upload android base code part4

android/hardware/qcom/audio/hal/audio_extn/audio_extn.c

@@ -0,0 +1,162 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "audio_hw_extn"
+/*#define LOG_NDEBUG 0*/
+#define LOG_NDDEBUG 0
+
+#include <stdlib.h>
+#include <errno.h>
+#include <dlfcn.h>
+#include <cutils/properties.h>
+#include <cutils/log.h>
+
+#include "audio_hw.h"
+#include "audio_extn.h"
+#include "platform.h"
+#include "platform_api.h"
+
+struct snd_card_split cur_snd_card_split = {
+    .device = {0},
+    .snd_card = {0},
+    .form_factor = {0},
+};
+
+struct snd_card_split *audio_extn_get_snd_card_split()
+{
+    return &cur_snd_card_split;
+}
+
+void audio_extn_set_snd_card_split(const char* in_snd_card_name)
+{
+    /* sound card name follows below mentioned convention
+       <target name>-<sound card name>-<form factor>-snd-card
+       parse target name, sound card name and form factor
+    */
+    char *snd_card_name = strdup(in_snd_card_name);
+    char *tmp = NULL;
+    char *device = NULL;
+    char *snd_card = NULL;
+    char *form_factor = NULL;
+
+    if (in_snd_card_name == NULL) {
+        ALOGE("%s: snd_card_name passed is NULL", __func__);
+        goto on_error;
+    }
+
+    device = strtok_r(snd_card_name, "-", &tmp);
+    if (device == NULL) {
+        ALOGE("%s: called on invalid snd card name", __func__);
+        goto on_error;
+    }
+    strlcpy(cur_snd_card_split.device, device, HW_INFO_ARRAY_MAX_SIZE);
+
+    snd_card = strtok_r(NULL, "-", &tmp);
+    if (snd_card == NULL) {
+        ALOGE("%s: called on invalid snd card name", __func__);
+        goto on_error;
+    }
+    strlcpy(cur_snd_card_split.snd_card, snd_card, HW_INFO_ARRAY_MAX_SIZE);
+
+    form_factor = strtok_r(NULL, "-", &tmp);
+    if (form_factor == NULL) {
+        ALOGE("%s: called on invalid snd card name", __func__);
+        goto on_error;
+    }
+    strlcpy(cur_snd_card_split.form_factor, form_factor, HW_INFO_ARRAY_MAX_SIZE);
+
+    ALOGI("%s: snd_card_name(%s) device(%s) snd_card(%s) form_factor(%s)",
+               __func__, in_snd_card_name, device, snd_card, form_factor);
+
+on_error:
+    if (snd_card_name)
+        free(snd_card_name);
+}
+
+#ifdef KPI_OPTIMIZE_ENABLED
+typedef int (*perf_lock_acquire_t)(int, int, int*, int);
+typedef int (*perf_lock_release_t)(int);
+
+static void *qcopt_handle;
+static perf_lock_acquire_t perf_lock_acq;
+static perf_lock_release_t perf_lock_rel;
+
+static int perf_lock_handle;
+char opt_lib_path[PROPERTY_VALUE_MAX] = {0};
+
+int perf_lock_opts[] = {0x101, 0x20E, 0x30E};
+
+int audio_extn_perf_lock_init(void)
+{
+    int ret = 0;
+    if (qcopt_handle == NULL) {
+        if (property_get("ro.vendor.extension_library",
+                         opt_lib_path, NULL) <= 0) {
+            ALOGE("%s: Failed getting perf property", __func__);
+            ret = -EINVAL;
+            goto err;
+        }
+        if ((qcopt_handle = dlopen(opt_lib_path, RTLD_NOW)) == NULL) {
+            ALOGE("%s: Failed to open perf handle", __func__);
+            ret = -EINVAL;
+            goto err;
+        } else {
+            perf_lock_acq = (perf_lock_acquire_t)dlsym(qcopt_handle,
+                                                       "perf_lock_acq");
+            if (perf_lock_acq == NULL) {
+                ALOGE("%s: Perf lock Acquire NULL", __func__);
+                dlclose(qcopt_handle);
+                qcopt_handle = NULL;
+                ret = -EINVAL;
+                goto err;
+            }
+            perf_lock_rel = (perf_lock_release_t)dlsym(qcopt_handle,
+                                                       "perf_lock_rel");
+            if (perf_lock_rel == NULL) {
+                ALOGE("%s: Perf lock Release NULL", __func__);
+                dlclose(qcopt_handle);
+                qcopt_handle = NULL;
+                ret = -EINVAL;
+                goto err;
+            }
+            ALOGD("%s: Perf lock handles Success", __func__);
+        }
+    }
+err:
+    return ret;
+}
+
+void audio_extn_perf_lock_acquire(void)
+{
+    if (perf_lock_acq) {
+        perf_lock_handle = perf_lock_acq(perf_lock_handle, 0, perf_lock_opts, 3);
+        ALOGV("%s: Perf lock acquired", __func__);
+    } else {
+        ALOGE("%s: Perf lock acquire error", __func__);
+    }
+}
+
+void audio_extn_perf_lock_release(void)
+{
+    if (perf_lock_rel && perf_lock_handle) {
+        perf_lock_rel(perf_lock_handle);
+        perf_lock_handle = 0;
+        ALOGV("%s: Perf lock released", __func__);
+    } else {
+        ALOGE("%s: Perf lock release error", __func__);
+    }
+}
+#endif /* KPI_OPTIMIZE_ENABLED */

android/hardware/qcom/audio/hal/audio_extn/audio_extn.h

@@ -0,0 +1,198 @@
+/*
+ * Copyright (C) 2014 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.
+ */
+
+#ifndef AUDIO_EXTN_H
+#define AUDIO_EXTN_H
+
+#include <cutils/str_parms.h>
+
+#define HW_INFO_ARRAY_MAX_SIZE 32
+
+struct snd_card_split {
+    char device[HW_INFO_ARRAY_MAX_SIZE];
+    char snd_card[HW_INFO_ARRAY_MAX_SIZE];
+    char form_factor[HW_INFO_ARRAY_MAX_SIZE];
+};
+
+void *audio_extn_extspk_init(struct audio_device *adev);
+void audio_extn_extspk_deinit(void *extn);
+void audio_extn_extspk_update(void* extn);
+void audio_extn_extspk_set_mode(void* extn, audio_mode_t mode);
+void audio_extn_extspk_set_voice_vol(void* extn, float vol);
+struct snd_card_split *audio_extn_get_snd_card_split();
+void audio_extn_set_snd_card_split(const char* in_snd_card_name);
+
+#ifndef SPKR_PROT_ENABLED
+#define audio_extn_spkr_prot_init(adev)       (0)
+#define audio_extn_spkr_prot_start_processing(snd_device)    (-EINVAL)
+#define audio_extn_spkr_prot_calib_cancel(adev) (0)
+#define audio_extn_spkr_prot_stop_processing(snd_device)     (0)
+#define audio_extn_spkr_prot_is_enabled() (false)
+#define audio_extn_spkr_prot_get_acdb_id(snd_device)         (-EINVAL)
+#define audio_extn_get_spkr_prot_snd_device(snd_device) (snd_device)
+#else
+void audio_extn_spkr_prot_init(void *adev);
+int audio_extn_spkr_prot_start_processing(snd_device_t snd_device);
+void audio_extn_spkr_prot_stop_processing(snd_device_t snd_device);
+bool audio_extn_spkr_prot_is_enabled();
+int audio_extn_spkr_prot_get_acdb_id(snd_device_t snd_device);
+int audio_extn_get_spkr_prot_snd_device(snd_device_t snd_device);
+void audio_extn_spkr_prot_calib_cancel(void *adev);
+#endif
+
+#ifndef HFP_ENABLED
+#define audio_extn_hfp_is_active(adev)                  (0)
+#define audio_extn_hfp_get_usecase()                    (-1)
+#define audio_extn_hfp_set_parameters(adev, params)     (0)
+#define audio_extn_hfp_set_mic_mute(adev, state)        (0)
+
+#else
+bool audio_extn_hfp_is_active(struct audio_device *adev);
+
+audio_usecase_t audio_extn_hfp_get_usecase();
+
+void audio_extn_hfp_set_parameters(struct audio_device *adev,
+                                    struct str_parms *parms);
+int audio_extn_hfp_set_mic_mute(struct audio_device *adev, bool state);
+
+#endif
+
+#ifndef USB_TUNNEL_ENABLED
+#define audio_extn_usb_init(adev)                                      (0)
+#define audio_extn_usb_deinit()                                        (0)
+#define audio_extn_usb_add_device(device, card)                        (0)
+#define audio_extn_usb_remove_device(device, card)                     (0)
+#define audio_extn_usb_is_config_supported(bit_width, sample_rate, ch, pb) (false)
+#define audio_extn_usb_enable_sidetone(device, enable)                 (0)
+#define audio_extn_usb_set_sidetone_gain(parms, value, len)            (0)
+#define audio_extn_usb_is_capture_supported()                          (false)
+#define audio_extn_usb_get_max_channels(dir)                           (0)
+#define audio_extn_usb_get_max_bit_width(dir)                          (0)
+#define audio_extn_usb_sup_sample_rates(t, s, l)                       (0)
+#define audio_extn_usb_alive(adev)                                     (false)
+#else
+void audio_extn_usb_init(void *adev);
+void audio_extn_usb_deinit();
+void audio_extn_usb_add_device(audio_devices_t device, int card);
+void audio_extn_usb_remove_device(audio_devices_t device, int card);
+bool audio_extn_usb_is_config_supported(unsigned int *bit_width,
+                                        unsigned int *sample_rate,
+                                        unsigned int *ch,
+                                        bool is_playback);
+int audio_extn_usb_enable_sidetone(int device, bool enable);
+int audio_extn_usb_set_sidetone_gain(struct str_parms *parms,
+                                     char *value, int len);
+bool audio_extn_usb_is_capture_supported();
+int audio_extn_usb_get_max_channels(bool is_playback);
+int audio_extn_usb_get_max_bit_width(bool is_playback);
+int audio_extn_usb_sup_sample_rates(bool is_playback, uint32_t *sr, uint32_t l);
+bool audio_extn_usb_alive(int card);
+#endif
+
+
+#ifndef SOUND_TRIGGER_ENABLED
+#define audio_extn_sound_trigger_init(adev)                            (0)
+#define audio_extn_sound_trigger_deinit(adev)                          (0)
+#define audio_extn_sound_trigger_update_device_status(snd_dev, event)  (0)
+#define audio_extn_sound_trigger_set_parameters(adev, parms)           (0)
+#define audio_extn_sound_trigger_check_and_get_session(in)             (0)
+#define audio_extn_sound_trigger_stop_lab(in)                          (0)
+#define audio_extn_sound_trigger_read(in, buffer, bytes)               (0)
+
+#else
+
+enum st_event_type {
+    ST_EVENT_SND_DEVICE_FREE,
+    ST_EVENT_SND_DEVICE_BUSY,
+    ST_EVENT_STREAM_FREE,
+    ST_EVENT_STREAM_BUSY
+};
+typedef enum st_event_type st_event_type_t;
+
+int audio_extn_sound_trigger_init(struct audio_device *adev);
+void audio_extn_sound_trigger_deinit(struct audio_device *adev);
+void audio_extn_sound_trigger_update_device_status(snd_device_t snd_device,
+                                     st_event_type_t event);
+void audio_extn_sound_trigger_set_parameters(struct audio_device *adev,
+                                             struct str_parms *parms);
+void audio_extn_sound_trigger_check_and_get_session(struct stream_in *in);
+void audio_extn_sound_trigger_stop_lab(struct stream_in *in);
+int audio_extn_sound_trigger_read(struct stream_in *in, void *buffer,
+                                  size_t bytes);
+#endif
+
+#ifndef DSM_FEEDBACK_ENABLED
+#define audio_extn_dsm_feedback_enable(adev, snd_device, benable)                (0)
+#else
+void audio_extn_dsm_feedback_enable(struct audio_device *adev,
+                         snd_device_t snd_device,
+                         bool benable);
+#endif
+
+void audio_extn_utils_send_default_app_type_cfg(void *platform, struct mixer *mixer);
+int audio_extn_utils_send_app_type_cfg(struct audio_device *adev,
+                                       struct audio_usecase *usecase);
+void audio_extn_utils_send_audio_calibration(struct audio_device *adev,
+                                             struct audio_usecase *usecase);
+int audio_extn_utils_send_app_type_gain(struct audio_device *adev,
+                                        int app_type,
+                                        int *gain);
+#ifndef HWDEP_CAL_ENABLED
+#define  audio_extn_hwdep_cal_send(snd_card, acdb_handle) (0)
+#else
+void audio_extn_hwdep_cal_send(int snd_card, void *acdb_handle);
+#endif
+
+#ifndef KPI_OPTIMIZE_ENABLED
+#define audio_extn_perf_lock_init() (0)
+#define audio_extn_perf_lock_acquire() (0)
+#define audio_extn_perf_lock_release() (0)
+#else
+int audio_extn_perf_lock_init(void);
+void audio_extn_perf_lock_acquire(void);
+void audio_extn_perf_lock_release(void);
+#endif /* KPI_OPTIMIZE_ENABLED */
+
+#ifndef HW_VARIANTS_ENABLED
+#define hw_info_init(snd_card_name)                  (0)
+#define hw_info_deinit(hw_info)                      (0)
+#define hw_info_append_hw_type(hw_info,\
+        snd_device, device_name)                     (0)
+#else
+void *hw_info_init(const char *snd_card_name);
+void hw_info_deinit(void *hw_info);
+void hw_info_append_hw_type(void *hw_info, snd_device_t snd_device,
+                             char *device_name);
+#endif /* HW_VARIANTS_ENABLED */
+
+typedef void (* snd_mon_cb)(void * stream, struct str_parms * parms);
+#ifndef SND_MONITOR_ENABLED
+#define audio_extn_snd_mon_init()           (0)
+#define audio_extn_snd_mon_deinit()         (0)
+#define audio_extn_snd_mon_register_listener(stream, cb) (0)
+#define audio_extn_snd_mon_unregister_listener(stream) (0)
+#else
+int audio_extn_snd_mon_init();
+int audio_extn_snd_mon_deinit();
+int audio_extn_snd_mon_register_listener(void *stream, snd_mon_cb cb);
+int audio_extn_snd_mon_unregister_listener(void *stream);
+#endif
+
+bool audio_extn_utils_resolve_config_file(char[]);
+int audio_extn_utils_get_platform_info(const char* snd_card_name,
+                                       char* platform_info_file);
+int audio_extn_utils_get_snd_card_num();
+#endif /* AUDIO_EXTN_H */

