parameters(headset) is received on low-latency
     * output, then we have to make sure deep-buffer is also switched to headset,
     * because of the limitation that both the devices cannot be enabled
     * at the same time as they share the same backend.
     */
    /* Disable all the usecases on the shared backend other than the
       specified usecase */
    for (i = 0; i < AUDIO_USECASE_MAX; i++)
        switch_device[i] = false;

    list_for_each(node, &adev->usecase_list) {
        usecase = node_to_item(node, struct audio_usecase, list);
        if (usecase->type == PCM_PLAYBACK &&
                usecase != uc_info &&
                usecase->out_snd_device != snd_device &&
                usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) {
            ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..",
                  __func__, use_case_table[usecase->id],
                  platform_get_snd_device_name(usecase->out_snd_device));
            disable_audio_route(adev, usecase, false);
            switch_device[usecase->id] = true;
            num_uc_to_switch++;
        }
    }

    if (num_uc_to_switch) {
        /* Make sure all the streams are de-routed before disabling the device */
        audio_route_update_mixer(adev->audio_route);

        list_for_each(node, &adev->usecase_list) {
            usecase = node_to_item(node, struct audio_usecase, list);
            if (switch_device[usecase->id]) {
                disable_snd_device(adev, usecase->out_snd_device, false);
                enable_snd_device(adev, snd_device, false);
            }
        }

        /* Make sure new snd device is enabled before re-routing the streams */
        audio_route_update_mixer(adev->audio_route);

        /* Re-route all the usecases on the shared backend other than the
           specified usecase to new snd devices */
        list_for_each(node, &adev->usecase_list) {
            usecase = node_to_item(node, struct audio_usecase, list);
            /* Update the out_snd_device only before enabling the audio route */
            if (switch_device[usecase->id] ) {
                usecase->out_snd_device = snd_device;
                enable_audio_route(adev, usecase, false);
            }
        }

        audio_route_update_mixer(adev->audio_route);
    }
}

static void check_and_route_capture_usecases(struct audio_device *adev,
                                             struct audio_usecase *uc_info,
                                             snd_device_t snd_device)
{
    struct listnode *node;
    struct audio_usecase *usecase;
    bool switch_device[AUDIO_USECASE_MAX];
    int i, num_uc_to_switch = 0;

    /*
     * This function is to make sure that all the active capture usecases
     * are always routed to the same input sound device.
     * For example, if audio-record and voice-call usecases are currently
     * active on speaker(rx) and speaker-mic (tx) and out_set_parameters(earpiece)
     * is received for voice call then we have to make sure that audio-record
     * usecase is also switched to earpiece i.e. voice-dmic-ef,
     * because of the limitation that two devices cannot be enabled
     * at the same time if they share the same backend.
     */
    for (i = 0; i < AUDIO_USECASE_MAX; i++)
        switch_device[i] = false;

    list_for_each(node, &adev->usecase_list) {
        usecase = node_to_item(node, struct audio_usecase, list);
        if (usecase->type == PCM_CAPTURE &&
                usecase != uc_info &&
                usecase->in_snd_device != snd_device) {
            ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..",
                  __func__, use_case_table[usecase->id],
                  platform_get_snd_device_name(usecase->in_snd_device));
            disable_audio_route(adev, usecase, false);
            switch_device[usecase->id] = true;
            num_uc_to_switch++;
        }
    }

    if (num_uc_to_switch) {
        /* Make sure all the streams are de-routed before disabling the device */
        audio_route_update_mixer(adev->audio_route);

        list_for_each(node, &adev->usecase_list) {
            usecase = node_to_item(node, struct audio_usecase, list);
            if (switch_device[usecase->id]) {
                disable_snd_device(adev, usecase->in_snd_device, false);
                enable_snd_device(adev, snd_device, false);
            }
        }

        /* Make sure new snd device is enabled before re-routing the streams */
        audio_route_update_mixer(adev->audio_route);

        /* Re-route all the usecases on the shared backend other than the
           specified usecase to new snd devices */
        list_for_each(node, &adev->usecase_list) {
            usecase = node_to_item(node, struct audio_usecase, list);
            /* Update the in_snd_device only before enabling the audio route */
            if (switch_device[usecase->id] ) {
                usecase->in_snd_device = snd_device;
                enable_audio_route(adev, usecase, false);
            }
        }

        audio_route_update_mixer(adev->audio_route);
    }
}

static int disable_all_usecases_of_type(struct audio_device *adev,
                                        usecase_type_t usecase_type,
                                        bool update_mixer)
{
    struct audio_usecase *usecase;
    struct listnode *node;
    int ret = 0;

    list_for_each(node, &adev->usecase_list) {
        usecase = node_to_item(node, struct audio_usecase, list);
        if (usecase->type == usecase_type) {
            ALOGV("%s: usecase id %d", __func__, usecase->id);
            ret = disable_audio_route(adev, usecase, update_mixer);
            if (ret) {
                ALOGE("%s: Failed to disable usecase id %d",
                      __func__, usecase->id);
            }
        }
    }

    return ret;
}

static int enable_all_usecases_of_type(struct audio_device *adev,
                                       usecase_type_t usecase_type,
                                       bool update_mixer)
{
    struct audio_usecase *usecase;
    struct listnode *node;
    int ret = 0;

    list_for_each(node, &adev->usecase_list) {
        usecase = node_to_item(node, struct audio_usecase, list);
        if (usecase->type == usecase_type) {
            ALOGV("%s: usecase id %d", __func__, usecase->id);
            ret = enable_audio_route(adev, usecase, update_mixer);
            if (ret) {
                ALOGE("%s: Failed to enable usecase id %d",
                      __func__, usecase->id);
            }
        }
    }

