hal/audio_hw.c

/*
 * Copyright (c) 2013-2017, 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.
 *
 * This file was modified by DTS, Inc. The portions of the
 * code modified by DTS, Inc are copyrighted and
 * licensed separately, as follows:
 *
 * (C) 2014 DTS, Inc.
 *
 * 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 ATRACE_TAG (ATRACE_TAG_AUDIO|ATRACE_TAG_HAL)
/*#define LOG_NDEBUG 0*/
/*#define VERY_VERY_VERBOSE_LOGGING*/
#ifdef VERY_VERY_VERBOSE_LOGGING
#define ALOGVV ALOGV
#else
#define ALOGVV(a...) do { } while(0)
#endif

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

#include <cutils/log.h>
#include <cutils/trace.h>
#include <cutils/str_parms.h>
#include <cutils/properties.h>
#include <cutils/atomic.h>
#include <cutils/sched_policy.h>

#include <hardware/audio_effect.h>
#include <system/thread_defs.h>
#include <audio_effects/effect_aec.h>
#include <audio_effects/effect_ns.h>
#include <audio_utils/format.h>
#include "audio_hw.h"
#include "platform_api.h"
#include <platform.h>
#include "audio_extn.h"
#include "voice_extn.h"

#include "sound/compress_params.h"
#include "sound/asound.h"

#ifdef DYNAMIC_LOG_ENABLED
#include <log_xml_parser.h>
#define LOG_MASK HAL_MOD_FILE_AUDIO_HW
#include <log_utils.h>
#endif

#define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4
/*DIRECT PCM has same buffer sizes as DEEP Buffer*/
#define DIRECT_PCM_NUM_FRAGMENTS 2
#define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
#define DSD_VOLUME_MIN_DB (-110)

#define PROXY_OPEN_RETRY_COUNT           100
#define PROXY_OPEN_WAIT_TIME             20

#ifdef USE_LL_AS_PRIMARY_OUTPUT
#define USECASE_AUDIO_PLAYBACK_PRIMARY USECASE_AUDIO_PLAYBACK_LOW_LATENCY
#define PCM_CONFIG_AUDIO_PLAYBACK_PRIMARY pcm_config_low_latency
#else
#define USECASE_AUDIO_PLAYBACK_PRIMARY USECASE_AUDIO_PLAYBACK_DEEP_BUFFER
#define PCM_CONFIG_AUDIO_PLAYBACK_PRIMARY pcm_config_deep_buffer
#endif

#define ULL_PERIOD_SIZE (DEFAULT_OUTPUT_SAMPLING_RATE/1000)

static unsigned int configured_low_latency_capture_period_size =
        LOW_LATENCY_CAPTURE_PERIOD_SIZE;

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,
};

static int af_period_multiplier = 4;
struct pcm_config pcm_config_rt = {
    .channels = 2,
    .rate = DEFAULT_OUTPUT_SAMPLING_RATE,
    .period_size = ULL_PERIOD_SIZE, //1 ms
    .period_count = 512, //=> buffer size is 512ms
    .format = PCM_FORMAT_S16_LE,
    .start_threshold = ULL_PERIOD_SIZE*8, //8ms
    .stop_threshold = INT_MAX,
    .silence_threshold = 0,
    .silence_size = 0,
    .avail_min = ULL_PERIOD_SIZE, //1 ms
};

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,
};

struct pcm_config pcm_config_audio_capture_rt = {
    .channels = 2,
    .rate = DEFAULT_OUTPUT_SAMPLING_RATE,
    .period_size = ULL_PERIOD_SIZE,
    .period_count = 512,
    .format = PCM_FORMAT_S16_LE,
    .start_threshold = 0,
    .stop_threshold = INT_MAX,
    .silence_threshold = 0,
    .silence_size = 0,
    .avail_min = ULL_PERIOD_SIZE, //1 ms
};

#define AFE_PROXY_CHANNEL_COUNT 2
#define AFE_PROXY_SAMPLING_RATE 48000

#define AFE_PROXY_PLAYBACK_PERIOD_SIZE  768
#define AFE_PROXY_PLAYBACK_PERIOD_COUNT 4

struct pcm_config pcm_config_afe_proxy_playback = {
    .channels = AFE_PROXY_CHANNEL_COUNT,
    .rate = AFE_PROXY_SAMPLING_RATE,
    .period_size = AFE_PROXY_PLAYBACK_PERIOD_SIZE,
    .period_count = AFE_PROXY_PLAYBACK_PERIOD_COUNT,
    .format = PCM_FORMAT_S16_LE,
    .start_threshold = AFE_PROXY_PLAYBACK_PERIOD_SIZE,
    .stop_threshold = INT_MAX,
    .avail_min = AFE_PROXY_PLAYBACK_PERIOD_SIZE,
};

#define AFE_PROXY_RECORD_PERIOD_SIZE  768
#define AFE_PROXY_RECORD_PERIOD_COUNT 4

struct pcm_config pcm_config_afe_proxy_record = {
    .channels = AFE_PROXY_CHANNEL_COUNT,
    .rate = AFE_PROXY_SAMPLING_RATE,
    .period_size = AFE_PROXY_RECORD_PERIOD_SIZE,
    .period_count = AFE_PROXY_RECORD_PERIOD_COUNT,
    .format = PCM_FORMAT_S16_LE,
    .start_threshold = AFE_PROXY_RECORD_PERIOD_SIZE,
    .stop_threshold = INT_MAX,
    .avail_min = AFE_PROXY_RECORD_PERIOD_SIZE,
};

#define AUDIO_MAX_PCM_FORMATS 7

const uint32_t format_to_bitwidth_table[AUDIO_MAX_PCM_FORMATS] = {
    [AUDIO_FORMAT_DEFAULT] = 0,
    [AUDIO_FORMAT_PCM_16_BIT] = sizeof(uint16_t),
    [AUDIO_FORMAT_PCM_8_BIT] = sizeof(uint8_t),
    [AUDIO_FORMAT_PCM_32_BIT] = sizeof(uint32_t),
    [AUDIO_FORMAT_PCM_8_24_BIT] = sizeof(uint32_t),
    [AUDIO_FORMAT_PCM_FLOAT] = sizeof(float),
    [AUDIO_FORMAT_PCM_24_BIT_PACKED] = sizeof(uint8_t) * 3,
};

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_ULL]         = "audio-ull-playback",
    [USECASE_AUDIO_PLAYBACK_MULTI_CH]    = "multi-channel-playback",
    [USECASE_AUDIO_PLAYBACK_OFFLOAD] = "compress-offload-playback",
    //Enabled for Direct_PCM
    [USECASE_AUDIO_PLAYBACK_OFFLOAD2] = "compress-offload-playback2",
    [USECASE_AUDIO_PLAYBACK_OFFLOAD3] = "compress-offload-playback3",
    [USECASE_AUDIO_PLAYBACK_OFFLOAD4] = "compress-offload-playback4",
    [USECASE_AUDIO_PLAYBACK_OFFLOAD5] = "compress-offload-playback5",
    [USECASE_AUDIO_PLAYBACK_OFFLOAD6] = "compress-offload-playback6",
    [USECASE_AUDIO_PLAYBACK_OFFLOAD7] = "compress-offload-playback7",
    [USECASE_AUDIO_PLAYBACK_OFFLOAD8] = "compress-offload-playback8",
    [USECASE_AUDIO_PLAYBACK_OFFLOAD9] = "compress-offload-playback9",

    [USECASE_AUDIO_RECORD] = "audio-record",
    [USECASE_AUDIO_RECORD_COMPRESS] = "audio-record-compress",
    [USECASE_AUDIO_RECORD_COMPRESS2] = "audio-record-compress2",
    [USECASE_AUDIO_RECORD_COMPRESS3] = "audio-record-compress3",
    [USECASE_AUDIO_RECORD_COMPRESS4] = "audio-record-compress4",
    [USECASE_AUDIO_RECORD_LOW_LATENCY] = "low-latency-record",
    [USECASE_AUDIO_RECORD_FM_VIRTUAL] = "fm-virtual-record",
    [USECASE_AUDIO_PLAYBACK_FM] = "play-fm",
    [USECASE_AUDIO_HFP_SCO] = "hfp-sco",
    [USECASE_AUDIO_HFP_SCO_WB] = "hfp-sco-wb",
    [USECASE_VOICE_CALL] = "voice-call",

    [USECASE_VOICE2_CALL] = "voice2-call",
    [USECASE_VOLTE_CALL] = "volte-call",
    [USECASE_QCHAT_CALL] = "qchat-call",
    [USECASE_VOWLAN_CALL] = "vowlan-call",
    [USECASE_VOICEMMODE1_CALL] = "voicemmode1-call",
    [USECASE_VOICEMMODE2_CALL] = "voicemmode2-call",
    [USECASE_COMPRESS_VOIP_CALL] = "compress-voip-call",
    [USECASE_INCALL_REC_UPLINK] = "incall-rec-uplink",
    [USECASE_INCALL_REC_DOWNLINK] = "incall-rec-downlink",
    [USECASE_INCALL_REC_UPLINK_AND_DOWNLINK] = "incall-rec-uplink-and-downlink",
    [USECASE_INCALL_REC_UPLINK_COMPRESS] = "incall-rec-uplink-compress",
    [USECASE_INCALL_REC_DOWNLINK_COMPRESS] = "incall-rec-downlink-compress",
    [USECASE_INCALL_REC_UPLINK_AND_DOWNLINK_COMPRESS] = "incall-rec-uplink-and-downlink-compress",

    [USECASE_INCALL_MUSIC_UPLINK] = "incall_music_uplink",
    [USECASE_INCALL_MUSIC_UPLINK2] = "incall_music_uplink2",
    [USECASE_AUDIO_SPKR_CALIB_RX] = "spkr-rx-calib",
    [USECASE_AUDIO_SPKR_CALIB_TX] = "spkr-vi-record",

    [USECASE_AUDIO_PLAYBACK_AFE_PROXY] = "afe-proxy-playback",
    [USECASE_AUDIO_RECORD_AFE_PROXY] = "afe-proxy-record",
    [USECASE_AUDIO_PLAYBACK_EXT_DISP_SILENCE] = "silence-playback",
    /* Transcode loopback cases */
    [USECASE_AUDIO_TRANSCODE_LOOPBACK] = "audio-transcode-loopback",
};

static const audio_usecase_t offload_usecases[] = {
    USECASE_AUDIO_PLAYBACK_OFFLOAD,
    USECASE_AUDIO_PLAYBACK_OFFLOAD2,
    USECASE_AUDIO_PLAYBACK_OFFLOAD3,
    USECASE_AUDIO_PLAYBACK_OFFLOAD4,
    USECASE_AUDIO_PLAYBACK_OFFLOAD5,
    USECASE_AUDIO_PLAYBACK_OFFLOAD6,
    USECASE_AUDIO_PLAYBACK_OFFLOAD7,
    USECASE_AUDIO_PLAYBACK_OFFLOAD8,
    USECASE_AUDIO_PLAYBACK_OFFLOAD9,
};

#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_2POINT1),
    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_QUAD),
    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_SURROUND),
    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_PENTA),
    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_6POINT1),
    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
};

static const struct string_to_enum out_formats_name_to_enum_table[] = {
    STRING_TO_ENUM(AUDIO_FORMAT_AC3),
    STRING_TO_ENUM(AUDIO_FORMAT_E_AC3),
    STRING_TO_ENUM(AUDIO_FORMAT_E_AC3_JOC),
    STRING_TO_ENUM(AUDIO_FORMAT_DOLBY_TRUEHD),
    STRING_TO_ENUM(AUDIO_FORMAT_DTS),
    STRING_TO_ENUM(AUDIO_FORMAT_DTS_HD),
    STRING_TO_ENUM(AUDIO_FORMAT_IEC61937)
};

//list of all supported sample rates by HDMI specification.
static const int out_hdmi_sample_rates[] = {
    32000, 44100, 48000, 88200, 96000, 176400, 192000,
};

static const struct string_to_enum out_hdmi_sample_rates_name_to_enum_table[] = {
    STRING_TO_ENUM(32000),
    STRING_TO_ENUM(44100),
    STRING_TO_ENUM(48000),
    STRING_TO_ENUM(88200),
    STRING_TO_ENUM(96000),
    STRING_TO_ENUM(176400),
    STRING_TO_ENUM(192000),
};

static struct audio_device *adev = NULL;
static pthread_mutex_t adev_init_lock;
static unsigned int audio_device_ref_count;
//cache last MBDRC cal step level
static int last_known_cal_step = -1 ;

static bool may_use_noirq_mode(struct audio_device *adev, audio_usecase_t uc_id,
                               int flags __unused)
{
    int dir = 0;
    switch (uc_id) {
        case USECASE_AUDIO_RECORD_LOW_LATENCY:
            dir = 1;
        case USECASE_AUDIO_PLAYBACK_ULL:
            break;
        default:
            return false;
    }

    int dev_id = platform_get_pcm_device_id(uc_id, dir == 0 ?
                                            PCM_PLAYBACK : PCM_CAPTURE);
    if (adev->adm_is_noirq_avail)
        return adev->adm_is_noirq_avail(adev->adm_data,
                                        adev->snd_card, dev_id, dir);
    return false;
}

static void register_out_stream(struct stream_out *out)
{
    struct audio_device *adev = out->dev;
    if (is_offload_usecase(out->usecase) ||
        !adev->adm_register_output_stream)
        return;

    // register stream first for backward compatibility
    adev->adm_register_output_stream(adev->adm_data,
                                     out->handle,
                                     out->flags);

    if (!adev->adm_set_config)
        return;

    if (out->realtime)
        adev->adm_set_config(adev->adm_data,
                             out->handle,
                             out->pcm, &out->config);
}

static void register_in_stream(struct stream_in *in)
{
    struct audio_device *adev = in->dev;
    if (!adev->adm_register_input_stream)
        return;

    adev->adm_register_input_stream(adev->adm_data,
                                    in->capture_handle,
                                    in->flags);

    if (!adev->adm_set_config)
        return;

    if (in->realtime)
        adev->adm_set_config(adev->adm_data,
                             in->capture_handle,
                             in->pcm,
                             &in->config);
}

static void request_out_focus(struct stream_out *out, long ns)
{
    struct audio_device *adev = out->dev;

    if (adev->adm_request_focus_v2)
        adev->adm_request_focus_v2(adev->adm_data, out->handle, ns);
    else if (adev->adm_request_focus)
        adev->adm_request_focus(adev->adm_data, out->handle);
}

static void request_in_focus(struct stream_in *in, long ns)
{
    struct audio_device *adev = in->dev;

    if (adev->adm_request_focus_v2)
        adev->adm_request_focus_v2(adev->adm_data, in->capture_handle, ns);
    else if (adev->adm_request_focus)
        adev->adm_request_focus(adev->adm_data, in->capture_handle);
}

static void release_out_focus(struct stream_out *out)
{
    struct audio_device *adev = out->dev;

    if (adev->adm_abandon_focus)
        adev->adm_abandon_focus(adev->adm_data, out->handle);
}

static void release_in_focus(struct stream_in *in)
{
    struct audio_device *adev = in->dev;
    if (adev->adm_abandon_focus)
        adev->adm_abandon_focus(adev->adm_data, in->capture_handle);
}

static int parse_snd_card_status(struct str_parms *parms, int *card,
                                 card_status_t *status)
{
    char value[32]={0};
    char state[32]={0};

    int  ret = str_parms_get_str(parms, "SND_CARD_STATUS", value, sizeof(value));
    if (ret < 0)
        return -1;

    // sscanf should be okay as value is of max length 32.
    // same as sizeof state.
    if (sscanf(value, "%d,%s", card, state) < 2)
        return -1;

    *status = !strcmp(state, "ONLINE") ? CARD_STATUS_ONLINE :
                                         CARD_STATUS_OFFLINE;
    return 0;
}

__attribute__ ((visibility ("default")))
bool audio_hw_send_gain_dep_calibration(int level) {
    bool ret_val = false;
    ALOGV("%s: called ...", __func__);

    pthread_mutex_lock(&adev_init_lock);

    if (adev != NULL && adev->platform != NULL) {
        pthread_mutex_lock(&adev->lock);
        ret_val = platform_send_gain_dep_cal(adev->platform, level);

        // cache level info for any of the use case which
        // was not started.
        last_known_cal_step = level;;

        pthread_mutex_unlock(&adev->lock);
    } else {
        ALOGE("%s: %s is NULL", __func__, adev == NULL ? "adev" : "adev->platform");
    }

    pthread_mutex_unlock(&adev_init_lock);

    return ret_val;
}

static int check_and_set_gapless_mode(struct audio_device *adev, bool enable_gapless)
{
    bool gapless_enabled = false;
    const char *mixer_ctl_name = "Compress Gapless Playback";
    struct mixer_ctl *ctl;

    ALOGV("%s:", __func__);
    gapless_enabled = property_get_bool("vendor.audio.offload.gapless.enabled", false);

    /*Disable gapless if its AV playback*/
    gapless_enabled = gapless_enabled && enable_gapless;

    ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
    if (!ctl) {
        ALOGE("%s: Could not get ctl for mixer cmd - %s",
                               __func__, mixer_ctl_name);
        return -EINVAL;
    }

    if (mixer_ctl_set_value(ctl, 0, gapless_enabled) < 0) {
        ALOGE("%s: Could not set gapless mode %d",
                       __func__, gapless_enabled);
         return -EINVAL;
    }
    return 0;
}

__attribute__ ((visibility ("default")))
int audio_hw_get_gain_level_mapping(struct amp_db_and_gain_table *mapping_tbl,
                                    int table_size) {
     int ret_val = 0;
     ALOGV("%s: enter ... ", __func__);

     pthread_mutex_lock(&adev_init_lock);
     if (adev == NULL) {
         ALOGW("%s: adev is NULL .... ", __func__);
         goto done;
     }

     pthread_mutex_lock(&adev->lock);
     ret_val = platform_get_gain_level_mapping(mapping_tbl, table_size);
     pthread_mutex_unlock(&adev->lock);
done:
     pthread_mutex_unlock(&adev_init_lock);
     ALOGV("%s: exit ... ", __func__);
     return ret_val;
}

static bool is_supported_format(audio_format_t format)
{
    if (format == AUDIO_FORMAT_MP3 ||
        format == AUDIO_FORMAT_MP2 ||
        format == AUDIO_FORMAT_AAC_LC ||
        format == AUDIO_FORMAT_AAC_HE_V1 ||
        format == AUDIO_FORMAT_AAC_HE_V2 ||
        format == AUDIO_FORMAT_AAC_ADTS_LC ||
        format == AUDIO_FORMAT_AAC_ADTS_HE_V1 ||
        format == AUDIO_FORMAT_AAC_ADTS_HE_V2 ||
        format == AUDIO_FORMAT_AAC_LATM_LC ||
        format == AUDIO_FORMAT_AAC_LATM_HE_V1 ||
        format == AUDIO_FORMAT_AAC_LATM_HE_V2 ||
        format == AUDIO_FORMAT_PCM_24_BIT_PACKED ||
        format == AUDIO_FORMAT_PCM_8_24_BIT ||
        format == AUDIO_FORMAT_PCM_FLOAT ||
        format == AUDIO_FORMAT_PCM_32_BIT ||
        format == AUDIO_FORMAT_PCM_16_BIT ||
        format == AUDIO_FORMAT_AC3 ||
        format == AUDIO_FORMAT_E_AC3 ||
        format == AUDIO_FORMAT_DOLBY_TRUEHD ||
        format == AUDIO_FORMAT_DTS ||
        format == AUDIO_FORMAT_DTS_HD ||
        format == AUDIO_FORMAT_FLAC ||
        format == AUDIO_FORMAT_ALAC ||
        format == AUDIO_FORMAT_APE ||
        format == AUDIO_FORMAT_DSD ||
        format == AUDIO_FORMAT_VORBIS ||
        format == AUDIO_FORMAT_WMA ||
        format == AUDIO_FORMAT_WMA_PRO ||
        format == AUDIO_FORMAT_APTX ||
        format == AUDIO_FORMAT_IEC61937)
           return true;

    return false;
}

static inline bool is_mmap_usecase(audio_usecase_t uc_id)
{
    return (uc_id == USECASE_AUDIO_RECORD_AFE_PROXY) ||
           (uc_id == USECASE_AUDIO_PLAYBACK_AFE_PROXY);
}

static int enable_audio_route_for_voice_usecases(struct audio_device *adev,
                                                 struct audio_usecase *uc_info)
{
    struct listnode *node;
    struct audio_usecase *usecase;

    if (uc_info == NULL)
        return -EINVAL;

    /* Re-route all voice 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);
        if ((usecase->type == VOICE_CALL) && (usecase != uc_info))
            enable_audio_route(adev, usecase);
    }
    return 0;
}

static void enable_asrc_mode(struct audio_device *adev)
{
    ALOGV("%s", __func__);
    audio_route_apply_and_update_path(adev->audio_route,
                                  "asrc-mode");
    adev->asrc_mode_enabled = true;
}

static void disable_asrc_mode(struct audio_device *adev)
{
    ALOGV("%s", __func__);
    audio_route_reset_and_update_path(adev->audio_route,
                                  "asrc-mode");
    adev->asrc_mode_enabled = false;
}