android/hardware/qcom/audio/hal/audio_extn/dsm_feedback.c

@@ -0,0 +1,105 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "audio_hw_dsm_feedback"
+/*#define LOG_NDEBUG 0*/
+#define LOG_NDDEBUG 0
+
+#include <errno.h>
+#include <math.h>
+#include <cutils/log.h>
+
+#include "audio_hw.h"
+#include "platform.h"
+#include "platform_api.h"
+#include <stdlib.h>
+
+
+static struct pcm_config pcm_config_dsm = {
+    .channels = 2,
+    .rate = 48000,
+    .period_size = 256,
+    .period_count = 4,
+    .format = PCM_FORMAT_S16_LE,
+    .start_threshold = 0,
+    .stop_threshold = INT_MAX,
+    .avail_min = 0,
+};
+
+int start_dsm_feedback_processing(struct audio_device *adev, int enable)
+{
+    int ret = 0;
+    int32_t pcm_dev_tx_id = -1;
+    static struct pcm *dsm_pcm_handle = NULL;
+
+    if (enable) {
+        /*do nothing if already enabled*/
+        if (dsm_pcm_handle)
+            return ret;
+
+        pcm_dev_tx_id = platform_get_pcm_device_id(USECASE_AUDIO_DSM_FEEDBACK, PCM_CAPTURE);
+        if (pcm_dev_tx_id < 0) {
+            ALOGE("%s: Invalid pcm device for usecase (%d)",
+                  __func__, USECASE_AUDIO_DSM_FEEDBACK);
+            ret = -ENODEV;
+            goto close;
+        }
+
+        dsm_pcm_handle = pcm_open(adev->snd_card,
+                                 pcm_dev_tx_id,
+                                 PCM_IN, &pcm_config_dsm);
+        if (dsm_pcm_handle && !pcm_is_ready(dsm_pcm_handle)) {
+            ALOGE("%s: %s", __func__, pcm_get_error(dsm_pcm_handle));
+            ret = -EIO;
+            goto close;
+        }
+
+        if (pcm_start(dsm_pcm_handle) < 0) {
+            ALOGE("%s: pcm start for RX failed", __func__);
+            ret = -EINVAL;
+            goto close;
+        }
+
+        return ret;
+    }
+
+close:
+    /*close pcm if disable or error happend in opening*/
+    if (dsm_pcm_handle) {
+        pcm_close(dsm_pcm_handle);
+        dsm_pcm_handle = NULL;
+    }
+
+    return ret;
+}
+
+void audio_extn_dsm_feedback_enable(struct audio_device *adev,
+                         snd_device_t snd_device,
+                         int benable)
+{
+    if ( NULL == adev )
+        return;
+
+    if( snd_device == SND_DEVICE_OUT_SPEAKER ||
+        snd_device == SND_DEVICE_OUT_SPEAKER_REVERSE ||
+        snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
+        snd_device == SND_DEVICE_OUT_SPEAKER_SAFE ||
+        snd_device == SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES ||
+        snd_device == SND_DEVICE_OUT_SPEAKER_AND_LINE ||
+        snd_device == SND_DEVICE_OUT_SPEAKER_SAFE_AND_HEADPHONES ||
+        snd_device == SND_DEVICE_OUT_SPEAKER_SAFE_AND_LINE )
+        start_dsm_feedback_processing(adev, benable);
+}

android/hardware/qcom/audio/hal/audio_extn/ext_speaker.c

@@ -0,0 +1,158 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "ext_speaker"
+/*#define LOG_NDEBUG 0*/
+#include <cutils/log.h>
+
+#include <stdlib.h>
+#include <audio_hw.h>
+#include <dlfcn.h>
+
+#ifdef __LP64__
+#define LIB_SPEAKER_BUNDLE "/vendor/lib64/soundfx/libspeakerbundle.so"
+#else
+#define LIB_SPEAKER_BUNDLE "/vendor/lib/soundfx/libspeakerbundle.so"
+#endif
+
+typedef void (*set_mode_t)(int);
+typedef void (*set_speaker_on_t)(bool);
+typedef void (*set_earpiece_on_t)(bool);
+typedef void (*set_voice_vol_t)(float);
+
+struct speaker_data {
+    struct audio_device *adev;
+    void *speaker_bundle;
+    set_mode_t set_mode;
+    set_speaker_on_t set_speaker_on;
+    set_earpiece_on_t set_earpiece_on;
+    set_voice_vol_t set_voice_vol;
+};
+
+static struct speaker_data* open_speaker_bundle()
+{
+    struct speaker_data *sd = calloc(1, sizeof(struct speaker_data));
+
+    sd->speaker_bundle = dlopen(LIB_SPEAKER_BUNDLE, RTLD_NOW);
+    if (sd->speaker_bundle == NULL) {
+        ALOGE("%s: DLOPEN failed for %s", __func__, LIB_SPEAKER_BUNDLE);
+        goto error;
+    } else {
+        ALOGV("%s: DLOPEN successful for %s", __func__, LIB_SPEAKER_BUNDLE);
+
+        sd->set_mode = (set_mode_t)dlsym(sd->speaker_bundle,
+                                             "set_mode");
+        if (sd->set_mode == NULL) {
+            ALOGE("%s: dlsym error %s for set_mode", __func__,
+                  dlerror());
+            goto error;
+        }
+        sd->set_speaker_on = (set_speaker_on_t)dlsym(sd->speaker_bundle,
+                                             "set_speaker_on");
+        if (sd->set_speaker_on == NULL) {
+            ALOGE("%s: dlsym error %s for set_speaker_on", __func__,
+                  dlerror());
+            goto error;
+        }
+        sd->set_earpiece_on = (set_earpiece_on_t)dlsym(sd->speaker_bundle,
+                                             "set_earpiece_on");
+        if (sd->set_earpiece_on == NULL) {
+            ALOGE("%s: dlsym error %s for set_earpiece_on", __func__,
+                  dlerror());
+            goto error;
+        }
+        sd->set_voice_vol = (set_voice_vol_t)dlsym(sd->speaker_bundle,
+                                             "set_voice_volume");
+        if (sd->set_voice_vol == NULL) {
+            ALOGE("%s: dlsym error %s for set_voice_volume",
+                  __func__, dlerror());
+            goto error;
+        }
+    }
+    return sd;
+
+error:
+    free(sd);
+    return 0;
+}
+
+static void close_speaker_bundle(struct speaker_data *sd)
+{
+    if (sd != NULL) {
+        dlclose(sd->speaker_bundle);
+        free(sd);
+        sd = NULL;
+    }
+}
+
+void *audio_extn_extspk_init(struct audio_device *adev)
+{
+    struct speaker_data *data = open_speaker_bundle();
+
+    if (data)
+        data->adev = adev;
+
+    return data;
+}
+
+void audio_extn_extspk_deinit(void *extn)
+{
+    struct speaker_data *data = (struct speaker_data*)extn;
+    close_speaker_bundle(data);
+}
+
+void audio_extn_extspk_update(void* extn)
+{
+    struct speaker_data *data = (struct speaker_data*)extn;
+
+    if (data) {
+        bool speaker_on = false;
+        bool earpiece_on = false;
+        struct listnode *node;
+        struct audio_usecase *usecase;
+        list_for_each(node, &data->adev->usecase_list) {
+            usecase = node_to_item(node, struct audio_usecase, list);
+            if (usecase->devices & AUDIO_DEVICE_OUT_EARPIECE) {
+                if(data->adev->snd_dev_ref_cnt[usecase->out_snd_device] != 0) {
+                    earpiece_on = true;
+                }
+            }
+            if (usecase->devices & AUDIO_DEVICE_OUT_SPEAKER) {
+                if(data->adev->snd_dev_ref_cnt[usecase->out_snd_device] != 0) {
+                    speaker_on = true;
+                }
+            }
+        }
+        data->set_earpiece_on(earpiece_on);
+        data->set_speaker_on(speaker_on);
+    }
+}
+
+void audio_extn_extspk_set_mode(void* extn, audio_mode_t mode)
+{
+    struct speaker_data *data = (struct speaker_data*)extn;
+
+    if (data)
+        data->set_mode(mode);
+}
+
+void audio_extn_extspk_set_voice_vol(void* extn, float vol)
+{
+    struct speaker_data *data = (struct speaker_data*)extn;
+
+    if (data)
+        data->set_voice_vol(vol);
+}