    return ret;
}

/* must be called with hw device mutex locked */
static int read_hdmi_channel_masks(struct stream_out *out)
{
    int ret = 0;
    int channels = platform_edid_get_max_channels(out->dev->platform);

    switch (channels) {
        /*
         * Do not handle stereo output in Multi-channel cases
         * Stereo case is handled in normal playback path
         */
    case 6:
        ALOGV("%s: HDMI supports 5.1", __func__);
        out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_5POINT1;
        break;
    case 8:
        ALOGV("%s: HDMI supports 5.1 and 7.1 channels", __func__);
        out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_5POINT1;
        out->supported_channel_masks[1] = AUDIO_CHANNEL_OUT_7POINT1;
        break;
    default:
        ALOGE("HDMI does not support multi channel playback");
        ret = -ENOSYS;
        break;
    }
    return ret;
}

static audio_usecase_t get_voice_usecase_id_from_list(struct audio_device *adev)
{
    struct audio_usecase *usecase;
    struct listnode *node;

    list_for_each(node, &adev->usecase_list) {
        usecase = node_to_item(node, struct audio_usecase, list);
        if (usecase->type == VOICE_CALL) {
            ALOGV("%s: usecase id %d", __func__, usecase->id);
            return usecase->id;
        }
    }
    return USECASE_INVALID;
}

struct audio_usecase *get_usecase_from_list(struct audio_device *adev,
                                            audio_usecase_t uc_id)
{
    struct audio_usecase *usecase;
    struct listnode *node;

    list_for_each(node, &adev->usecase_list) {
        usecase = node_to_item(node, struct audio_usecase, list);
        if (usecase->id == uc_id)
            return usecase;
    }
    return NULL;
}

int select_devices(struct audio_device *adev, audio_usecase_t uc_id)
{
    snd_device_t out_snd_device = SND_DEVICE_NONE;
    snd_device_t in_snd_device = SND_DEVICE_NONE;
    struct audio_usecase *usecase = NULL;
    struct audio_usecase *vc_usecase = NULL;
    struct listnode *node;
    int status = 0;

    usecase = get_usecase_from_list(adev, uc_id);
    if (usecase == NULL) {
        ALOGE("%s: Could not find the usecase(%d)", __func__, uc_id);
        return -EINVAL;
    }

    if (usecase->type == VOICE_CALL) {
        out_snd_device = platform_get_output_snd_device(adev->platform,
                                                        usecase->stream.out->devices);
        in_snd_device = platform_get_input_snd_device(adev->platform, usecase->stream.out->devices);
        usecase->devices = usecase->stream.out->devices;
    } else {
        /*
         * If the voice call is active, use the sound devices of voice call usecase
         * so that it would not result any device switch. All the usecases will
         * be switched to new device when select_devices() is called for voice call
         * usecase. This is to avoid switching devices for voice call when
         * check_usecases_codec_backend() is called below.
         */
        if (voice_is_in_call(adev)) {
            vc_usecase = get_usecase_from_list(adev,
                                               get_voice_usecase_id_from_list(adev));
            if (vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) {
                in_snd_device = vc_usecase->in_snd_device;
                out_snd_device = vc_usecase->out_snd_device;
            }
        }
        if (usecase->type == PCM_PLAYBACK) {
            usecase->devices = usecase->stream.out->devices;
            in_snd_device = SND_DEVICE_NONE;
            if (out_snd_device == SND_DEVICE_NONE) {
                out_snd_device = platform_get_output_snd_device(adev->platform,
                                            usecase->stream.out->devices);
                if (usecase->stream.out == adev->primary_output &&
                        adev->active_input &&
                        adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION) {
                    select_devices(adev, adev->active_input->usecase);
                }
            }
        } else if (usecase->type == PCM_CAPTURE) {
            usecase->devices = usecase->stream.in->device;
            out_snd_device = SND_DEVICE_NONE;
            if (in_snd_device == SND_DEVICE_NONE) {
                if (adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION &&
                        adev->primary_output && !adev->primary_output->standby) {
                    in_snd_device = platform_get_input_snd_device(adev->platform,
                                        adev->primary_output->devices);
                } else {
                    in_snd_device = platform_get_input_snd_device(adev->platform,
                                                                  AUDIO_DEVICE_NONE);
                }
            }
        }
    }

    if (out_snd_device == usecase->out_snd_device &&
        in_snd_device == usecase->in_snd_device) {
        return 0;
    }

    ALOGD("%s: out_snd_device(%d: %s) in_snd_device(%d: %s)", __func__,
          out_snd_device, platform_get_snd_device_name(out_snd_device),
          in_snd_device,  platform_get_snd_device_name(in_snd_device));

    /*
     * Limitation: While in call, to do a device switch we need to disable
     * and enable both RX and TX devices though one of them is same as current
     * device.
     */
    if (usecase->type == VOICE_CALL) {
        disable_all_usecases_of_type(adev, VOICE_CALL, true);
        status = platform_switch_voice_call_device_pre(adev->platform);
    }

    /* Disable current sound devices */
    if (usecase->out_snd_device != SND_DEVICE_NONE) {
        disable_audio_route(adev, usecase, true);
        disable_snd_device(adev, usecase->out_snd_device, false);
    }

    if (usecase->in_snd_device != SND_DEVICE_NONE) {
        disable_audio_route(adev, usecase, true);
        disable_snd_device(adev, usecase->in_snd_device, false);
    }