/*
 * - Enable ASRC mode for incoming mix path use case(Headphone backend)if Headphone
 *   44.1 or Native DSD backends are enabled for any of current use case.
 *   e.g. 48-> + (Naitve DSD or Headphone 44.1)
 * - Disable current mix path use case(Headphone backend) and re-enable it with
 *   ASRC mode for incoming Headphone 44.1 or Native DSD use case.
 *   e.g. Naitve DSD or Headphone 44.1 -> + 48
 */
static void check_and_set_asrc_mode(struct audio_device *adev,
                                          struct audio_usecase *uc_info,
                                          snd_device_t snd_device)
{
    ALOGV("%s snd device %d", __func__, snd_device);
    int i, num_new_devices = 0;
    snd_device_t split_new_snd_devices[SND_DEVICE_OUT_END];
    /*
    *Split snd device for new combo use case
    *e.g. Headphopne 44.1-> + Ringtone (Headphone + Speaker)
    */
    if (platform_split_snd_device(adev->platform,
                                 snd_device,
                                 &num_new_devices,
                                 split_new_snd_devices) == 0) {
        for (i = 0; i < num_new_devices; i++)
            check_and_set_asrc_mode(adev, uc_info, split_new_snd_devices[i]);
    } else {
        int new_backend_idx = platform_get_backend_index(snd_device);
        if (((new_backend_idx == HEADPHONE_BACKEND) ||
                (new_backend_idx == HEADPHONE_44_1_BACKEND) ||
                (new_backend_idx == DSD_NATIVE_BACKEND)) &&
                !adev->asrc_mode_enabled) {
            struct listnode *node = NULL;
            struct audio_usecase *uc = NULL;
            struct stream_out *curr_out = NULL;
            int usecase_backend_idx = DEFAULT_CODEC_BACKEND;
            int i, num_devices, ret = 0;
            snd_device_t split_snd_devices[SND_DEVICE_OUT_END];

            list_for_each(node, &adev->usecase_list) {
                uc = node_to_item(node, struct audio_usecase, list);
                curr_out = (struct stream_out*) uc->stream.out;
                if (curr_out && PCM_PLAYBACK == uc->type && uc != uc_info) {
                    /*
                    *Split snd device for existing combo use case
                    *e.g. Ringtone (Headphone + Speaker) + Headphopne 44.1
                    */
                    ret = platform_split_snd_device(adev->platform,
                                             uc->out_snd_device,
                                             &num_devices,
                                             split_snd_devices);
                    if (ret < 0 || num_devices == 0) {
                        ALOGV("%s: Unable to split uc->out_snd_device: %d",__func__, uc->out_snd_device);
                        split_snd_devices[0] = uc->out_snd_device;
                        num_devices = 1;
                    }
                    for (i = 0; i < num_devices; i++) {
                        usecase_backend_idx = platform_get_backend_index(split_snd_devices[i]);
                        ALOGD("%s:snd_dev %d usecase_backend_idx %d",__func__, split_snd_devices[i],usecase_backend_idx);
                        if((new_backend_idx == HEADPHONE_BACKEND) &&
                               ((usecase_backend_idx == HEADPHONE_44_1_BACKEND) ||
                               (usecase_backend_idx == DSD_NATIVE_BACKEND))) {
                            ALOGD("%s:DSD or native stream detected enabling asrcmode in hardware",
                                  __func__);
                            enable_asrc_mode(adev);
                            break;
                        } else if(((new_backend_idx == HEADPHONE_44_1_BACKEND) ||
                                  (new_backend_idx == DSD_NATIVE_BACKEND)) &&
                                  (usecase_backend_idx == HEADPHONE_BACKEND)) {
                            ALOGD("%s:48K stream detected, disabling and enabling it with asrcmode in hardware",
                                  __func__);
                            disable_audio_route(adev, uc);
                            disable_snd_device(adev, uc->out_snd_device);
                            // Apply true-high-quality-mode if DSD or > 44.1KHz or >=24-bit
                            if (new_backend_idx == DSD_NATIVE_BACKEND)
                                audio_route_apply_and_update_path(adev->audio_route,
                                                        "hph-true-highquality-mode");
                            else if ((new_backend_idx == HEADPHONE_44_1_BACKEND) &&
                                     (curr_out->bit_width >= 24))
                                audio_route_apply_and_update_path(adev->audio_route,
                                                             "hph-highquality-mode");
                            enable_asrc_mode(adev);
                            enable_snd_device(adev, uc->out_snd_device);
                            enable_audio_route(adev, uc);
                            break;
                        }
                    }
                    // reset split devices count
                    num_devices = 0;
                }
                if (adev->asrc_mode_enabled)
                    break;
            }
        }
    }
}

int pcm_ioctl(struct pcm *pcm, int request, ...)
{
    va_list ap;
    void * arg;
    int pcm_fd = *(int*)pcm;

    va_start(ap, request);
    arg = va_arg(ap, void *);
    va_end(ap);

    return ioctl(pcm_fd, request, arg);
}

int enable_audio_route(struct audio_device *adev,
                       struct audio_usecase *usecase)
{
    snd_device_t snd_device;
    char mixer_path[MIXER_PATH_MAX_LENGTH];
    struct stream_out *out = NULL;

    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;

#ifdef DS1_DOLBY_DAP_ENABLED
    audio_extn_dolby_set_dmid(adev);
    audio_extn_dolby_set_endpoint(adev);
#endif
    audio_extn_dolby_ds2_set_endpoint(adev);
    audio_extn_sound_trigger_update_stream_status(usecase, ST_EVENT_STREAM_BUSY);
    audio_extn_listen_update_stream_status(usecase, LISTEN_EVENT_STREAM_BUSY);
    audio_extn_utils_send_app_type_cfg(adev, usecase);
    audio_extn_utils_send_audio_calibration(adev, usecase);
    if ((usecase->type == PCM_PLAYBACK) && is_offload_usecase(usecase->id)) {
        out = usecase->stream.out;
        if (out && out->compr)
            audio_extn_utils_compress_set_clk_rec_mode(usecase);
    }

    strlcpy(mixer_path, use_case_table[usecase->id], MIXER_PATH_MAX_LENGTH);
    platform_add_backend_name(mixer_path, snd_device, usecase);
    ALOGD("%s: apply mixer and update path: %s", __func__, mixer_path);
    audio_route_apply_and_update_path(adev->audio_route, mixer_path);
    ALOGV("%s: exit", __func__);
    return 0;
}

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

    if (usecase == NULL || usecase->id == USECASE_INVALID)
        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;
    strlcpy(mixer_path, use_case_table[usecase->id], MIXER_PATH_MAX_LENGTH);
    platform_add_backend_name(mixer_path, snd_device, usecase);
    ALOGD("%s: reset and update mixer path: %s", __func__, mixer_path);
    audio_route_reset_and_update_path(adev->audio_route, mixer_path);
    audio_extn_sound_trigger_update_stream_status(usecase, ST_EVENT_STREAM_FREE);
    audio_extn_listen_update_stream_status(usecase, LISTEN_EVENT_STREAM_FREE);
    ALOGV("%s: exit", __func__);
    return 0;
}

int enable_snd_device(struct audio_device *adev,
                      snd_device_t snd_device)
{
    int i, num_devices = 0;
    snd_device_t new_snd_devices[SND_DEVICE_OUT_END];
    char device_name[DEVICE_NAME_MAX_SIZE] = {0};

    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(platform_get_snd_device_name_extn(adev->platform, snd_device, device_name) < 0 ) {
        ALOGE("%s: Invalid sound device returned", __func__);
        return -EINVAL;
    }
    if (adev->snd_dev_ref_cnt[snd_device] > 1) {
        ALOGV("%s: snd_device(%d: %s) is already active",
              __func__, snd_device, device_name);
        return 0;
    }


    if (audio_extn_spkr_prot_is_enabled())
         audio_extn_spkr_prot_calib_cancel(adev);

    if (platform_can_enable_spkr_prot_on_device(snd_device) &&
         audio_extn_spkr_prot_is_enabled()) {
       if (platform_get_spkr_prot_acdb_id(snd_device) < 0) {
           adev->snd_dev_ref_cnt[snd_device]--;
           return -EINVAL;
       }
       audio_extn_dev_arbi_acquire(snd_device);
       if (audio_extn_spkr_prot_start_processing(snd_device)) {
            ALOGE("%s: spkr_start_processing failed", __func__);
            audio_extn_dev_arbi_release(snd_device);
            return -EINVAL;
        }
    } else if (platform_split_snd_device(adev->platform,
                                         snd_device,
                                         &num_devices,
                                         new_snd_devices) == 0) {
        for (i = 0; i < num_devices; i++) {
            enable_snd_device(adev, new_snd_devices[i]);
        }
    } else {
        ALOGD("%s: snd_device(%d: %s)", __func__, snd_device, device_name);


       if ((SND_DEVICE_OUT_BT_A2DP == snd_device) &&
           (audio_extn_a2dp_start_playback() < 0)) {
           ALOGE(" fail to configure A2dp control path ");
           return -EINVAL;
       }

        /* due to the possibility of calibration overwrite between listen
            and audio, notify listen hal before audio calibration is sent */
        audio_extn_sound_trigger_update_device_status(snd_device,
                                        ST_EVENT_SND_DEVICE_BUSY);
        audio_extn_listen_update_device_status(snd_device,
                                        LISTEN_EVENT_SND_DEVICE_BUSY);
        if (platform_get_snd_device_acdb_id(snd_device) < 0) {
            adev->snd_dev_ref_cnt[snd_device]--;
            audio_extn_sound_trigger_update_device_status(snd_device,
                                            ST_EVENT_SND_DEVICE_FREE);
            audio_extn_listen_update_device_status(snd_device,
                                        LISTEN_EVENT_SND_DEVICE_FREE);
            return -EINVAL;
        }
        audio_extn_dev_arbi_acquire(snd_device);
        audio_route_apply_and_update_path(adev->audio_route, device_name);

        if (SND_DEVICE_OUT_HEADPHONES == snd_device &&
            !adev->native_playback_enabled &&
            audio_is_true_native_stream_active(adev)) {
            ALOGD("%s: %d: napb: enabling native mode in hardware",
                  __func__, __LINE__);
            audio_route_apply_and_update_path(adev->audio_route,
                                              "true-native-mode");
            adev->native_playback_enabled = true;
        }
    }
    return 0;
}

int disable_snd_device(struct audio_device *adev,
                       snd_device_t snd_device)
{
    int i, num_devices = 0;
    snd_device_t new_snd_devices[SND_DEVICE_OUT_END];
    char device_name[DEVICE_NAME_MAX_SIZE] = {0};

    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(platform_get_snd_device_name_extn(adev->platform, snd_device, device_name) < 0) {
        ALOGE("%s: Invalid sound device returned", __func__);
        return -EINVAL;
    }

    if (adev->snd_dev_ref_cnt[snd_device] == 0) {
        ALOGD("%s: snd_device(%d: %s)", __func__, snd_device, device_name);

        if (platform_can_enable_spkr_prot_on_device(snd_device) &&
             audio_extn_spkr_prot_is_enabled()) {
            audio_extn_spkr_prot_stop_processing(snd_device);
        } else if (platform_split_snd_device(adev->platform,
                                             snd_device,
                                             &num_devices,
                                             new_snd_devices) == 0) {
            for (i = 0; i < num_devices; i++) {
                disable_snd_device(adev, new_snd_devices[i]);
            }
        } else {
            audio_route_reset_and_update_path(adev->audio_route, device_name);
        }

        if (SND_DEVICE_OUT_BT_A2DP == snd_device)
            audio_extn_a2dp_stop_playback();

        if (snd_device == SND_DEVICE_OUT_HDMI || snd_device == SND_DEVICE_OUT_DISPLAY_PORT)
            adev->is_channel_status_set = false;
        else if (SND_DEVICE_OUT_HEADPHONES == snd_device &&
                 adev->native_playback_enabled) {
            ALOGD("%s: %d: napb: disabling native mode in hardware",
                  __func__, __LINE__);
            audio_route_reset_and_update_path(adev->audio_route,
                                              "true-native-mode");
            adev->native_playback_enabled = false;
        } else if (SND_DEVICE_OUT_HEADPHONES == snd_device &&
                 adev->asrc_mode_enabled) {
            ALOGD("%s: %d: disabling asrc mode in hardware", __func__, __LINE__);
            disable_asrc_mode(adev);
            audio_route_apply_and_update_path(adev->audio_route, "hph-lowpower-mode");
        }

        audio_extn_dev_arbi_release(snd_device);
        audio_extn_sound_trigger_update_device_status(snd_device,
                                        ST_EVENT_SND_DEVICE_FREE);
        audio_extn_listen_update_device_status(snd_device,
                                        LISTEN_EVENT_SND_DEVICE_FREE);
    }

    return 0;
}

/*
  legend:
  uc - existing usecase
  new_uc - new usecase
  d1, d11, d2 - SND_DEVICE enums
  a1, a2 - corresponding ANDROID device enums
  B1, B2 - backend strings

case 1
  uc->dev  d1 (a1)               B1
  new_uc->dev d1 (a1), d2 (a2)   B1, B2

  resolution: disable and enable uc->dev on d1

case 2
  uc->dev d1 (a1)        B1
  new_uc->dev d11 (a1)   B1

  resolution: need to switch uc since d1 and d11 are related
  (e.g. speaker and voice-speaker)
  use ANDROID_DEVICE_OUT enums to match devices since SND_DEVICE enums may vary

case 3
  uc->dev d1 (a1)        B1
  new_uc->dev d2 (a2)    B2

  resolution: no need to switch uc

case 4
  uc->dev d1 (a1)      B1
  new_uc->dev d2 (a2)  B1

  resolution: disable enable uc-dev on d2 since backends match
  we cannot enable two streams on two different devices if they
  share the same backend. e.g. if offload is on speaker device using
  QUAD_MI2S backend and a low-latency stream is started on voice-handset
  using the same backend, offload must also be switched to voice-handset.

case 5
  uc->dev  d1 (a1)                  B1
  new_uc->dev d1 (a1), d2 (a2)      B1

  resolution: disable enable uc-dev on d2 since backends match
  we cannot enable two streams on two different devices if they
  share the same backend.

case 6
  uc->dev  d1 (a1)    B1
  new_uc->dev d2 (a1) B2

  resolution: no need to switch

case 7
  uc->dev d1 (a1), d2 (a2)       B1, B2
  new_uc->dev d1 (a1)            B1

  resolution: no need to switch

*/
static snd_device_t derive_playback_snd_device(void * platform,
                                               struct audio_usecase *uc,
                                               struct audio_usecase *new_uc,
                                               snd_device_t new_snd_device)
{
    audio_devices_t a1, a2;

    snd_device_t d1 = uc->out_snd_device;
    snd_device_t d2 = new_snd_device;

    switch (uc->type) {
        case TRANSCODE_LOOPBACK :
            a1 = uc->stream.inout->out_config.devices;
            a2 = new_uc->stream.inout->out_config.devices;
            break;
        default :
            a1 = uc->stream.out->devices;
            a2 = new_uc->stream.out->devices;
            break;
    }

    // Treat as a special case when a1 and a2 are not disjoint
    if ((a1 != a2) && (a1 & a2)) {
        snd_device_t d3[2];
        int num_devices = 0;
        int ret = platform_split_snd_device(platform,
                                            popcount(a1) > 1 ? d1 : d2,
                                            &num_devices,
                                            d3);
        if (ret < 0) {
            if (ret != -ENOSYS) {
                ALOGW("%s failed to split snd_device %d",
                      __func__,
                      popcount(a1) > 1 ? d1 : d2);
            }
            goto end;
        }

        // NB: case 7 is hypothetical and isn't a practical usecase yet.
        // But if it does happen, we need to give priority to d2 if
        // the combo devices active on the existing usecase share a backend.
        // This is because we cannot have a usecase active on a combo device
        // and a new usecase requests one device in this combo pair.
        if (platform_check_backends_match(d3[0], d3[1])) {
            return d2; // case 5
        } else {
            return d1; // case 1
        }
    } else {
        if (platform_check_backends_match(d1, d2)) {
            return d2; // case 2, 4
        } else {
            return d1; // case 6, 3
        }
    }

end:
    return d2; // return whatever was calculated before.
}

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];
    snd_device_t uc_derive_snd_device;
    snd_device_t derive_snd_device[AUDIO_USECASE_MAX];
    int i, num_uc_to_switch = 0;
    int status = 0;
    bool force_restart_session = false;
    /*
     * 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.
     */
    /*
     * This call is to check if we need to force routing for a particular stream
     * If there is a backend configuration change for the device when a
     * new stream starts, then ADM needs to be closed and re-opened with the new
     * configuraion. This call check if we need to re-route all the streams
     * associated with the backend. Touch tone + 24 bit + native playback.
     */
    bool force_routing = platform_check_and_set_codec_backend_cfg(adev, uc_info,
                         snd_device);
    /* For a2dp device reconfigure all active sessions
     * with new AFE encoder format based on a2dp state
     */
    if ((SND_DEVICE_OUT_BT_A2DP == snd_device ||
         SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP == snd_device) &&
         audio_extn_a2dp_is_force_device_switch()) {
         force_routing = true;
         force_restart_session = true;
    }
    ALOGD("%s:becf: force routing %d", __func__, force_routing);

    /* 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);

        ALOGD("%s:becf: (%d) check_usecases curr device: %s, usecase device:%s "
            "backends match %d",__func__, i,
              platform_get_snd_device_name(snd_device),
              platform_get_snd_device_name(usecase->out_snd_device),
              platform_check_backends_match(snd_device, usecase->out_snd_device));
        if ((usecase->type != PCM_CAPTURE) && (usecase != uc_info)) {
            uc_derive_snd_device = derive_playback_snd_device(adev->platform,
                                               usecase, uc_info, snd_device);
            if (((uc_derive_snd_device != usecase->out_snd_device) || force_routing) &&
                ((usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) ||
                (usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) ||
                (usecase->devices & AUDIO_DEVICE_OUT_USB_DEVICE) ||
                (usecase->devices & AUDIO_DEVICE_OUT_ALL_A2DP) ||
                (usecase->devices & AUDIO_DEVICE_OUT_ALL_SCO)) &&
                ((force_restart_session) ||
                (platform_check_backends_match(snd_device, usecase->out_snd_device)))) {
                ALOGD("%s:becf: check_usecases (%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);
                switch_device[usecase->id] = true;
                /* Enable existing usecase on derived playback device */
                derive_snd_device[usecase->id] = uc_derive_snd_device;
                num_uc_to_switch++;
            }
        }
    }

    ALOGD("%s:becf: check_usecases num.of Usecases to switch %d", __func__,
        num_uc_to_switch);

    if (num_uc_to_switch) {
        /* All streams have been de-routed. Disable the device */

        /* Make sure the previous devices to be disabled first and then enable the
           selected devices */
        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);
            }
        }

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

        /* 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 = derive_snd_device[usecase->id];
                if (usecase->type != VOICE_CALL) {
                    ALOGD("%s:becf: enabling usecase (%s) on (%s)", __func__,
                         use_case_table[usecase->id],
                         platform_get_snd_device_name(usecase->out_snd_device));
                    /* Update voc calibration before enabling VoIP route */
                    if (usecase->type == VOIP_CALL)
                        status = platform_switch_voice_call_device_post(adev->platform,
                                                                        usecase->out_snd_device,
                                                                        platform_get_input_snd_device(adev->platform, uc_info->devices));
                    enable_audio_route(adev, usecase);
                }
            }
        }
    }
}

static void check_usecases_capture_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;
    int backend_check_cond = AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND;
    int status = 0;

    bool force_routing = platform_check_and_set_capture_codec_backend_cfg(adev, uc_info,
                         snd_device);
    ALOGD("%s:becf: force routing %d", __func__, force_routing);

    /*
     * Make sure out devices is checked against out codec backend device and
     * also in devices against in codec backend. Checking out device against in
     * codec backend or vice versa causes issues.
     */
    if (uc_info->type == PCM_CAPTURE)
        backend_check_cond = AUDIO_DEVICE_IN_ALL_CODEC_BACKEND;
    /*
     * 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);
        /*
         * TODO: Enhance below condition to handle BT sco/USB multi recording
         */
        if (usecase->type != PCM_PLAYBACK &&
                usecase != uc_info &&
                (usecase->in_snd_device != snd_device || force_routing) &&
                ((uc_info->devices & backend_check_cond) &&
                 (((usecase->devices & ~AUDIO_DEVICE_BIT_IN) & AUDIO_DEVICE_IN_ALL_CODEC_BACKEND) ||
                  (usecase->type == VOIP_CALL))) &&
                (usecase->id != USECASE_AUDIO_SPKR_CALIB_TX)) {
            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);
            switch_device[usecase->id] = true;
            num_uc_to_switch++;
        }
    }

    if (num_uc_to_switch) {
        /* All streams have been de-routed. Disable the device */

        /* Make sure the previous devices to be disabled first and then enable the
           selected devices */
        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);
            }
        }

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

        /* 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;
                if (usecase->type != VOICE_CALL) {
                    /* Update voc calibration before enabling VoIP route */
                    if (usecase->type == VOIP_CALL)
                        status = platform_switch_voice_call_device_post(adev->platform,
                                                                        platform_get_output_snd_device(adev->platform, uc_info->stream.out),
                                                                        usecase->in_snd_device);
                    enable_audio_route(adev, usecase);
                }
            }
        }
    }
}

static void reset_hdmi_sink_caps(struct stream_out *out) {
    int i = 0;

    for (i = 0; i<= MAX_SUPPORTED_CHANNEL_MASKS; i++) {
        out->supported_channel_masks[i] = 0;
    }
    for (i = 0; i<= MAX_SUPPORTED_FORMATS; i++) {
        out->supported_formats[i] = 0;
    }
    for (i = 0; i<= MAX_SUPPORTED_SAMPLE_RATES; i++) {
        out->supported_sample_rates[i] = 0;
    }
}

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

    reset_hdmi_sink_caps(out);

    /* Cache ext disp type */
    if (platform_get_ext_disp_type(adev->platform) <= 0) {
        ALOGE("%s: Failed to query disp type, ret:%d", __func__, ret);
        return -EINVAL;
    }

    switch (channels) {
    case 8:
        ALOGV("%s: HDMI supports 7.1 channels", __func__);
        out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_7POINT1;
        out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_6POINT1;
    case 6:
        ALOGV("%s: HDMI supports 5.1 channels", __func__);
        out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_5POINT1;
        out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_PENTA;
        out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_QUAD;
        out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_SURROUND;
        out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_2POINT1;
        break;
    default:
        ALOGE("invalid/nonstandard channal count[%d]",channels);
        ret = -ENOSYS;
        break;
    }

    // check channel format caps
    i = 0;
    if (platform_is_edid_supported_format(out->dev->platform, AUDIO_FORMAT_AC3)) {
        ALOGV(":%s HDMI supports AC3/EAC3 formats", __func__);
        out->supported_formats[i++] = AUDIO_FORMAT_AC3;
        //Adding EAC3/EAC3_JOC formats if AC3 is supported by the sink.
        //EAC3/EAC3_JOC will be converted to AC3 for decoding if needed
        out->supported_formats[i++] = AUDIO_FORMAT_E_AC3;
        out->supported_formats[i++] = AUDIO_FORMAT_E_AC3_JOC;
    }

    if (platform_is_edid_supported_format(out->dev->platform, AUDIO_FORMAT_DOLBY_TRUEHD)) {
        ALOGV(":%s HDMI supports TRUE HD format", __func__);
        out->supported_formats[i++] = AUDIO_FORMAT_DOLBY_TRUEHD;
    }

    if (platform_is_edid_supported_format(out->dev->platform, AUDIO_FORMAT_DTS)) {
        ALOGV(":%s HDMI supports DTS format", __func__);
        out->supported_formats[i++] = AUDIO_FORMAT_DTS;
    }

    if (platform_is_edid_supported_format(out->dev->platform, AUDIO_FORMAT_DTS_HD)) {
        ALOGV(":%s HDMI supports DTS HD format", __func__);
        out->supported_formats[i++] = AUDIO_FORMAT_DTS_HD;
    }

    if (platform_is_edid_supported_format(out->dev->platform, AUDIO_FORMAT_IEC61937)) {
        ALOGV(":%s HDMI supports IEC61937 format", __func__);
        out->supported_formats[i++] = AUDIO_FORMAT_IEC61937;
    }