android/hardware/qcom/audio/hal/audio_extn/hfp.c

@@ -0,0 +1,509 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "audio_hw_hfp"
+/*#define LOG_NDEBUG 0*/
+#define LOG_NDDEBUG 0
+
+#include <errno.h>
+#include <math.h>
+#include <cutils/log.h>
+
+#include "audio_hw.h"
+#include "platform.h"
+#include "platform_api.h"
+#include <stdlib.h>
+#include <cutils/str_parms.h>
+#include "audio_extn/tfa_98xx.h"
+#include "audio_extn.h"
+
+#define AUDIO_PARAMETER_HFP_ENABLE            "hfp_enable"
+#define AUDIO_PARAMETER_HFP_SET_SAMPLING_RATE "hfp_set_sampling_rate"
+#define AUDIO_PARAMETER_KEY_HFP_VOLUME        "hfp_volume"
+#define AUDIO_PARAMETER_HFP_VOL_MIXER_CTL     "hfp_vol_mixer_ctl"
+#define AUDIO_PARAMATER_HFP_VALUE_MAX         128
+
+#define AUDIO_PARAMETER_KEY_HFP_MIC_VOLUME "hfp_mic_volume"
+#define PLAYBACK_VOLUME_MAX 0x2000
+#define CAPTURE_VOLUME_DEFAULT                (15.0)
+
+static int32_t start_hfp(struct audio_device *adev,
+                               struct str_parms *parms);
+
+static int32_t stop_hfp(struct audio_device *adev);
+
+struct hfp_module {
+    struct pcm *hfp_sco_rx;
+    struct pcm *hfp_sco_tx;
+    struct pcm *hfp_pcm_rx;
+    struct pcm *hfp_pcm_tx;
+    float  hfp_volume;
+    float  mic_volume;
+    char   hfp_vol_mixer_ctl[AUDIO_PARAMATER_HFP_VALUE_MAX];
+    bool   is_hfp_running;
+    bool   mic_mute;
+    audio_usecase_t ucid;
+};
+
+static struct hfp_module hfpmod = {
+    .hfp_sco_rx = NULL,
+    .hfp_sco_tx = NULL,
+    .hfp_pcm_rx = NULL,
+    .hfp_pcm_tx = NULL,
+    .hfp_volume = 0,
+    .mic_volume = CAPTURE_VOLUME_DEFAULT,
+    .hfp_vol_mixer_ctl = {0, },
+    .is_hfp_running = 0,
+    .mic_mute = 0,
+    .ucid = USECASE_AUDIO_HFP_SCO,
+};
+static struct pcm_config pcm_config_hfp = {
+    .channels = 1,
+    .rate = 8000,
+    .period_size = 240,
+    .period_count = 2,
+    .format = PCM_FORMAT_S16_LE,
+    .start_threshold = 0,
+    .stop_threshold = INT_MAX,
+    .avail_min = 0,
+};
+
+static int32_t hfp_set_volume(struct audio_device *adev, float value)
+{
+    int32_t vol, ret = 0;
+    struct mixer_ctl *ctl;
+
+    ALOGV("%s: entry", __func__);
+    ALOGD("%s: (%f)\n", __func__, value);
+
+    hfpmod.hfp_volume = value;
+    audio_extn_tfa_98xx_set_voice_vol(value);
+
+    if (value < 0.0) {
+        ALOGW("%s: (%f) Under 0.0, assuming 0.0\n", __func__, value);
+        value = 0.0;
+    } else {
+        value = ((value > 15.000000) ? 1.0 : (value / 15));
+        ALOGW("%s: Volume brought with in range (%f)\n", __func__, value);
+    }
+    vol  = lrint((value * 0x2000) + 0.5);
+
+    if (!hfpmod.is_hfp_running) {
+        ALOGV("%s: HFP not active, ignoring set_hfp_volume call", __func__);
+        return -EIO;
+    }
+
+    ALOGD("%s: Setting HFP volume to %d \n", __func__, vol);
+    if (0 == hfpmod.hfp_vol_mixer_ctl[0]) {
+#ifdef EXTERNAL_BT_SUPPORTED
+        strcpy(hfpmod.hfp_vol_mixer_ctl, "PRI AUXPCM LOOPBACK Volume");
+#else
+        strcpy(hfpmod.hfp_vol_mixer_ctl, "Internal HFP RX Volume");
+#endif
+        ALOGW("%s: Defaulting hfp mixer control to: %s",
+                 __func__, hfpmod.hfp_vol_mixer_ctl);
+    }
+    ctl = mixer_get_ctl_by_name(adev->mixer, hfpmod.hfp_vol_mixer_ctl);
+    if (!ctl) {
+        ALOGE("%s: Could not get ctl for mixer cmd - %s",
+              __func__, hfpmod.hfp_vol_mixer_ctl);
+        return -EINVAL;
+    }
+    if(mixer_ctl_set_value(ctl, 0, vol) < 0) {
+        ALOGE("%s: Couldn't set HFP Volume: [%d]", __func__, vol);
+        return -EINVAL;
+    }
+
+    ALOGV("%s: exit", __func__);
+    return ret;
+}
+
+
+/*Set mic volume to value.
+*
+* This interface is used for mic volume control, set mic volume as value(range 0 ~ 15).
+*/
+static int hfp_set_mic_volume(struct audio_device *adev, float value)
+{
+    int volume, ret = 0;
+    char mixer_ctl_name[128];
+    struct mixer_ctl *ctl;
+    int pcm_device_id = HFP_ASM_RX_TX;
+
+    if (!hfpmod.is_hfp_running) {
+        ALOGE("%s: HFP not active, ignoring set_hfp_mic_volume call", __func__);
+        return -EIO;
+    }
+
+    if (value < 0.0) {
+        ALOGW("%s: (%f) Under 0.0, assuming 0.0\n", __func__, value);
+        value = 0.0;
+    } else if (value > CAPTURE_VOLUME_DEFAULT) {
+        value = CAPTURE_VOLUME_DEFAULT;
+        ALOGW("%s: Volume brought within range (%f)\n", __func__, value);
+    }
+
+    value = value / CAPTURE_VOLUME_DEFAULT;
+    memset(mixer_ctl_name, 0, sizeof(mixer_ctl_name));
+    snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
+             "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 = (int)(value * PLAYBACK_VOLUME_MAX);
+
+    ALOGD("%s: Setting volume to %d (%s)\n", __func__, volume, mixer_ctl_name);
+    if (mixer_ctl_set_value(ctl, 0, volume) < 0) {
+        ALOGE("%s: Couldn't set HFP Volume: [%d]", __func__, volume);
+        return -EINVAL;
+    }
+
+    return ret;
+}
+
+static float hfp_get_mic_volume(struct audio_device *adev)
+{
+    int volume, ret = 0;
+    char mixer_ctl_name[128];
+    struct mixer_ctl *ctl;
+    int pcm_device_id = HFP_ASM_RX_TX;
+    float value = 0.0;
+
+    if (!hfpmod.is_hfp_running) {
+        ALOGE("%s: HFP not active, ignoring set_hfp_mic_volume call", __func__);
+        return -EIO;
+    }
+
+    memset(mixer_ctl_name, 0, sizeof(mixer_ctl_name));
+    snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
+             "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 = mixer_ctl_get_value(ctl, 0);
+    if ( volume < 0) {
+        ALOGE("%s: Couldn't set HFP Volume: [%d]", __func__, volume);
+        return -EINVAL;
+    }
+    ALOGD("%s: getting mic volume %d \n", __func__, volume);
+
+    value = (volume / PLAYBACK_VOLUME_MAX) * CAPTURE_VOLUME_DEFAULT;
+    if (value < 0.0) {
+        ALOGW("%s: (%f) Under 0.0, assuming 0.0\n", __func__, value);
+        value = 0.0;
+    } else if (value > CAPTURE_VOLUME_DEFAULT) {
+        value = CAPTURE_VOLUME_DEFAULT;
+        ALOGW("%s: Volume brought within range (%f)\n", __func__, value);
+    }
+
+    return value;
+}
+
+/*Set mic mute state.
+*
+* This interface is used for mic mute state control
+*/
+int audio_extn_hfp_set_mic_mute(struct audio_device *adev, bool state)
+{
+    int rc = 0;
+
+    if (state == hfpmod.mic_mute)
+        return rc;
+
+    if (state == true) {
+        hfpmod.mic_volume = hfp_get_mic_volume(adev);
+    }
+    rc = hfp_set_mic_volume(adev, (state == true) ? 0.0 : hfpmod.mic_volume);
+    adev->voice.mic_mute = state;
+    hfpmod.mic_mute = state;
+    ALOGD("%s: Setting mute state %d, rc %d\n", __func__, state, rc);
+    return rc;
+}
+
+static int32_t start_hfp(struct audio_device *adev,
+                         struct str_parms *parms __unused)
+{
+    int32_t i, ret = 0;
+    struct audio_usecase *uc_info;
+    int32_t pcm_dev_rx_id, pcm_dev_tx_id, pcm_dev_asm_rx_id, pcm_dev_asm_tx_id;
+
+    ALOGD("%s: enter", __func__);
+
+    if (adev->enable_hfp == true) {
+        ALOGD("%s: HFP is already active!\n", __func__);
+        return 0;
+    }
+    adev->enable_hfp = true;
+    platform_set_mic_mute(adev->platform, false);
+
+    uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
+    uc_info->id = hfpmod.ucid;
+    uc_info->type = PCM_HFP_CALL;
+    uc_info->stream.out = adev->primary_output;
+    uc_info->devices = adev->primary_output->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_tfa_98xx_set_mode_bt();
+
+    select_devices(adev, hfpmod.ucid);
+
+    pcm_dev_rx_id = platform_get_pcm_device_id(uc_info->id, PCM_PLAYBACK);
+    pcm_dev_tx_id = platform_get_pcm_device_id(uc_info->id, PCM_CAPTURE);
+    pcm_dev_asm_rx_id = HFP_ASM_RX_TX;
+    pcm_dev_asm_tx_id = HFP_ASM_RX_TX;
+    if (pcm_dev_rx_id < 0 || pcm_dev_tx_id < 0 ||
+        pcm_dev_asm_rx_id < 0 || pcm_dev_asm_tx_id < 0 ) {
+        ALOGE("%s: Invalid PCM devices (rx: %d tx: %d asm: rx tx %d) for the usecase(%d)",
+              __func__, pcm_dev_rx_id, pcm_dev_tx_id, pcm_dev_asm_rx_id, uc_info->id);
+        ret = -EIO;
+        goto exit;
+    }
+
+    ALOGV("%s: HFP PCM devices (hfp rx tx: %d pcm rx tx: %d) for the usecase(%d)",
+              __func__, pcm_dev_rx_id, pcm_dev_tx_id, uc_info->id);
+
+    ALOGV("%s: Opening PCM playback device card_id(%d) device_id(%d)",
+          __func__, adev->snd_card, pcm_dev_rx_id);
+    hfpmod.hfp_sco_rx = pcm_open(adev->snd_card,
+                                  pcm_dev_asm_rx_id,
+                                  PCM_OUT, &pcm_config_hfp);
+    if (hfpmod.hfp_sco_rx && !pcm_is_ready(hfpmod.hfp_sco_rx)) {
+        ALOGE("%s: %s", __func__, pcm_get_error(hfpmod.hfp_sco_rx));
+        ret = -EIO;
+        goto exit;
+    }
+    ALOGD("%s: Opening PCM capture device card_id(%d) device_id(%d)",
+          __func__, adev->snd_card, pcm_dev_tx_id);
+
+    if (audio_extn_tfa_98xx_is_supported() == false) {
+        hfpmod.hfp_pcm_rx = pcm_open(adev->snd_card,
+                                       pcm_dev_rx_id,
+                                       PCM_OUT, &pcm_config_hfp);
+        if (hfpmod.hfp_pcm_rx && !pcm_is_ready(hfpmod.hfp_pcm_rx)) {
+            ALOGE("%s: %s", __func__, pcm_get_error(hfpmod.hfp_pcm_rx));
+            ret = -EIO;
+            goto exit;
+        }
+    }
+    hfpmod.hfp_sco_tx = pcm_open(adev->snd_card,
+                                  pcm_dev_asm_tx_id,
+                                  PCM_IN, &pcm_config_hfp);
+    if (hfpmod.hfp_sco_tx && !pcm_is_ready(hfpmod.hfp_sco_tx)) {
+        ALOGE("%s: %s", __func__, pcm_get_error(hfpmod.hfp_sco_tx));
+        ret = -EIO;
+        goto exit;
+    }
+    ALOGV("%s: Opening PCM capture device card_id(%d) device_id(%d)",
+          __func__, adev->snd_card, pcm_dev_tx_id);
+
+    if (audio_extn_tfa_98xx_is_supported() == false) {
+        hfpmod.hfp_pcm_tx = pcm_open(adev->snd_card,
+                                       pcm_dev_tx_id,
+                                       PCM_IN, &pcm_config_hfp);
+        if (hfpmod.hfp_pcm_tx && !pcm_is_ready(hfpmod.hfp_pcm_tx)) {
+            ALOGE("%s: %s", __func__, pcm_get_error(hfpmod.hfp_pcm_tx));
+            ret = -EIO;
+            goto exit;
+        }
+    }
+    pcm_start(hfpmod.hfp_sco_rx);
+    pcm_start(hfpmod.hfp_sco_tx);
+    if (audio_extn_tfa_98xx_is_supported() == false) {
+        pcm_start(hfpmod.hfp_pcm_rx);
+        pcm_start(hfpmod.hfp_pcm_tx);
+    }
+
+    audio_extn_tfa_98xx_enable_speaker();
+
+    hfpmod.is_hfp_running = true;
+    hfp_set_volume(adev, hfpmod.hfp_volume);
+
+    /* Set mic volume by mute status, we don't provide set mic volume in phone app, only
+    provide mute and unmute. */
+    audio_extn_hfp_set_mic_mute(adev, adev->mic_muted);
+
+    ALOGD("%s: exit: status(%d)", __func__, ret);
+    return 0;
+
+exit:
+    stop_hfp(adev);
+    ALOGE("%s: Problem in HFP start: status(%d)", __func__, ret);
+    return ret;
+}
+
+static int32_t stop_hfp(struct audio_device *adev)
+{
+    int32_t i, ret = 0;
+    struct audio_usecase *uc_info;
+
+    ALOGD("%s: enter", __func__);
+    hfpmod.is_hfp_running = false;
+
+    /* 1. Close the PCM devices */
+    if (hfpmod.hfp_sco_rx) {
+        pcm_close(hfpmod.hfp_sco_rx);
+        hfpmod.hfp_sco_rx = NULL;
+    }
+    if (hfpmod.hfp_sco_tx) {
+        pcm_close(hfpmod.hfp_sco_tx);
+        hfpmod.hfp_sco_tx = NULL;
+    }
+    if (hfpmod.hfp_pcm_rx) {
+        pcm_close(hfpmod.hfp_pcm_rx);
+        hfpmod.hfp_pcm_rx = NULL;
+    }
+    if (hfpmod.hfp_pcm_tx) {
+        pcm_close(hfpmod.hfp_pcm_tx);
+        hfpmod.hfp_pcm_tx = NULL;
+    }
+
+    uc_info = get_usecase_from_list(adev, hfpmod.ucid);
+    if (uc_info == NULL) {
+        ALOGE("%s: Could not find the usecase (%d) in the list",
+              __func__, hfpmod.ucid);
+        return -EINVAL;
+    }
+
+    /* 2. Get and set stream specific mixer controls */
+    disable_audio_route(adev, uc_info);
+
+    /* 3. Disable the rx and tx devices */
+    disable_snd_device(adev, uc_info->out_snd_device);
+    disable_snd_device(adev, uc_info->in_snd_device);
+
+    /* Disable the echo reference for HFP Tx */
+    platform_set_echo_reference(adev, false, AUDIO_DEVICE_NONE);
+
+    /* Set the unmute Tx mixer control */
+    if (voice_get_mic_mute(adev)) {
+        platform_set_mic_mute(adev->platform, false);
+        ALOGD("%s: unMute HFP Tx", __func__);
+    }
+    adev->enable_hfp = false;
+
+    list_remove(&uc_info->list);
+    free(uc_info);
+
+    ALOGD("%s: exit: status(%d)", __func__, ret);
+    return ret;
+}
+
+bool audio_extn_hfp_is_active(struct audio_device *adev)
+{
+    struct audio_usecase *hfp_usecase = NULL;
+    hfp_usecase = get_usecase_from_list(adev, hfpmod.ucid);
+
+    if (hfp_usecase != NULL)
+        return true;
+    else
+        return false;
+}
+
+audio_usecase_t audio_extn_hfp_get_usecase()
+{
+    return hfpmod.ucid;
+}
+
+void audio_extn_hfp_set_parameters(struct audio_device *adev, struct str_parms *parms)
+{
+    int ret;
+    int rate;
+    int val;
+    float vol;
+    char value[AUDIO_PARAMATER_HFP_VALUE_MAX] = {0, };
+
+    ret = str_parms_get_str(parms, AUDIO_PARAMETER_HFP_ENABLE, value,
+                            sizeof(value));
+    if (ret >= 0) {
+           if (!strncmp(value,"true",sizeof(value)))
+               ret = start_hfp(adev,parms);
+           else
+               stop_hfp(adev);
+    }
+    memset(value, 0, sizeof(value));
+    ret = str_parms_get_str(parms,AUDIO_PARAMETER_HFP_SET_SAMPLING_RATE, value,
+                            sizeof(value));
+    if (ret >= 0) {
+           rate = atoi(value);
+           if (rate == 8000){
+               hfpmod.ucid = USECASE_AUDIO_HFP_SCO;
+               pcm_config_hfp.rate = rate;
+           } else if (rate == 16000){
+               hfpmod.ucid = USECASE_AUDIO_HFP_SCO_WB;
+               pcm_config_hfp.rate = rate;
+           } else
+               ALOGE("Unsupported rate..");
+    }
+
+    if (hfpmod.is_hfp_running) {
+        memset(value, 0, sizeof(value));
+        ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING,
+                                value, sizeof(value));
+        if (ret >= 0) {
+            val = atoi(value);
+            if (val > 0)
+                select_devices(adev, hfpmod.ucid);
+        }
+    }
+
+    memset(value, 0, sizeof(value));
+    ret = str_parms_get_str(parms, AUDIO_PARAMETER_HFP_VOL_MIXER_CTL,
+                          value, sizeof(value));
+    if (ret >= 0) {
+        ALOGD("%s: mixer ctl name: %s", __func__, value);
+        strcpy(hfpmod.hfp_vol_mixer_ctl, value);
+        str_parms_del(parms, AUDIO_PARAMETER_HFP_VOL_MIXER_CTL);
+    }
+
+    memset(value, 0, sizeof(value));
+    ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_HFP_VOLUME,
+                            value, sizeof(value));
+    if (ret >= 0) {
+        if (sscanf(value, "%f", &vol) != 1){
+            ALOGE("%s: error in retrieving hfp volume", __func__);
+            ret = -EIO;
+            goto exit;
+        }
+        ALOGD("%s: set_hfp_volume usecase, Vol: [%f]", __func__, vol);
+        hfp_set_volume(adev, vol);
+    }
+
+    memset(value, 0, sizeof(value));
+    ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_HFP_MIC_VOLUME,
+                            value, sizeof(value));
+    if (ret >= 0) {
+        if (sscanf(value, "%f", &vol) != 1){
+            ALOGE("%s: error in retrieving hfp mic volume", __func__);
+            ret = -EIO;
+            goto exit;
+        }
+        ALOGD("%s: set_hfp_mic_volume usecase, Vol: [%f]", __func__, vol);
+        hfp_set_mic_volume(adev, vol);
+    }
+
+exit:
+    ALOGV("%s Exit",__func__);
+}