    /* Enable new sound devices */
    if (out_snd_device != SND_DEVICE_NONE) {
        if (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND)
            check_usecases_codec_backend(adev, usecase, out_snd_device);
        enable_snd_device(adev, out_snd_device, false);
    }

    if (in_snd_device != SND_DEVICE_NONE) {
        check_and_route_capture_usecases(adev, usecase, in_snd_device);
        enable_snd_device(adev, in_snd_device, false);
    }

    if (usecase->type == VOICE_CALL)
        status = platform_switch_voice_call_device_post(adev->platform,
                                                        out_snd_device,
                                                        in_snd_device);

    audio_route_update_mixer(adev->audio_route);

    usecase->in_snd_device = in_snd_device;
    usecase->out_snd_device = out_snd_device;

    if (usecase->type == VOICE_CALL)
        enable_all_usecases_of_type(adev, VOICE_CALL, true);
    else
        enable_audio_route(adev, usecase, true);

    return status;
}

static int stop_input_stream(struct stream_in *in)
{
    int i, ret = 0;
    struct audio_usecase *uc_info;
    struct audio_device *adev = in->dev;

    adev->active_input = NULL;

    ALOGV("%s: enter: usecase(%d: %s)", __func__,
          in->usecase, use_case_table[in->usecase]);
    uc_info = get_usecase_from_list(adev, in->usecase);
    if (uc_info == NULL) {
        ALOGE("%s: Could not find the usecase (%d) in the list",
              __func__, in->usecase);
        return -EINVAL;
    }

    /* 1. Disable stream specific mixer controls */
    disable_audio_route(adev, uc_info, true);

    /* 2. Disable the tx device */
    disable_snd_device(adev, uc_info->in_snd_device, true);

    list_remove(&uc_info->list);
    free(uc_info);

    ALOGV("%s: exit: status(%d)", __func__, ret);
    return ret;
}

int start_input_stream(struct stream_in *in)
{
    /* 1. Enable output device and stream routing controls */
    int ret = 0;
    struct audio_usecase *uc_info;
    struct audio_device *adev = in->dev;

    ALOGV("%s: enter: usecase(%d)", __func__, in->usecase);
    in->pcm_device_id = platform_get_pcm_device_id(in->usecase, PCM_CAPTURE);
    if (in->pcm_device_id < 0) {
        ALOGE("%s: Could not find PCM device id for the usecase(%d)",
              __func__, in->usecase);
        ret = -EINVAL;
        goto error_config;
    }

    adev->active_input = in;
    uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
    uc_info->id = in->usecase;
    uc_info->type = PCM_CAPTURE;
    uc_info->stream.in = in;
    uc_info->devices = in->device;
    uc_info->in_snd_device = SND_DEVICE_NONE;
    uc_info->out_snd_device = SND_DEVICE_NONE;

    list_add_tail(&adev->usecase_list, &uc_info->list);
    select_devices(adev, in->usecase);

    ALOGV("%s: Opening PCM device card_id(%d) device_id(%d), channels %d",
          __func__, SOUND_CARD, in->pcm_device_id, in->config.channels);
    in->pcm = pcm_open(SOUND_CARD, in->pcm_device_id,
                           PCM_IN, &in->config);
    if (in->pcm && !pcm_is_ready(in->pcm)) {
        ALOGE("%s: %s", __func__, pcm_get_error(in->pcm));
        pcm_close(in->pcm);
        in->pcm = NULL;
        ret = -EIO;
        goto error_open;
    }
    ALOGV("%s: exit", __func__);
    return ret;

error_open:
    stop_input_stream(in);

error_config:
    adev->active_input = NULL;
    ALOGD("%s: exit: status(%d)", __func__, ret);

    return ret;
}

static int stop_output_stream(struct stream_out *out)
{
    int i, ret = 0;
    struct audio_usecase *uc_info;
    struct audio_device *adev = out->dev;

    ALOGV("%s: enter: usecase(%d: %s)", __func__,
          out->usecase, use_case_table[out->usecase]);
    uc_info = get_usecase_from_list(adev, out->usecase);
    if (uc_info == NULL) {
        ALOGE("%s: Could not find the usecase (%d) in the list",
              __func__, out->usecase);
        return -EINVAL;
    }

    /* 1. Get and set stream specific mixer controls */
    disable_audio_route(adev, uc_info, true);

    /* 2. Disable the rx device */
    disable_snd_device(adev, uc_info->out_snd_device, true);

    list_remove(&uc_info->list);
    free(uc_info);

    ALOGV("%s: exit: status(%d)", __func__, ret);
    return ret;
}

int start_output_stream(struct stream_out *out)
{
    int ret = 0;
    struct audio_usecase *uc_info;
    struct audio_device *adev = out->dev;

    ALOGV("%s: enter: usecase(%d: %s) devices(%#x)",
          __func__, out->usecase, use_case_table[out->usecase], out->devices);
    out->pcm_device_id = platform_get_pcm_device_id(out->usecase, PCM_PLAYBACK);
    if (out->pcm_device_id < 0) {
        ALOGE("%s: Invalid PCM device id(%d) for the usecase(%d)",
              __func__, out->pcm_device_id, out->usecase);
        ret = -EINVAL;
        goto error_config;
    }

    uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
    uc_info->id = out->usecase;
    uc_info->type = PCM_PLAYBACK;
    uc_info->stream.out = out;
    uc_info->devices = out->devices;
    uc_info->in_snd_device = SND_DEVICE_NONE;
    uc_info->out_snd_device = SND_DEVICE_NONE;

    list_add_tail(&adev->usecase_list, &uc_info->list);

    select_devices(adev, out->usecase);