    // check sample rate caps
    i = 0;
    for (j = 0; j < MAX_SUPPORTED_SAMPLE_RATES; j++) {
        if (platform_is_edid_supported_sample_rate(out->dev->platform, out_hdmi_sample_rates[j])) {
            ALOGV(":%s HDMI supports sample rate:%d", __func__, out_hdmi_sample_rates[j]);
            out->supported_sample_rates[i++] = out_hdmi_sample_rates[j];
        }
    }

    return ret;
}

audio_usecase_t get_usecase_id_from_usecase_type(const struct audio_device *adev,
                                                 usecase_type_t type)
{
    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 == type) {
            ALOGV("%s: usecase id %d", __func__, usecase->id);
            return usecase->id;
        }
    }
    return USECASE_INVALID;
}

struct audio_usecase *get_usecase_from_list(const 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;
}

struct stream_in *get_next_active_input(const struct audio_device *adev)
{
    struct audio_usecase *usecase;
    struct listnode *node;

    list_for_each_reverse(node, &adev->usecase_list) {
        usecase = node_to_item(node, struct audio_usecase, list);
        if (usecase->type == PCM_CAPTURE)
            return usecase->stream.in;
    }
    return NULL;
}

/*
 * is a true native playback active
 */
bool audio_is_true_native_stream_active(struct audio_device *adev)
{
    bool active = false;
    int i = 0;
    struct listnode *node;

    if (NATIVE_AUDIO_MODE_TRUE_44_1 != platform_get_native_support()) {
        ALOGV("%s:napb: not in true mode or non hdphones device",
               __func__);
        active = false;
        goto exit;
    }

    list_for_each(node, &adev->usecase_list) {
        struct audio_usecase *uc;
        uc = node_to_item(node, struct audio_usecase, list);
        struct stream_out *curr_out =
            (struct stream_out*) uc->stream.out;

        if (curr_out && PCM_PLAYBACK == uc->type) {
            ALOGD("%s:napb: (%d) (%s)id (%d) sr %d bw "
                  "(%d) device %s", __func__, i++, use_case_table[uc->id],
                  uc->id, curr_out->sample_rate,
                  curr_out->bit_width,
                  platform_get_snd_device_name(uc->out_snd_device));

            if (is_offload_usecase(uc->id) &&
                (curr_out->sample_rate == OUTPUT_SAMPLING_RATE_44100)) {
                active = true;
                ALOGD("%s:napb:native stream detected", __func__);
            }
        }
    }
exit:
    return active;
}

uint32_t adev_get_dsp_bit_width_enforce_mode()
{
    if (adev == NULL) {
        ALOGE("%s: adev is null. Disable DSP bit width enforce mode.\n", __func__);
        return 0;
    }
    return adev->dsp_bit_width_enforce_mode;
}

static uint32_t adev_init_dsp_bit_width_enforce_mode(struct mixer *mixer)
{
    char value[PROPERTY_VALUE_MAX];
    int trial;
    uint32_t dsp_bit_width_enforce_mode = 0;

    if (!mixer) {
        ALOGE("%s: adev mixer is null. cannot update DSP bitwidth.\n",
                __func__);
        return 0;
    }

    if (property_get("persist.vendor.audio_hal.dsp_bit_width_enforce_mode",
                        value, NULL) > 0) {
        trial = atoi(value);
        switch (trial) {
        case 16:
            dsp_bit_width_enforce_mode = 16;
            break;
        case 24:
            dsp_bit_width_enforce_mode = 24;
            break;
        case 32:
            dsp_bit_width_enforce_mode = 32;
            break;
       default:
            dsp_bit_width_enforce_mode = 0;
            ALOGD("%s Dynamic DSP bitwidth config is disabled.", __func__);
            break;
        }
    }

    return dsp_bit_width_enforce_mode;
}

static void audio_enable_asm_bit_width_enforce_mode(struct mixer *mixer,
                                                uint32_t enforce_mode,
                                                bool enable)
{
    struct mixer_ctl *ctl = NULL;
    const char *mixer_ctl_name = "ASM Bit Width";
    uint32_t asm_bit_width_mode = 0;

    if (enforce_mode == 0) {
        ALOGD("%s: DSP bitwidth feature is disabled.", __func__);
        return;
    }

    ctl = mixer_get_ctl_by_name(mixer, mixer_ctl_name);
    if (!ctl) {
        ALOGE("%s: Could not get ctl for mixer cmd - %s",
                __func__, mixer_ctl_name);
        return;
    }

    if (enable)
        asm_bit_width_mode = enforce_mode;
    else
        asm_bit_width_mode = 0;

    ALOGV("%s DSP bit width feature status is %d width=%d",
        __func__, enable, asm_bit_width_mode);
    if (mixer_ctl_set_value(ctl, 0, asm_bit_width_mode) < 0)
        ALOGE("%s: Could not set ASM biwidth %d", __func__,
                asm_bit_width_mode);

    return;
}

/*
 * if native DSD playback active
 */
bool audio_is_dsd_native_stream_active(struct audio_device *adev)
{
    bool active = false;
    struct listnode *node = NULL;
    struct audio_usecase *uc = NULL;
    struct stream_out *curr_out = NULL;

    list_for_each(node, &adev->usecase_list) {
        uc = node_to_item(node, struct audio_usecase, list);
        curr_out = (struct stream_out*) uc->stream.out;

        if (curr_out && PCM_PLAYBACK == uc->type &&
               (DSD_NATIVE_BACKEND == platform_get_backend_index(uc->out_snd_device))) {
            active = true;
            ALOGV("%s:DSD playback is active", __func__);
            break;
        }
    }
    return active;
}

static bool force_device_switch(struct audio_usecase *usecase)
{
    bool ret = false;
    bool is_it_true_mode = false;

    if (is_offload_usecase(usecase->id) &&
        (usecase->stream.out) &&
        (usecase->stream.out->sample_rate == OUTPUT_SAMPLING_RATE_44100) &&
        (usecase->stream.out->devices == AUDIO_DEVICE_OUT_WIRED_HEADSET ||
         usecase->stream.out->devices == AUDIO_DEVICE_OUT_WIRED_HEADPHONE)) {
        is_it_true_mode = (NATIVE_AUDIO_MODE_TRUE_44_1 == platform_get_native_support()? true : false);
         if ((is_it_true_mode && !adev->native_playback_enabled) ||
             (!is_it_true_mode && adev->native_playback_enabled)){
            ret = true;
            ALOGD("napb: time to toggle native mode");
        }
    }

    // Force all a2dp output devices to reconfigure for proper AFE encode format
    //Also handle a case where in earlier a2dp start failed as A2DP stream was
    //in suspended state, hence try to trigger a retry when we again get a routing request.
    if((usecase->stream.out) &&
       (usecase->stream.out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) &&
        audio_extn_a2dp_is_force_device_switch()) {
         ALOGD("Force a2dp device switch to update new encoder config");
         ret = true;
     }

    return ret;
}

bool is_btsco_device(snd_device_t out_snd_device, snd_device_t in_snd_device)
{
   bool ret=false;
   if ((out_snd_device == SND_DEVICE_OUT_BT_SCO ||
        out_snd_device == SND_DEVICE_OUT_BT_SCO_WB) ||
        in_snd_device == SND_DEVICE_IN_BT_SCO_MIC_WB_NREC ||
        in_snd_device == SND_DEVICE_IN_BT_SCO_MIC_WB ||
        in_snd_device == SND_DEVICE_IN_BT_SCO_MIC_NREC ||
        in_snd_device == SND_DEVICE_IN_BT_SCO_MIC)
        ret = true;

   return ret;
}

bool is_a2dp_device(snd_device_t out_snd_device)
{
   bool ret=false;
   if (out_snd_device == SND_DEVICE_OUT_BT_A2DP)
        ret = true;

   return ret;
}

bool is_bt_soc_on(struct audio_device *adev)
{
    struct mixer_ctl *ctl;
    char *mixer_ctl_name = "BT SOC status";
    ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
    bool bt_soc_status = true;
    if (!ctl) {
        ALOGE("%s: Could not get ctl for mixer cmd - %s",
              __func__, mixer_ctl_name);
        /*This is to ensure we dont break targets which dont have the kernel change*/
        return true;
    }
    bt_soc_status = mixer_ctl_get_value(ctl, 0);
    ALOGD("BT SOC status: %d",bt_soc_status);
    return bt_soc_status;
}

int out_standby_l(struct audio_stream *stream);

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 audio_usecase *voip_usecase = NULL;
    struct audio_usecase *hfp_usecase = NULL;
    audio_usecase_t hfp_ucid;
    int status = 0;

    ALOGD("%s for use case (%s)", __func__, use_case_table[uc_id]);

    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) ||
        (usecase->type == VOIP_CALL)  ||
        (usecase->type == PCM_HFP_CALL)) {
        if(usecase->stream.out == NULL) {
            ALOGE("%s: stream.out is NULL", __func__);
            return -EINVAL;
        }
        out_snd_device = platform_get_output_snd_device(adev->platform,
                                                        usecase->stream.out);
        in_snd_device = platform_get_input_snd_device(adev->platform, usecase->stream.out->devices);
        usecase->devices = usecase->stream.out->devices;
    } else if (usecase->type == TRANSCODE_LOOPBACK ) {
        if (usecase->stream.inout == NULL) {
            ALOGE("%s: stream.inout is NULL", __func__);
            return -EINVAL;
        }
        out_snd_device = usecase->stream.inout->out_config.devices;
        in_snd_device = usecase->stream.inout->in_config.devices;
        usecase->devices = (out_snd_device | in_snd_device);
    } 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.
         * choose voice call device only if the use case device is
         * also using the codec backend
         */
        if (voice_is_in_call(adev) && adev->mode != AUDIO_MODE_NORMAL) {
            vc_usecase = get_usecase_from_list(adev,
                                               get_usecase_id_from_usecase_type(adev, VOICE_CALL));
            if ((vc_usecase) && (((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
                                 (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND)) ||
                                 ((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
                                 (usecase->devices & AUDIO_DEVICE_IN_ALL_CODEC_BACKEND)) ||
                                (usecase->devices == AUDIO_DEVICE_IN_VOICE_CALL))) {
                in_snd_device = vc_usecase->in_snd_device;
                out_snd_device = vc_usecase->out_snd_device;
            }
        } else if (voice_extn_compress_voip_is_active(adev)) {
            bool out_snd_device_backend_match = true;
            voip_usecase = get_usecase_from_list(adev, USECASE_COMPRESS_VOIP_CALL);
            if ((voip_usecase != NULL) &&
                (usecase->type == PCM_PLAYBACK) &&
                (usecase->stream.out != NULL)) {
                out_snd_device_backend_match = platform_check_backends_match(
                                                   voip_usecase->out_snd_device,
                                                   platform_get_output_snd_device(
                                                       adev->platform,
                                                       usecase->stream.out));
            }
            if ((voip_usecase) && ((voip_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
                ((usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) ||
                  ((usecase->devices & ~AUDIO_DEVICE_BIT_IN) & AUDIO_DEVICE_IN_ALL_CODEC_BACKEND)) &&
                out_snd_device_backend_match &&
                 (voip_usecase->stream.out != adev->primary_output))) {
                    in_snd_device = voip_usecase->in_snd_device;
                    out_snd_device = voip_usecase->out_snd_device;
            }
        } else if (audio_extn_hfp_is_active(adev)) {
            hfp_ucid = audio_extn_hfp_get_usecase();
            hfp_usecase = get_usecase_from_list(adev, hfp_ucid);
            if ((hfp_usecase) && (hfp_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND)) {
                   in_snd_device = hfp_usecase->in_snd_device;
                   out_snd_device = hfp_usecase->out_snd_device;
            }
        }
        if (usecase->type == PCM_PLAYBACK) {
            if (usecase->stream.out == NULL) {
                ALOGE("%s: stream.out is NULL", __func__);
                return -EINVAL;
            }
            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);
                if (usecase->stream.out == adev->primary_output &&
                        adev->active_input &&
                        out_snd_device != usecase->out_snd_device) {
                    select_devices(adev, adev->active_input->usecase);
                }
            }
        } else if (usecase->type == PCM_CAPTURE) {
            if (usecase->stream.in == NULL) {
                ALOGE("%s: stream.in is NULL", __func__);
                return -EINVAL;
            }
            usecase->devices = usecase->stream.in->device;
            out_snd_device = SND_DEVICE_NONE;
            if (in_snd_device == SND_DEVICE_NONE) {
                audio_devices_t out_device = AUDIO_DEVICE_NONE;
                if (adev->active_input &&
                    (adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION ||
                    (adev->mode == AUDIO_MODE_IN_COMMUNICATION &&
                     adev->active_input->source == AUDIO_SOURCE_MIC)) &&
                     adev->primary_output && !adev->primary_output->standby) {
                    out_device = adev->primary_output->devices;
                    platform_set_echo_reference(adev, false, AUDIO_DEVICE_NONE);
                } else if (usecase->id == USECASE_AUDIO_RECORD_AFE_PROXY) {
                    out_device = AUDIO_DEVICE_OUT_TELEPHONY_TX;
                }
                in_snd_device = platform_get_input_snd_device(adev->platform, out_device);
            }
        }
    }

    if (out_snd_device == usecase->out_snd_device &&
        in_snd_device == usecase->in_snd_device) {

        if (!force_device_switch(usecase))
            return 0;
    }

    if ((is_btsco_device(out_snd_device,in_snd_device) && !adev->bt_sco_on) ||
         (is_a2dp_device(out_snd_device) && !audio_extn_a2dp_is_ready())) {
          ALOGD("SCO/A2DP is selected but they are not connected/ready hence dont route");
          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) &&
        (usecase->in_snd_device != SND_DEVICE_NONE) &&
        (usecase->out_snd_device != SND_DEVICE_NONE)) {
        status = platform_switch_voice_call_device_pre(adev->platform);
    }

    if (((usecase->type == VOICE_CALL) ||
         (usecase->type == VOIP_CALL)) &&
        (usecase->out_snd_device != SND_DEVICE_NONE)) {
        /* Disable sidetone only if voice/voip call already exists */
        if (voice_is_call_state_active(adev) ||
            voice_extn_compress_voip_is_started(adev))
            voice_set_sidetone(adev, usecase->out_snd_device, false);

        /* Disable aanc only if voice call exists */
        if (voice_is_call_state_active(adev))
            voice_check_and_update_aanc_path(adev, usecase->out_snd_device, false);
    }

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

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

    /* Applicable only on the targets that has external modem.
     * New device information should be sent to modem before enabling
     * the devices to reduce in-call device switch time.
     */
    if ((usecase->type == VOICE_CALL) &&
        (usecase->in_snd_device != SND_DEVICE_NONE) &&
        (usecase->out_snd_device != SND_DEVICE_NONE)) {
        status = platform_switch_voice_call_enable_device_config(adev->platform,
                                                                 out_snd_device,
                                                                 in_snd_device);
    }

    /* Enable new sound devices */
    if (out_snd_device != SND_DEVICE_NONE) {
        check_usecases_codec_backend(adev, usecase, out_snd_device);
        if (platform_check_codec_asrc_support(adev->platform))
            check_and_set_asrc_mode(adev, usecase, out_snd_device);
        enable_snd_device(adev, out_snd_device);
    }

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

    if (usecase->type == VOICE_CALL || usecase->type == VOIP_CALL) {
        status = platform_switch_voice_call_device_post(adev->platform,
                                                        out_snd_device,
                                                        in_snd_device);
        enable_audio_route_for_voice_usecases(adev, usecase);
    }

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

    audio_extn_utils_update_stream_app_type_cfg_for_usecase(adev,
                                                            usecase);
    if (usecase->type == PCM_PLAYBACK) {
        if ((24 == usecase->stream.out->bit_width) &&
                (usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER)) {
            usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
        } else if ((out_snd_device == SND_DEVICE_OUT_HDMI ||
                    out_snd_device == SND_DEVICE_OUT_USB_HEADSET ||
                    out_snd_device == SND_DEVICE_OUT_DISPLAY_PORT) &&
                   (usecase->stream.out->sample_rate >= OUTPUT_SAMPLING_RATE_44100)) {
            /*
             * To best utlize DSP, check if the stream sample rate is supported/multiple of
             * configured device sample rate, if not update the COPP rate to be equal to the
             * device sample rate, else open COPP at stream sample rate
             */
            platform_check_and_update_copp_sample_rate(adev->platform, out_snd_device,
                    usecase->stream.out->sample_rate,
                    &usecase->stream.out->app_type_cfg.sample_rate);
        } else if (((out_snd_device != SND_DEVICE_OUT_HEADPHONES_44_1 &&
                     !audio_is_true_native_stream_active(adev)) &&
                    usecase->stream.out->sample_rate == OUTPUT_SAMPLING_RATE_44100) ||
                    (usecase->stream.out->sample_rate < OUTPUT_SAMPLING_RATE_44100)) {
            usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
        }

        /* Notify device change info to effect clients registered */
        pthread_mutex_unlock(&adev->lock);
        audio_extn_gef_notify_device_config(
                usecase->stream.out->devices,
                usecase->stream.out->channel_mask,
                usecase->stream.out->app_type_cfg.sample_rate,
                platform_get_snd_device_acdb_id(usecase->out_snd_device));
        pthread_mutex_lock(&adev->lock);
    }
    enable_audio_route(adev, usecase);

    if (usecase->type == VOICE_CALL || usecase->type == VOIP_CALL) {
        /* Enable aanc only if voice call exists */
        if (voice_is_call_state_active(adev))
            voice_check_and_update_aanc_path(adev, out_snd_device, true);

        /* Enable sidetone only if other voice/voip call already exists */
        if (voice_is_call_state_active(adev) ||
            voice_extn_compress_voip_is_started(adev))
            voice_set_sidetone(adev, out_snd_device, true);
    }

    /* Applicable only on the targets that has external modem.
     * Enable device command should be sent to modem only after
     * enabling voice call mixer controls
     */
    if (usecase->type == VOICE_CALL)
        status = platform_switch_voice_call_usecase_route_post(adev->platform,
                                                               out_snd_device,
                                                               in_snd_device);

    if (is_btsco_device(out_snd_device, in_snd_device) || is_a2dp_device(out_snd_device)) {

         if (usecase->type == VOIP_CALL) {
             if (adev->active_input != NULL &&
                 !adev->active_input->standby) {
                 if (is_bt_soc_on(adev) == false){
                      ALOGD("BT SCO MIC disconnected while in connection");
                      if (adev->active_input->pcm != NULL)
                          pcm_stop(adev->active_input->pcm);
                 }
             }
             if ((usecase->stream.out != NULL) && (usecase->stream.out != adev->primary_output)
                  && usecase->stream.out->started) {
                  if (is_bt_soc_on(adev) == false) {
                      ALOGD("BT SCO/A2DP disconnected while in connection");
                      out_standby_l(&usecase->stream.out->stream.common);
                  }
             }
         } else if ((usecase->stream.out != NULL) &&
              !(usecase->stream.out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
              usecase->stream.out->started) {
              if (is_bt_soc_on(adev) == false) {
                  ALOGD("BT SCO/A2dp disconnected while in connection");
                  out_standby_l(&usecase->stream.out->stream.common);
              }
         }
    }

    ALOGD("%s: done",__func__);

    return status;
}

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

    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;
    }

    /* Close in-call recording streams */
    voice_check_and_stop_incall_rec_usecase(adev, in);

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

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

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

    adev->active_input = get_next_active_input(adev);

    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;

    int usecase = platform_update_usecase_from_source(in->source,in->usecase);
    if (get_usecase_from_list(adev, usecase) == NULL)
        in->usecase = usecase;
    ALOGD("%s: enter: stream(%p)usecase(%d: %s)",
          __func__, &in->stream, in->usecase, use_case_table[in->usecase]);


    if (CARD_STATUS_OFFLINE == in->card_status||
        CARD_STATUS_OFFLINE == adev->card_status) {
        ALOGW("in->card_status or adev->card_status offline, try again");
        ret = -EIO;
        goto error_config;
    }

    if (audio_is_bluetooth_sco_device(in->device)) {
        if (!adev->bt_sco_on) {
            ALOGE("%s: SCO profile is not ready, return error", __func__);
            ret = -EIO;
            goto error_config;
        }
    }

    /* Check if source matches incall recording usecase criteria */
    ret = voice_check_and_set_incall_rec_usecase(adev, in);
    if (ret)
        goto error_config;
    else
        ALOGV("%s: usecase(%d)", __func__, in->usecase);

    if (get_usecase_from_list(adev, in->usecase) != NULL) {
        ALOGE("%s: use case assigned already in use, stream(%p)usecase(%d: %s)",
            __func__, &in->stream, in->usecase, use_case_table[in->usecase]);
        return -EINVAL;
    }

    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));

    if (!uc_info) {
        ret = -ENOMEM;
        goto error_config;
    }

    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);
    audio_extn_perf_lock_acquire(&adev->perf_lock_handle, 0,
                                 adev->perf_lock_opts,
                                 adev->perf_lock_opts_size);
    select_devices(adev, in->usecase);

    ALOGV("%s: Opening PCM device card_id(%d) device_id(%d), channels %d format %d",
          __func__, adev->snd_card, in->pcm_device_id, in->config.channels, in->config.format);

   if (audio_extn_cin_attached_usecase(in->usecase)) {
       ret = audio_extn_cin_start_input_stream(in);
       if (ret)
           goto error_open;
       else
           goto done_open;
   }

    unsigned int flags = PCM_IN;
    unsigned int pcm_open_retry_count = 0;

    if (in->usecase == USECASE_AUDIO_RECORD_AFE_PROXY) {
        flags |= PCM_MMAP | PCM_NOIRQ;
        pcm_open_retry_count = PROXY_OPEN_RETRY_COUNT;
    } else if (in->realtime) {
        flags |= PCM_MMAP | PCM_NOIRQ;
    }

    while (1) {
        in->pcm = pcm_open(adev->snd_card, in->pcm_device_id,
                           flags, &in->config);
        if (in->pcm == NULL || !pcm_is_ready(in->pcm)) {
            ALOGE("%s: %s", __func__, pcm_get_error(in->pcm));
            if (in->pcm != NULL) {
                pcm_close(in->pcm);
                in->pcm = NULL;
            }
            if (pcm_open_retry_count-- == 0) {
                ret = -EIO;
                goto error_open;
            }
            usleep(PROXY_OPEN_WAIT_TIME * 1000);
            continue;
        }
        break;
    }