android/hardware/qcom/audio/hal/audio_extn/hwdep_cal.c

@@ -0,0 +1,147 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "hardware_cal"
+/*#define LOG_NDEBUG 0*/
+#define LOG_NDDEBUG 0
+
+#ifdef HWDEP_CAL_ENABLED
+
+#include <stdlib.h>
+#include <dlfcn.h>
+#include <fcntl.h>
+#include <sys/ioctl.h>
+#include <unistd.h>
+#include <cutils/log.h>
+#include <audio_hw.h>
+#include "audio_extn.h"
+#include "sound/msmcal-hwdep.h"
+
+#define SOUND_TRIGGER_DEVICE_HANDSET_MONO_LOW_POWER_ACDB_ID (100)
+#define MAX_CAL_NAME 20
+
+typedef struct acdb_audio_cal_cfg {
+    uint32_t             persist;
+    uint32_t             snd_dev_id;
+    audio_devices_t      dev_id;
+    int32_t              acdb_dev_id;
+    uint32_t             app_type;
+    uint32_t             topo_id;
+    uint32_t             sampling_rate;
+    uint32_t             cal_type;
+    uint32_t             module_id;
+    uint32_t             param_id;
+} acdb_audio_cal_cfg_t;
+
+struct param_data {
+    int    use_case;
+    int    acdb_id;
+    int    get_size;
+    int    buff_size;
+    int    data_size;
+    void   *buff;
+};
+
+char cal_name_info[WCD9XXX_MAX_CAL][MAX_CAL_NAME] = {
+        [WCD9XXX_ANC_CAL] = "anc_cal",
+        [WCD9XXX_MBHC_CAL] = "mbhc_cal",
+        [WCD9XXX_MAD_CAL] = "mad_cal",
+};
+
+typedef int (*acdb_get_calibration_t)(char *attr, int size, void *data);
+acdb_get_calibration_t     acdb_get_calibration;
+
+static int hw_util_open(int card_no)
+{
+    int fd = -1;
+    char dev_name[256];
+
+    snprintf(dev_name, sizeof(dev_name), "/dev/snd/hwC%uD%u",
+                               card_no, WCD9XXX_CODEC_HWDEP_NODE);
+    ALOGD("%s: Opening device %s\n", __func__, dev_name);
+    fd = open(dev_name, O_WRONLY);
+    if (fd < 0) {
+        ALOGE("%s: cannot open device '%s'\n", __func__, dev_name);
+        return fd;
+    }
+    ALOGD("%s: success", __func__);
+    return fd;
+}
+
+static int send_codec_cal(acdb_get_calibration_t acdb_loader_get_calibration, int fd)
+{
+    int ret = 0, type;
+
+    for (type = WCD9XXX_ANC_CAL; type < WCD9XXX_MAX_CAL; type++) {
+        struct wcdcal_ioctl_buffer codec_buffer;
+        struct param_data calib;
+
+        if (!strcmp(cal_name_info[type], "mad_cal"))
+            calib.acdb_id = SOUND_TRIGGER_DEVICE_HANDSET_MONO_LOW_POWER_ACDB_ID;
+        calib.get_size = 1;
+        ret = acdb_loader_get_calibration(cal_name_info[type], sizeof(struct param_data),
+                                                                 &calib);
+        if (ret < 0) {
+            ALOGE("%s get_calibration failed\n", __func__);
+            return ret;
+        }
+        calib.get_size = 0;
+        calib.buff = malloc(calib.buff_size);
+        ret = acdb_loader_get_calibration(cal_name_info[type],
+                              sizeof(struct param_data), &calib);
+        if (ret < 0) {
+            ALOGE("%s get_calibration failed\n", __func__);
+            free(calib.buff);
+            return ret;
+        }
+        codec_buffer.buffer = calib.buff;
+        codec_buffer.size = calib.data_size;
+        codec_buffer.cal_type = type;
+        if (ioctl(fd, SNDRV_CTL_IOCTL_HWDEP_CAL_TYPE, &codec_buffer) < 0)
+            ALOGE("Failed to call ioctl  for %s err=%d",
+                                  cal_name_info[type], errno);
+        ALOGD("%s cal sent for %s", __func__, cal_name_info[type]);
+        free(calib.buff);
+    }
+    return ret;
+}
+
+
+void audio_extn_hwdep_cal_send(int snd_card, void *acdb_handle)
+{
+    int fd;
+
+    fd = hw_util_open(snd_card);
+    if (fd == -1) {
+        ALOGE("%s error open\n", __func__);
+        return;
+    }
+
+    acdb_get_calibration = (acdb_get_calibration_t)
+          dlsym(acdb_handle, "acdb_loader_get_calibration");
+
+    if (acdb_get_calibration == NULL) {
+        ALOGE("%s: ERROR. dlsym Error:%s acdb_loader_get_calibration", __func__,
+           dlerror());
+        return;
+    }
+    if (send_codec_cal(acdb_get_calibration, fd) < 0)
+        ALOGE("%s: Could not send anc cal", __FUNCTION__);
+
+    close(fd);
+}
+
+#endif

android/hardware/qcom/audio/hal/audio_extn/sndmonitor.c

@@ -0,0 +1,670 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "audio_hw_sndmonitor"
+/*#define LOG_NDEBUG 0*/
+#define LOG_NDDEBUG 0
+
+/* monitor sound card, cpe state
+
+   audio_dev registers for a callback from this module in adev_open
+   Each stream in audio_hal registers for a callback in
+   adev_open_*_stream.
+
+   A thread is spawned to poll() on sound card state files in /proc.
+   On observing a sound card state change, this thread invokes the
+   callbacks registered.
+
+   Callbacks are deregistered in adev_close_*_stream and adev_close
+*/
+#include <stdlib.h>
+#include <dirent.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/stat.h>
+#include <sys/poll.h>
+#include <pthread.h>
+#include <cutils/list.h>
+#include <cutils/hashmap.h>
+#include <cutils/log.h>
+#include <cutils/str_parms.h>
+#include <ctype.h>
+
+#include "audio_hw.h"
+#include "audio_extn.h"
+
+//#define MONITOR_DEVICE_EVENTS
+#define CPE_MAGIC_NUM 0x2000
+#define MAX_CPE_SLEEP_RETRY 2
+#define CPE_SLEEP_WAIT 100
+
+#define MAX_SLEEP_RETRY 100
+#define AUDIO_INIT_SLEEP_WAIT 100 /* 100 ms */
+
+#define AUDIO_PARAMETER_KEY_EXT_AUDIO_DEVICE "ext_audio_device"
+
+typedef enum {
+    audio_event_on,
+    audio_event_off
+} audio_event_status;
+
+typedef struct {
+    int card;
+    int fd;
+    struct listnode node; // membership in sndcards list
+    card_status_t status;
+} sndcard_t;
+
+typedef struct {
+    char * dev;
+    int fd;
+    int status;
+    struct listnode node; // membership in deviceevents list;
+} dev_event_t;
+
+typedef void (* notifyfn)(const void * target, const char * msg);
+
+typedef struct {
+    const void * target;
+    notifyfn notify;
+    struct listnode cards;
+    unsigned int num_cards;
+    struct listnode dev_events;
+    unsigned int num_dev_events;
+    pthread_t monitor_thread;
+    int intpipe[2];
+    Hashmap * listeners; // from stream * -> callback func
+    bool initcheck;
+} sndmonitor_state_t;
+
+static sndmonitor_state_t sndmonitor;
+
+static char * read_state(int fd)
+{
+    struct stat buf;
+    if (fstat(fd, &buf) < 0)
+        return NULL;
+
+    off_t pos = lseek(fd, 0, SEEK_CUR);
+    off_t avail = buf.st_size - pos;
+    if (avail <= 0) {
+        ALOGD("avail %ld", avail);
+        return NULL;
+    }
+
+    char * state = (char *)calloc(avail+1, sizeof(char));
+    if (!state)
+        return NULL;
+
+    ssize_t bytes=read(fd, state, avail);
+    if (bytes <= 0)
+        return NULL;
+
+    // trim trailing whitespace
+    while (bytes && isspace(*(state+bytes-1))) {
+        *(state + bytes - 1) = '\0';
+        --bytes;
+    }
+    lseek(fd, 0, SEEK_SET);
+    return state;
+}
+
+static int add_new_sndcard(int card, int fd)
+{
+    sndcard_t * s = (sndcard_t *)calloc(sizeof(sndcard_t), 1);
+
+    if (!s)
+        return -1;
+
+    s->card = card;
+    s->fd = fd; // dup?
+
+    char * state = read_state(fd);
+    bool online = state && !strcmp(state, "ONLINE");
+
+    ALOGV("card %d initial state %s %d", card, state, online);
+
+    if (state)
+        free(state);
+
+    s->status = online ? CARD_STATUS_ONLINE : CARD_STATUS_OFFLINE;
+    list_add_tail(&sndmonitor.cards, &s->node);
+    return 0;
+}
+
+static int enum_sndcards()
+{
+    const char* cards = "/proc/asound/cards";
+    int tries = 10;
+    char *line = NULL;
+    size_t len = 0;
+    ssize_t bytes_read;
+    char path[128] = {0};
+    char *ptr, *saveptr, *card_id;
+    int line_no=0;
+    unsigned int num_cards=0, num_cpe=0;
+    FILE *fp;
+    int fd, ret;
+
+    while (--tries) {
+        if ((fp = fopen(cards, "r")) == NULL) {
+            ALOGE("Cannot open %s file to get list of sound cards", cards);
+            usleep(100000);
+            continue;
+        }
+        break;
+    }
+
+    if (!tries)
+        return -ENODEV;
+
+    while ((bytes_read = getline(&line, &len, fp) != -1)) {
+        // skip every other line to to match
+        // the output format of /proc/asound/cards
+        if (line_no++ % 2)
+            continue;
+
+        ptr = strtok_r(line, " [", &saveptr);
+        if (!ptr)
+            continue;
+
+        card_id = strtok_r(saveptr+1, "]", &saveptr);
+        if (!card_id)
+            continue;
+
+        // Limit to sound cards associated with ADSP
+        if ((strncasecmp(card_id, "msm", 3) != 0) &&
+            (strncasecmp(card_id, "sdm", 3) != 0) &&
+            (strncasecmp(card_id, "sdc", 3) != 0) &&
+            (strncasecmp(card_id, "apq", 3) != 0)) {
+            ALOGW("Skip over non-ADSP snd card %s", card_id);
+            continue;
+        }
+
+        snprintf(path, sizeof(path), "/proc/asound/card%s/state", ptr);
+        ALOGV("Opening sound card state : %s", path);
+
+        fd = open(path, O_RDONLY);
+        if (fd == -1) {
+            ALOGE("Open %s failed : %s", path, strerror(errno));
+            continue;
+        }
+
+        ret = add_new_sndcard(atoi(ptr), fd);
+        if (ret != 0) {
+            close(fd); // card state fd ownership is taken by sndcard on success
+            continue;
+        }
+
+        num_cards++;
+
+        // query cpe state for this card as well
+        tries=MAX_CPE_SLEEP_RETRY;
+        snprintf(path, sizeof(path), "/proc/asound/card%s/cpe0_state", ptr);
+
+        if (access(path, R_OK) < 0) {
+            ALOGW("access %s failed w/ err %s", path, strerror(errno));
+            continue;
+        }
+
+        ALOGV("Open cpe state card state %s", path);
+        while (--tries) {
+            if ((fd = open(path, O_RDONLY)) < 0) {
+                ALOGW("Open cpe state card state failed, retry : %s", path);
+                usleep(CPE_SLEEP_WAIT*1000);
+                continue;
+            }
+            break;
+        }
+
+        if (!tries)
+            continue;
+
+        ret = add_new_sndcard(CPE_MAGIC_NUM+num_cpe, fd);
+        if (ret != 0) {
+            close(fd); // card state fd ownership is taken by sndcard on success
+            continue;
+        }
+
+        num_cpe++;
+        num_cards++;
+    }
+    if (line)
+        free(line);
+    fclose(fp);
+    ALOGV("sndmonitor registerer num_cards %d", num_cards);
+    sndmonitor.num_cards = num_cards;
+    return num_cards ? 0 : -1;
+}
+
+static void free_sndcards()
+{
+    while (!list_empty(&sndmonitor.cards)) {
+        struct listnode * n = list_head(&sndmonitor.cards);
+        sndcard_t * s = node_to_item(n, sndcard_t, node);
+        list_remove(n);
+        close(s->fd);
+        free(s);
+    }
+}
+
+static int add_new_dev_event(char * d_name, int fd)
+{
+    dev_event_t * d = (dev_event_t *)calloc(sizeof(dev_event_t), 1);
+
+    if (!d)
+        return -1;
+