    ALOGV("%s: Opening PCM device card_id(%d) device_id(%d)",
          __func__, 0, out->pcm_device_id);
    out->pcm = pcm_open(SOUND_CARD, out->pcm_device_id,
                           PCM_OUT, &out->config);
    if (out->pcm && !pcm_is_ready(out->pcm)) {
        ALOGE("%s: %s", __func__, pcm_get_error(out->pcm));
        pcm_close(out->pcm);
        out->pcm = NULL;
        ret = -EIO;
        goto error_pcm_open;
    }
    ALOGV("%s: exit", __func__);
    return 0;
error_pcm_open:
    stop_output_stream(out);
error_config:
    return ret;
}

static int check_input_parameters(uint32_t sample_rate,
                                  audio_format_t format,
                                  int channel_count)
{
    if (format != AUDIO_FORMAT_PCM_16_BIT) return -EINVAL;

    if ((channel_count < 1) || (channel_count > 2)) return -EINVAL;

    switch (sample_rate) {
    case 8000:
    case 11025:
    case 12000:
    case 16000:
    case 22050:
    case 24000:
    case 32000:
    case 44100:
    case 48000:
        break;
    default:
        return -EINVAL;
    }

    return 0;
}

static size_t get_input_buffer_size(uint32_t sample_rate,
                                    audio_format_t format,
                                    int channel_count)
{
    size_t size = 0;

    if (check_input_parameters(sample_rate, format, channel_count) != 0)
        return 0;

    size = (sample_rate * AUDIO_CAPTURE_PERIOD_DURATION_MSEC) / 1000;
    /* ToDo: should use frame_size computed based on the format and
       channel_count here. */
    size *= sizeof(short) * channel_count;

    /* make sure the size is multiple of 64 */
    size += 0x3f;
    size &= ~0x3f;

    return size;
}

static uint32_t out_get_sample_rate(const struct audio_stream *stream)
{
    struct stream_out *out = (struct stream_out *)stream;

    return out->config.rate;
}

static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
    return -ENOSYS;
}

static size_t out_get_buffer_size(const struct audio_stream *stream)
{
    struct stream_out *out = (struct stream_out *)stream;

    return out->config.period_size * audio_stream_frame_size(stream);
}

static uint32_t out_get_channels(const struct audio_stream *stream)
{
    struct stream_out *out = (struct stream_out *)stream;

    return out->channel_mask;
}

static audio_format_t out_get_format(const struct audio_stream *stream)
{
    return AUDIO_FORMAT_PCM_16_BIT;
}

static int out_set_format(struct audio_stream *stream, audio_format_t format)
{
    return -ENOSYS;
}

static int out_standby(struct audio_stream *stream)
{
    struct stream_out *out = (struct stream_out *)stream;
    struct audio_device *adev = out->dev;
    ALOGV("%s: enter: usecase(%d: %s)", __func__,
          out->usecase, use_case_table[out->usecase]);
    pthread_mutex_lock(&out->lock);

    if (!out->standby) {
        out->standby = true;
        if (out->pcm) {
            pcm_close(out->pcm);
            out->pcm = NULL;
        }
        pthread_mutex_lock(&adev->lock);
        stop_output_stream(out);
        pthread_mutex_unlock(&adev->lock);
    }
    pthread_mutex_unlock(&out->lock);
    ALOGV("%s: exit", __func__);
    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)
{
    struct stream_out *out = (struct stream_out *)stream;
    struct audio_device *adev = out->dev;
    struct audio_usecase *usecase;
    struct listnode *node;
    struct str_parms *parms;
    char value[32];
    int ret, val = 0;
    bool select_new_device = false;

    ALOGD("%s: enter: usecase(%d: %s) kvpairs: %s",
          __func__, out->usecase, use_case_table[out->usecase], kvpairs);
    parms = str_parms_create_str(kvpairs);
    ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
    if (ret >= 0) {
        val = atoi(value);
        pthread_mutex_lock(&out->lock);
        pthread_mutex_lock(&adev->lock);

        /*
         * When HDMI cable is unplugged the music playback is paused and
         * the policy manager sends routing=0. But the audioflinger
         * continues to write data until standby time (3sec).
         * As the HDMI core is turned off, the write gets blocked.
         * Avoid this by routing audio to speaker until standby.
         */
        if (out->devices == AUDIO_DEVICE_OUT_AUX_DIGITAL &&
                val == AUDIO_DEVICE_NONE) {
            val = AUDIO_DEVICE_OUT_SPEAKER;
        }

        /*
         * select_devices() call below switches all the usecases on the same
         * backend to the new device. Refer to check_usecases_codec_backend() in
         * the select_devices(). But how do we undo this?
         *
         * For example, music playback is active on headset (deep-buffer usecase)
         * and if we go to ringtones and select a ringtone, low-latency usecase
         * will be started on headset+speaker. As we can't enable headset+speaker
         * and headset devices at the same time, select_devices() switches the music
         * playback to headset+speaker while starting low-lateny usecase for ringtone.
         * So when the ringtone playback is completed, how do we undo the same?
         *
         * We are relying on the out_set_parameters() call on deep-buffer output,
         * once the ringtone playback is ended.
         * NOTE: We should not check if the current devices are same as new devices.
         *       Because select_devices() must be called to switch back the music
         *       playback to headset.
         */
        if (val != 0) {
            out->devices = val;

            if (!out->standby)
                select_devices(adev, out->usecase);

            if ((adev->mode == AUDIO_MODE_IN_CALL) &&
                    !voice_is_in_call(adev) &&
                    (out == adev->primary_output)) {
                voice_start_call(adev);
            } else if ((adev->mode == AUDIO_MODE_IN_CALL) &&
                            voice_is_in_call(adev) &&
                            (out == adev->primary_output)) {
                select_devices(adev, get_voice_usecase_id_from_list(adev));
            }
        }