    ALOGV("%s: pcm_prepare", __func__);
    ret = pcm_prepare(in->pcm);
    if (ret < 0) {
        ALOGE("%s: pcm_prepare returned %d", __func__, ret);
        pcm_close(in->pcm);
        in->pcm = NULL;
        goto error_open;
    }

    register_in_stream(in);
    if (in->realtime) {
        ret = pcm_start(in->pcm);
        if (ret < 0)
            goto error_open;
    }

done_open:
    audio_extn_perf_lock_release(&adev->perf_lock_handle);
    ALOGD("%s: exit", __func__);

    return ret;

error_open:
    audio_extn_perf_lock_release(&adev->perf_lock_handle);
    stop_input_stream(in);
error_config:
    adev->active_input = get_next_active_input(adev);
    /*
     * sleep 50ms to allow sufficient time for kernel
     * drivers to recover incases like SSR.
     */
    usleep(50000);
    ALOGD("%s: exit: status(%d)", __func__, ret);

    return ret;
}

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);
}

/* must be called with out->lock locked */
static int send_offload_cmd_l(struct stream_out* out, int command)
{
    struct offload_cmd *cmd = (struct offload_cmd *)calloc(1, sizeof(struct offload_cmd));

    if (!cmd) {
        ALOGE("failed to allocate mem for command 0x%x", command);
        return -ENOMEM;
    }

    ALOGVV("%s %d", __func__, command);

    cmd->cmd = command;
    list_add_tail(&out->offload_cmd_list, &cmd->node);
    pthread_cond_signal(&out->offload_cond);
    return 0;
}

/* must be called iwth out->lock locked */
static void stop_compressed_output_l(struct stream_out *out)
{
    out->offload_state = OFFLOAD_STATE_IDLE;
    out->playback_started = 0;
    out->send_new_metadata = 1;
    if (out->compr != NULL) {
        compress_stop(out->compr);
        while (out->offload_thread_blocked) {
            pthread_cond_wait(&out->cond, &out->lock);
        }
    }
}

bool is_offload_usecase(audio_usecase_t uc_id)
{
    unsigned int i;
    for (i = 0; i < sizeof(offload_usecases)/sizeof(offload_usecases[0]); i++) {
        if (uc_id == offload_usecases[i])
            return true;
    }
    return false;
}

static audio_usecase_t get_offload_usecase(struct audio_device *adev, bool is_compress)
{
    audio_usecase_t ret_uc = USECASE_INVALID;
    unsigned int offload_uc_index;
    unsigned int num_usecase = sizeof(offload_usecases)/sizeof(offload_usecases[0]);
    if (!adev->multi_offload_enable) {
        if (!is_compress)
            ret_uc = USECASE_AUDIO_PLAYBACK_OFFLOAD2;
        else
            ret_uc = USECASE_AUDIO_PLAYBACK_OFFLOAD;

        pthread_mutex_lock(&adev->lock);
        if (get_usecase_from_list(adev, ret_uc) != NULL)
           ret_uc = USECASE_INVALID;
        pthread_mutex_unlock(&adev->lock);

        return ret_uc;
    }

    ALOGV("%s: num_usecase: %d", __func__, num_usecase);
    for (offload_uc_index = 0; offload_uc_index < num_usecase; offload_uc_index++) {
        if (!(adev->offload_usecases_state & (0x1 << offload_uc_index))) {
            adev->offload_usecases_state |= 0x1 << offload_uc_index;
            ret_uc = offload_usecases[offload_uc_index];
            break;
        }
    }

    ALOGV("%s: offload usecase is %d", __func__, ret_uc);
    return ret_uc;
}

static void free_offload_usecase(struct audio_device *adev,
                                 audio_usecase_t uc_id)
{
    unsigned int offload_uc_index;
    unsigned int num_usecase = sizeof(offload_usecases)/sizeof(offload_usecases[0]);

    if (!adev->multi_offload_enable)
        return;

    for (offload_uc_index = 0; offload_uc_index < num_usecase; offload_uc_index++) {
        if (offload_usecases[offload_uc_index] == uc_id) {
            adev->offload_usecases_state &= ~(0x1 << offload_uc_index);
            break;
        }
    }
    ALOGV("%s: free offload usecase %d", __func__, uc_id);
}

static void *offload_thread_loop(void *context)
{
    struct stream_out *out = (struct stream_out *) context;
    struct listnode *item;
    int ret = 0;

    setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO);
    set_sched_policy(0, SP_FOREGROUND);
    prctl(PR_SET_NAME, (unsigned long)"Offload Callback", 0, 0, 0);

    ALOGV("%s", __func__);
    lock_output_stream(out);
    for (;;) {
        struct offload_cmd *cmd = NULL;
        stream_callback_event_t event;
        bool send_callback = false;

        ALOGVV("%s offload_cmd_list %d out->offload_state %d",
              __func__, list_empty(&out->offload_cmd_list),
              out->offload_state);
        if (list_empty(&out->offload_cmd_list)) {
            ALOGV("%s SLEEPING", __func__);
            pthread_cond_wait(&out->offload_cond, &out->lock);
            ALOGV("%s RUNNING", __func__);
            continue;
        }

        item = list_head(&out->offload_cmd_list);
        cmd = node_to_item(item, struct offload_cmd, node);
        list_remove(item);

        ALOGVV("%s STATE %d CMD %d out->compr %p",
               __func__, out->offload_state, cmd->cmd, out->compr);

        if (cmd->cmd == OFFLOAD_CMD_EXIT) {
            free(cmd);
            break;
        }

        if (out->compr == NULL) {
            ALOGE("%s: Compress handle is NULL", __func__);
            free(cmd);
            pthread_cond_signal(&out->cond);
            continue;
        }
        out->offload_thread_blocked = true;
        pthread_mutex_unlock(&out->lock);
        send_callback = false;
        switch(cmd->cmd) {
        case OFFLOAD_CMD_WAIT_FOR_BUFFER:
            ALOGD("copl(%p):calling compress_wait", out);
            compress_wait(out->compr, -1);
            ALOGD("copl(%p):out of compress_wait", out);
            send_callback = true;
            event = STREAM_CBK_EVENT_WRITE_READY;
            break;
        case OFFLOAD_CMD_PARTIAL_DRAIN:
            ret = compress_next_track(out->compr);
            if(ret == 0) {
                ALOGD("copl(%p):calling compress_partial_drain", out);
                ret = compress_partial_drain(out->compr);
                ALOGD("copl(%p):out of compress_partial_drain", out);
                if (ret < 0)
                    ret = -errno;
            }
            else if (ret == -ETIMEDOUT)
                compress_drain(out->compr);
            else
                ALOGE("%s: Next track returned error %d",__func__, ret);
            if (ret != -ENETRESET) {
                send_callback = true;
                pthread_mutex_lock(&out->lock);
                out->send_new_metadata = 1;
                out->send_next_track_params = true;
                pthread_mutex_unlock(&out->lock);
                event = STREAM_CBK_EVENT_DRAIN_READY;
                ALOGV("copl(%p):send drain callback, ret %d", out, ret);
            } else
                ALOGI("%s: Block drain ready event during SSR", __func__);
            break;
        case OFFLOAD_CMD_DRAIN:
            ALOGD("copl(%p):calling compress_drain", out);
            compress_drain(out->compr);
            ALOGD("copl(%p):calling compress_drain", out);
            send_callback = true;
            event = STREAM_CBK_EVENT_DRAIN_READY;
            break;
        case OFFLOAD_CMD_ERROR:
            ALOGD("copl(%p): sending error callback to AF", out);
            send_callback = true;
            event = STREAM_CBK_EVENT_ERROR;
            break;
        default:
            ALOGE("%s unknown command received: %d", __func__, cmd->cmd);
            break;
        }
        lock_output_stream(out);
        out->offload_thread_blocked = false;
        pthread_cond_signal(&out->cond);
        if (send_callback && out->client_callback) {
            ALOGVV("%s: sending client_callback event %d", __func__, event);
            out->client_callback(event, NULL, out->client_cookie);
        }
        free(cmd);
    }

    pthread_cond_signal(&out->cond);
    while (!list_empty(&out->offload_cmd_list)) {
        item = list_head(&out->offload_cmd_list);
        list_remove(item);
        free(node_to_item(item, struct offload_cmd, node));
    }
    pthread_mutex_unlock(&out->lock);

    return NULL;
}

static int create_offload_callback_thread(struct stream_out *out)
{
    pthread_cond_init(&out->offload_cond, (const pthread_condattr_t *) NULL);
    list_init(&out->offload_cmd_list);
    pthread_create(&out->offload_thread, (const pthread_attr_t *) NULL,
                    offload_thread_loop, out);
    return 0;
}

static int destroy_offload_callback_thread(struct stream_out *out)
{
    lock_output_stream(out);
    stop_compressed_output_l(out);
    send_offload_cmd_l(out, OFFLOAD_CMD_EXIT);

    pthread_mutex_unlock(&out->lock);
    pthread_join(out->offload_thread, (void **) NULL);
    pthread_cond_destroy(&out->offload_cond);

    return 0;
}

static int stop_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)", __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;
    }

    if (is_offload_usecase(out->usecase) &&
        !(audio_extn_passthru_is_passthrough_stream(out))) {
        if (adev->visualizer_stop_output != NULL)
            adev->visualizer_stop_output(out->handle, out->pcm_device_id);

        audio_extn_dts_remove_state_notifier_node(out->usecase);

        if (adev->offload_effects_stop_output != NULL)
            adev->offload_effects_stop_output(out->handle, out->pcm_device_id);
    }

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

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

    if (is_offload_usecase(out->usecase)) {
        audio_enable_asm_bit_width_enforce_mode(adev->mixer,
                                                adev->dsp_bit_width_enforce_mode,
                                                false);
    }

    list_remove(&uc_info->list);
    free(uc_info);
    out->started = 0;
    if (is_offload_usecase(out->usecase) &&
        (audio_extn_passthru_is_passthrough_stream(out))) {
        ALOGV("Disable passthrough , reset mixer to pcm");
        /* NO_PASSTHROUGH */
        out->compr_config.codec->compr_passthr = 0;
        audio_extn_passthru_on_stop(out);
        audio_extn_dolby_set_dap_bypass(adev, DAP_STATE_ON);
    }

    /* Must be called after removing the usecase from list */
    if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)
        audio_extn_keep_alive_start();

    if (audio_extn_ip_hdlr_intf_supported(out->format) && out->ip_hdlr_handle) {
        ret = audio_extn_ip_hdlr_intf_close(out->ip_hdlr_handle, true, out);
        if (ret < 0)
            ALOGE("%s: audio_extn_ip_hdlr_intf_close failed %d",__func__, ret);
    }

    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;
    char mixer_ctl_name[128];
    struct mixer_ctl *ctl = NULL;
    char* perf_mode[] = {"ULL", "ULL_PP", "LL"};

    if ((out->usecase < 0) || (out->usecase >= AUDIO_USECASE_MAX)) {
        ret = -EINVAL;
        goto error_config;
    }

    ALOGD("%s: enter: stream(%p)usecase(%d: %s) devices(%#x)",
          __func__, &out->stream, out->usecase, use_case_table[out->usecase],
          out->devices);

    if (CARD_STATUS_OFFLINE == out->card_status ||
        CARD_STATUS_OFFLINE == adev->card_status) {
        ALOGW("out->card_status or adev->card_status offline, try again");
        ret = -EIO;
        goto error_config;
    }

    if (out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
        if (!audio_extn_a2dp_is_ready()) {
            if (out->devices & AUDIO_DEVICE_OUT_SPEAKER) {
                //combo usecase just by pass a2dp
                ALOGW("%s: A2DP profile is not ready, route it to speaker", __func__);
                out->devices = AUDIO_DEVICE_OUT_SPEAKER;
            } else {
                ALOGE("%s: A2DP profile is not ready, return error", __func__);
                ret = -EAGAIN;
                goto error_config;
            }
        }
    }
    if (out->devices & AUDIO_DEVICE_OUT_ALL_SCO) {
        if (!adev->bt_sco_on) {
            if (out->devices & AUDIO_DEVICE_OUT_SPEAKER) {
                //combo usecase just by pass a2dp
                ALOGW("%s: SCO is not connected, route it to speaker", __func__);
                out->devices = AUDIO_DEVICE_OUT_SPEAKER;
            } else {
                ALOGE("%s: SCO profile is not ready, return error", __func__);
                ret = -EAGAIN;
                goto error_config;
            }
        }
    }

    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_open;
    }

    uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));

    if (!uc_info) {
        ret = -ENOMEM;
        goto error_config;
    }

    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);

    audio_extn_perf_lock_acquire(&adev->perf_lock_handle, 0,
                                 adev->perf_lock_opts,
                                 adev->perf_lock_opts_size);

    if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
        audio_extn_keep_alive_stop();
        if (audio_extn_passthru_is_enabled() &&
            audio_extn_passthru_is_passthrough_stream(out)) {
            audio_extn_passthru_on_start(out);
            audio_extn_passthru_update_stream_configuration(adev, out);
        }
    }

    select_devices(adev, out->usecase);

    ALOGV("%s: Opening PCM device card_id(%d) device_id(%d) format(%#x)",
          __func__, adev->snd_card, out->pcm_device_id, out->config.format);
    if (!is_offload_usecase(out->usecase)) {
        unsigned int flags = PCM_OUT;
        unsigned int pcm_open_retry_count = 0;
        if (out->usecase == USECASE_AUDIO_PLAYBACK_AFE_PROXY) {
            flags |= PCM_MMAP | PCM_NOIRQ;
            pcm_open_retry_count = PROXY_OPEN_RETRY_COUNT;
        } else if (out->realtime) {
            flags |= PCM_MMAP | PCM_NOIRQ;
        } else
            flags |= PCM_MONOTONIC;

        if ((adev->vr_audio_mode_enabled) &&
            (out->flags & AUDIO_OUTPUT_FLAG_RAW)) {
            snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
                    "PCM_Dev %d Topology", out->pcm_device_id);
            ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
            if (!ctl) {
                ALOGI("%s: Could not get ctl for mixer cmd might be ULL - %s",
                      __func__, mixer_ctl_name);
            } else {
                //if success use ULLPP
                ALOGI("%s: mixer ctrl %s succeeded setting up ULL for %d",
                    __func__, mixer_ctl_name, out->pcm_device_id);
                //There is a still a possibility that some sessions
                // that request for FAST|RAW when 3D audio is active
                //can go through ULLPP. Ideally we expects apps to
                //listen to audio focus and stop concurrent playback
                //Also, we will look for mode flag (voice_in_communication)
                //before enabling the realtime flag.
                mixer_ctl_set_enum_by_string(ctl, perf_mode[1]);
            }
        }

        while (1) {
            out->pcm = pcm_open(adev->snd_card, out->pcm_device_id,
                               flags, &out->config);
            if (out->pcm == NULL || !pcm_is_ready(out->pcm)) {
                ALOGE("%s: %s", __func__, pcm_get_error(out->pcm));
                if (out->pcm != NULL) {
                    pcm_close(out->pcm);
                    out->pcm = NULL;
                }
                if (pcm_open_retry_count-- == 0) {
                    ret = -EIO;
                    goto error_open;
                }
                usleep(PROXY_OPEN_WAIT_TIME * 1000);
                continue;
            }
            break;
        }

        ALOGV("%s: pcm_prepare", __func__);
        if (pcm_is_ready(out->pcm)) {
            ret = pcm_prepare(out->pcm);
            if (ret < 0) {
                ALOGE("%s: pcm_prepare returned %d", __func__, ret);
                pcm_close(out->pcm);
                out->pcm = NULL;
                goto error_open;
            }
        }
        platform_set_stream_channel_map(adev->platform, out->channel_mask,
                   out->pcm_device_id, &out->channel_map_param.channel_map[0]);

    } else {
        platform_set_stream_channel_map(adev->platform, out->channel_mask,
                   out->pcm_device_id, &out->channel_map_param.channel_map[0]);
        audio_enable_asm_bit_width_enforce_mode(adev->mixer,
                                                adev->dsp_bit_width_enforce_mode,
                                                true);
        out->pcm = NULL;
        out->compr = compress_open(adev->snd_card,
                                   out->pcm_device_id,
                                   COMPRESS_IN, &out->compr_config);
        if (out->compr && !is_compress_ready(out->compr)) {
            ALOGE("%s: %s", __func__, compress_get_error(out->compr));
            compress_close(out->compr);
            out->compr = NULL;
            ret = -EIO;
            goto error_open;
        }
        /* compress_open sends params of the track, so reset the flag here */
        out->is_compr_metadata_avail = false;

        if (out->client_callback)
            compress_nonblock(out->compr, out->non_blocking);

        /* Since small bufs uses blocking writes, a write will be blocked
           for the default max poll time (20s) in the event of an SSR.
           Reduce the poll time to observe and deal with SSR faster.
        */
        if (!out->non_blocking) {
            compress_set_max_poll_wait(out->compr, 1000);
        }

        audio_extn_utils_compress_set_render_mode(out);
        audio_extn_utils_compress_set_clk_rec_mode(uc_info);

        audio_extn_dts_create_state_notifier_node(out->usecase);
        audio_extn_dts_notify_playback_state(out->usecase, 0, out->sample_rate,
                                             popcount(out->channel_mask),
                                             out->playback_started);

#ifdef DS1_DOLBY_DDP_ENABLED
        if (audio_extn_is_dolby_format(out->format))
            audio_extn_dolby_send_ddp_endp_params(adev);
#endif
        if (!(audio_extn_passthru_is_passthrough_stream(out)) &&
                (out->sample_rate != 176400 && out->sample_rate <= 192000)) {
            if (adev->visualizer_start_output != NULL)
                adev->visualizer_start_output(out->handle, out->pcm_device_id);
            if (adev->offload_effects_start_output != NULL)
                adev->offload_effects_start_output(out->handle, out->pcm_device_id, adev->mixer);
            audio_extn_check_and_set_dts_hpx_state(adev);
        }
    }

    if (ret == 0) {
        register_out_stream(out);
        if (out->realtime) {
            ret = pcm_start(out->pcm);
            if (ret < 0)
                goto error_open;
        }
    }

    audio_extn_perf_lock_release(&adev->perf_lock_handle);
    ALOGD("%s: exit", __func__);

    if (audio_extn_ip_hdlr_intf_supported(out->format) && out->ip_hdlr_handle) {
        ret = audio_extn_ip_hdlr_intf_open(out->ip_hdlr_handle, true, out, out->usecase);
        if (ret < 0)
            ALOGE("%s: audio_extn_ip_hdlr_intf_open failed %d",__func__, ret);
    }

    return ret;
error_open:
    audio_extn_perf_lock_release(&adev->perf_lock_handle);
    stop_output_stream(out);
error_config:
    /*
     * sleep 50ms to allow sufficient time for kernel
     * drivers to recover incases like SSR.
     */
    usleep(50000);
    return ret;
}

static int check_input_parameters(uint32_t sample_rate,
                                  audio_format_t format,
                                  int channel_count)
{
    int ret = 0;

    if (((format != AUDIO_FORMAT_PCM_16_BIT) && (format != AUDIO_FORMAT_PCM_8_24_BIT) &&
        (format != AUDIO_FORMAT_PCM_24_BIT_PACKED) && (format != AUDIO_FORMAT_PCM_32_BIT) &&
        (format != AUDIO_FORMAT_PCM_FLOAT)) &&
        !voice_extn_compress_voip_is_format_supported(format) &&
        !audio_extn_compr_cap_format_supported(format))  ret = -EINVAL;

    switch (channel_count) {
    case 1:
    case 2:
    case 3:
    case 4:
    case 6:
        break;
    default:
        ret = -EINVAL;
    }

    switch (sample_rate) {
    case 8000:
    case 11025:
    case 12000:
    case 16000:
    case 22050:
    case 24000:
    case 32000:
    case 44100:
    case 48000:
    case 96000:
    case 192000:
        break;
    default:
        ret = -EINVAL;
    }

    return ret;
}

static size_t get_input_buffer_size(uint32_t sample_rate,
                                    audio_format_t format,
                                    int channel_count,
                                    bool is_low_latency)
{
    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;
    if (is_low_latency)
        size = configured_low_latency_capture_period_size;

    size *= audio_bytes_per_sample(format) * channel_count;

    /* make sure the size is multiple of 32 bytes
     * At 48 kHz mono 16-bit PCM:
     *  5.000 ms = 240 frames = 15*16*1*2 = 480, a whole multiple of 32 (15)
     *  3.333 ms = 160 frames = 10*16*1*2 = 320, a whole multiple of 32 (10)
     */
    size += 0x1f;
    size &= ~0x1f;

    return size;
}

static size_t get_output_period_size(uint32_t sample_rate,
                                    audio_format_t format,
                                    int channel_count,
                                    int duration /*in millisecs*/)
{
    size_t size = 0;
    uint32_t bytes_per_sample = audio_bytes_per_sample(format);

    if ((duration == 0) || (sample_rate == 0) ||
        (bytes_per_sample == 0) || (channel_count == 0)) {
        ALOGW("Invalid config duration %d sr %d bps %d ch %d", duration, sample_rate,
               bytes_per_sample, channel_count);
        return -EINVAL;
    }

    size = (sample_rate *
            duration *
            bytes_per_sample *
            channel_count) / 1000;
    /*
     * To have same PCM samples for all channels, the buffer size requires to
     * be multiple of (number of channels * bytes per sample)
     * For writes to succeed, the buffer must be written at address which is multiple of 32
     */
    size = ALIGN(size, (bytes_per_sample * channel_count * 32));

    return (size/(channel_count * bytes_per_sample));
}

static uint64_t get_actual_pcm_frames_rendered(struct stream_out *out)
{
    uint64_t actual_frames_rendered = 0;
    size_t kernel_buffer_size = out->compr_config.fragment_size * out->compr_config.fragments;

    /* This adjustment accounts for buffering after app processor.
     * It is based on estimated DSP latency per use case, rather than exact.
     */
    int64_t platform_latency =  platform_render_latency(out->usecase) *
                                out->sample_rate / 1000000LL;

    /* not querying actual state of buffering in kernel as it would involve an ioctl call
     * which then needs protection, this causes delay in TS query for pcm_offload usecase
     * hence only estimate.
     */
    int64_t signed_frames = out->written - kernel_buffer_size;

    signed_frames = signed_frames / (audio_bytes_per_sample(out->format) * popcount(out->channel_mask)) - platform_latency;

    if (signed_frames > 0)
        actual_frames_rendered = signed_frames;