+    d->dev = strdup(d_name);
+    d->fd = fd;
+    list_add_tail(&sndmonitor.dev_events, &d->node);
+    return 0;
+}
+
+static int enum_dev_events()
+{
+    const char* events_dir = "/sys/class/switch/";
+    DIR *dp;
+    struct dirent* in_file;
+    int fd;
+    char path[128] = {0};
+    unsigned int num_dev_events = 0;
+
+    if ((dp = opendir(events_dir)) == NULL) {
+        ALOGE("Cannot open switch directory %s err %s",
+              events_dir, strerror(errno));
+        return -1;
+    }
+
+    while ((in_file = readdir(dp)) != NULL) {
+        if (!strstr(in_file->d_name, "qc_"))
+            continue;
+
+        snprintf(path, sizeof(path), "%s/%s/state",
+                 events_dir, in_file->d_name);
+
+        ALOGV("Opening audio dev event state : %s ", path);
+        fd = open(path, O_RDONLY);
+        if (fd == -1) {
+            ALOGE("Open %s failed : %s", path, strerror(errno));
+        } else {
+            if (!add_new_dev_event(in_file->d_name, fd))
+                num_dev_events++;
+        }
+    }
+    closedir(dp);
+    sndmonitor.num_dev_events = num_dev_events;
+    return num_dev_events ? 0 : -1;
+}
+
+static void free_dev_events()
+{
+    while (!list_empty(&sndmonitor.dev_events)) {
+        struct listnode * n = list_head(&sndmonitor.dev_events);
+        dev_event_t * d = node_to_item(n, dev_event_t, node);
+        list_remove(n);
+        close(d->fd);
+        free(d->dev);
+        free(d);
+    }
+}
+
+static int notify(const struct str_parms * params)
+{
+    if (!params)
+        return -1;
+
+    char * str = str_parms_to_str((struct str_parms *)params);
+
+    if (!str)
+        return -1;
+
+    if (sndmonitor.notify)
+        sndmonitor.notify(sndmonitor.target, str);
+
+    ALOGV("%s", str);
+    free(str);
+    return 0;
+}
+
+int on_dev_event(dev_event_t * dev_event)
+{
+    char state_buf[2];
+    if (read(dev_event->fd, state_buf, 1) <= 0)
+        return -1;
+
+    lseek(dev_event->fd, 0, SEEK_SET);
+    state_buf[1]='\0';
+    if (atoi(state_buf) == dev_event->status)
+        return 0;
+
+    dev_event->status = atoi(state_buf);
+
+    struct str_parms * params = str_parms_create();
+
+    if (!params)
+        return -1;
+
+    char val[32] = {0};
+    snprintf(val, sizeof(val), "%s,%s", dev_event->dev,
+             dev_event->status ? "ON" : "OFF");
+
+    if (str_parms_add_str(params, AUDIO_PARAMETER_KEY_EXT_AUDIO_DEVICE, val) < 0)
+        return -1;
+
+    int ret = notify(params);
+    str_parms_destroy(params);
+    return ret;
+}
+
+bool on_sndcard_state_update(sndcard_t * s)
+{
+    char rd_buf[9]={0};
+    card_status_t status;
+
+    if (read(s->fd, rd_buf, 8) <= 0)
+        return -1;
+
+    rd_buf[8] = '\0';
+    lseek(s->fd, 0, SEEK_SET);
+
+    ALOGV("card num %d, new state %s", s->card, rd_buf);
+
+    bool is_cpe = (s->card >= CPE_MAGIC_NUM);
+    if (strstr(rd_buf, "OFFLINE"))
+        status = CARD_STATUS_OFFLINE;
+    else if (strstr(rd_buf, "ONLINE"))
+        status = CARD_STATUS_ONLINE;
+    else {
+        ALOGE("unknown state");
+        return 0;
+    }
+
+    if (status == s->status) // no change
+        return 0;
+
+    s->status = status;
+
+    struct str_parms * params = str_parms_create();
+
+    if (!params)
+        return -1;
+
+    char val[32] = {0};
+    // cpe actual card num is (card - MAGIC_NUM). so subtract accordingly
+    snprintf(val, sizeof(val), "%d,%s", s->card - (is_cpe ? CPE_MAGIC_NUM : 0),
+                 status == CARD_STATUS_ONLINE ? "ONLINE" : "OFFLINE");
+
+    if (str_parms_add_str(params, is_cpe ? "CPE_STATUS" : "SND_CARD_STATUS",
+                          val) < 0)
+        return -1;
+
+    int ret = notify(params);
+    str_parms_destroy(params);
+    return ret;
+}
+
+void * monitor_thread_loop(void * args __unused)
+{
+    ALOGV("Start threadLoop()");
+    unsigned int num_poll_fds = sndmonitor.num_cards +
+                                sndmonitor.num_dev_events + 1/*pipe*/;
+    struct pollfd * pfd = (struct pollfd *)calloc(sizeof(struct pollfd),
+                                                  num_poll_fds);
+    if (!pfd)
+        return NULL;
+
+    pfd[0].fd = sndmonitor.intpipe[0];
+    pfd[0].events = POLLPRI|POLLIN;
+
+    int i=1;
+    struct listnode *node;
+    list_for_each(node, &sndmonitor.cards) {
+        sndcard_t * s = node_to_item(node, sndcard_t, node);
+        pfd[i].fd = s->fd;
+        pfd[i].events = POLLPRI;
+        ++i;
+    }
+
+    list_for_each(node, &sndmonitor.dev_events) {
+        dev_event_t * d = node_to_item(node, dev_event_t, node);
+        pfd[i].fd = d->fd;
+        pfd[i].events = POLLPRI;
+        ++i;
+    }
+
+    while (1) {
+        if (poll(pfd, num_poll_fds, -1) < 0) {
+            int errno_ = errno;
+            ALOGE("poll() failed w/ err %s", strerror(errno));
+            switch (errno_) {
+            case EINTR:
+            case ENOMEM:
+                sleep(2);
+                continue;
+            default:
+                /* above errors can be caused due to current system
+                   state .. any other error is not expected */
+                LOG_ALWAYS_FATAL("unxpected poll() system call failure");
+                break;
+            }
+        }
+        ALOGV("out of poll()");
+
+#define READY_TO_READ(p) ((p)->revents & (POLLIN|POLLPRI))
+#define ERROR_IN_FD(p) ((p)->revents & (POLLERR|POLLHUP|POLLNVAL))
+
+        // check if requested to exit
+        if (READY_TO_READ(&pfd[0])) {
+            char buf[2]={0};
+            read(pfd[0].fd, buf, 1);
+            if (!strcmp(buf, "Q"))
+                break;
+        } else if (ERROR_IN_FD(&pfd[0])) {
+            // do not consider for poll again
+            // POLLERR - can this happen?
+            // POLLHUP - adev must not close pipe
+            // POLLNVAL - fd is valid
+            LOG_ALWAYS_FATAL("unxpected error in pipe poll fd 0x%x",
+                             pfd[0].revents);
+            pfd[0].fd *= -1;
+        }
+
+        i=1;
+        list_for_each(node, &sndmonitor.cards) {
+            sndcard_t * s = node_to_item(node, sndcard_t, node);
+            if (READY_TO_READ(&pfd[i]))
+                on_sndcard_state_update(s);
+            else if (ERROR_IN_FD(&pfd[i])) {
+                // do not consider for poll again
+                // POLLERR - can this happen as we are reading from a fs?
+                // POLLHUP - not valid for cardN/state
+                // POLLNVAL - fd is valid
+                LOG_ALWAYS_FATAL("unxpected error in card poll fd 0x%x",
+                                 pfd[i].revents);
+                pfd[i].fd *= -1;
+            }
+            ++i;
+        }
+
+        list_for_each(node, &sndmonitor.dev_events) {
+            dev_event_t * d = node_to_item(node, dev_event_t, node);
+            if (READY_TO_READ(&pfd[i]))
+                on_dev_event(d);
+            else if (ERROR_IN_FD(&pfd[i])) {
+                // do not consider for poll again
+                // POLLERR - can this happen as we are reading from a fs?
+                // POLLHUP - not valid for switch/state
+                // POLLNVAL - fd is valid
+                LOG_ALWAYS_FATAL("unxpected error in dev poll fd 0x%x",
+                                 pfd[i].revents);
+                pfd[i].fd *= -1;
+            }
+            ++i;
+        }
+    }
+
+    return NULL;
+}
+
+// ---- listener static APIs ---- //
+static int hashfn(void * key)
+{
+    return (int)key;
+}
+
+static bool hasheq(void * key1, void *key2)
+{
+    return key1 == key2;
+}
+
+static bool snd_cb(void* key, void* value, void* context)
+{
+    snd_mon_cb cb = (snd_mon_cb)value;
+    cb(key, context);
+    return true;
+}
+
+static void snd_mon_update(const void * target __unused, const char * msg)
+{
+    // target can be used to check if this message is intended for the
+    // recipient or not. (using some statically saved state)
+
+    struct str_parms *parms = str_parms_create_str(msg);
+
+    if (!parms)
+        return;
+
+    hashmapLock(sndmonitor.listeners);
+    hashmapForEach(sndmonitor.listeners, snd_cb, parms);
+    hashmapUnlock(sndmonitor.listeners);
+
+    str_parms_destroy(parms);
+}
+
+static int listeners_init()
+{
+    sndmonitor.listeners = hashmapCreate(5, hashfn, hasheq);
+    if (!sndmonitor.listeners)
+        return -1;
+    return 0;
+}
+
+static int listeners_deinit()
+{
+    // XXX TBD
+    return -1;
+}
+
+static int add_listener(void *stream, snd_mon_cb cb)
+{
+    Hashmap * map = sndmonitor.listeners;
+    hashmapLock(map);
+    hashmapPut(map, stream, cb);
+    hashmapUnlock(map);
+    return 0;
+}
+
+static int del_listener(void * stream)
+{
+    Hashmap * map = sndmonitor.listeners;
+    hashmapLock(map);
+    hashmapRemove(map, stream);
+    hashmapUnlock(map);
+    return 0;
+}
+
+// --- public APIs --- //
+
+int audio_extn_snd_mon_deinit()
+{
+    if (!sndmonitor.initcheck)
+        return -1;
+
+    write(sndmonitor.intpipe[1], "Q", 1);
+    pthread_join(sndmonitor.monitor_thread, (void **) NULL);
+    free_dev_events();
+    listeners_deinit();
+    free_sndcards();
+    close(sndmonitor.intpipe[0]);
+    close(sndmonitor.intpipe[1]);
+
+    sndmonitor.initcheck = 0;
+    return 0;
+}
+
+int audio_extn_snd_mon_init()
+{
+    sndmonitor.notify = snd_mon_update;
+    sndmonitor.target = NULL; // unused for now
+    list_init(&sndmonitor.cards);
+    list_init(&sndmonitor.dev_events);
+    sndmonitor.initcheck = false;
+
+    if (pipe(sndmonitor.intpipe) < 0)
+        goto pipe_error;
+
+    if (enum_sndcards() < 0)
+        goto enum_sncards_error;
+
+    if (listeners_init() < 0)
+        goto listeners_error;
+
+#ifdef MONITOR_DEVICE_EVENTS
+    enum_dev_events(); // failure here isn't fatal
+#endif
+
+    int ret = pthread_create(&sndmonitor.monitor_thread,
+                             (const pthread_attr_t *) NULL,
+                             monitor_thread_loop, NULL);
+
+    if (ret) {
+        goto monitor_thread_create_error;
+    }
+    sndmonitor.initcheck = true;
+    return 0;
+
+monitor_thread_create_error:
+    listeners_deinit();
+listeners_error:
+    free_sndcards();
+enum_sncards_error:
+    close(sndmonitor.intpipe[0]);
+    close(sndmonitor.intpipe[1]);
+pipe_error:
+    return -ENODEV;
+}
+
+int audio_extn_snd_mon_register_listener(void *stream, snd_mon_cb cb)
+{
+    if (!sndmonitor.initcheck) {
+        ALOGW("sndmonitor initcheck failed, cannot register");
+        return -1;
+    }
+
+    return add_listener(stream, cb);
+}
+
+int audio_extn_snd_mon_unregister_listener(void * stream)
+{
+    if (!sndmonitor.initcheck) {
+        ALOGW("sndmonitor initcheck failed, cannot deregister");
+        return -1;
+    }
+
+    ALOGV("deregister listener for stream %p ", stream);
+    return del_listener(stream);
+}