        if ((adev->mode == AUDIO_MODE_NORMAL) &&
                voice_is_in_call(adev) &&
                (out == adev->primary_output)) {
            voice_stop_call(adev);
        }

        pthread_mutex_unlock(&adev->lock);
        pthread_mutex_unlock(&out->lock);
    }

    if (out == adev->primary_output) {
        pthread_mutex_lock(&adev->lock);
        audio_extn_set_parameters(adev, parms);
        pthread_mutex_unlock(&adev->lock);
    }

    str_parms_destroy(parms);
    ALOGV("%s: exit: code(%d)", __func__, ret);
    return ret;
}

static char* out_get_parameters(const struct audio_stream *stream, const char *keys)
{
    struct stream_out *out = (struct stream_out *)stream;
    struct str_parms *query = str_parms_create_str(keys);
    char *str;
    char value[256];
    struct str_parms *reply = str_parms_create();
    size_t i, j;
    int ret;
    bool first = true;
    ALOGV("%s: enter: keys - %s", __func__, keys);
    ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value, sizeof(value));
    if (ret >= 0) {
        value[0] = '\0';
        i = 0;
        while (out->supported_channel_masks[i] != 0) {
            for (j = 0; j < ARRAY_SIZE(out_channels_name_to_enum_table); j++) {
                if (out_channels_name_to_enum_table[j].value == out->supported_channel_masks[i]) {
                    if (!first) {
                        strcat(value, "|");
                    }
                    strcat(value, out_channels_name_to_enum_table[j].name);
                    first = false;
                    break;
                }
            }
            i++;
        }
        str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value);
        str = str_parms_to_str(reply);
    } else {
        str = strdup(keys);
    }
    str_parms_destroy(query);
    str_parms_destroy(reply);
    ALOGV("%s: exit: returns - %s", __func__, str);
    return str;
}

static uint32_t out_get_latency(const struct audio_stream_out *stream)
{
    struct stream_out *out = (struct stream_out *)stream;

    return (out->config.period_count * out->config.period_size * 1000) / (out->config.rate);
}

static int out_set_volume(struct audio_stream_out *stream, float left,
                          float right)
{
    struct stream_out *out = (struct stream_out *)stream;
    if (out->usecase == USECASE_AUDIO_PLAYBACK_MULTI_CH) {
        /* only take left channel into account: the API is for stereo anyway */
        out->muted = (left == 0.0f);
        return 0;
    }
    return -ENOSYS;
}

static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
                         size_t bytes)
{
    struct stream_out *out = (struct stream_out *)stream;
    struct audio_device *adev = out->dev;
    int i, ret = -1;

    pthread_mutex_lock(&out->lock);
    if (out->standby) {
        out->standby = false;
        pthread_mutex_lock(&adev->lock);
        ret = start_output_stream(out);
        pthread_mutex_unlock(&adev->lock);
        if (ret != 0) {
            out->standby = true;
            goto exit;
        }
    }

    if (out->pcm) {
        if (out->muted)
            memset((void *)buffer, 0, bytes);
        //ALOGV("%s: writing buffer (%d bytes) to pcm device", __func__, bytes);
        ret = pcm_write(out->pcm, (void *)buffer, bytes);
    }

exit:
    pthread_mutex_unlock(&out->lock);

    if (ret != 0) {
        if (out->pcm)
            ALOGE("%s: error %d - %s", __func__, ret, pcm_get_error(out->pcm));
        out_standby(&out->stream.common);
        usleep(bytes * 1000000 / audio_stream_frame_size(&out->stream.common) /
               out_get_sample_rate(&out->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_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
    return 0;
}

static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
    return 0;
}

static int out_get_next_write_timestamp(const struct audio_stream_out *stream,
                                        int64_t *timestamp)
{
    return -EINVAL;
}

/** audio_stream_in implementation **/
static uint32_t in_get_sample_rate(const struct audio_stream *stream)
{
    struct stream_in *in = (struct stream_in *)stream;

    return in->config.rate;
}

static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
    return -ENOSYS;
}

static size_t in_get_buffer_size(const struct audio_stream *stream)
{
    struct stream_in *in = (struct stream_in *)stream;

    return in->config.period_size * audio_stream_frame_size(stream);
}

static uint32_t in_get_channels(const struct audio_stream *stream)
{
    struct stream_in *in = (struct stream_in *)stream;

    return in->channel_mask;
}

static audio_format_t in_get_format(const struct audio_stream *stream)
{
    return AUDIO_FORMAT_PCM_16_BIT;
}

static int in_set_format(struct audio_stream *stream, audio_format_t format)
{
    return -ENOSYS;
}

static int in_standby(struct audio_stream *stream)
{
    struct stream_in *in = (struct stream_in *)stream;
    struct audio_device *adev = in->dev;
    int status = 0;
    ALOGV("%s: enter", __func__);
    pthread_mutex_lock(&in->lock);
    if (!in->standby) {
        in->standby = true;
        if (in->pcm) {
            pcm_close(in->pcm);
            in->pcm = NULL;
        }
        pthread_mutex_lock(&adev->lock);
        status = stop_input_stream(in);
        pthread_mutex_unlock(&adev->lock);
    }
    pthread_mutex_unlock(&in->lock);
    ALOGV("%s: exit:  status(%d)", __func__, status);
    return status;
}

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)
{
    struct stream_in *in = (struct stream_in *)stream;
    struct audio_device *adev = in->dev;
    struct str_parms *parms;
    char *str;
    char value[32];
    int ret, val = 0;

    ALOGV("%s: enter: kvpairs=%s", __func__, kvpairs);
    parms = str_parms_create_str(kvpairs);

    ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_INPUT_SOURCE, value, sizeof(value));

    pthread_mutex_lock(&in->lock);
    pthread_mutex_lock(&adev->lock);
    if (ret >= 0) {
        val = atoi(value);
        /* no audio source uses val == 0 */
        if ((in->source != val) && (val != 0)) {
            in->source = val;
        }
    }

    ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
    if (ret >= 0) {
        val = atoi(value);
        if ((in->device != val) && (val != 0)) {
            in->device = val;
            /* If recording is in progress, change the tx device to new device */
            if (!in->standby)
                ret = select_devices(adev, in->usecase);
        }
    }

    pthread_mutex_unlock(&adev->lock);
    pthread_mutex_unlock(&in->lock);

    str_parms_destroy(parms);
    ALOGV("%s: exit: status(%d)", __func__, ret);
    return ret;
}

static char* in_get_parameters(const struct audio_stream *stream,
                               const char *keys)
{
    return strdup("");
}

static int in_set_gain(struct audio_stream_in *stream, float gain)
{
    return 0;
}

static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
                       size_t bytes)
{
    struct stream_in *in = (struct stream_in *)stream;
    struct audio_device *adev = in->dev;
    int i, ret = -1;

    pthread_mutex_lock(&in->lock);
    if (in->standby) {
        pthread_mutex_lock(&adev->lock);
        ret = start_input_stream(in);
        pthread_mutex_unlock(&adev->lock);
        if (ret != 0) {
            goto exit;
        }
        in->standby = 0;
    }

    if (in->pcm) {
        ret = pcm_read(in->pcm, buffer, bytes);
    }

    /*
     * Instead of writing zeroes here, we could trust the hardware
     * to always provide zeroes when muted.
     */
    if (ret == 0 && voice_get_mic_mute(adev))
        memset(buffer, 0, bytes);

exit:
    pthread_mutex_unlock(&in->lock);

    if (ret != 0) {
        in_standby(&in->stream.common);
        ALOGV("%s: read failed - sleeping for buffer duration", __func__);
        usleep(bytes * 1000000 / audio_stream_frame_size(&in->stream.common) /
               in_get_sample_rate(&in->stream.common));
    }
    return bytes;
}

static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
{
    return 0;
}

static int add_remove_audio_effect(const struct audio_stream *stream,
                                   effect_handle_t effect,
                                   bool enable)
{
    struct stream_in *in = (struct stream_in *)stream;
    int status = 0;
    effect_descriptor_t desc;

    status = (*effect)->get_descriptor(effect, &desc);
    if (status != 0)
        return status;

    pthread_mutex_lock(&in->lock);
    pthread_mutex_lock(&in->dev->lock);
    if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) &&
            in->enable_aec != enable &&
            (memcmp(&desc.type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0)) {
        in->enable_aec = enable;
        if (!in->standby)
            select_devices(in->dev, in->usecase);
    }
    pthread_mutex_unlock(&in->dev->lock);
    pthread_mutex_unlock(&in->lock);

    return 0;
}

static int in_add_audio_effect(const struct audio_stream *stream,
                               effect_handle_t effect)
{
    ALOGV("%s: effect %p", __func__, effect);
    return add_remove_audio_effect(stream, effect, true);
}

static int in_remove_audio_effect(const struct audio_stream *stream,
                                  effect_handle_t effect)
{
    ALOGV("%s: effect %p", __func__, effect);
    return add_remove_audio_effect(stream, effect, false);
}

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)
{
    struct audio_device *adev = (struct audio_device *)dev;
    struct stream_out *out;
    int i, ret;

    ALOGV("%s: enter: sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x)",
          __func__, config->sample_rate, config->channel_mask, devices, flags);
    *stream_out = NULL;
    out = (struct stream_out *)calloc(1, sizeof(struct stream_out));

    if (devices == AUDIO_DEVICE_NONE)
        devices = AUDIO_DEVICE_OUT_SPEAKER;

    out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_STEREO;
    out->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
    out->flags = flags;
    out->devices = devices;
    out->dev = adev;

    /* Init use case and pcm_config */
    if (out->flags & AUDIO_OUTPUT_FLAG_DIRECT &&
        out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
        pthread_mutex_lock(&adev->lock);
        ret = read_hdmi_channel_masks(out);
        pthread_mutex_unlock(&adev->lock);
        if (ret != 0) {
            /* If HDMI does not support multi channel playback, set the default */
            out->config.channels = popcount(out->channel_mask);
            platform_set_hdmi_channels(adev->platform, out->config.channels);
            goto error_open;
        }

        if (config->sample_rate == 0)
            config->sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
        if (config->channel_mask == 0)
            config->channel_mask = AUDIO_CHANNEL_OUT_5POINT1;

        out->channel_mask = config->channel_mask;
        out->usecase = USECASE_AUDIO_PLAYBACK_MULTI_CH;
        out->config = pcm_config_hdmi_multi;
        out->config.rate = config->sample_rate;
        out->config.channels = popcount(out->channel_mask);
        out->config.period_size = HDMI_MULTI_PERIOD_BYTES / (out->config.channels * 2);
        platform_set_hdmi_channels(adev->platform, out->config.channels);
    } else if (out->flags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) {
        out->usecase = USECASE_AUDIO_PLAYBACK_DEEP_BUFFER;
        out->config = pcm_config_deep_buffer;
    } else {
        out->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY;
        out->config = pcm_config_low_latency;
    }

    if (flags & AUDIO_OUTPUT_FLAG_PRIMARY) {
        if(adev->primary_output == NULL)
            adev->primary_output = out;
        else {
            ALOGE("%s: Primary output is already opened", __func__);
            ret = -EEXIST;
            goto error_open;
        }
    }