    ALOGVV("%s signed frames %lld out_written %lld kernel_buffer_size %d"
            "bytes/sample %zu channel count %d", __func__,(long long int)signed_frames,
             (long long int)out->written, (int)kernel_buffer_size,
             audio_bytes_per_sample(out->compr_config.codec->format),
             popcount(out->channel_mask));

    return actual_frames_rendered;
}

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

    return out->sample_rate;
}

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

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

    if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
        if (out->flags & AUDIO_OUTPUT_FLAG_TIMESTAMP)
            return out->compr_config.fragment_size - sizeof(struct snd_codec_metadata);
        else
            return out->compr_config.fragment_size;
    } else if(out->usecase == USECASE_COMPRESS_VOIP_CALL)
        return voice_extn_compress_voip_out_get_buffer_size(out);
    else if (is_offload_usecase(out->usecase) &&
             out->flags == AUDIO_OUTPUT_FLAG_DIRECT)
        return out->hal_fragment_size;

    return out->config.period_size * out->af_period_multiplier *
                audio_stream_out_frame_size((const struct audio_stream_out *)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)
{
    struct stream_out *out = (struct stream_out *)stream;

    return out->format;
}

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

static int out_standby(struct audio_stream *stream)
{
    struct stream_out *out = (struct stream_out *)stream;
    struct audio_device *adev = out->dev;

    ALOGD("%s: enter: stream (%p) usecase(%d: %s)", __func__,
          stream, out->usecase, use_case_table[out->usecase]);

    lock_output_stream(out);
    if (!out->standby) {
        if (adev->adm_deregister_stream)
            adev->adm_deregister_stream(adev->adm_data, out->handle);

        if (is_offload_usecase(out->usecase))
            stop_compressed_output_l(out);

        pthread_mutex_lock(&adev->lock);
        out->standby = true;
        if (out->usecase == USECASE_COMPRESS_VOIP_CALL) {
            voice_extn_compress_voip_close_output_stream(stream);
            out->started = 0;
            pthread_mutex_unlock(&adev->lock);
            pthread_mutex_unlock(&out->lock);
            ALOGD("VOIP output entered standby");
            return 0;
        } else if (!is_offload_usecase(out->usecase)) {
            if (out->pcm) {
                pcm_close(out->pcm);
                out->pcm = NULL;
            }
        } else {
            ALOGD("copl(%p):standby", out);
            out->send_next_track_params = false;
            out->is_compr_metadata_avail = false;
            out->gapless_mdata.encoder_delay = 0;
            out->gapless_mdata.encoder_padding = 0;
            if (out->compr != NULL) {
                compress_close(out->compr);
                out->compr = NULL;
            }
        }
        stop_output_stream(out);
        pthread_mutex_unlock(&adev->lock);
    }
    pthread_mutex_unlock(&out->lock);
    ALOGD("%s: exit", __func__);
    return 0;
}

static int out_on_error(struct audio_stream *stream)
{
    struct stream_out *out = (struct stream_out *)stream;
    bool do_standby = false;

    lock_output_stream(out);
    if (!out->standby) {
        if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
            stop_compressed_output_l(out);
            send_offload_cmd_l(out, OFFLOAD_CMD_ERROR);
        } else
            do_standby = true;
    }
    pthread_mutex_unlock(&out->lock);

    if (do_standby)
        return out_standby(&out->stream.common);

    return 0;
}

/*
 *standby implementation without locks, assumes that the callee already
 *has taken adev and out lock.
 */
int out_standby_l(struct audio_stream *stream)
{
    struct stream_out *out = (struct stream_out *)stream;
    struct audio_device *adev = out->dev;

    ALOGD("%s: enter: stream (%p) usecase(%d: %s)", __func__,
          stream, out->usecase, use_case_table[out->usecase]);

    if (!out->standby) {
        if (adev->adm_deregister_stream)
            adev->adm_deregister_stream(adev->adm_data, out->handle);

        if (is_offload_usecase(out->usecase))
            stop_compressed_output_l(out);

        out->standby = true;
        if (out->usecase == USECASE_COMPRESS_VOIP_CALL) {
            voice_extn_compress_voip_close_output_stream(stream);
            out->started = 0;
            ALOGD("VOIP output entered standby");
            return 0;
        } else if (!is_offload_usecase(out->usecase)) {
            if (out->pcm) {
                pcm_close(out->pcm);
                out->pcm = NULL;
            }
        } else {
            ALOGD("copl(%p):standby", out);
            out->send_next_track_params = false;
            out->is_compr_metadata_avail = false;
            out->gapless_mdata.encoder_delay = 0;
            out->gapless_mdata.encoder_padding = 0;
            if (out->compr != NULL) {
                compress_close(out->compr);
                out->compr = NULL;
            }
        }
        stop_output_stream(out);
    }
    ALOGD("%s: exit", __func__);
    return 0;
}

static int out_dump(const struct audio_stream *stream __unused,
                    int fd __unused)
{
    return 0;
}

static int parse_compress_metadata(struct stream_out *out, struct str_parms *parms)
{
    int ret = 0;
    char value[32];

    if (!out || !parms) {
        ALOGE("%s: return invalid ",__func__);
        return -EINVAL;
    }

    ret = audio_extn_parse_compress_metadata(out, parms);

    ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_DELAY_SAMPLES, value, sizeof(value));
    if (ret >= 0) {
        out->gapless_mdata.encoder_delay = atoi(value); //whats a good limit check?
    }
    ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_PADDING_SAMPLES, value, sizeof(value));
    if (ret >= 0) {
        out->gapless_mdata.encoder_padding = atoi(value);
    }

    ALOGV("%s new encoder delay %u and padding %u", __func__,
          out->gapless_mdata.encoder_delay, out->gapless_mdata.encoder_padding);

    return 0;
}

static bool output_drives_call(struct audio_device *adev, struct stream_out *out)
{
    return out == adev->primary_output || out == adev->voice_tx_output;
}

// note: this call is safe only if the stream_cb is
// removed first in close_output_stream (as is done now).
static void out_snd_mon_cb(void * stream, struct str_parms * parms)
{
    if (!stream || !parms)
        return;

    struct stream_out *out = (struct stream_out *)stream;
    struct audio_device *adev = out->dev;

    card_status_t status;
    int card;
    if (parse_snd_card_status(parms, &card, &status) < 0)
        return;

    pthread_mutex_lock(&adev->lock);
    bool valid_cb = (card == adev->snd_card);
    pthread_mutex_unlock(&adev->lock);

    if (!valid_cb)
        return;

    lock_output_stream(out);
    if (out->card_status != status)
        out->card_status = status;
    pthread_mutex_unlock(&out->lock);

    ALOGI("out_snd_mon_cb for card %d usecase %s, status %s", card,
          use_case_table[out->usecase],
          status == CARD_STATUS_OFFLINE ? "offline" : "online");

    if (status == CARD_STATUS_OFFLINE)
        out_on_error(stream);

    return;
}

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 str_parms *parms;
    char value[32];
    int ret = 0, val = 0, err;

    ALOGD("%s: enter: usecase(%d: %s) kvpairs: %s",
          __func__, out->usecase, use_case_table[out->usecase], kvpairs);
    parms = str_parms_create_str(kvpairs);
    if (!parms)
        goto error;
    err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
    if (err >= 0) {
        val = atoi(value);
        lock_output_stream(out);
        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) &&
                !audio_extn_passthru_is_passthrough_stream(out) &&
                (platform_get_edid_info(adev->platform) != 0) /* HDMI disconnected */) {
            val = AUDIO_DEVICE_OUT_SPEAKER;
        }
        /*
         * When A2DP is disconnected the
         * music playback is paused and the policy manager sends routing=0
         * But the audioflingercontinues to write data until standby time
         * (3sec). As BT is turned off, the write gets blocked.
         * Avoid this by routing audio to speaker until standby.
         */
        if ((out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) &&
                (val == AUDIO_DEVICE_NONE)) {
                val = AUDIO_DEVICE_OUT_SPEAKER;
        }
        /* To avoid a2dp to sco overlapping / BT device improper state
         * check with BT lib about a2dp streaming support before routing
         */
        if (val & AUDIO_DEVICE_OUT_ALL_A2DP) {
            if (!audio_extn_a2dp_is_ready()) {
                if (val & AUDIO_DEVICE_OUT_SPEAKER) {
                    //combo usecase just by pass a2dp
                    ALOGW("%s: A2DP profile is not ready,routing to speaker only", __func__);
                    val = AUDIO_DEVICE_OUT_SPEAKER;
                } else {
                    ALOGE("%s: A2DP profile is not ready,ignoring routing request", __func__);
                    /* update device to a2dp and don't route as BT returned error
                     * However it is still possible a2dp routing called because
                     * of current active device disconnection (like wired headset)
                     */
                    out->devices = val;
                    pthread_mutex_unlock(&out->lock);
                    pthread_mutex_unlock(&adev->lock);
                    goto error;
                }
            }
        }
        /*
         * 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) {
            audio_devices_t new_dev = val;
            bool same_dev = out->devices == new_dev;
            out->devices = new_dev;

            if (output_drives_call(adev, out)) {
                if(!voice_is_in_call(adev)) {
                    if (adev->mode == AUDIO_MODE_IN_CALL) {
                        adev->current_call_output = out;
                        ret = voice_start_call(adev);
                    }
                } else {
                    adev->current_call_output = out;
                    voice_update_devices_for_all_voice_usecases(adev);
                }
            }

            if (!out->standby) {
                if (!same_dev) {
                    ALOGV("update routing change");
                    audio_extn_perf_lock_acquire(&adev->perf_lock_handle, 0,
                                                 adev->perf_lock_opts,
                                                 adev->perf_lock_opts_size);
                    if (adev->adm_on_routing_change)
                        adev->adm_on_routing_change(adev->adm_data,
                                                    out->handle);
                }
                select_devices(adev, out->usecase);
                if (!same_dev)
                    audio_extn_perf_lock_release(&adev->perf_lock_handle);
            }
        }

        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);
    }
    if (is_offload_usecase(out->usecase)) {
        lock_output_stream(out);
        parse_compress_metadata(out, parms);

        audio_extn_dts_create_state_notifier_node(out->usecase);
        audio_extn_dts_notify_playback_state(out->usecase, 0, out->sample_rate,
                                                 popcount(out->channel_mask),
                                                 out->playback_started);

        pthread_mutex_unlock(&out->lock);
    }

    err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_PROFILE, value, sizeof(value));
    if (err >= 0) {
        strlcpy(out->profile, value, sizeof(out->profile));
        ALOGV("updating stream profile with value '%s'", out->profile);
        lock_output_stream(out);
        audio_extn_utils_update_stream_output_app_type_cfg(adev->platform,
                                                          &adev->streams_output_cfg_list,
                                                          out->devices, out->flags, out->format,
                                                          out->sample_rate, out->bit_width,
                                                          out->channel_mask, out->profile,
                                                          &out->app_type_cfg);
        pthread_mutex_unlock(&out->lock);
    }

    str_parms_destroy(parms);
error:
    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*) NULL;
    char value[256];
    struct str_parms *reply = str_parms_create();
    size_t i, j;
    int ret;
    bool first = true;

    if (!query || !reply) {
        if (reply) {
            str_parms_destroy(reply);
        }
        if (query) {
            str_parms_destroy(query);
        }
        ALOGE("out_get_parameters: failed to allocate mem for query or reply");
        return NULL;
    }

    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) {
                        strlcat(value, "|", sizeof(value));
                    }
                    strlcat(value, out_channels_name_to_enum_table[j].name, sizeof(value));
                    first = false;
                    break;
                }
            }
            i++;
        }
        str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value);
        str = str_parms_to_str(reply);
    } else {
        voice_extn_out_get_parameters(out, query, reply);
        str = str_parms_to_str(reply);
    }


    ret = str_parms_get_str(query, "is_direct_pcm_track", value, sizeof(value));
    if (ret >= 0) {
        value[0] = '\0';
        if (out->flags & AUDIO_OUTPUT_FLAG_DIRECT &&
            !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) {
            ALOGV("in direct_pcm");
            strlcat(value, "true", sizeof(value));
        } else {
            ALOGV("not in direct_pcm");
            strlcat(value, "false", sizeof(value));
        }
        str_parms_add_str(reply, "is_direct_pcm_track", value);
        if (str)
            free(str);
        str = str_parms_to_str(reply);
    }

    ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS, value, sizeof(value));
    if (ret >= 0) {
        value[0] = '\0';
        i = 0;
        first = true;
        while (out->supported_formats[i] != 0) {
            for (j = 0; j < ARRAY_SIZE(out_formats_name_to_enum_table); j++) {
                if (out_formats_name_to_enum_table[j].value == out->supported_formats[i]) {
                    if (!first) {
                        strlcat(value, "|", sizeof(value));
                    }
                    strlcat(value, out_formats_name_to_enum_table[j].name, sizeof(value));
                    first = false;
                    break;
                }
            }
            i++;
        }
        str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_FORMATS, value);
        if (str)
            free(str);
        str = str_parms_to_str(reply);
    }

    ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES, value, sizeof(value));
    if (ret >= 0) {
        value[0] = '\0';
        i = 0;
        first = true;
        while (out->supported_sample_rates[i] != 0) {
            for (j = 0; j < ARRAY_SIZE(out_hdmi_sample_rates_name_to_enum_table); j++) {
                if (out_hdmi_sample_rates_name_to_enum_table[j].value == out->supported_sample_rates[i]) {
                    if (!first) {
                        strlcat(value, "|", sizeof(value));
                    }
                    strlcat(value, out_hdmi_sample_rates_name_to_enum_table[j].name, sizeof(value));
                    first = false;
                    break;
                }
            }
            i++;
        }
        str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES, value);
        if (str)
            free(str);
        str = str_parms_to_str(reply);
    }

    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)
{
    uint32_t period_ms;
    struct stream_out *out = (struct stream_out *)stream;
    uint32_t latency = 0;

    if (is_offload_usecase(out->usecase)) {
        lock_output_stream(out);
        latency = audio_extn_utils_compress_get_dsp_latency(out);
        pthread_mutex_unlock(&out->lock);
    } else if (out->realtime) {
        // since the buffer won't be filled up faster than realtime,
        // return a smaller number
        if (out->config.rate)
            period_ms = (out->af_period_multiplier * out->config.period_size *
                         1000) / (out->config.rate);
        else
            period_ms = 0;
        latency = period_ms + platform_render_latency(out->usecase)/1000;
    } else {
        latency = (out->config.period_count * out->config.period_size * 1000) /
           (out->config.rate);
    }

    if ((AUDIO_DEVICE_OUT_BLUETOOTH_A2DP == out->devices) &&
            !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))
        latency += audio_extn_a2dp_get_encoder_latency();

    ALOGV("%s: Latency %d", __func__, latency);
    return latency;
}

static float AmpToDb(float amplification)
{
    float db = DSD_VOLUME_MIN_DB;
    if (amplification > 0) {
        db = 20 * log10(amplification);
        if(db < DSD_VOLUME_MIN_DB)
            return DSD_VOLUME_MIN_DB;
    }
    return db;
}

static int out_set_volume(struct audio_stream_out *stream, float left,
                          float right)
{
    struct stream_out *out = (struct stream_out *)stream;
    int volume[2];

    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;
    } else if (is_offload_usecase(out->usecase)) {
        if (audio_extn_passthru_is_passthrough_stream(out)) {
            /*
             * Set mute or umute on HDMI passthrough stream.
             * Only take left channel into account.
             * Mute is 0 and unmute 1
             */
            audio_extn_passthru_set_volume(out, (left == 0.0f));
        } else if (out->format == AUDIO_FORMAT_DSD){
            char mixer_ctl_name[128] =  "DSD Volume";
            struct audio_device *adev = out->dev;
            struct mixer_ctl *ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);

            if (!ctl) {
                ALOGE("%s: Could not get ctl for mixer cmd - %s",
                      __func__, mixer_ctl_name);
                return -EINVAL;
            }
            volume[0] = (int)(AmpToDb(left));
            volume[1] = (int)(AmpToDb(right));
            mixer_ctl_set_array(ctl, volume, sizeof(volume)/sizeof(volume[0]));
            return 0;
        } else {
            char mixer_ctl_name[128];
            struct audio_device *adev = out->dev;
            struct mixer_ctl *ctl;
            int pcm_device_id = platform_get_pcm_device_id(out->usecase,
                                                       PCM_PLAYBACK);

            snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
                     "Compress Playback %d Volume", pcm_device_id);
            ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
            if (!ctl) {
                ALOGE("%s: Could not get ctl for mixer cmd - %s",
                      __func__, mixer_ctl_name);
                return -EINVAL;
            }
            volume[0] = (int)(left * COMPRESS_PLAYBACK_VOLUME_MAX);
            volume[1] = (int)(right * COMPRESS_PLAYBACK_VOLUME_MAX);
            mixer_ctl_set_array(ctl, volume, sizeof(volume)/sizeof(volume[0]));
            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;
    ssize_t ret = 0;

    lock_output_stream(out);

    if (CARD_STATUS_OFFLINE == out->card_status) {

        if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
            /*during SSR for compress usecase we should return error to flinger*/
            ALOGD(" copl %s: sound card is not active/SSR state", __func__);
            pthread_mutex_unlock(&out->lock);
            return -ENETRESET;
        } else {
            /* increase written size during SSR to avoid mismatch
             * with the written frames count in AF
             */
            // bytes per frame
            size_t bpf = audio_bytes_per_sample(out->format) *
                         audio_channel_count_from_out_mask(out->channel_mask);
            if (bpf != 0)
                out->written += bytes / bpf;
            ALOGD(" %s: sound card is not active/SSR state", __func__);
            ret= -EIO;
            goto exit;
        }
    }

    if (audio_extn_passthru_should_drop_data(out)) {
        ALOGV(" %s : Drop data as compress passthrough session is going on", __func__);
        if ((audio_bytes_per_sample(out->format) != 0) && (out->config.channels != 0))
            out->written += bytes / (out->config.channels * audio_bytes_per_sample(out->format));
        ret = -EIO;
        goto exit;
    }

    if (out->standby) {
        out->standby = false;
        pthread_mutex_lock(&adev->lock);
        if (out->usecase == USECASE_COMPRESS_VOIP_CALL)
            ret = voice_extn_compress_voip_start_output_stream(out);
        else
            ret = start_output_stream(out);
        pthread_mutex_unlock(&adev->lock);
        /* ToDo: If use case is compress offload should return 0 */
        if (ret != 0) {
            out->standby = true;
            goto exit;
        }
        out->started = 1;
        if (last_known_cal_step != -1) {
            ALOGD("%s: retry previous failed cal level set", __func__);
            audio_hw_send_gain_dep_calibration(last_known_cal_step);
            last_known_cal_step = -1;
        }
    }

    if (adev->is_channel_status_set == false && (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)){
        audio_utils_set_hdmi_channel_status(out, (void *)buffer, bytes);
        adev->is_channel_status_set = true;
    }

    if (is_offload_usecase(out->usecase)) {
        ALOGVV("copl(%p): writing buffer (%zu bytes) to compress device", out, bytes);
        if (out->send_new_metadata) {
            ALOGD("copl(%p):send new gapless metadata", out);
            compress_set_gapless_metadata(out->compr, &out->gapless_mdata);
            out->send_new_metadata = 0;
            if (out->send_next_track_params && out->is_compr_metadata_avail) {
                ALOGD("copl(%p):send next track params in gapless", out);
                compress_set_next_track_param(out->compr, &(out->compr_config.codec->options));
                out->send_next_track_params = false;
                out->is_compr_metadata_avail = false;
            }
        }
        if (!(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
                      (out->convert_buffer) != NULL) {

            if ((bytes > out->hal_fragment_size)) {
                ALOGW("Error written bytes %zu > %d (fragment_size)",
                       bytes, out->hal_fragment_size);
                pthread_mutex_unlock(&out->lock);
                return -EINVAL;
            } else {
                audio_format_t dst_format = out->hal_op_format;
                audio_format_t src_format = out->hal_ip_format;

                uint32_t frames = bytes / format_to_bitwidth_table[src_format];
                uint32_t bytes_to_write = frames * format_to_bitwidth_table[dst_format];

                memcpy_by_audio_format(out->convert_buffer,
                                       dst_format,
                                       buffer,
                                       src_format,
                                       frames);

                ret = compress_write(out->compr, out->convert_buffer,
                                     bytes_to_write);

                /*Convert written bytes in audio flinger format*/
                if (ret > 0)
                    ret = ((ret * format_to_bitwidth_table[out->format]) /
                           format_to_bitwidth_table[dst_format]);
            }
        } else
            ret = compress_write(out->compr, buffer, bytes);

        if (ret < 0)
            ret = -errno;
        ALOGVV("%s: writing buffer (%zu bytes) to compress device returned %zd", __func__, bytes, ret);
        /*msg to cb thread only if non blocking write is enabled*/
        if (ret >= 0 && ret < (ssize_t)bytes && out->non_blocking) {
            ALOGD("No space available in compress driver, post msg to cb thread");
            send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER);
        } else if (-ENETRESET == ret) {
            ALOGE("copl %s: received sound card offline state on compress write", __func__);
            out->card_status = CARD_STATUS_OFFLINE;
            pthread_mutex_unlock(&out->lock);
            return ret;
        }
        if ( ret == (ssize_t)bytes && !out->non_blocking)
            out->written += bytes;

        /* Call compr start only when non-zero bytes of data is there to be rendered */
        if (!out->playback_started && ret > 0) {
            int status = compress_start(out->compr);
            if (status < 0) {
                ret = status;
                ALOGE("%s: compr start failed with err %d", __func__, errno);
                goto exit;
            }
            audio_extn_dts_eagle_fade(adev, true, out);
            out->playback_started = 1;
            out->offload_state = OFFLOAD_STATE_PLAYING;

            audio_extn_dts_notify_playback_state(out->usecase, 0, out->sample_rate,
                                                     popcount(out->channel_mask),
                                                     out->playback_started);
        }
        pthread_mutex_unlock(&out->lock);
        return ret;
    } else {
        if (out->pcm) {
            if (out->muted)
                memset((void *)buffer, 0, bytes);