android/hardware/qcom/audio/hal/audio_extn/soundtrigger.c

@@ -0,0 +1,369 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#define LOG_TAG "soundtrigger"
+/* #define LOG_NDEBUG 0 */
+#define LOG_NDDEBUG 0
+
+#include <errno.h>
+#include <stdbool.h>
+#include <stdlib.h>
+#include <dlfcn.h>
+#include <pthread.h>
+#include <unistd.h>
+#include <cutils/log.h>
+#include "audio_hw.h"
+#include "audio_extn.h"
+#include "platform.h"
+#include "platform_api.h"
+#include "sound_trigger_prop_intf.h"
+
+#define XSTR(x) STR(x)
+#define STR(x) #x
+
+#ifdef __LP64__
+#define SOUND_TRIGGER_LIBRARY_PATH "/system/vendor/lib64/hw/sound_trigger.primary.%s.so"
+#else
+#define SOUND_TRIGGER_LIBRARY_PATH "/system/vendor/lib/hw/sound_trigger.primary.%s.so"
+#endif
+
+struct sound_trigger_info  {
+    struct sound_trigger_session_info st_ses;
+    bool lab_stopped;
+    struct listnode list;
+};
+
+struct sound_trigger_audio_device {
+    void *lib_handle;
+    struct audio_device *adev;
+    sound_trigger_hw_call_back_t st_callback;
+    struct listnode st_ses_list;
+    pthread_mutex_t lock;
+};
+
+static struct sound_trigger_audio_device *st_dev;
+
+static struct sound_trigger_info *
+get_sound_trigger_info(int capture_handle)
+{
+    struct sound_trigger_info  *st_ses_info = NULL;
+    struct listnode *node;
+    ALOGV("%s: list %d capture_handle %d", __func__,
+           list_empty(&st_dev->st_ses_list), capture_handle);
+    list_for_each(node, &st_dev->st_ses_list) {
+        st_ses_info = node_to_item(node, struct sound_trigger_info , list);
+        if (st_ses_info->st_ses.capture_handle == capture_handle)
+            return st_ses_info;
+    }
+    return NULL;
+}
+
+static void stdev_snd_mon_cb(void * stream __unused, struct str_parms * parms)
+{
+    if (!parms)
+        return;
+
+    audio_extn_sound_trigger_set_parameters(NULL, parms);
+    return;
+}
+
+int audio_hw_call_back(sound_trigger_event_type_t event,
+                       sound_trigger_event_info_t* config)
+{
+    int status = 0;
+    struct sound_trigger_info  *st_ses_info;
+
+    if (!st_dev)
+       return -EINVAL;
+
+    pthread_mutex_lock(&st_dev->lock);
+    switch (event) {
+    case ST_EVENT_SESSION_REGISTER:
+        if (!config) {
+            ALOGE("%s: NULL config", __func__);
+            status = -EINVAL;
+            break;
+        }
+        st_ses_info= calloc(1, sizeof(struct sound_trigger_info ));
+        if (!st_ses_info) {
+            ALOGE("%s: st_ses_info alloc failed", __func__);
+            status = -ENOMEM;
+            break;
+        }
+        memcpy(&st_ses_info->st_ses, &config->st_ses, sizeof (config->st_ses));
+        ALOGV("%s: add capture_handle %d pcm %p", __func__,
+              st_ses_info->st_ses.capture_handle, st_ses_info->st_ses.pcm);
+        list_add_tail(&st_dev->st_ses_list, &st_ses_info->list);
+        break;
+
+    case ST_EVENT_SESSION_DEREGISTER:
+        if (!config) {
+            ALOGE("%s: NULL config", __func__);
+            status = -EINVAL;
+            break;
+        }
+        st_ses_info = get_sound_trigger_info(config->st_ses.capture_handle);
+        if (!st_ses_info) {
+            ALOGE("%s: pcm %p not in the list!", __func__, config->st_ses.pcm);
+            status = -EINVAL;
+            break;
+        }
+        ALOGV("%s: remove capture_handle %d pcm %p", __func__,
+              st_ses_info->st_ses.capture_handle, st_ses_info->st_ses.pcm);
+        list_remove(&st_ses_info->list);
+        free(st_ses_info);
+        break;
+    default:
+        ALOGW("%s: Unknown event %d", __func__, event);
+        break;
+    }
+    pthread_mutex_unlock(&st_dev->lock);
+    return status;
+}
+
+int audio_extn_sound_trigger_read(struct stream_in *in, void *buffer,
+                       size_t bytes)
+{
+    int ret = -1;
+    struct sound_trigger_info  *st_info = NULL;
+    audio_event_info_t event;
+
+    if (!st_dev)
+       return ret;
+
+    if (!in->is_st_session_active) {
+        ALOGE(" %s: Sound trigger is not active", __func__);
+        goto exit;
+    }
+    if (in->standby)
+        in->standby = false;
+
+    pthread_mutex_lock(&st_dev->lock);
+    st_info = get_sound_trigger_info(in->capture_handle);
+    pthread_mutex_unlock(&st_dev->lock);
+    if (st_info) {
+        event.u.aud_info.ses_info = &st_info->st_ses;
+        event.u.aud_info.buf = buffer;
+        event.u.aud_info.num_bytes = bytes;
+        ret = st_dev->st_callback(AUDIO_EVENT_READ_SAMPLES, &event);
+    }
+
+exit:
+    if (ret) {
+        if (-ENETRESET == ret)
+            in->is_st_session_active = false;
+        memset(buffer, 0, bytes);
+        ALOGV("%s: read failed status %d - sleep", __func__, ret);
+        usleep((bytes * 1000000) / (audio_stream_in_frame_size((struct audio_stream_in *)in) *
+                                   in->config.rate));
+    }
+    return ret;
+}
+
+void audio_extn_sound_trigger_stop_lab(struct stream_in *in)
+{
+    int status = 0;
+    struct sound_trigger_info  *st_ses_info = NULL;
+    audio_event_info_t event;
+
+    if (!st_dev || !in)
+       return;
+
+    pthread_mutex_lock(&st_dev->lock);
+    st_ses_info = get_sound_trigger_info(in->capture_handle);
+    pthread_mutex_unlock(&st_dev->lock);
+    if (st_ses_info) {
+        event.u.ses_info = st_ses_info->st_ses;
+        ALOGV("%s: AUDIO_EVENT_STOP_LAB pcm %p", __func__, st_ses_info->st_ses.pcm);
+        st_dev->st_callback(AUDIO_EVENT_STOP_LAB, &event);
+    }
+}
+void audio_extn_sound_trigger_check_and_get_session(struct stream_in *in)
+{
+    struct sound_trigger_info  *st_ses_info = NULL;
+    struct listnode *node;
+
+    if (!st_dev || !in)
+       return;
+
+    pthread_mutex_lock(&st_dev->lock);
+    in->is_st_session = false;
+    ALOGV("%s: list %d capture_handle %d", __func__,
+          list_empty(&st_dev->st_ses_list), in->capture_handle);
+    list_for_each(node, &st_dev->st_ses_list) {
+        st_ses_info = node_to_item(node, struct sound_trigger_info , list);
+        if (st_ses_info->st_ses.capture_handle == in->capture_handle) {
+            in->pcm = st_ses_info->st_ses.pcm;
+            in->config = st_ses_info->st_ses.config;
+            in->channel_mask = audio_channel_in_mask_from_count(in->config.channels);
+            in->is_st_session = true;
+            in->is_st_session_active = true;
+            ALOGV("%s: capture_handle %d is sound trigger", __func__, in->capture_handle);
+            break;
+        }
+    }
+    pthread_mutex_unlock(&st_dev->lock);
+}
+
+void audio_extn_sound_trigger_update_device_status(snd_device_t snd_device,
+                                     st_event_type_t event)
+{
+    int device_type = -1;
+
+    if (!st_dev)
+       return;
+
+    if (snd_device >= SND_DEVICE_OUT_BEGIN &&
+        snd_device < SND_DEVICE_OUT_END) {
+        device_type = PCM_PLAYBACK;
+    } else if (snd_device >= SND_DEVICE_IN_BEGIN &&
+        snd_device < SND_DEVICE_IN_END) {
+        if (snd_device == SND_DEVICE_IN_CAPTURE_VI_FEEDBACK)
+            return;
+        device_type = PCM_CAPTURE;
+    } else {
+        ALOGE("%s: invalid device 0x%x, for event %d",
+                           __func__, snd_device, event);
+        return;
+    }
+
+    ALOGV("%s: device 0x%x of type %d for Event %d",
+        __func__, snd_device, device_type, event);
+    if (device_type == PCM_CAPTURE) {
+        switch(event) {
+        case ST_EVENT_SND_DEVICE_FREE:
+            st_dev->st_callback(AUDIO_EVENT_CAPTURE_DEVICE_INACTIVE, NULL);
+            break;
+        case ST_EVENT_SND_DEVICE_BUSY:
+            st_dev->st_callback(AUDIO_EVENT_CAPTURE_DEVICE_ACTIVE, NULL);
+            break;
+        default:
+            ALOGW("%s:invalid event %d for device 0x%x",
+                                  __func__, event, snd_device);
+        }
+    }/*Events for output device, if required can be placed here in else*/
+}
+
+void audio_extn_sound_trigger_set_parameters(struct audio_device *adev __unused,
+                               struct str_parms *params)
+{
+    audio_event_info_t event;
+    char value[32];
+    int ret, val;
+
+    if(!st_dev || !params) {
+        ALOGE("%s: str_params NULL", __func__);
+        return;
+    }
+
+    ret = str_parms_get_str(params, "SND_CARD_STATUS", value,
+                            sizeof(value));
+    if (ret > 0) {
+        if (strstr(value, "OFFLINE")) {
+            event.u.status = SND_CARD_STATUS_OFFLINE;
+            st_dev->st_callback(AUDIO_EVENT_SSR, &event);
+        }
+        else if (strstr(value, "ONLINE")) {
+            event.u.status = SND_CARD_STATUS_ONLINE;
+            st_dev->st_callback(AUDIO_EVENT_SSR, &event);
+        }
+        else
+            ALOGE("%s: unknown snd_card_status", __func__);
+    }
+
+    ret = str_parms_get_str(params, "CPE_STATUS", value, sizeof(value));
+    if (ret > 0) {
+        if (strstr(value, "OFFLINE")) {
+            event.u.status = CPE_STATUS_OFFLINE;
+            st_dev->st_callback(AUDIO_EVENT_SSR, &event);
+        }
+        else if (strstr(value, "ONLINE")) {
+            event.u.status = CPE_STATUS_ONLINE;
+            st_dev->st_callback(AUDIO_EVENT_SSR, &event);
+        }
+        else
+            ALOGE("%s: unknown CPE status", __func__);
+    }
+
+    ret = str_parms_get_int(params, "SVA_NUM_SESSIONS", &val);
+    if (ret >= 0) {
+        event.u.value = val;
+        st_dev->st_callback(AUDIO_EVENT_NUM_ST_SESSIONS, &event);
+    }
+}
+
+int audio_extn_sound_trigger_init(struct audio_device *adev)
+{
+    int status = 0;
+    char sound_trigger_lib[100];
+    void *lib_handle;
+
+    ALOGV("%s: Enter", __func__);
+
+    st_dev = (struct sound_trigger_audio_device*)
+                        calloc(1, sizeof(struct sound_trigger_audio_device));
+    if (!st_dev) {
+        ALOGE("%s: ERROR. sound trigger alloc failed", __func__);
+        return -ENOMEM;
+    }
+
+    snprintf(sound_trigger_lib, sizeof(sound_trigger_lib),
+             SOUND_TRIGGER_LIBRARY_PATH,
+              XSTR(SOUND_TRIGGER_PLATFORM_NAME));
+
+    st_dev->lib_handle = dlopen(sound_trigger_lib, RTLD_NOW);
+
+    if (st_dev->lib_handle == NULL) {
+        ALOGE("%s: DLOPEN failed for %s. error = %s", __func__, sound_trigger_lib,
+                dlerror());
+        status = -EINVAL;
+        goto cleanup;
+    }
+    ALOGV("%s: DLOPEN successful for %s", __func__, sound_trigger_lib);
+
+    st_dev->st_callback = (sound_trigger_hw_call_back_t)
+              dlsym(st_dev->lib_handle, "sound_trigger_hw_call_back");
+
+    if (st_dev->st_callback == NULL) {
+       ALOGE("%s: ERROR. dlsym Error:%s sound_trigger_hw_call_back", __func__,
+               dlerror());
+       goto cleanup;
+    }
+
+    st_dev->adev = adev;
+    list_init(&st_dev->st_ses_list);
+    audio_extn_snd_mon_register_listener(st_dev, stdev_snd_mon_cb);
+
+    return 0;
+
+cleanup:
+    if (st_dev->lib_handle)
+        dlclose(st_dev->lib_handle);
+    free(st_dev);
+    st_dev = NULL;
+    return status;
+
+}
+
+void audio_extn_sound_trigger_deinit(struct audio_device *adev)
+{
+    ALOGV("%s: Enter", __func__);
+    if (st_dev && (st_dev->adev == adev) && st_dev->lib_handle) {
+        audio_extn_snd_mon_unregister_listener(st_dev);
+        dlclose(st_dev->lib_handle);
+        free(st_dev);
+        st_dev = NULL;
+    }
+}