    /* Check if this usecase is already existing */
    pthread_mutex_lock(&adev->lock);
    if (get_usecase_from_list(adev, out->usecase) != NULL) {
        ALOGE("%s: Usecase (%d) is already present", __func__, out->usecase);
        pthread_mutex_unlock(&adev->lock);
        ret = -EEXIST;
        goto error_open;
    }
    pthread_mutex_unlock(&adev->lock);

    out->stream.common.get_sample_rate = out_get_sample_rate;
    out->stream.common.set_sample_rate = out_set_sample_rate;
    out->stream.common.get_buffer_size = out_get_buffer_size;
    out->stream.common.get_channels = out_get_channels;
    out->stream.common.get_format = out_get_format;
    out->stream.common.set_format = out_set_format;
    out->stream.common.standby = out_standby;
    out->stream.common.dump = out_dump;
    out->stream.common.set_parameters = out_set_parameters;
    out->stream.common.get_parameters = out_get_parameters;
    out->stream.common.add_audio_effect = out_add_audio_effect;
    out->stream.common.remove_audio_effect = out_remove_audio_effect;
    out->stream.get_latency = out_get_latency;
    out->stream.set_volume = out_set_volume;
    out->stream.write = out_write;
    out->stream.get_render_position = out_get_render_position;
    out->stream.get_next_write_timestamp = out_get_next_write_timestamp;

    out->standby = 1;
    /* out->muted = false; by calloc() */

    config->format = out->stream.common.get_format(&out->stream.common);
    config->channel_mask = out->stream.common.get_channels(&out->stream.common);
    config->sample_rate = out->stream.common.get_sample_rate(&out->stream.common);

    *stream_out = &out->stream;
    ALOGV("%s: exit", __func__);
    return 0;

error_open:
    free(out);
    *stream_out = NULL;
    ALOGD("%s: exit: ret %d", __func__, ret);
    return ret;
}

static void adev_close_output_stream(struct audio_hw_device *dev,
                                     struct audio_stream_out *stream)
{
    ALOGV("%s: enter", __func__);
    out_standby(&stream->common);
    free(stream);
    ALOGV("%s: exit", __func__);
}

static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
{
    struct audio_device *adev = (struct audio_device *)dev;
    struct str_parms *parms;
    char *str;
    char value[32];
    int val;
    int ret;

    ALOGV("%s: enter: %s", __func__, kvpairs);
    parms = str_parms_create_str(kvpairs);

    voice_set_parameters(adev, parms);
    platform_set_parameters(adev->platform, parms);

    ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_BT_NREC, value, sizeof(value));
    if (ret >= 0) {
        /* When set to false, HAL should disable EC and NS
         * But it is currently not supported.
         */
        if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0)
            adev->bluetooth_nrec = true;
        else
            adev->bluetooth_nrec = false;
    }

    ret = str_parms_get_str(parms, "screen_state", value, sizeof(value));
    if (ret >= 0) {
        if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0)
            adev->screen_off = false;
        else
            adev->screen_off = true;
    }

    ret = str_parms_get_int(parms, "rotation", &val);
    if (ret >= 0) {
        bool reverse_speakers = false;
        switch(val) {
        // FIXME: note that the code below assumes that the speakers are in the correct placement
        //   relative to the user when the device is rotated 90deg from its default rotation. This
        //   assumption is device-specific, not platform-specific like this code.
        case 270:
            reverse_speakers = true;
            break;
        case 0:
        case 90:
        case 180:
            break;
        default:
            ALOGE("%s: unexpected rotation of %d", __func__, val);
        }
        pthread_mutex_lock(&adev->lock);
        if (adev->speaker_lr_swap != reverse_speakers) {
            adev->speaker_lr_swap = reverse_speakers;
            // only update the selected device if there is active pcm playback
            struct audio_usecase *usecase;
            struct listnode *node;
            list_for_each(node, &adev->usecase_list) {
                usecase = node_to_item(node, struct audio_usecase, list);
                if (usecase->type == PCM_PLAYBACK) {
                    select_devices(adev, usecase->id);
                    break;
                }
            }
        }
        pthread_mutex_unlock(&adev->lock);
    }

    audio_extn_set_parameters(adev, parms);
    str_parms_destroy(parms);
    ALOGV("%s: exit with code(%d)", __func__, ret);
    return ret;
}

static char* adev_get_parameters(const struct audio_hw_device *dev,
                                 const char *keys)
{
    struct audio_device *adev = (struct audio_device *)dev;
    struct str_parms *reply = str_parms_create();
    struct str_parms *query = str_parms_create_str(keys);
    char *str;

    audio_extn_get_parameters(adev, query, reply);
    platform_get_parameters(adev->platform, query, reply);
    str = str_parms_to_str(reply);
    str_parms_destroy(query);
    str_parms_destroy(reply);

    ALOGV("%s: exit: returns - %s", __func__, str);
    return str;
}

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 voice_set_volume((struct audio_device *)dev, volume);
}

static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
{
    return -ENOSYS;
}

static int adev_get_master_volume(struct audio_hw_device *dev,
                                  float *volume)
{
    return -ENOSYS;
}

static int adev_set_master_mute(struct audio_hw_device *dev, bool muted)
{
    return -ENOSYS;
}

static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted)
{
    return -ENOSYS;
}

static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
{
    struct audio_device *adev = (struct audio_device *)dev;
    pthread_mutex_lock(&adev->lock);
    if (adev->mode != mode) {
        adev->mode = mode;
    }
    pthread_mutex_unlock(&adev->lock);
    return 0;
}

static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
{
    return voice_set_mic_mute((struct audio_device *)dev, state);
}

static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
{
    *state = voice_get_mic_mute((struct audio_device *)dev);
    return 0;
}

static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
                                         const struct audio_config *config)
{
    int channel_count = popcount(config->channel_mask);

    return get_input_buffer_size(config->sample_rate, config->format, channel_count);
}