            ALOGVV("%s: writing buffer (%zu bytes) to pcm device", __func__, bytes);

            long ns = 0;

            if (out->config.rate)
                ns = pcm_bytes_to_frames(out->pcm, bytes)*1000000000LL/
                                                     out->config.rate;

            bool use_mmap = is_mmap_usecase(out->usecase) || out->realtime;

            request_out_focus(out, ns);

            if (use_mmap)
                ret = pcm_mmap_write(out->pcm, (void *)buffer, bytes);
            else if (out->hal_op_format != out->hal_ip_format &&
                       out->convert_buffer != NULL) {

                memcpy_by_audio_format(out->convert_buffer,
                                       out->hal_op_format,
                                       buffer,
                                       out->hal_ip_format,
                                       out->config.period_size * out->config.channels);

                ret = pcm_write(out->pcm, out->convert_buffer,
                                 (out->config.period_size *
                                 out->config.channels *
                                 format_to_bitwidth_table[out->hal_op_format]));
            } else {
                /*
                 * To avoid underrun in DSP when the application is not pumping
                 * data at required rate, check for the no. of bytes and ignore
                 * pcm_write if it is less than actual buffer size.
                 * It is a work around to a change in compress VOIP driver.
                 */
                if ((out->flags & AUDIO_OUTPUT_FLAG_VOIP_RX) &&
                    bytes < (out->config.period_size * out->config.channels *
                    audio_bytes_per_sample(out->format))) {
                    size_t voip_buf_size =
                        out->config.period_size * out->config.channels *
                        audio_bytes_per_sample(out->format);
                    ALOGE("%s:VOIP underrun: bytes received %zu, required:%zu\n",
                            __func__, bytes, voip_buf_size);
                    usleep(((uint64_t)voip_buf_size - bytes) *
                           1000000 / audio_stream_out_frame_size(stream) /
                           out_get_sample_rate(&out->stream.common));
                    ret = 0;
                } else
                    ret = pcm_write(out->pcm, (void *)buffer, bytes);
            }

            release_out_focus(out);

            if (ret < 0)
                ret = -errno;
            else if (ret == 0 && (audio_bytes_per_sample(out->format) != 0))
                out->written += bytes / (out->config.channels * audio_bytes_per_sample(out->format));
            else
                ret = -EINVAL;
        }
    }

exit:
    if (-ENETRESET == ret) {
        out->card_status = CARD_STATUS_OFFLINE;
    }
    pthread_mutex_unlock(&out->lock);

    if (ret != 0) {
        if (out->pcm)
            ALOGE("%s: error %d, %s", __func__, (int)ret, pcm_get_error(out->pcm));
        if (out->usecase == USECASE_COMPRESS_VOIP_CALL) {
            pthread_mutex_lock(&adev->lock);
            voice_extn_compress_voip_close_output_stream(&out->stream.common);
            out->started = 0;
            pthread_mutex_unlock(&adev->lock);
            out->standby = true;
        }
        out_on_error(&out->stream.common);
        if (!(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))
            usleep((uint64_t)bytes * 1000000 / audio_stream_out_frame_size(stream) /
                            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)
{
    struct stream_out *out = (struct stream_out *)stream;

    if (dsp_frames == NULL)
        return -EINVAL;

    *dsp_frames = 0;
    if (is_offload_usecase(out->usecase)) {
        ssize_t ret = 0;

        /* Below piece of code is not guarded against any lock beacuse audioFliner serializes
         * this operation and adev_close_output_stream(where out gets reset).
         */
        if (!out->non_blocking && !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) {
            *dsp_frames = get_actual_pcm_frames_rendered(out);
             ALOGVV("dsp_frames %d sampleRate %d",(int)*dsp_frames,out->sample_rate);
             return 0;
        }

        lock_output_stream(out);
        if (out->compr != NULL && out->non_blocking) {
            ret = compress_get_tstamp(out->compr, (unsigned long *)dsp_frames,
                    &out->sample_rate);
            if (ret < 0)
                ret = -errno;
            ALOGVV("%s rendered frames %d sample_rate %d",
                    __func__, *dsp_frames, out->sample_rate);
        }
        if (-ENETRESET == ret) {
            ALOGE(" ERROR: sound card not active Unable to get time stamp from compress driver");
            out->card_status = CARD_STATUS_OFFLINE;
            ret = -EINVAL;
        } else if(ret < 0) {
            ALOGE(" ERROR: Unable to get time stamp from compress driver");
            ret = -EINVAL;
        } else if (out->card_status == CARD_STATUS_OFFLINE) {
            /*
             * Handle corner case where compress session is closed during SSR
             * and timestamp is queried
             */
            ALOGE(" ERROR: sound card not active, return error");
            ret = -EINVAL;
        } else {
            ret = 0;
        }
        pthread_mutex_unlock(&out->lock);
        return ret;
    } else if (audio_is_linear_pcm(out->format)) {
        *dsp_frames = out->written;
        return 0;
    } else
        return -EINVAL;
}

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

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

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

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;
    int ret = -1;
    unsigned long dsp_frames;

    /* below piece of code is not guarded against any lock because audioFliner serializes
     * this operation and adev_close_output_stream( where out gets reset).
     */
    if (is_offload_usecase(out->usecase) && !out->non_blocking &&
        !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) {
        *frames = get_actual_pcm_frames_rendered(out);
        /* this is the best we can do */
        clock_gettime(CLOCK_MONOTONIC, timestamp);
        ALOGVV("frames %lld playedat %lld",(long long int)*frames,
             timestamp->tv_sec * 1000000LL + timestamp->tv_nsec / 1000);
        return 0;
    }

    lock_output_stream(out);

    if (is_offload_usecase(out->usecase) && out->compr != NULL && out->non_blocking) {
        ret = compress_get_tstamp(out->compr, &dsp_frames,
                 &out->sample_rate);
        ALOGVV("%s rendered frames %ld sample_rate %d",
               __func__, dsp_frames, out->sample_rate);
        *frames = dsp_frames;
        if (ret < 0)
            ret = -errno;
        if (-ENETRESET == ret) {
            ALOGE(" ERROR: sound card not active Unable to get time stamp from compress driver");
            out->card_status = CARD_STATUS_OFFLINE;
            ret = -EINVAL;
        } else
            ret = 0;
         /* this is the best we can do */
        clock_gettime(CLOCK_MONOTONIC, timestamp);
    } else {
        if (out->pcm) {
            unsigned int avail;
            if (pcm_get_htimestamp(out->pcm, &avail, timestamp) == 0) {
                size_t kernel_buffer_size = out->config.period_size * out->config.period_count;
                int64_t signed_frames = out->written - kernel_buffer_size + avail;
                // This adjustment accounts for buffering after app processor.
                // It is based on estimated DSP latency per use case, rather than exact.
                signed_frames -=
                    (platform_render_latency(out->usecase) * out->sample_rate / 1000000LL);

                // It would be unusual for this value to be negative, but check just in case ...
                if (signed_frames >= 0) {
                    *frames = signed_frames;
                    ret = 0;
                }
            }
        } else if (out->card_status == CARD_STATUS_OFFLINE) {
            *frames = out->written;
            clock_gettime(CLOCK_MONOTONIC, timestamp);
            ret = 0;
        }
    }
    pthread_mutex_unlock(&out->lock);
    return ret;
}

static int out_set_callback(struct audio_stream_out *stream,
            stream_callback_t callback, void *cookie)
{
    struct stream_out *out = (struct stream_out *)stream;
    int ret;

    ALOGV("%s", __func__);
    lock_output_stream(out);
    out->client_callback = callback;
    out->client_cookie = cookie;
    if (out->adsp_hdlr_stream_handle) {
        ret = audio_extn_adsp_hdlr_stream_set_callback(
                                out->adsp_hdlr_stream_handle,
                                callback,
                                cookie);
        if (ret)
            ALOGW("%s:adsp hdlr callback registration failed %d",
                   __func__, ret);
    }
    pthread_mutex_unlock(&out->lock);
    return 0;
}

static int out_pause(struct audio_stream_out* stream)
{
    struct stream_out *out = (struct stream_out *)stream;
    int status = -ENOSYS;
    ALOGV("%s", __func__);
    if (is_offload_usecase(out->usecase)) {
        ALOGD("copl(%p):pause compress driver", out);
        lock_output_stream(out);
        if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PLAYING) {
            if (out->card_status != CARD_STATUS_OFFLINE)
                status = compress_pause(out->compr);

            out->offload_state = OFFLOAD_STATE_PAUSED;

            if (audio_extn_passthru_is_active()) {
                ALOGV("offload use case, pause passthru");
                audio_extn_passthru_on_pause(out);
            }

            audio_extn_dts_eagle_fade(adev, false, out);
            audio_extn_dts_notify_playback_state(out->usecase, 0,
                                                 out->sample_rate, popcount(out->channel_mask),
                                                 0);
        }
        pthread_mutex_unlock(&out->lock);
    }
    return status;
}

static int out_resume(struct audio_stream_out* stream)
{
    struct stream_out *out = (struct stream_out *)stream;
    int status = -ENOSYS;
    ALOGV("%s", __func__);
    if (is_offload_usecase(out->usecase)) {
        ALOGD("copl(%p):resume compress driver", out);
        status = 0;
        lock_output_stream(out);
        if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PAUSED) {
            if (out->card_status != CARD_STATUS_OFFLINE) {
                status = compress_resume(out->compr);
            }
            if (!status) {
                out->offload_state = OFFLOAD_STATE_PLAYING;
            }
            audio_extn_dts_eagle_fade(adev, true, out);
            audio_extn_dts_notify_playback_state(out->usecase, 0, out->sample_rate,
                                                     popcount(out->channel_mask), 1);
        }
        pthread_mutex_unlock(&out->lock);
    }
    return status;
}

static int out_drain(struct audio_stream_out* stream, audio_drain_type_t type )
{
    struct stream_out *out = (struct stream_out *)stream;
    int status = -ENOSYS;
    ALOGV("%s", __func__);
    if (is_offload_usecase(out->usecase)) {
        lock_output_stream(out);
        if (type == AUDIO_DRAIN_EARLY_NOTIFY)
            status = send_offload_cmd_l(out, OFFLOAD_CMD_PARTIAL_DRAIN);
        else
            status = send_offload_cmd_l(out, OFFLOAD_CMD_DRAIN);
        pthread_mutex_unlock(&out->lock);
    }
    return status;
}

static int out_flush(struct audio_stream_out* stream)
{
    struct stream_out *out = (struct stream_out *)stream;
    ALOGV("%s", __func__);
    if (is_offload_usecase(out->usecase)) {
        ALOGD("copl(%p):calling compress flush", out);
        lock_output_stream(out);
        if (out->offload_state == OFFLOAD_STATE_PAUSED) {
            stop_compressed_output_l(out);
            out->written = 0;
        } else {
            ALOGW("%s called in invalid state %d", __func__, out->offload_state);
        }
        pthread_mutex_unlock(&out->lock);
        ALOGD("copl(%p):out of compress flush", out);
        return 0;
    }
    return -ENOSYS;
}

/** 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 __unused,
                              uint32_t rate __unused)
{
    return -ENOSYS;
}

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

    if(in->usecase == USECASE_COMPRESS_VOIP_CALL)
        return voice_extn_compress_voip_in_get_buffer_size(in);
    else if(audio_extn_compr_cap_usecase_supported(in->usecase))
        return audio_extn_compr_cap_get_buffer_size(in->config.format);
    else if(audio_extn_cin_attached_usecase(in->usecase))
        return audio_extn_cin_get_buffer_size(in);

    return in->config.period_size * in->af_period_multiplier *
        audio_stream_in_frame_size((const struct audio_stream_in *)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)
{
    struct stream_in *in = (struct stream_in *)stream;

    return in->format;
}

static int in_set_format(struct audio_stream *stream __unused,
                         audio_format_t format __unused)
{
    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;
    ALOGD("%s: enter: stream (%p) usecase(%d: %s)", __func__,
          stream, in->usecase, use_case_table[in->usecase]);

    lock_input_stream(in);
    if (!in->standby && in->is_st_session) {
        ALOGD("%s: sound trigger pcm stop lab", __func__);
        audio_extn_sound_trigger_stop_lab(in);
        in->standby = 1;
    }

    if (!in->standby) {
        if (adev->adm_deregister_stream)
            adev->adm_deregister_stream(adev->adm_data, in->capture_handle);

        pthread_mutex_lock(&adev->lock);
        in->standby = true;
        if (in->usecase == USECASE_COMPRESS_VOIP_CALL) {
            voice_extn_compress_voip_close_input_stream(stream);
            ALOGD("VOIP input entered standby");
        } else {
            if (audio_extn_cin_attached_usecase(in->usecase))
                audio_extn_cin_stop_input_stream(in);
            if (in->pcm) {
                pcm_close(in->pcm);
                in->pcm = NULL;
            }
            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 __unused,
                   int fd __unused)
{
    return 0;
}

static void in_snd_mon_cb(void * stream, struct str_parms * parms)
{
    if (!stream || !parms)
        return;

    struct stream_in *in = (struct stream_in *)stream;
    struct audio_device *adev = in->dev;

    card_status_t status;
    int card;
    if (parse_snd_card_status(parms, &card, &status) < 0)
        return;

    pthread_mutex_lock(&adev->lock);
    bool valid_cb = (card == adev->snd_card);
    pthread_mutex_unlock(&adev->lock);

    if (!valid_cb)
        return;

    lock_input_stream(in);
    if (in->card_status != status)
        in->card_status = status;
    pthread_mutex_unlock(&in->lock);

    ALOGW("in_snd_mon_cb for card %d usecase %s, status %s", card,
          use_case_table[in->usecase],
          status == CARD_STATUS_OFFLINE ? "offline" : "online");

    // a better solution would be to report error back to AF and let
    // it put the stream to standby
    if (status == CARD_STATUS_OFFLINE)
        in_standby(&in->stream.common);

    return;
}

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 value[32];
    int ret = 0, val = 0, err;

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

    if (!parms)
        goto error;
    lock_input_stream(in);
    pthread_mutex_lock(&adev->lock);

    err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_INPUT_SOURCE, value, sizeof(value));
    if (err >= 0) {
        val = atoi(value);
        /* no audio source uses val == 0 */
        if ((in->source != val) && (val != 0)) {
            in->source = val;
            if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) &&
                (in->dev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
                (voice_extn_compress_voip_is_format_supported(in->format)) &&
                (in->config.rate == 8000 || in->config.rate == 16000 ||
                 in->config.rate == 32000 || in->config.rate == 48000 ) &&
                (audio_channel_count_from_in_mask(in->channel_mask) == 1)) {
                err = voice_extn_compress_voip_open_input_stream(in);
                if (err != 0) {
                    ALOGE("%s: Compress voip input cannot be opened, error:%d",
                          __func__, err);
                }
            }
        }
    }

    err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
    if (err >= 0) {
        val = atoi(value);
        if (((int)in->device != val) && (val != 0)) {
            in->device = val;
            /* If recording is in progress, change the tx device to new device */
            if (!in->standby && !in->is_st_session) {
                ALOGV("update input routing change");
                if (adev->adm_on_routing_change)
                        adev->adm_on_routing_change(adev->adm_data,
                                                    in->capture_handle);
                ret = select_devices(adev, in->usecase);
            }
        }
    }

    err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_PROFILE, value, sizeof(value));
    if (err >= 0) {
        strlcpy(in->profile, value, sizeof(in->profile));
        ALOGV("updating stream profile with value '%s'", in->profile);
        audio_extn_utils_update_stream_input_app_type_cfg(adev->platform,
                                                          &adev->streams_input_cfg_list,
                                                          in->device, in->flags, in->format,
                                                          in->sample_rate, in->bit_width,
                                                          in->profile, &in->app_type_cfg);
    }

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

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

static char* in_get_parameters(const struct audio_stream *stream,
                               const char *keys)
{
    struct stream_in *in = (struct stream_in *)stream;
    struct str_parms *query = str_parms_create_str(keys);
    char *str;
    struct str_parms *reply = str_parms_create();

    if (!query || !reply) {
        if (reply) {
            str_parms_destroy(reply);
        }
        if (query) {
            str_parms_destroy(query);
        }
        ALOGE("in_get_parameters: failed to create query or reply");
        return NULL;
    }

    ALOGV("%s: enter: keys - %s", __func__, keys);

    voice_extn_in_get_parameters(in, 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 in_set_gain(struct audio_stream_in *stream __unused,
                       float gain __unused)
{
    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 ret = -1;
    size_t bytes_read = 0;

    lock_input_stream(in);

    if (in->is_st_session) {
        ALOGVV(" %s: reading on st session bytes=%zu", __func__, bytes);
        /* Read from sound trigger HAL */
        audio_extn_sound_trigger_read(in, buffer, bytes);
        pthread_mutex_unlock(&in->lock);
        return bytes;
    }

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

    // what's the duration requested by the client?
    long ns = 0;

    if (in->pcm && in->config.rate)
        ns = pcm_bytes_to_frames(in->pcm, bytes)*1000000000LL/
                                             in->config.rate;

    request_in_focus(in, ns);
    bool use_mmap = is_mmap_usecase(in->usecase) || in->realtime;

    if (audio_extn_cin_attached_usecase(in->usecase)) {
        ret = audio_extn_cin_read(in, buffer, bytes, &bytes_read);
    } else if (in->pcm) {
        if (audio_extn_ssr_get_stream() == in) {
            ret = audio_extn_ssr_read(stream, buffer, bytes);
        } else if (audio_extn_compr_cap_usecase_supported(in->usecase)) {
            ret = audio_extn_compr_cap_read(in, buffer, bytes);
        } else if (use_mmap) {
            ret = pcm_mmap_read(in->pcm, buffer, bytes);
        } else {
            ret = pcm_read(in->pcm, buffer, bytes);
            /* data from DSP comes in 24_8 format, convert it to 8_24 */
            if (!ret && bytes > 0 && (in->format == AUDIO_FORMAT_PCM_8_24_BIT)) {
                if (audio_extn_utils_convert_format_24_8_to_8_24(buffer, bytes)
                    != bytes) {
                    ret = -EINVAL;
                    goto exit;
                }
            } else if (ret < 0) {
                ret = -errno;
            }
        }
        /* bytes read is always set to bytes for non compress usecases */
        bytes_read = bytes;
    }

    release_in_focus(in);

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

exit:
    if (-ENETRESET == ret)
        in->card_status = CARD_STATUS_OFFLINE;
    pthread_mutex_unlock(&in->lock);

    if (ret != 0) {
        if (in->usecase == USECASE_COMPRESS_VOIP_CALL) {
            pthread_mutex_lock(&adev->lock);
            voice_extn_compress_voip_close_input_stream(&in->stream.common);
            pthread_mutex_unlock(&adev->lock);
            in->standby = true;
        }
        if (!audio_extn_cin_attached_usecase(in->usecase)) {
            bytes_read = bytes;
            memset(buffer, 0, bytes);
        }
        in_standby(&in->stream.common);
        ALOGV("%s: read failed status %d- sleeping for buffer duration", __func__, ret);
        usleep((uint64_t)bytes * 1000000 / audio_stream_in_frame_size(stream) /
                                   in_get_sample_rate(&in->stream.common));
    }
    return bytes_read;
}

static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream __unused)
{
    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;

    lock_input_stream(in);
    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);
    }
    if (in->enable_ns != enable &&
            (memcmp(&desc.type, FX_IID_NS, sizeof(effect_uuid_t)) == 0)) {
        in->enable_ns = 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);
}

int adev_open_output_stream(struct audio_hw_device *dev,
                                   audio_io_handle_t handle,
                                   audio_devices_t devices,
                                   audio_output_flags_t flags,
                                   struct audio_config *config,
                                   struct audio_stream_out **stream_out,
                                   const char *address __unused)
{
    struct audio_device *adev = (struct audio_device *)dev;
    struct stream_out *out;
    int ret = 0;
    audio_format_t format;
    struct adsp_hdlr_stream_cfg hdlr_stream_cfg;

    *stream_out = NULL;

    out = (struct stream_out *)calloc(1, sizeof(struct stream_out));

    ALOGD("%s: enter: format(%#x) sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x)\
        stream_handle(%p)", __func__, config->format, config->sample_rate, config->channel_mask,
        devices, flags, &out->stream);


    if (!out) {
        return -ENOMEM;
    }

    pthread_mutex_init(&out->lock, (const pthread_mutexattr_t *) NULL);
    pthread_mutex_init(&out->pre_lock, (const pthread_mutexattr_t *) NULL);
    pthread_cond_init(&out->cond, (const pthread_condattr_t *) NULL);

    if (devices == AUDIO_DEVICE_NONE)
        devices = AUDIO_DEVICE_OUT_SPEAKER;

    out->flags = flags;
    out->devices = devices;
    out->dev = adev;
    format = out->format = config->format;
    out->sample_rate = config->sample_rate;
    out->channel_mask = config->channel_mask;
    out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_STEREO;
    out->handle = handle;
    out->bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
    out->non_blocking = 0;
    out->convert_buffer = NULL;
    out->started = 0;

    if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL &&
        (flags & AUDIO_OUTPUT_FLAG_DIRECT)) {
        pthread_mutex_lock(&adev->lock);
        ALOGV("AUDIO_DEVICE_OUT_AUX_DIGITAL and DIRECT|OFFLOAD, check hdmi caps");
        ret = read_hdmi_sink_caps(out);
        pthread_mutex_unlock(&adev->lock);
        if (ret != 0) {
            if (ret == -ENOSYS) {
                /* ignore and go with default */
                ret = 0;
            } else {
                ALOGE("error reading hdmi sink caps");
                goto error_open;
            }
        }
    }

    /* Init use case and pcm_config */
    if ((out->dev->mode == AUDIO_MODE_IN_COMMUNICATION || voice_extn_compress_voip_is_active(out->dev)) &&
               (out->flags == (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_VOIP_RX)) &&
               (voice_extn_compress_voip_is_config_supported(config))) {
        ret = voice_extn_compress_voip_open_output_stream(out);
        if (ret != 0) {
            ALOGE("%s: Compress voip output cannot be opened, error:%d",
                  __func__, ret);
            goto error_open;
        }
    } else if ((out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) ||
               (out->flags == AUDIO_OUTPUT_FLAG_DIRECT)) {
        pthread_mutex_lock(&adev->lock);
        bool offline = (adev->card_status == CARD_STATUS_OFFLINE);
        pthread_mutex_unlock(&adev->lock);

        // reject offload during card offline to allow
        // fallback to s/w paths
        if (offline) {
            ret = -ENODEV;
            goto error_open;
        }

        if (config->offload_info.version != AUDIO_INFO_INITIALIZER.version ||
            config->offload_info.size != AUDIO_INFO_INITIALIZER.size) {
            ALOGE("%s: Unsupported Offload information", __func__);
            ret = -EINVAL;
            goto error_open;
        }

        if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
            if(config->offload_info.format == 0)
                config->offload_info.format = out->supported_formats[0];
            if (config->offload_info.sample_rate == 0)
                config->offload_info.sample_rate = out->supported_sample_rates[0];
        }

        if (!is_supported_format(config->offload_info.format) &&
                !audio_extn_passthru_is_supported_format(config->offload_info.format)) {
            ALOGE("%s: Unsupported audio format %x " , __func__, config->offload_info.format);
            ret = -EINVAL;
            goto error_open;
        }