android/hardware/qcom/audio/hal/audio_extn/spkr_protection.c

@@ -0,0 +1,907 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "audio_hw_spkr_prot"
+/*#define LOG_NDEBUG 0*/
+//#define LOG_NDDEBUG 0
+
+#include <errno.h>
+#include <math.h>
+#include <cutils/log.h>
+#include <fcntl.h>
+#include "audio_hw.h"
+#include "platform.h"
+#include "platform_api.h"
+#include <sys/stat.h>
+#include <stdlib.h>
+#include <dlfcn.h>
+#include <math.h>
+#include <cutils/properties.h>
+#include "audio_extn.h"
+#include <linux/msm_audio_calibration.h>
+
+#define THERMAL_CLIENT_LIBRARY_PATH "libthermalclient.so"
+
+#ifdef SPKR_PROT_ENABLED
+
+/*Range of spkr temparatures -30C to 80C*/
+#define MIN_SPKR_TEMP_Q6 (-30 * (1 << 6))
+#define MAX_SPKR_TEMP_Q6 (80 * (1 << 6))
+#define VI_FEED_CHANNEL "VI_FEED_TX Channels"
+
+/*Set safe temp value to 40C*/
+#define SAFE_SPKR_TEMP 40
+#define SAFE_SPKR_TEMP_Q6 (SAFE_SPKR_TEMP * (1 << 6))
+
+/*Range of resistance values 2ohms to 40 ohms*/
+#define MIN_RESISTANCE_SPKR_Q24 (2 * (1 << 24))
+#define MAX_RESISTANCE_SPKR_Q24 (40 * (1 << 24))
+
+/*Path where the calibration file will be stored*/
+#define CALIB_FILE "/data/misc/audio/audio.cal"
+
+/*Time between retries for calibartion or intial wait time
+  after boot up*/
+#define WAIT_TIME_SPKR_CALIB (60 * 1000 * 1000)
+
+#define MIN_SPKR_IDLE_SEC (60 * 30)
+
+/*Once calibration is started sleep for 1 sec to allow
+  the calibration to kick off*/
+#define SLEEP_AFTER_CALIB_START (3000)
+
+/*If calibration is in progress wait for 200 msec before querying
+  for status again*/
+#define WAIT_FOR_GET_CALIB_STATUS (200)
+#define GET_SPKR_PROT_CAL_TIMEOUT_MSEC (5000)
+
+/*Speaker states*/
+#define SPKR_NOT_CALIBRATED -1
+#define SPKR_CALIBRATED 1
+
+/*Speaker processing state*/
+#define SPKR_PROCESSING_IN_PROGRESS 1
+#define SPKR_PROCESSING_IN_IDLE 0
+
+/*Modes of Speaker Protection*/
+enum speaker_protection_mode {
+    SPKR_PROTECTION_DISABLED = -1,
+    SPKR_PROTECTION_MODE_PROCESSING = 0,
+    SPKR_PROTECTION_MODE_CALIBRATE = 1,
+};
+
+struct speaker_prot_session {
+    int spkr_prot_mode;
+    int spkr_processing_state;
+    int thermal_client_handle;
+    pthread_mutex_t mutex_spkr_prot;
+    pthread_t spkr_calibration_thread;
+    pthread_mutex_t spkr_prot_thermalsync_mutex;
+    pthread_cond_t spkr_prot_thermalsync;
+    int cancel_spkr_calib;
+    pthread_cond_t spkr_calib_cancel;
+    pthread_mutex_t spkr_calib_cancelack_mutex;
+    pthread_cond_t spkr_calibcancel_ack;
+    pthread_t speaker_prot_threadid;
+    void *thermal_handle;
+    void *adev_handle;
+    int spkr_prot_t0;
+    struct pcm *pcm_rx;
+    struct pcm *pcm_tx;
+    int (*thermal_client_register_callback)
+    (char *client_name, int (*callback)(int), void *data);
+    void (*thermal_client_unregister_callback)(int handle);
+    int (*thermal_client_request)(char *client_name, int req_data);
+    bool spkr_prot_enable;
+    bool spkr_in_use;
+   struct timespec spkr_last_time_used;
+};
+
+static struct pcm_config pcm_config_skr_prot = {
+    .channels = 4,
+    .rate = 48000,
+    .period_size = 256,
+    .period_count = 4,
+    .format = PCM_FORMAT_S16_LE,
+    .start_threshold = 0,
+    .stop_threshold = INT_MAX,
+    .avail_min = 0,
+};
+
+static struct speaker_prot_session handle;
+static int vi_feed_no_channels;
+
+static void spkr_prot_set_spkrstatus(bool enable)
+{
+    struct timespec ts;
+    if (enable)
+       handle.spkr_in_use = true;
+    else {
+       handle.spkr_in_use = false;
+       clock_gettime(CLOCK_BOOTTIME, &handle.spkr_last_time_used);
+   }
+}
+
+void audio_extn_spkr_prot_calib_cancel(void *adev)
+{
+    pthread_t threadid;
+    struct audio_usecase *uc_info;
+    int count = 0;
+    threadid = pthread_self();
+    ALOGV("%s: Entry", __func__);
+    if (pthread_equal(handle.speaker_prot_threadid, threadid) || !adev) {
+        ALOGV("%s: Calibration not in progress.. nothihg to cancel", __func__);
+        return;
+    }
+    uc_info = get_usecase_from_list(adev, USECASE_AUDIO_SPKR_CALIB_RX);
+    if (uc_info) {
+            pthread_mutex_lock(&handle.mutex_spkr_prot);
+            pthread_mutex_lock(&handle.spkr_calib_cancelack_mutex);
+            handle.cancel_spkr_calib = 1;
+            pthread_cond_signal(&handle.spkr_calib_cancel);
+            pthread_mutex_unlock(&handle.mutex_spkr_prot);
+            pthread_cond_wait(&handle.spkr_calibcancel_ack,
+            &handle.spkr_calib_cancelack_mutex);
+            pthread_mutex_unlock(&handle.spkr_calib_cancelack_mutex);
+    }
+    ALOGV("%s: Exit", __func__);
+}
+
+static bool is_speaker_in_use(unsigned long *sec)
+{
+    struct timespec temp;
+    if (!sec) {
+        ALOGE("%s: Invalid params", __func__);
+        return true;
+    }
+     if (handle.spkr_in_use) {
+        *sec = 0;
+         return true;
+     } else {
+         clock_gettime(CLOCK_BOOTTIME, &temp);
+         *sec = temp.tv_sec - handle.spkr_last_time_used.tv_sec;
+         return false;
+     }
+}
+
+
+static int get_spkr_prot_cal(int cal_fd,
+				struct audio_cal_info_msm_spk_prot_status *status)
+{
+    int ret = 0;
+    struct audio_cal_fb_spk_prot_status    cal_data;
+
+    if (cal_fd < 0) {
+        ALOGE("%s: Error: cal_fd = %d", __func__, cal_fd);
+        ret = -EINVAL;
+        goto done;
+    }
+
+    if (status == NULL) {
+        ALOGE("%s: Error: status NULL", __func__);
+        ret = -EINVAL;
+        goto done;
+    }
+
+    cal_data.hdr.data_size = sizeof(cal_data);
+    cal_data.hdr.version = VERSION_0_0;
+    cal_data.hdr.cal_type = AFE_FB_SPKR_PROT_CAL_TYPE;
+    cal_data.hdr.cal_type_size = sizeof(cal_data.cal_type);
+    cal_data.cal_type.cal_hdr.version = VERSION_0_0;
+    cal_data.cal_type.cal_hdr.buffer_number = 0;
+    cal_data.cal_type.cal_data.mem_handle = -1;
+
+    if (ioctl(cal_fd, AUDIO_GET_CALIBRATION, &cal_data)) {
+        ALOGE("%s: Error: AUDIO_GET_CALIBRATION failed!",
+            __func__);
+        ret = -ENODEV;
+        goto done;
+    }
+
+    status->r0[SP_V2_SPKR_1] = cal_data.cal_type.cal_info.r0[SP_V2_SPKR_1];
+    status->r0[SP_V2_SPKR_2] = cal_data.cal_type.cal_info.r0[SP_V2_SPKR_2];
+    status->status = cal_data.cal_type.cal_info.status;
+done:
+    return ret;
+}
+
+static int set_spkr_prot_cal(int cal_fd,
+				struct audio_cal_info_spk_prot_cfg *protCfg)
+{
+    int ret = 0;
+    struct audio_cal_fb_spk_prot_cfg    cal_data;
+    char value[PROPERTY_VALUE_MAX];
+
+    if (cal_fd < 0) {
+        ALOGE("%s: Error: cal_fd = %d", __func__, cal_fd);
+        ret = -EINVAL;
+        goto done;
+    }
+
+    if (protCfg == NULL) {
+        ALOGE("%s: Error: status NULL", __func__);
+        ret = -EINVAL;
+        goto done;
+    }
+
+    memset(&cal_data, 0, sizeof(cal_data));
+    cal_data.hdr.data_size = sizeof(cal_data);
+    cal_data.hdr.version = VERSION_0_0;
+    cal_data.hdr.cal_type = AFE_FB_SPKR_PROT_CAL_TYPE;
+    cal_data.hdr.cal_type_size = sizeof(cal_data.cal_type);
+    cal_data.cal_type.cal_hdr.version = VERSION_0_0;
+    cal_data.cal_type.cal_hdr.buffer_number = 0;
+    cal_data.cal_type.cal_info.r0[SP_V2_SPKR_1] = protCfg->r0[SP_V2_SPKR_1];
+    cal_data.cal_type.cal_info.r0[SP_V2_SPKR_2] = protCfg->r0[SP_V2_SPKR_2];
+    cal_data.cal_type.cal_info.t0[SP_V2_SPKR_1] = protCfg->t0[SP_V2_SPKR_1];
+    cal_data.cal_type.cal_info.t0[SP_V2_SPKR_2] = protCfg->t0[SP_V2_SPKR_2];
+    cal_data.cal_type.cal_info.mode = protCfg->mode;
+    property_get("persist.spkr.cal.duration", value, "0");
+    if (atoi(value) > 0) {
+        ALOGD("%s: quick calibration enabled", __func__);
+        cal_data.cal_type.cal_info.quick_calib_flag = 1;
+    } else {
+        ALOGD("%s: quick calibration disabled", __func__);
+        cal_data.cal_type.cal_info.quick_calib_flag = 0;
+    }
+
+    cal_data.cal_type.cal_data.mem_handle = -1;
+
+    if (ioctl(cal_fd, AUDIO_SET_CALIBRATION, &cal_data)) {
+        ALOGE("%s: Error: AUDIO_SET_CALIBRATION failed!",
+            __func__);
+        ret = -ENODEV;
+        goto done;
+    }
+done:
+    return ret;
+}
+
+static int vi_feed_get_channels(struct audio_device *adev)
+{
+    struct mixer_ctl *ctl;
+    const char *mixer_ctl_name = VI_FEED_CHANNEL;
+    int value;
+
+    ALOGV("%s: entry", __func__);
+    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);
+        goto error;
+    }
+    value = mixer_ctl_get_value(ctl, 0);
+    if (value < 0)
+        goto error;
+    else
+        return value+1;
+error:
+     return -EINVAL;
+}
+
+// must be called with adev->lock acquired
+static int spkr_calibrate(int t0)
+{
+    struct audio_device *adev = handle.adev_handle;
+    struct audio_cal_info_spk_prot_cfg protCfg;
+    struct audio_cal_info_msm_spk_prot_status status;
+    bool cleanup = false, disable_rx = false, disable_tx = false;
+    int acdb_fd = -1;
+    struct audio_usecase *uc_info_rx = NULL, *uc_info_tx = NULL;
+    int32_t pcm_dev_rx_id = -1, pcm_dev_tx_id = -1;
+    struct timespec ts;
+    int retry_duration;
+
+    if (!adev) {
+        ALOGE("%s: Invalid params", __func__);
+        return -EINVAL;
+    }
+    if (!list_empty(&adev->usecase_list)) {
+        ALOGD("%s: Usecase present retry speaker protection", __func__);
+        return -EAGAIN;
+    }
+    acdb_fd = open("/dev/msm_audio_cal",O_RDWR | O_NONBLOCK);
+    if (acdb_fd < 0) {
+        ALOGE("%s: spkr_prot_thread open msm_acdb failed", __func__);
+        return -ENODEV;
+    } else {
+        protCfg.mode = MSM_SPKR_PROT_CALIBRATION_IN_PROGRESS;
+        /* HAL for speaker protection gets only one Temperature */
+        protCfg.t0[SP_V2_SPKR_1] = t0;
+        protCfg.t0[SP_V2_SPKR_2] = t0;
+        if (set_spkr_prot_cal(acdb_fd, &protCfg)) {
+            ALOGE("%s: spkr_prot_thread set failed AUDIO_SET_SPEAKER_PROT",
+            __func__);
+            status.status = -ENODEV;
+            goto exit;
+        }
+    }
+    uc_info_rx = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
+    if (!uc_info_rx) {
+        return -ENOMEM;
+    }
+    uc_info_rx->id = USECASE_AUDIO_SPKR_CALIB_RX;
+    uc_info_rx->type = PCM_PLAYBACK;
+    uc_info_rx->in_snd_device = SND_DEVICE_NONE;
+    uc_info_rx->stream.out = adev->primary_output;
+    uc_info_rx->out_snd_device = SND_DEVICE_OUT_SPEAKER_PROTECTED;
+    disable_rx = true;
+    list_add_tail(&adev->usecase_list, &uc_info_rx->list);
+    enable_snd_device(adev, SND_DEVICE_OUT_SPEAKER_PROTECTED);
+    enable_audio_route(adev, uc_info_rx);
+
+    pcm_dev_rx_id = platform_get_pcm_device_id(uc_info_rx->id, PCM_PLAYBACK);
+    ALOGV("%s: pcm device id %d", __func__, pcm_dev_rx_id);
+    if (pcm_dev_rx_id < 0) {
+        ALOGE("%s: Invalid pcm device for usecase (%d)",
+              __func__, uc_info_rx->id);
+        status.status = -ENODEV;
+        goto exit;
+    }
+    handle.pcm_rx = handle.pcm_tx = NULL;
+    handle.pcm_rx = pcm_open(adev->snd_card,
+                             pcm_dev_rx_id,
+                             PCM_OUT, &pcm_config_skr_prot);
+    if (handle.pcm_rx && !pcm_is_ready(handle.pcm_rx)) {
+        ALOGE("%s: %s", __func__, pcm_get_error(handle.pcm_rx));
+        status.status = -EIO;
+        goto exit;
+    }
+    uc_info_tx = (struct audio_usecase *)
+    calloc(1, sizeof(struct audio_usecase));
+    if (!uc_info_tx) {
+        status.status = -ENOMEM;
+        goto exit;
+    }
+    uc_info_tx->id = USECASE_AUDIO_SPKR_CALIB_TX;
+    uc_info_tx->type = PCM_CAPTURE;
+    uc_info_tx->in_snd_device = SND_DEVICE_IN_CAPTURE_VI_FEEDBACK;
+    uc_info_tx->out_snd_device = SND_DEVICE_NONE;
+
+    disable_tx = true;
+    list_add_tail(&adev->usecase_list, &uc_info_tx->list);
+    enable_snd_device(adev, SND_DEVICE_IN_CAPTURE_VI_FEEDBACK);
+    enable_audio_route(adev, uc_info_tx);
+
+    pcm_dev_tx_id = platform_get_pcm_device_id(uc_info_tx->id, PCM_CAPTURE);
+    if (pcm_dev_tx_id < 0) {
+        ALOGE("%s: Invalid pcm device for usecase (%d)",
+              __func__, uc_info_tx->id);
+        status.status = -ENODEV;
+        goto exit;
+    }
+    handle.pcm_tx = pcm_open(adev->snd_card,
+                             pcm_dev_tx_id,
+                             PCM_IN, &pcm_config_skr_prot);
+    if (handle.pcm_tx && !pcm_is_ready(handle.pcm_tx)) {
+        ALOGE("%s: %s", __func__, pcm_get_error(handle.pcm_tx));
+        status.status = -EIO;
+        goto exit;
+    }
+    if (pcm_start(handle.pcm_rx) < 0) {
+        ALOGE("%s: pcm start for RX failed", __func__);
+        status.status = -EINVAL;
+        goto exit;
+    }
+    if (pcm_start(handle.pcm_tx) < 0) {
+        ALOGE("%s: pcm start for TX failed", __func__);
+        status.status = -EINVAL;
+        goto exit;
+    }
+    cleanup = true;
+    clock_gettime(CLOCK_REALTIME, &ts);
+    ts.tv_sec += (SLEEP_AFTER_CALIB_START/1000);
+    ts.tv_nsec = 0;
+    pthread_mutex_lock(&handle.mutex_spkr_prot);
+    pthread_mutex_unlock(&adev->lock);
+
+    (void)pthread_cond_timedwait(&handle.spkr_calib_cancel,
+                                 &handle.mutex_spkr_prot, &ts);
+    ALOGD("%s: Speaker calibration done", __func__);
+    pthread_mutex_lock(&handle.spkr_calib_cancelack_mutex);
+    if (handle.cancel_spkr_calib) {
+        status.status = -EAGAIN;
+        goto exit;
+    }
+
+    if (acdb_fd >= 0) {
+        status.status = -EINVAL;
+        retry_duration = 0;
+        while (!get_spkr_prot_cal(acdb_fd, &status) &&
+               retry_duration < GET_SPKR_PROT_CAL_TIMEOUT_MSEC) {
+            if (!status.status) {
+                ALOGD("%s: spkr_prot_thread calib Success R0 %d %d",
+                 __func__, status.r0[SP_V2_SPKR_1], status.r0[SP_V2_SPKR_2]);
+                FILE *fp;
+
+                vi_feed_no_channels = vi_feed_get_channels(adev);
+                ALOGD("%s: vi_feed_no_channels %d", __func__, vi_feed_no_channels);
+                if (vi_feed_no_channels < 0) {
+                    ALOGE("%s: no of channels negative !!", __func__);
+                    /* limit the number of channels to 2*/
+                    vi_feed_no_channels = 2;
+                }
+
+                fp = fopen(CALIB_FILE,"wb");
+                if (!fp) {
+                    ALOGE("%s: spkr_prot_thread File open failed %s",
+                    __func__, strerror(errno));
+                    status.status = -ENODEV;
+                } else {
+                    int i;
+                    /* HAL for speaker protection is always calibrating for stereo usecase*/
+                    for (i = 0; i < vi_feed_no_channels; i++) {
+                        fwrite(&status.r0[i], sizeof(status.r0[i]), 1, fp);
+                        fwrite(&protCfg.t0[i], sizeof(protCfg.t0[i]), 1, fp);
+                    }
+                    fclose(fp);
+                }
+                break;
+            } else if (status.status == -EAGAIN) {
+                  ALOGD("%s: spkr_prot_thread try again", __func__);
+                  usleep(WAIT_FOR_GET_CALIB_STATUS * 1000);
+                  retry_duration += WAIT_FOR_GET_CALIB_STATUS;
+            } else {
+                ALOGE("%s: spkr_prot_thread get failed status %d",
+                __func__, status.status);
+                break;
+            }
+        }
+    }
+
+exit:
+    if (handle.pcm_rx)
+        pcm_close(handle.pcm_rx);
+    handle.pcm_rx = NULL;
+
+    if (handle.pcm_tx)
+        pcm_close(handle.pcm_tx);
+    handle.pcm_tx = NULL;
+
+    /* Clear TX calibration to handset mic */
+    platform_send_audio_calibration(adev->platform, SND_DEVICE_IN_HANDSET_MIC);
+    if (!status.status) {
+        protCfg.mode = MSM_SPKR_PROT_CALIBRATED;
+        protCfg.r0[SP_V2_SPKR_1] = status.r0[SP_V2_SPKR_1];
+        protCfg.r0[SP_V2_SPKR_2] = status.r0[SP_V2_SPKR_2];