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)
{
    struct audio_device *adev = (struct audio_device *)dev;
    struct stream_in *in;
    int ret, buffer_size, frame_size;
    int channel_count = popcount(config->channel_mask);

    ALOGV("%s: enter", __func__);
    *stream_in = NULL;
    if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0)
        return -EINVAL;

    in = (struct stream_in *)calloc(1, sizeof(struct stream_in));

    in->stream.common.get_sample_rate = in_get_sample_rate;
    in->stream.common.set_sample_rate = in_set_sample_rate;
    in->stream.common.get_buffer_size = in_get_buffer_size;
    in->stream.common.get_channels = in_get_channels;
    in->stream.common.get_format = in_get_format;
    in->stream.common.set_format = in_set_format;
    in->stream.common.standby = in_standby;
    in->stream.common.dump = in_dump;
    in->stream.common.set_parameters = in_set_parameters;
    in->stream.common.get_parameters = in_get_parameters;
    in->stream.common.add_audio_effect = in_add_audio_effect;
    in->stream.common.remove_audio_effect = in_remove_audio_effect;
    in->stream.set_gain = in_set_gain;
    in->stream.read = in_read;
    in->stream.get_input_frames_lost = in_get_input_frames_lost;

    in->device = devices;
    in->source = AUDIO_SOURCE_DEFAULT;
    in->dev = adev;
    in->standby = 1;
    in->channel_mask = config->channel_mask;

    /* Update config params with the requested sample rate and channels */
    in->usecase = USECASE_AUDIO_RECORD;
    in->config = pcm_config_audio_capture;
    in->config.channels = channel_count;
    in->config.rate = config->sample_rate;

    frame_size = audio_stream_frame_size((struct audio_stream *)in);
    buffer_size = get_input_buffer_size(config->sample_rate,
                                        config->format,
                                        channel_count);
    in->config.period_size = buffer_size / frame_size;

    *stream_in = &in->stream;
    ALOGV("%s: exit", __func__);
    return 0;

err_open:
    free(in);
    *stream_in = NULL;
    return ret;
}

static void adev_close_input_stream(struct audio_hw_device *dev,
                                    struct audio_stream_in *stream)
{
    ALOGV("%s", __func__);

    in_standby(&stream->common);
    free(stream);

    return;
}

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;
    audio_route_free(adev->audio_route);
    free(adev->snd_dev_ref_cnt);
    platform_deinit(adev->platform);
    free(device);
    return 0;
}

static int adev_open(const hw_module_t *module, const char *name,
                     hw_device_t **device)
{
    int i, ret;

    ALOGD("%s: enter", __func__);
    if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) return -EINVAL;

    pthread_mutex_lock(&adev_init_lock);
    if (is_adev_initialised == true){
            *device = &adev->device.common;
            ALOGD("%s: returning existing instance of adev", __func__);
            ALOGD("%s: exit", __func__);
            pthread_mutex_unlock(&adev_init_lock);
            return 0;
    }

    adev = calloc(1, sizeof(struct audio_device));

    adev->device.common.tag = HARDWARE_DEVICE_TAG;
    adev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
    adev->device.common.module = (struct hw_module_t *)module;
    adev->device.common.close = adev_close;

    adev->device.init_check = adev_init_check;
    adev->device.set_voice_volume = adev_set_voice_volume;
    adev->device.set_master_volume = adev_set_master_volume;
    adev->device.get_master_volume = adev_get_master_volume;
    adev->device.set_master_mute = adev_set_master_mute;
    adev->device.get_master_mute = adev_get_master_mute;
    adev->device.set_mode = adev_set_mode;
    adev->device.set_mic_mute = adev_set_mic_mute;
    adev->device.get_mic_mute = adev_get_mic_mute;
    adev->device.set_parameters = adev_set_parameters;
    adev->device.get_parameters = adev_get_parameters;
    adev->device.get_input_buffer_size = adev_get_input_buffer_size;
    adev->device.open_output_stream = adev_open_output_stream;
    adev->device.close_output_stream = adev_close_output_stream;
    adev->device.open_input_stream = adev_open_input_stream;
    adev->device.close_input_stream = adev_close_input_stream;
    adev->device.dump = adev_dump;

    /* Set the default route before the PCM stream is opened */
    adev->mode = AUDIO_MODE_NORMAL;
    adev->active_input = NULL;
    adev->primary_output = NULL;
    adev->out_device = AUDIO_DEVICE_NONE;
    adev->bluetooth_nrec = true;
    adev->acdb_settings = TTY_MODE_OFF;
    adev->snd_dev_ref_cnt = calloc(SND_DEVICE_MAX, sizeof(int));
    voice_init(adev);
    list_init(&adev->usecase_list);

    /* Loads platform specific libraries dynamically */
    adev->platform = platform_init(adev);
    if (!adev->platform) {
        free(adev->snd_dev_ref_cnt);
        free(adev);
        ALOGE("%s: Failed to init platform data, aborting.", __func__);
        *device = NULL;
        return -EINVAL;
    }
    *device = &adev->device.common;

    /* update init flag*/
    is_adev_initialised = true;
    pthread_mutex_unlock(&adev_init_lock);

    ALOGV("%s: exit", __func__);
    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 = "QCOM Audio HAL",
        .author = "Code Aurora Forum",
        .methods = &hal_module_methods,
    },
};