        /* TrueHD only supported for 48k multiples (48k, 96k, 192k) */
        if ((config->offload_info.format == AUDIO_FORMAT_DOLBY_TRUEHD) &&
                (audio_extn_passthru_is_passthrough_stream(out)) &&
                !((config->sample_rate == 48000) ||
                  (config->sample_rate == 96000) ||
                  (config->sample_rate == 192000))) {
            ALOGE("%s: Unsupported sample rate %d for audio format %x",
                    __func__, config->sample_rate, config->offload_info.format);
            ret = -EINVAL;
            goto error_open;
        }

        out->compr_config.codec = (struct snd_codec *)
                                    calloc(1, sizeof(struct snd_codec));

        if (!out->compr_config.codec) {
            ret = -ENOMEM;
            goto error_open;
        }

        out->stream.pause = out_pause;
        out->stream.resume = out_resume;
        out->stream.flush = out_flush;
        if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
            out->stream.set_callback = out_set_callback;
            out->stream.drain = out_drain;
            out->usecase = get_offload_usecase(adev, true /* is_compress */);
            ALOGV("Compress Offload usecase .. usecase selected %d", out->usecase);
        } else {
            out->usecase = get_offload_usecase(adev, false /* is_compress */);
            ALOGV("non-offload DIRECT_usecase ... usecase selected %d ", out->usecase);
        }

        if (out->usecase == USECASE_INVALID) {
            if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL &&
                    config->format == 0 && config->sample_rate == 0 &&
                    config->channel_mask == 0) {
                ALOGI("%s dummy open to query sink capability",__func__);
                out->usecase = USECASE_AUDIO_PLAYBACK_OFFLOAD;
            } else {
                ALOGE("%s, Max allowed OFFLOAD usecase reached ... ", __func__);
                ret = -EEXIST;
                goto error_open;
            }
        }

        if (config->offload_info.channel_mask)
            out->channel_mask = config->offload_info.channel_mask;
        else if (config->channel_mask) {
            out->channel_mask = config->channel_mask;
            config->offload_info.channel_mask = config->channel_mask;
        } else {
            ALOGE("out->channel_mask not set for OFFLOAD/DIRECT usecase");
            ret = -EINVAL;
            goto error_open;
        }

        format = out->format = config->offload_info.format;
        out->sample_rate = config->offload_info.sample_rate;

        out->bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;

        out->compr_config.codec->id = get_snd_codec_id(config->offload_info.format);
        if (audio_extn_is_dolby_format(config->offload_info.format)) {
            audio_extn_dolby_send_ddp_endp_params(adev);
            audio_extn_dolby_set_dmid(adev);
        }

        out->compr_config.codec->sample_rate =
                    config->offload_info.sample_rate;
        out->compr_config.codec->bit_rate =
                    config->offload_info.bit_rate;
        out->compr_config.codec->ch_in =
                audio_channel_count_from_out_mask(out->channel_mask);
        out->compr_config.codec->ch_out = out->compr_config.codec->ch_in;
        /* Update bit width only for non passthrough usecases.
         * For passthrough usecases, the output will always be opened @16 bit
         */
        if (!audio_extn_passthru_is_passthrough_stream(out))
            out->bit_width = AUDIO_OUTPUT_BIT_WIDTH;

        if (out->flags & AUDIO_OUTPUT_FLAG_TIMESTAMP)
            out->compr_config.codec->flags |= COMPRESSED_TIMESTAMP_FLAG;
        ALOGVV("%s : out->compr_config.codec->flags -> (%#x) ", __func__, out->compr_config.codec->flags);

        /*TODO: Do we need to change it for passthrough */
        out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_RAW;

        if ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC)
             out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_RAW;
        else if ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS)
            out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_MP4ADTS;
        else if ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_LATM)
            out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_MP4LATM;

        if ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) ==
             AUDIO_FORMAT_PCM) {

            /*Based on platform support, configure appropriate alsa format for corresponding
             *hal input format.
             */
            out->compr_config.codec->format = hal_format_to_alsa(
                                              config->offload_info.format);

            out->hal_op_format = alsa_format_to_hal(
                                                  out->compr_config.codec->format);
            out->hal_ip_format = out->format;

            /*for direct non-compress playback populate bit_width based on selected alsa format as
             *hal input format and alsa format might differ based on platform support.
             */
            out->bit_width = audio_bytes_per_sample(
                             out->hal_op_format) << 3;

            out->compr_config.fragments = DIRECT_PCM_NUM_FRAGMENTS;

            /* Check if alsa session is configured with the same format as HAL input format,
             * if not then derive correct fragment size needed to accomodate the
             * conversion of HAL input format to alsa format.
             */
            audio_extn_utils_update_direct_pcm_fragment_size(out);

            /*if hal input and output fragment size is different this indicates HAL input format is
             *not same as the alsa format
             */
            if (out->hal_fragment_size != out->compr_config.fragment_size) {
                /*Allocate a buffer to convert input data to the alsa configured format.
                 *size of convert buffer is equal to the size required to hold one fragment size
                 *worth of pcm data, this is because flinger does not write more than fragment_size
                 */
                out->convert_buffer = calloc(1,out->compr_config.fragment_size);
                if (out->convert_buffer == NULL){
                    ALOGE("Allocation failed for convert buffer for size %d", out->compr_config.fragment_size);
                    ret = -ENOMEM;
                    goto error_open;
                }
            }
        } else if (audio_extn_passthru_is_passthrough_stream(out)) {
            out->compr_config.fragment_size =
                   audio_extn_passthru_get_buffer_size(&config->offload_info);
            out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
        } else {
            out->compr_config.fragment_size =
                  platform_get_compress_offload_buffer_size(&config->offload_info);
            out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
        }

        if (out->flags & AUDIO_OUTPUT_FLAG_TIMESTAMP) {
            out->compr_config.fragment_size += sizeof(struct snd_codec_metadata);
        }
        if (config->offload_info.format == AUDIO_FORMAT_FLAC)
            out->compr_config.codec->options.flac_dec.sample_size = AUDIO_OUTPUT_BIT_WIDTH;

        if (config->offload_info.format == AUDIO_FORMAT_APTX) {
            audio_extn_send_aptx_dec_bt_addr_to_dsp(out);
        }

        if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING)
            out->non_blocking = 1;

        if ((flags & AUDIO_OUTPUT_FLAG_TIMESTAMP) &&
            (flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC)) {
            out->render_mode = RENDER_MODE_AUDIO_STC_MASTER;
        } else if(flags & AUDIO_OUTPUT_FLAG_TIMESTAMP) {
            out->render_mode = RENDER_MODE_AUDIO_MASTER;
        } else {
            out->render_mode = RENDER_MODE_AUDIO_NO_TIMESTAMP;
        }

        memset(&out->channel_map_param, 0,
                sizeof(struct audio_out_channel_map_param));

        out->send_new_metadata = 1;
        out->send_next_track_params = false;
        out->is_compr_metadata_avail = false;
        out->offload_state = OFFLOAD_STATE_IDLE;
        out->playback_started = 0;

        audio_extn_dts_create_state_notifier_node(out->usecase);

        ALOGV("%s: offloaded output offload_info version %04x bit rate %d",
                __func__, config->offload_info.version,
                config->offload_info.bit_rate);

        /* Check if DSD audio format is supported in codec
         * and there is no active native DSD use case
         */

        if ((config->format == AUDIO_FORMAT_DSD) &&
                (!platform_check_codec_dsd_support(adev->platform) ||
                audio_is_dsd_native_stream_active(adev))) {
            ret = -EINVAL;
            goto error_open;
        }

        /* Disable gapless if any of the following is true
         * passthrough playback
         * AV playback
         * non compressed Direct playback
         */
        if (audio_extn_passthru_is_passthrough_stream(out) ||
                (config->format == AUDIO_FORMAT_DSD) ||
                (config->format == AUDIO_FORMAT_IEC61937) ||
                config->offload_info.has_video ||
                !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) {
            check_and_set_gapless_mode(adev, false);
        } else
            check_and_set_gapless_mode(adev, true);

        if (audio_extn_passthru_is_passthrough_stream(out)) {
            out->flags |= AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH;
        }
        if (config->format == AUDIO_FORMAT_DSD) {
            out->flags |= AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH;
            out->compr_config.codec->compr_passthr = PASSTHROUGH_DSD;
        }

        create_offload_callback_thread(out);

    } else if (out->flags & AUDIO_OUTPUT_FLAG_INCALL_MUSIC) {
        ret = voice_extn_check_and_set_incall_music_usecase(adev, out);
        if (ret != 0) {
            ALOGE("%s: Incall music delivery usecase cannot be set error:%d",
                  __func__, ret);
            goto error_open;
        }
    } else  if (out->devices == AUDIO_DEVICE_OUT_TELEPHONY_TX) {
        if (config->sample_rate == 0)
            config->sample_rate = AFE_PROXY_SAMPLING_RATE;
        if (config->sample_rate != 48000 && config->sample_rate != 16000 &&
                config->sample_rate != 8000) {
            config->sample_rate = AFE_PROXY_SAMPLING_RATE;
            ret = -EINVAL;
            goto error_open;
        }
        out->sample_rate = config->sample_rate;
        out->config.rate = config->sample_rate;
        if (config->format == AUDIO_FORMAT_DEFAULT)
            config->format = AUDIO_FORMAT_PCM_16_BIT;
        if (config->format != AUDIO_FORMAT_PCM_16_BIT) {
            config->format = AUDIO_FORMAT_PCM_16_BIT;
            ret = -EINVAL;
            goto error_open;
        }
        out->format = config->format;
        out->usecase = USECASE_AUDIO_PLAYBACK_AFE_PROXY;
        out->config = pcm_config_afe_proxy_playback;
        adev->voice_tx_output = out;
    } else {
        unsigned int channels = 0;
        /*Update config params to default if not set by the caller*/
        if (config->sample_rate == 0)
            config->sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
        if (config->channel_mask == AUDIO_CHANNEL_NONE)
            config->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
        if (config->format == AUDIO_FORMAT_DEFAULT)
            config->format = AUDIO_FORMAT_PCM_16_BIT;

        channels = audio_channel_count_from_out_mask(out->channel_mask);

        if (out->flags & AUDIO_OUTPUT_FLAG_RAW) {
            out->usecase = USECASE_AUDIO_PLAYBACK_ULL;
            out->realtime = may_use_noirq_mode(adev, USECASE_AUDIO_PLAYBACK_ULL,
                                               out->flags);
            out->config = out->realtime ? pcm_config_rt : pcm_config_low_latency;
        } else if (out->flags & AUDIO_OUTPUT_FLAG_FAST) {
            out->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY;
            out->config = pcm_config_low_latency;
        } else if (out->flags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) {
            out->usecase = USECASE_AUDIO_PLAYBACK_DEEP_BUFFER;
            out->config = pcm_config_deep_buffer;
            out->config.period_size = get_output_period_size(config->sample_rate, out->format,
                                                 channels, DEEP_BUFFER_OUTPUT_PERIOD_DURATION);
            if (out->config.period_size <= 0) {
                ALOGE("Invalid configuration period size is not valid");
                ret = -EINVAL;
                goto error_open;
            }
        } else {
            /* primary path is the default path selected if no other outputs are available/suitable */
            out->usecase = USECASE_AUDIO_PLAYBACK_PRIMARY;
            out->config = PCM_CONFIG_AUDIO_PLAYBACK_PRIMARY;
        }
        out->hal_ip_format = format = out->format;
        out->config.format = hal_format_to_pcm(out->hal_ip_format);
        out->hal_op_format = pcm_format_to_hal(out->config.format);
        out->bit_width = format_to_bitwidth_table[out->hal_op_format] << 3;
        out->config.rate = config->sample_rate;
        out->sample_rate = out->config.rate;
        out->config.channels = channels;
        if (out->hal_ip_format != out->hal_op_format) {
            uint32_t buffer_size = out->config.period_size *
                                   format_to_bitwidth_table[out->hal_op_format] *
                                   out->config.channels;
            out->convert_buffer = calloc(1, buffer_size);
            if (out->convert_buffer == NULL){
                ALOGE("Allocation failed for convert buffer for size %d",
                       out->compr_config.fragment_size);
                ret = -ENOMEM;
                goto error_open;
            }
            ALOGD("Convert buffer allocated of size %d", buffer_size);
        }
    }

    ALOGV("%s devices:%d, format:%x, out->sample_rate:%d,out->bit_width:%d out->format:%d out->flags:%x, flags: %x usecase %d",
          __func__, devices, format, out->sample_rate, out->bit_width, out->format, out->flags, flags, out->usecase);

    /* TODO remove this hardcoding and check why width is zero*/
    if (out->bit_width == 0)
        out->bit_width = 16;
    audio_extn_utils_update_stream_output_app_type_cfg(adev->platform,
                                                &adev->streams_output_cfg_list,
                                                devices, out->flags, format, out->sample_rate,
                                                out->bit_width, out->channel_mask, out->profile,
                                                &out->app_type_cfg);
    if ((out->usecase == USECASE_AUDIO_PLAYBACK_PRIMARY) ||
        (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
        /* Ensure the default output is not selected twice */
        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) &&
        (out->usecase != USECASE_COMPRESS_VOIP_CALL)) {
        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->stream.get_presentation_position = out_get_presentation_position;

    out->af_period_multiplier  = out->realtime ? af_period_multiplier : 1;
    out->standby = 1;
    /* out->muted = false; by calloc() */
    /* out->written = 0; 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);

    /*
       By locking output stream before registering, we allow the callback
       to update stream's state only after stream's initial state is set to
       adev state.
    */
    lock_output_stream(out);
    audio_extn_snd_mon_register_listener(out, out_snd_mon_cb);
    pthread_mutex_lock(&adev->lock);
    out->card_status = adev->card_status;
    pthread_mutex_unlock(&adev->lock);
    pthread_mutex_unlock(&out->lock);

    *stream_out = &out->stream;
    ALOGD("%s: Stream (%p) picks up usecase (%s)", __func__, &out->stream,
           use_case_table[out->usecase]);

    if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
        audio_extn_dts_notify_playback_state(out->usecase, 0, out->sample_rate,
                                             popcount(out->channel_mask), out->playback_started);
    /* setup a channel for client <--> adsp communication for stream events */
    if ((out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) ||
            (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) ||
            (audio_extn_ip_hdlr_intf_supported(config->format))) {
        hdlr_stream_cfg.pcm_device_id = platform_get_pcm_device_id(
                out->usecase, PCM_PLAYBACK);
        hdlr_stream_cfg.flags = out->flags;
        hdlr_stream_cfg.type = PCM_PLAYBACK;
        ret = audio_extn_adsp_hdlr_stream_open(&out->adsp_hdlr_stream_handle,
                &hdlr_stream_cfg);
        if (ret) {
            ALOGE("%s: adsp_hdlr_stream_open failed %d",__func__, ret);
            out->adsp_hdlr_stream_handle = NULL;
        }
    }
    if (audio_extn_ip_hdlr_intf_supported(config->format)) {
        ret = audio_extn_ip_hdlr_intf_init(&out->ip_hdlr_handle, NULL, NULL);
        if (ret < 0) {
            ALOGE("%s: audio_extn_ip_hdlr_intf_init failed %d",__func__, ret);
            out->ip_hdlr_handle = NULL;
        }
    }
    ALOGV("%s: exit", __func__);
    return 0;

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

void adev_close_output_stream(struct audio_hw_device *dev __unused,
                                     struct audio_stream_out *stream)
{
    struct stream_out *out = (struct stream_out *)stream;
    struct audio_device *adev = out->dev;
    int ret = 0;

    ALOGD("%s: enter:stream_handle(%p)",__func__, out);

    // must deregister from sndmonitor first to prevent races
    // between the callback and close_stream
    audio_extn_snd_mon_unregister_listener(out);

    /* close adsp hdrl session before standby */
    if (out->adsp_hdlr_stream_handle) {
        ret = audio_extn_adsp_hdlr_stream_close(out->adsp_hdlr_stream_handle);
        if (ret)
            ALOGE("%s: adsp_hdlr_stream_close failed %d",__func__, ret);
        out->adsp_hdlr_stream_handle = NULL;
    }

    if (audio_extn_ip_hdlr_intf_supported(out->format) && out->ip_hdlr_handle) {
        audio_extn_ip_hdlr_intf_deinit(out->ip_hdlr_handle);
        out->ip_hdlr_handle = NULL;
    }

    if (out->usecase == USECASE_COMPRESS_VOIP_CALL) {
        pthread_mutex_lock(&adev->lock);
        ret = voice_extn_compress_voip_close_output_stream(&stream->common);
        out->started = 0;
        pthread_mutex_unlock(&adev->lock);
        if(ret != 0)
            ALOGE("%s: Compress voip output cannot be closed, error:%d",
                  __func__, ret);
    } else
        out_standby(&stream->common);

    if (is_offload_usecase(out->usecase)) {
        audio_extn_dts_remove_state_notifier_node(out->usecase);
        destroy_offload_callback_thread(out);
        free_offload_usecase(adev, out->usecase);
        if (out->compr_config.codec != NULL)
            free(out->compr_config.codec);
    }

    if (out->convert_buffer != NULL) {
        free(out->convert_buffer);
        out->convert_buffer = NULL;
    }

    if (adev->voice_tx_output == out)
        adev->voice_tx_output = NULL;

    if (adev->primary_output == out)
        adev->primary_output = NULL;

    pthread_cond_destroy(&out->cond);
    pthread_mutex_destroy(&out->lock);
    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 value[32];
    int val;
    int ret;
    int status = 0;

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

    if (!parms)
        goto error;

    ret = str_parms_get_str(parms, "BT_SCO", value, sizeof(value));
    if (ret >= 0) {
        /* When set to false, HAL should disable EC and NS */
        if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0)
            adev->bt_sco_on = true;
        else
            adev->bt_sco_on = false;
    }

    pthread_mutex_lock(&adev->lock);
    status = voice_set_parameters(adev, parms);
    if (status != 0)
        goto done;

    status = platform_set_parameters(adev->platform, parms);
    if (status != 0)
        goto done;

    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 */
        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);
            status = -EINVAL;
        }
        if (status == 0) {
            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;
                    }
                }
            }
        }
    }

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

    ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_CONNECT, value, sizeof(value));
    if (ret >= 0) {
        val = atoi(value);
        if (val & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
            ALOGV("cache new ext disp type and edid");
            ret = platform_get_ext_disp_type(adev->platform);
            if (ret < 0) {
                ALOGE("%s: Failed to query disp type, ret:%d", __func__, ret);
                status = ret;
                goto done;
            }
            platform_cache_edid(adev->platform);
        } else if ((val & AUDIO_DEVICE_OUT_USB_DEVICE) ||
                   !(val ^ AUDIO_DEVICE_IN_USB_DEVICE)) {
            /*
             * Do not allow AFE proxy port usage by WFD source when USB headset is connected.
             * Per AudioPolicyManager, USB device is higher priority than WFD.
             * For Voice call over USB headset, voice call audio is routed to AFE proxy ports.
             * If WFD use case occupies AFE proxy, it may result unintended behavior while
             * starting voice call on USB
             */
            ret = str_parms_get_str(parms, "card", value, sizeof(value));
            if (ret >= 0) {
                audio_extn_usb_add_device(AUDIO_DEVICE_OUT_USB_DEVICE, atoi(value));
                audio_extn_usb_add_device(AUDIO_DEVICE_IN_USB_DEVICE, atoi(value));
            }
            ALOGV("detected USB connect .. disable proxy");
            adev->allow_afe_proxy_usage = false;
        }
    }

    ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_DISCONNECT, value, sizeof(value));
    if (ret >= 0) {
        val = atoi(value);
        /*
         * The HDMI / Displayport disconnect handling has been moved to
         * audio extension to ensure that its parameters are not
         * invalidated prior to updating sysfs of the disconnect event
         * Invalidate will be handled by audio_extn_ext_disp_set_parameters()
         */
        if ((val & AUDIO_DEVICE_OUT_USB_DEVICE) ||
                   !(val ^ AUDIO_DEVICE_IN_USB_DEVICE)) {
            ret = str_parms_get_str(parms, "card", value, sizeof(value));
            if (ret >= 0) {
                audio_extn_usb_remove_device(AUDIO_DEVICE_OUT_USB_DEVICE, atoi(value));
                audio_extn_usb_remove_device(AUDIO_DEVICE_IN_USB_DEVICE, atoi(value));
            }
            ALOGV("detected USB disconnect .. enable proxy");
            adev->allow_afe_proxy_usage = true;
        }
    }

    ret = str_parms_get_str(parms,"reconfigA2dp", value, sizeof(value));
    if (ret >= 0) {
        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) &&
                (usecase->devices & AUDIO_DEVICE_OUT_ALL_A2DP)){
                ALOGD("reconfigure a2dp... forcing device switch");

                pthread_mutex_unlock(&adev->lock);
                lock_output_stream(usecase->stream.out);
                pthread_mutex_lock(&adev->lock);
                audio_extn_a2dp_set_handoff_mode(true);
                //force device switch to re configure encoder
                select_devices(adev, usecase->id);
                audio_extn_a2dp_set_handoff_mode(false);
                pthread_mutex_unlock(&usecase->stream.out->lock);
                break;
            }
        }
    }

    //handle vr audio setparam
    ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_VR_AUDIO_MODE,
        value, sizeof(value));
    if (ret >= 0) {
        ALOGI("Setting vr mode to be %s", value);
        if (!strncmp(value, "true", 4)) {
            adev->vr_audio_mode_enabled = true;
            ALOGI("Setting vr mode to true");
        } else if (!strncmp(value, "false", 5)) {
            adev->vr_audio_mode_enabled = false;
            ALOGI("Setting vr mode to false");
        } else {
            ALOGI("wrong vr mode set");
        }
    }

    audio_extn_set_parameters(adev, parms);
done:
    str_parms_destroy(parms);
    pthread_mutex_unlock(&adev->lock);
error:
    ALOGV("%s: exit with code(%d)", __func__, status);
    return status;
}

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;
    char value[256] = {0};
    int ret = 0;

    if (!query || !reply) {
        if (reply) {
            str_parms_destroy(reply);
        }
        if (query) {
            str_parms_destroy(query);
        }
        ALOGE("adev_get_parameters: failed to create query or reply");
        return NULL;
    }

    //handle vr audio getparam

    ret = str_parms_get_str(query,
        AUDIO_PARAMETER_KEY_VR_AUDIO_MODE,
        value, sizeof(value));

    if (ret >= 0) {
        bool vr_audio_enabled = false;
        pthread_mutex_lock(&adev->lock);
        vr_audio_enabled = adev->vr_audio_mode_enabled;
        pthread_mutex_unlock(&adev->lock);

        ALOGI("getting vr mode to %d", vr_audio_enabled);

        if (vr_audio_enabled) {
            str_parms_add_str(reply, AUDIO_PARAMETER_KEY_VR_AUDIO_MODE,
                "true");
            goto exit;
        } else {
            str_parms_add_str(reply, AUDIO_PARAMETER_KEY_VR_AUDIO_MODE,
                "false");
            goto exit;
        }
    }

    pthread_mutex_lock(&adev->lock);
    audio_extn_get_parameters(adev, query, reply);
    voice_get_parameters(adev, query, reply);
    platform_get_parameters(adev->platform, query, reply);
    pthread_mutex_unlock(&adev->lock);

exit:
    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 __unused)
{
    return 0;
}

static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
{
    int ret;
    struct audio_device *adev = (struct audio_device *)dev;
    pthread_mutex_lock(&adev->lock);
    /* cache volume */
    ret = voice_set_volume(adev, volume);
    pthread_mutex_unlock(&adev->lock);
    return ret;
}

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

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

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

static int adev_get_master_mute(struct audio_hw_device *dev __unused,
                                bool *muted __unused)
{
    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) {
        ALOGD("%s: mode %d\n", __func__, mode);
        adev->mode = mode;
        if ((mode == AUDIO_MODE_NORMAL) && voice_is_in_call(adev)) {
            voice_stop_call(adev);
            platform_set_gsm_mode(adev->platform, false);
            adev->current_call_output = NULL;
        }
    }
    pthread_mutex_unlock(&adev->lock);
    return 0;
}

static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
{
    int ret;

    pthread_mutex_lock(&adev->lock);
    ALOGD("%s state %d\n", __func__, state);
    ret = voice_set_mic_mute((struct audio_device *)dev, state);
    pthread_mutex_unlock(&adev->lock);

    return ret;
}

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 __unused,
                                         const struct audio_config *config)
{
    int channel_count = audio_channel_count_from_in_mask(config->channel_mask);

    return get_input_buffer_size(config->sample_rate, config->format, channel_count,
            false /* is_low_latency: since we don't know, be conservative */);
}

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,
                                  const char *address __unused,
                                  audio_source_t source)
{
    struct audio_device *adev = (struct audio_device *)dev;
    struct stream_in *in;
    int ret = 0, buffer_size, frame_size;
    int channel_count = audio_channel_count_from_in_mask(config->channel_mask);
    bool is_low_latency = false;
    bool channel_mask_updated = false;

    *stream_in = NULL;
    if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0) {
        ALOGE("%s: invalid input parameters", __func__);
        return -EINVAL;
    }

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

    if (!in) {
        ALOGE("failed to allocate input stream");
        return -ENOMEM;
    }

    ALOGD("%s: enter: sample_rate(%d) channel_mask(%#x) devices(%#x)\
        stream_handle(%p) io_handle(%d) source(%d) format %x",__func__, config->sample_rate,
        config->channel_mask, devices, &in->stream, handle, source, config->format);
    pthread_mutex_init(&in->lock, (const pthread_mutexattr_t *) NULL);
    pthread_mutex_init(&in->pre_lock, (const pthread_mutexattr_t *) NULL);

    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 = source;
    in->dev = adev;
    in->standby = 1;
    in->channel_mask = config->channel_mask;
    in->capture_handle = handle;
    in->flags = flags;

    in->usecase = USECASE_AUDIO_RECORD;
    if (config->sample_rate == LOW_LATENCY_CAPTURE_SAMPLE_RATE &&
            (flags & AUDIO_INPUT_FLAG_FAST) != 0) {
        is_low_latency = true;
#if LOW_LATENCY_CAPTURE_USE_CASE
        in->usecase = USECASE_AUDIO_RECORD_LOW_LATENCY;
#endif
        in->realtime = may_use_noirq_mode(adev, in->usecase, in->flags);
    }

    in->format = config->format;
    if (in->realtime) {
        in->config = pcm_config_audio_capture_rt;
        in->sample_rate = in->config.rate;
        in->af_period_multiplier = af_period_multiplier;
    } else {
        in->config = pcm_config_audio_capture;
        in->config.rate = config->sample_rate;
        in->sample_rate = config->sample_rate;
        in->af_period_multiplier = 1;
    }
    in->bit_width = 16;

    /* restrict 24 bit capture for unprocessed source only
     * for other sources if 24 bit requested reject 24 and set 16 bit capture only
     */
    if (config->format == AUDIO_FORMAT_DEFAULT) {
        config->format = AUDIO_FORMAT_PCM_16_BIT;
    } else if ((config->format == AUDIO_FORMAT_PCM_FLOAT) ||
               (config->format == AUDIO_FORMAT_PCM_32_BIT) ||
               (config->format == AUDIO_FORMAT_PCM_24_BIT_PACKED) ||
               (config->format == AUDIO_FORMAT_PCM_8_24_BIT)) {
        bool ret_error = false;
        in->bit_width = 24;
        /* 24 bit is restricted to UNPROCESSED source only,also format supported
           from HAL is 24_packed and 8_24
         *> In case of UNPROCESSED source, for 24 bit, if format requested is other than
            24_packed return error indicating supported format is 24_packed
         *> In case of any other source requesting 24 bit or float return error
            indicating format supported is 16 bit only.

            on error flinger will retry with supported format passed
         */
        if ((source != AUDIO_SOURCE_UNPROCESSED) &&
            (source != AUDIO_SOURCE_CAMCORDER)) {
            config->format = AUDIO_FORMAT_PCM_16_BIT;
            if (config->sample_rate > 48000)
                config->sample_rate = 48000;
            ret_error = true;
        } else if (config->format == AUDIO_FORMAT_PCM_24_BIT_PACKED) {
            in->config.format = PCM_FORMAT_S24_3LE;
        } else if (config->format == AUDIO_FORMAT_PCM_8_24_BIT) {
            in->config.format = PCM_FORMAT_S24_LE;
        } else {
            config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
            ret_error = true;
        }

        if (ret_error) {
            ret = -EINVAL;
            goto err_open;
        }
    }

    /* Update config params with the requested sample rate and channels */
    if ((in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) &&
          (adev->mode != AUDIO_MODE_IN_CALL)) {
        ret = -EINVAL;
        goto err_open;
    }

    if ((in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) ||
             (in->device == AUDIO_DEVICE_IN_PROXY)) {
        if (config->sample_rate == 0)
            config->sample_rate = AFE_PROXY_SAMPLING_RATE;
        if (config->sample_rate != 48000 && config->sample_rate != 16000 &&
                config->sample_rate != 8000) {
            config->sample_rate = AFE_PROXY_SAMPLING_RATE;
            ret = -EINVAL;
            goto err_open;
        }
        if (config->format == AUDIO_FORMAT_DEFAULT)
            config->format = AUDIO_FORMAT_PCM_16_BIT;
        if (config->format != AUDIO_FORMAT_PCM_16_BIT) {
            config->format = AUDIO_FORMAT_PCM_16_BIT;
            ret = -EINVAL;
            goto err_open;
        }

        in->usecase = USECASE_AUDIO_RECORD_AFE_PROXY;
        in->config = pcm_config_afe_proxy_record;
        in->config.channels = channel_count;
        in->config.rate = config->sample_rate;
        in->sample_rate = config->sample_rate;
    } else if (!audio_extn_check_and_set_multichannel_usecase(adev,
                in, config, &channel_mask_updated)) {
        if (channel_mask_updated == true) {
            ALOGD("%s: return error to retry with updated channel mask (%#x)",
                   __func__, config->channel_mask);
            ret = -EINVAL;
            goto err_open;
        }
        ALOGD("%s: created surround sound session succesfully",__func__);
    } else if (audio_extn_compr_cap_enabled() &&
            audio_extn_compr_cap_format_supported(config->format) &&
            (in->dev->mode != AUDIO_MODE_IN_COMMUNICATION)) {
        audio_extn_compr_cap_init(in);
    } else if (audio_extn_cin_applicable_stream(in)) {
        ret = audio_extn_cin_configure_input_stream(in);
        if (ret)
            goto err_open;
    } else {
        in->config.channels = channel_count;
        if (!in->realtime) {
            in->format = config->format;
            frame_size = audio_stream_in_frame_size(&in->stream);
            buffer_size = get_input_buffer_size(config->sample_rate,
                                                config->format,
                                                channel_count,
                                                is_low_latency);
            in->config.period_size = buffer_size / frame_size;
        }

        if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) &&
               (in->dev->mode == AUDIO_MODE_IN_COMMUNICATION || voice_extn_compress_voip_is_active(in->dev)) &&
               (voice_extn_compress_voip_is_format_supported(in->format)) &&
               (in->config.rate == 8000 || in->config.rate == 16000 ||
                in->config.rate == 32000 || in->config.rate == 48000) &&
               (audio_channel_count_from_in_mask(in->channel_mask) == 1)) {
            voice_extn_compress_voip_open_input_stream(in);
        }
    }

    audio_extn_utils_update_stream_input_app_type_cfg(adev->platform,
                                                &adev->streams_input_cfg_list,
                                                devices, flags, in->format, in->sample_rate,
                                                in->bit_width, in->profile, &in->app_type_cfg);

    /* This stream could be for sound trigger lab,
       get sound trigger pcm if present */
    audio_extn_sound_trigger_check_and_get_session(in);

    lock_input_stream(in);
    audio_extn_snd_mon_register_listener(in, in_snd_mon_cb);
    pthread_mutex_lock(&adev->lock);
    in->card_status = adev->card_status;
    pthread_mutex_unlock(&adev->lock);
    pthread_mutex_unlock(&in->lock);

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

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)
{
    int ret;
    struct stream_in *in = (struct stream_in *)stream;
    struct audio_device *adev = (struct audio_device *)dev;

    ALOGD("%s: enter:stream_handle(%p)",__func__, in);

    // must deregister from sndmonitor first to prevent races
    // between the callback and close_stream
    audio_extn_snd_mon_unregister_listener(stream);

    /* Disable echo reference while closing input stream */
    platform_set_echo_reference(adev, false, AUDIO_DEVICE_NONE);

    if (in->usecase == USECASE_COMPRESS_VOIP_CALL) {
        pthread_mutex_lock(&adev->lock);
        ret = voice_extn_compress_voip_close_input_stream(&stream->common);
        pthread_mutex_unlock(&adev->lock);
        if (ret != 0)
            ALOGE("%s: Compress voip input cannot be closed, error:%d",
                  __func__, ret);
    } else
        in_standby(&stream->common);

    if (audio_extn_ssr_get_stream() == in) {
        audio_extn_ssr_deinit();
    }

    if (audio_extn_compr_cap_enabled() &&
            audio_extn_compr_cap_format_supported(in->config.format))
        audio_extn_compr_cap_deinit();

    if (audio_extn_cin_attached_usecase(in->usecase))
        audio_extn_cin_close_input_stream(in);

    if (in->is_st_session) {
        ALOGV("%s: sound trigger pcm stop lab", __func__);
        audio_extn_sound_trigger_stop_lab(in);
    }
    free(stream);
    return;
}

int adev_create_audio_patch(struct audio_hw_device *dev,
                            unsigned int num_sources,
                            const struct audio_port_config *sources,
                            unsigned int num_sinks,
                            const struct audio_port_config *sinks,
                            audio_patch_handle_t *handle)
{


     return audio_extn_hw_loopback_create_audio_patch(dev,
                                         num_sources,
                                         sources,
                                         num_sinks,
                                         sinks,
                                         handle);

}

int adev_release_audio_patch(struct audio_hw_device *dev,
                           audio_patch_handle_t handle)
{
    return audio_extn_hw_loopback_release_audio_patch(dev, handle);
}

int adev_get_audio_port(struct audio_hw_device *dev, struct audio_port *config)
{
    return audio_extn_hw_loopback_get_audio_port(dev, config);
}

int adev_set_audio_port_config(struct audio_hw_device *dev,
                        const struct audio_port_config *config)
{
    return audio_extn_hw_loopback_set_audio_port_config(dev, config);
}

static int adev_dump(const audio_hw_device_t *device __unused,
                     int fd __unused)
{
    return 0;
}

static int adev_close(hw_device_t *device)
{
    struct audio_device *adev = (struct audio_device *)device;

    if (!adev)
        return 0;

    pthread_mutex_lock(&adev_init_lock);

    if ((--audio_device_ref_count) == 0) {
        audio_extn_snd_mon_unregister_listener(adev);
        audio_extn_sound_trigger_deinit(adev);
        audio_extn_listen_deinit(adev);
        audio_extn_utils_release_streams_cfg_lists(
                      &adev->streams_output_cfg_list,
                      &adev->streams_input_cfg_list);
        if (audio_extn_qaf_is_enabled())
            audio_extn_qaf_deinit();
        audio_route_free(adev->audio_route);
        audio_extn_gef_deinit();
        free(adev->snd_dev_ref_cnt);
        platform_deinit(adev->platform);
        if (adev->adm_deinit)
            adev->adm_deinit(adev->adm_data);
        qahwi_deinit(device);
        audio_extn_adsp_hdlr_deinit();
        audio_extn_snd_mon_deinit();
        audio_extn_loopback_deinit(adev);
        free(device);
        adev = NULL;
    }
    pthread_mutex_unlock(&adev_init_lock);

    return 0;
}

/* This returns 1 if the input parameter looks at all plausible as a low latency period size,
 * or 0 otherwise.  A return value of 1 doesn't mean the value is guaranteed to work,
 * just that it _might_ work.
 */
static int period_size_is_plausible_for_low_latency(int period_size)
{
    switch (period_size) {
    case 160:
    case 192:
    case 240:
    case 320:
    case 480:
        return 1;
    default:
        return 0;
    }
}

static void adev_snd_mon_cb(void *cookie, struct str_parms *parms)
{
    bool is_snd_card_status = false;
    bool is_ext_device_status = false;
    char value[32];
    int card = -1;
    card_status_t status;

    if (cookie != adev || !parms)
        return;

    if (!parse_snd_card_status(parms, &card, &status)) {
        is_snd_card_status = true;
    } else if (0 < str_parms_get_str(parms, "ext_audio_device", value, sizeof(value))) {
        is_ext_device_status = true;
    } else {
        // not a valid event
        return;
    }

    pthread_mutex_lock(&adev->lock);
    if (card == adev->snd_card || is_ext_device_status) {
        if (is_snd_card_status && adev->card_status != status) {
            adev->card_status = status;
            platform_snd_card_update(adev->platform, status);
            audio_extn_fm_set_parameters(adev, parms);
        } else if (is_ext_device_status) {
            platform_set_parameters(adev->platform, parms);
        }
    }
    pthread_mutex_unlock(&adev->lock);
    return;
}

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

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

    pthread_mutex_lock(&adev_init_lock);
    if (audio_device_ref_count != 0){
            *device = &adev->device.common;
            audio_device_ref_count++;
            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));

    if (!adev) {
        pthread_mutex_unlock(&adev_init_lock);
        return -ENOMEM;
    }

    pthread_mutex_init(&adev->lock, (const pthread_mutexattr_t *) NULL);

#ifdef DYNAMIC_LOG_ENABLED
    register_for_dynamic_logging("hal");
#endif

    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.create_audio_patch = adev_create_audio_patch;
    adev->device.release_audio_patch = adev_release_audio_patch;
    adev->device.get_audio_port = adev_get_audio_port;
    adev->device.set_audio_port_config = adev_set_audio_port_config;
    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->allow_afe_proxy_usage = true;
    adev->bt_sco_on = false;
    /* adev->cur_hdmi_channels = 0;  by calloc() */
    adev->snd_dev_ref_cnt = calloc(SND_DEVICE_MAX, sizeof(int));
    voice_init(adev);
    list_init(&adev->usecase_list);
    adev->cur_wfd_channels = 2;
    adev->offload_usecases_state = 0;
    adev->is_channel_status_set = false;
    adev->perf_lock_opts[0] = 0x101;
    adev->perf_lock_opts[1] = 0x20E;
    adev->perf_lock_opts_size = 2;
    adev->dsp_bit_width_enforce_mode = 0;

    /* 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;
        pthread_mutex_unlock(&adev_init_lock);
        pthread_mutex_destroy(&adev->lock);
        return -EINVAL;
    }

    if (audio_extn_qaf_is_enabled()) {
        ret = audio_extn_qaf_init(adev);
        if (ret < 0) {
            free(adev);
            ALOGE("%s: Failed to init platform data, aborting.", __func__);
            *device = NULL;
            pthread_mutex_unlock(&adev_init_lock);
            pthread_mutex_destroy(&adev->lock);
            return ret;
        }

        adev->device.open_output_stream = audio_extn_qaf_open_output_stream;
        adev->device.close_output_stream = audio_extn_qaf_close_output_stream;
    }

    if (access(VISUALIZER_LIBRARY_PATH, R_OK) == 0) {
        adev->visualizer_lib = dlopen(VISUALIZER_LIBRARY_PATH, RTLD_NOW);
        if (adev->visualizer_lib == NULL) {
            ALOGE("%s: DLOPEN failed for %s", __func__, VISUALIZER_LIBRARY_PATH);
        } else {
            ALOGV("%s: DLOPEN successful for %s", __func__, VISUALIZER_LIBRARY_PATH);
            adev->visualizer_start_output =
                        (int (*)(audio_io_handle_t, int))dlsym(adev->visualizer_lib,
                                                        "visualizer_hal_start_output");
            adev->visualizer_stop_output =
                        (int (*)(audio_io_handle_t, int))dlsym(adev->visualizer_lib,
                                                        "visualizer_hal_stop_output");
        }
    }
    audio_extn_init(adev);
    audio_extn_listen_init(adev, adev->snd_card);
    audio_extn_gef_init(adev);
    audio_extn_loopback_init(adev);

    if (access(OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH, R_OK) == 0) {
        adev->offload_effects_lib = dlopen(OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH, RTLD_NOW);
        if (adev->offload_effects_lib == NULL) {
            ALOGE("%s: DLOPEN failed for %s", __func__,
                  OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH);
        } else {
            ALOGV("%s: DLOPEN successful for %s", __func__,
                  OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH);
            adev->offload_effects_start_output =
                        (int (*)(audio_io_handle_t, int, struct mixer *))dlsym(adev->offload_effects_lib,
                                         "offload_effects_bundle_hal_start_output");
            adev->offload_effects_stop_output =
                        (int (*)(audio_io_handle_t, int))dlsym(adev->offload_effects_lib,
                                         "offload_effects_bundle_hal_stop_output");
            adev->offload_effects_set_hpx_state =
                        (int (*)(bool))dlsym(adev->offload_effects_lib,
                                         "offload_effects_bundle_set_hpx_state");
            adev->offload_effects_get_parameters =
                        (void (*)(struct str_parms *, struct str_parms *))
                                         dlsym(adev->offload_effects_lib,
                                         "offload_effects_bundle_get_parameters");
            adev->offload_effects_set_parameters =
                        (void (*)(struct str_parms *))dlsym(adev->offload_effects_lib,
                                         "offload_effects_bundle_set_parameters");
        }
    }

    if (access(ADM_LIBRARY_PATH, R_OK) == 0) {
        adev->adm_lib = dlopen(ADM_LIBRARY_PATH, RTLD_NOW);
        if (adev->adm_lib == NULL) {
            ALOGE("%s: DLOPEN failed for %s", __func__, ADM_LIBRARY_PATH);
        } else {
            ALOGV("%s: DLOPEN successful for %s", __func__, ADM_LIBRARY_PATH);
            adev->adm_init = (adm_init_t)
                                    dlsym(adev->adm_lib, "adm_init");
            adev->adm_deinit = (adm_deinit_t)
                                    dlsym(adev->adm_lib, "adm_deinit");
            adev->adm_register_input_stream = (adm_register_input_stream_t)
                                    dlsym(adev->adm_lib, "adm_register_input_stream");
            adev->adm_register_output_stream = (adm_register_output_stream_t)
                                    dlsym(adev->adm_lib, "adm_register_output_stream");
            adev->adm_deregister_stream = (adm_deregister_stream_t)
                                    dlsym(adev->adm_lib, "adm_deregister_stream");
            adev->adm_request_focus = (adm_request_focus_t)
                                    dlsym(adev->adm_lib, "adm_request_focus");
            adev->adm_abandon_focus = (adm_abandon_focus_t)
                                    dlsym(adev->adm_lib, "adm_abandon_focus");
            adev->adm_set_config = (adm_set_config_t)
                                    dlsym(adev->adm_lib, "adm_set_config");
            adev->adm_request_focus_v2 = (adm_request_focus_v2_t)
                                    dlsym(adev->adm_lib, "adm_request_focus_v2");
            adev->adm_is_noirq_avail = (adm_is_noirq_avail_t)
                                    dlsym(adev->adm_lib, "adm_is_noirq_avail");
            adev->adm_on_routing_change = (adm_on_routing_change_t)
                                    dlsym(adev->adm_lib, "adm_on_routing_change");
        }
    }

    adev->bt_wb_speech_enabled = false;
    //initialize this to false for now,
    //this will be set to true through set param
    adev->vr_audio_mode_enabled = false;

    audio_extn_ds2_enable(adev);
    *device = &adev->device.common;
    adev->dsp_bit_width_enforce_mode =
        adev_init_dsp_bit_width_enforce_mode(adev->mixer);

    audio_extn_utils_update_streams_cfg_lists(adev->platform, adev->mixer,
                                             &adev->streams_output_cfg_list,
                                             &adev->streams_input_cfg_list);

    audio_device_ref_count++;

    char value[PROPERTY_VALUE_MAX];
    int trial;
    if (property_get("vendor.audio_hal.period_size", value, NULL) > 0) {
        trial = atoi(value);
        if (period_size_is_plausible_for_low_latency(trial)) {
            pcm_config_low_latency.period_size = trial;
            pcm_config_low_latency.start_threshold = trial / 4;
            pcm_config_low_latency.avail_min = trial / 4;
            configured_low_latency_capture_period_size = trial;
        }
    }
    if (property_get("vendor.audio_hal.in_period_size", value, NULL) > 0) {
        trial = atoi(value);
        if (period_size_is_plausible_for_low_latency(trial)) {
            configured_low_latency_capture_period_size = trial;
        }
    }

    if (property_get("vendor.audio_hal.period_multiplier", value, NULL) > 0) {
        af_period_multiplier = atoi(value);
        if (af_period_multiplier < 0)
            af_period_multiplier = 2;
        else if (af_period_multiplier > 4)
            af_period_multiplier = 4;

        ALOGV("new period_multiplier = %d", af_period_multiplier);
    }

    adev->multi_offload_enable = property_get_bool("vendor.audio.offload.multiple.enabled", false);
    pthread_mutex_unlock(&adev_init_lock);

    if (adev->adm_init)
        adev->adm_data = adev->adm_init();

    qahwi_init(*device);
    audio_extn_perf_lock_init();
    audio_extn_adsp_hdlr_init(adev->mixer);

    audio_extn_snd_mon_init();
    pthread_mutex_lock(&adev->lock);
    audio_extn_snd_mon_register_listener(adev, adev_snd_mon_cb);
    adev->card_status = CARD_STATUS_ONLINE;
    pthread_mutex_unlock(&adev->lock);
    audio_extn_sound_trigger_init(adev); /* dependent on snd_mon_init() */

    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 = "The Linux Foundation",
        .methods = &hal_module_methods,
    },
};