+        if (set_spkr_prot_cal(acdb_fd, &protCfg))
+            ALOGE("%s: spkr_prot_thread disable calib mode", __func__);
+        else
+            handle.spkr_prot_mode = MSM_SPKR_PROT_CALIBRATED;
+    } else {
+        protCfg.mode = MSM_SPKR_PROT_NOT_CALIBRATED;
+        handle.spkr_prot_mode = MSM_SPKR_PROT_NOT_CALIBRATED;
+        if (set_spkr_prot_cal(acdb_fd, &protCfg))
+            ALOGE("%s: spkr_prot_thread disable calib mode failed", __func__);
+    }
+    if (acdb_fd >= 0)
+        close(acdb_fd);
+
+    if (!handle.cancel_spkr_calib && cleanup) {
+        pthread_mutex_unlock(&handle.spkr_calib_cancelack_mutex);
+        pthread_cond_wait(&handle.spkr_calib_cancel, &handle.mutex_spkr_prot);
+        pthread_mutex_lock(&handle.spkr_calib_cancelack_mutex);
+    }
+    if (disable_rx) {
+        list_remove(&uc_info_rx->list);
+        disable_snd_device(adev, SND_DEVICE_OUT_SPEAKER_PROTECTED);
+        disable_audio_route(adev, uc_info_rx);
+    }
+    if (disable_tx) {
+        list_remove(&uc_info_tx->list);
+        disable_snd_device(adev, SND_DEVICE_IN_CAPTURE_VI_FEEDBACK);
+        disable_audio_route(adev, uc_info_tx);
+    }
+    if (uc_info_rx) free(uc_info_rx);
+    if (uc_info_tx) free(uc_info_tx);
+    if (cleanup) {
+        if (handle.cancel_spkr_calib)
+            pthread_cond_signal(&handle.spkr_calibcancel_ack);
+        handle.cancel_spkr_calib = 0;
+        pthread_mutex_unlock(&handle.spkr_calib_cancelack_mutex);
+        pthread_mutex_unlock(&handle.mutex_spkr_prot);
+        pthread_mutex_lock(&adev->lock);
+    }
+
+    return status.status;
+}
+
+static void* spkr_calibration_thread()
+{
+    unsigned long sec = 0;
+    int t0;
+    bool goahead = false;
+    struct audio_cal_info_spk_prot_cfg protCfg;
+    FILE *fp;
+    int acdb_fd;
+    struct audio_device *adev = handle.adev_handle;
+    unsigned long min_idle_time = MIN_SPKR_IDLE_SEC;
+    char value[PROPERTY_VALUE_MAX];
+
+    /* If the value of this persist.spkr.cal.duration is 0
+     * then it means it will take 30min to calibrate
+     * and if the value is greater than zero then it would take
+     * that much amount of time to calibrate.
+     */
+    property_get("persist.spkr.cal.duration", value, "0");
+    if (atoi(value) > 0)
+        min_idle_time = atoi(value);
+    handle.speaker_prot_threadid = pthread_self();
+    ALOGD("spkr_prot_thread enable prot Entry");
+    acdb_fd = open("/dev/msm_audio_cal",O_RDWR | O_NONBLOCK);
+    if (acdb_fd >= 0) {
+        /*Set processing mode with t0/r0*/
+        protCfg.mode = MSM_SPKR_PROT_NOT_CALIBRATED;
+        if (set_spkr_prot_cal(acdb_fd, &protCfg)) {
+            ALOGE("%s: spkr_prot_thread enable prot failed", __func__);
+            handle.spkr_prot_mode = MSM_SPKR_PROT_DISABLED;
+            close(acdb_fd);
+        } else
+            handle.spkr_prot_mode = MSM_SPKR_PROT_NOT_CALIBRATED;
+    } else {
+        handle.spkr_prot_mode = MSM_SPKR_PROT_DISABLED;
+        ALOGE("%s: Failed to open acdb node", __func__);
+    }
+    if (handle.spkr_prot_mode == MSM_SPKR_PROT_DISABLED) {
+        ALOGD("%s: Speaker protection disabled", __func__);
+        pthread_exit(0);
+        return NULL;
+    }
+
+    fp = fopen(CALIB_FILE,"rb");
+    if (fp) {
+        int i;
+        bool spkr_calibrated = true;
+        /* HAL for speaker protection is always calibrating for stereo usecase*/
+        vi_feed_no_channels = vi_feed_get_channels(adev);
+        ALOGD("%s: vi_feed_no_channels %d", __func__, vi_feed_no_channels);
+        if (vi_feed_no_channels < 0) {
+            ALOGE("%s: no of channels negative !!", __func__);
+            /* limit the number of channels to 2*/
+            vi_feed_no_channels = 2;
+        }
+        for (i = 0; i < vi_feed_no_channels; i++) {
+            fread(&protCfg.r0[i], sizeof(protCfg.r0[i]), 1, fp);
+            fread(&protCfg.t0[i], sizeof(protCfg.t0[i]), 1, fp);
+        }
+        ALOGD("%s: spkr_prot_thread r0 value %d %d",
+               __func__, protCfg.r0[SP_V2_SPKR_1], protCfg.r0[SP_V2_SPKR_2]);
+        ALOGD("%s: spkr_prot_thread t0 value %d %d",
+               __func__, protCfg.t0[SP_V2_SPKR_1], protCfg.t0[SP_V2_SPKR_2]);
+        fclose(fp);
+        /*Valid tempature range: -30C to 80C(in q6 format)
+          Valid Resistance range: 2 ohms to 40 ohms(in q24 format)*/
+        for (i = 0; i < vi_feed_no_channels; i++) {
+            if (!((protCfg.t0[i] > MIN_SPKR_TEMP_Q6) && (protCfg.t0[i] < MAX_SPKR_TEMP_Q6)
+                && (protCfg.r0[i] >= MIN_RESISTANCE_SPKR_Q24)
+                && (protCfg.r0[i] < MAX_RESISTANCE_SPKR_Q24))) {
+                spkr_calibrated = false;
+                break;
+            }
+        }
+        if (spkr_calibrated) {
+            ALOGD("%s: Spkr calibrated", __func__);
+            protCfg.mode = MSM_SPKR_PROT_CALIBRATED;
+            if (set_spkr_prot_cal(acdb_fd, &protCfg)) {
+                ALOGE("%s: enable prot failed", __func__);
+                handle.spkr_prot_mode = MSM_SPKR_PROT_DISABLED;
+            } else
+                handle.spkr_prot_mode = MSM_SPKR_PROT_CALIBRATED;
+            close(acdb_fd);
+            pthread_exit(0);
+            return NULL;
+        }
+        close(acdb_fd);
+    }
+
+    while (1) {
+        ALOGV("%s: start calibration", __func__);
+        if (!handle.thermal_client_request("spkr",1)) {
+            ALOGD("%s: wait for callback from thermal daemon", __func__);
+            pthread_mutex_lock(&handle.spkr_prot_thermalsync_mutex);
+            pthread_cond_wait(&handle.spkr_prot_thermalsync,
+            &handle.spkr_prot_thermalsync_mutex);
+            /*Convert temp into q6 format*/
+            t0 = (handle.spkr_prot_t0 * (1 << 6));
+            pthread_mutex_unlock(&handle.spkr_prot_thermalsync_mutex);
+            if (t0 < MIN_SPKR_TEMP_Q6 || t0 > MAX_SPKR_TEMP_Q6) {
+                ALOGE("%s: Calibration temparature error %d", __func__,
+                      handle.spkr_prot_t0);
+                continue;
+            }
+            ALOGD("%s: Request t0 success value %d", __func__,
+            handle.spkr_prot_t0);
+        } else {
+            ALOGE("%s: Request t0 failed", __func__);
+            /*Assume safe value for temparature*/
+            t0 = SAFE_SPKR_TEMP_Q6;
+        }
+        goahead = false;
+        pthread_mutex_lock(&adev->lock);
+        if (is_speaker_in_use(&sec)) {
+            ALOGD("%s: Speaker in use retry calibration", __func__);
+            pthread_mutex_unlock(&adev->lock);
+            continue;
+        } else {
+            ALOGD("%s: speaker idle %ld min time %ld", __func__, sec, min_idle_time);
+            if (sec < min_idle_time) {
+                ALOGD("%s: speaker idle is less retry", __func__);
+                pthread_mutex_unlock(&adev->lock);
+                continue;
+            }
+            goahead = true;
+        }
+        if (!list_empty(&adev->usecase_list)) {
+            ALOGD("%s: Usecase active re-try calibration", __func__);
+            goahead = false;
+            pthread_mutex_unlock(&adev->lock);
+        }
+        if (goahead) {
+                int status;
+                status = spkr_calibrate(t0);
+                pthread_mutex_unlock(&adev->lock);
+                if (status == -EAGAIN) {
+                    ALOGE("%s: failed to calibrate try again %s",
+                    __func__, strerror(status));
+                    continue;
+                } else {
+                    ALOGE("%s: calibrate status %s", __func__, strerror(status));
+                }
+                ALOGD("%s: spkr_prot_thread end calibration", __func__);
+                break;
+        }
+    }
+    if (handle.thermal_client_handle)
+        handle.thermal_client_unregister_callback(handle.thermal_client_handle);
+    handle.thermal_client_handle = 0;
+    if (handle.thermal_handle)
+        dlclose(handle.thermal_handle);
+    handle.thermal_handle = NULL;
+    pthread_exit(0);
+    return NULL;
+}
+
+static int thermal_client_callback(int temp)
+{
+    pthread_mutex_lock(&handle.spkr_prot_thermalsync_mutex);
+    ALOGD("%s: spkr_prot set t0 %d and signal", __func__, temp);
+    if (handle.spkr_prot_mode == MSM_SPKR_PROT_NOT_CALIBRATED)
+        handle.spkr_prot_t0 = temp;
+    pthread_cond_signal(&handle.spkr_prot_thermalsync);
+    pthread_mutex_unlock(&handle.spkr_prot_thermalsync_mutex);
+    return 0;
+}
+
+void audio_extn_spkr_prot_init(void *adev)
+{
+    char value[PROPERTY_VALUE_MAX];
+    ALOGD("%s: Initialize speaker protection module", __func__);
+    memset(&handle, 0, sizeof(handle));
+    if (!adev) {
+        ALOGE("%s: Invalid params", __func__);
+        return;
+    }
+    property_get("persist.speaker.prot.enable", value, "");
+    handle.spkr_prot_enable = false;
+    if (!strncmp("true", value, 4))
+       handle.spkr_prot_enable = true;
+    if (!handle.spkr_prot_enable) {
+        ALOGD("%s: Speaker protection disabled", __func__);
+        return;
+    }
+    handle.adev_handle = adev;
+    handle.spkr_prot_mode = MSM_SPKR_PROT_DISABLED;
+    handle.spkr_processing_state = SPKR_PROCESSING_IN_IDLE;
+    handle.spkr_prot_t0 = -1;
+    pthread_cond_init(&handle.spkr_prot_thermalsync, NULL);
+    pthread_cond_init(&handle.spkr_calib_cancel, NULL);
+    pthread_cond_init(&handle.spkr_calibcancel_ack, NULL);
+    pthread_mutex_init(&handle.mutex_spkr_prot, NULL);
+    pthread_mutex_init(&handle.spkr_calib_cancelack_mutex, NULL);
+    pthread_mutex_init(&handle.spkr_prot_thermalsync_mutex, NULL);
+    handle.thermal_handle = dlopen(THERMAL_CLIENT_LIBRARY_PATH,
+            RTLD_NOW);
+    if (!handle.thermal_handle) {
+        ALOGE("%s: DLOPEN for thermal client failed", __func__);
+    } else {
+        /*Query callback function symbol*/
+        handle.thermal_client_register_callback =
+       (int (*)(char *, int (*)(int),void *))
+        dlsym(handle.thermal_handle, "thermal_client_register_callback");
+        handle.thermal_client_unregister_callback =
+        (void (*)(int) )
+        dlsym(handle.thermal_handle, "thermal_client_unregister_callback");
+        if (!handle.thermal_client_register_callback ||
+            !handle.thermal_client_unregister_callback) {
+            ALOGE("%s: DLSYM thermal_client_register_callback failed", __func__);
+        } else {
+            /*Register callback function*/
+            handle.thermal_client_handle =
+            handle.thermal_client_register_callback("spkr", thermal_client_callback, NULL);
+            if (!handle.thermal_client_handle) {
+                ALOGE("%s: thermal_client_register_callback failed", __func__);
+            } else {
+                ALOGD("%s: spkr_prot thermal_client_register_callback success", __func__);
+                handle.thermal_client_request = (int (*)(char *, int))
+                dlsym(handle.thermal_handle, "thermal_client_request");
+            }
+        }
+    }
+    if (handle.thermal_client_request) {
+        ALOGD("%s: Create calibration thread", __func__);
+        (void)pthread_create(&handle.spkr_calibration_thread,
+        (const pthread_attr_t *) NULL, spkr_calibration_thread, &handle);
+    } else {
+        ALOGE("%s: thermal_client_request failed", __func__);
+        if (handle.thermal_client_handle &&
+            handle.thermal_client_unregister_callback)
+            handle.thermal_client_unregister_callback(handle.thermal_client_handle);
+        if (handle.thermal_handle)
+            dlclose(handle.thermal_handle);
+        handle.thermal_handle = NULL;
+        handle.spkr_prot_enable = false;
+    }
+
+    if (handle.spkr_prot_enable) {
+        char platform[PROPERTY_VALUE_MAX];
+        property_get("ro.board.platform", platform, "");
+        if (!strncmp("apq8084", platform, sizeof("apq8084"))) {
+            platform_set_snd_device_backend(SND_DEVICE_OUT_VOICE_SPEAKER,
+                                            "speaker-protected",
+                                            "SLIMBUS_0_RX");
+        }
+    }
+}
+
+int audio_extn_spkr_prot_get_acdb_id(snd_device_t snd_device)
+{
+    int acdb_id;
+
+    switch(snd_device) {
+    case SND_DEVICE_OUT_SPEAKER:
+        acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_SPEAKER_PROTECTED);
+        break;
+    case SND_DEVICE_OUT_VOICE_SPEAKER:
+        acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED);
+        break;
+    default:
+        acdb_id = -EINVAL;
+        break;
+    }
+    return acdb_id;
+}
+
+int audio_extn_get_spkr_prot_snd_device(snd_device_t snd_device)
+{
+    if (!handle.spkr_prot_enable)
+        return snd_device;
+
+    switch(snd_device) {
+    case SND_DEVICE_OUT_SPEAKER:
+        return SND_DEVICE_OUT_SPEAKER_PROTECTED;
+    case SND_DEVICE_OUT_VOICE_SPEAKER:
+        return SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
+    default:
+        return snd_device;
+    }
+}
+
+int audio_extn_spkr_prot_start_processing(snd_device_t snd_device)
+{
+    struct audio_usecase *uc_info_tx;
+    struct audio_device *adev = handle.adev_handle;
+    int32_t pcm_dev_tx_id = -1, ret = 0;
+
+    ALOGV("%s: Entry", __func__);
+    if (!adev) {
+       ALOGE("%s: Invalid params", __func__);
+       return -EINVAL;
+    }
+    snd_device = audio_extn_get_spkr_prot_snd_device(snd_device);
+    spkr_prot_set_spkrstatus(true);
+    uc_info_tx = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
+    if (!uc_info_tx) {
+        return -ENOMEM;
+    }
+    ALOGV("%s: snd_device(%d: %s)", __func__, snd_device,
+           platform_get_snd_device_name(snd_device));
+    audio_route_apply_and_update_path(adev->audio_route,
+           platform_get_snd_device_name(snd_device));
+
+    pthread_mutex_lock(&handle.mutex_spkr_prot);
+    if (handle.spkr_processing_state == SPKR_PROCESSING_IN_IDLE) {
+        uc_info_tx->id = USECASE_AUDIO_SPKR_CALIB_TX;
+        uc_info_tx->type = PCM_CAPTURE;
+        uc_info_tx->in_snd_device = SND_DEVICE_IN_CAPTURE_VI_FEEDBACK;
+        uc_info_tx->out_snd_device = SND_DEVICE_NONE;
+        handle.pcm_tx = NULL;
+        list_add_tail(&adev->usecase_list, &uc_info_tx->list);
+        enable_snd_device(adev, SND_DEVICE_IN_CAPTURE_VI_FEEDBACK);
+        enable_audio_route(adev, uc_info_tx);
+
+        pcm_dev_tx_id = platform_get_pcm_device_id(uc_info_tx->id, PCM_CAPTURE);
+        if (pcm_dev_tx_id < 0) {
+            ALOGE("%s: Invalid pcm device for usecase (%d)",
+                  __func__, uc_info_tx->id);
+            ret = -ENODEV;
+            goto exit;
+        }
+        handle.pcm_tx = pcm_open(adev->snd_card,
+                                 pcm_dev_tx_id,
+                                 PCM_IN, &pcm_config_skr_prot);
+        if (handle.pcm_tx && !pcm_is_ready(handle.pcm_tx)) {
+            ALOGE("%s: %s", __func__, pcm_get_error(handle.pcm_tx));
+            ret = -EIO;
+            goto exit;
+        }
+        if (pcm_start(handle.pcm_tx) < 0) {
+            ALOGE("%s: pcm start for TX failed", __func__);
+            ret = -EINVAL;
+        }
+    }
+
+exit:
+    /* Clear VI feedback cal and replace with handset MIC  */
+    platform_send_audio_calibration(adev->platform, SND_DEVICE_IN_HANDSET_MIC);
+    if (ret) {
+        if (handle.pcm_tx)
+            pcm_close(handle.pcm_tx);
+        handle.pcm_tx = NULL;
+        list_remove(&uc_info_tx->list);
+        disable_snd_device(adev, SND_DEVICE_IN_CAPTURE_VI_FEEDBACK);
+        disable_audio_route(adev, uc_info_tx);
+        free(uc_info_tx);
+    } else
+        handle.spkr_processing_state = SPKR_PROCESSING_IN_PROGRESS;
+    pthread_mutex_unlock(&handle.mutex_spkr_prot);
+    ALOGV("%s: Exit", __func__);
+    return ret;
+}
+
+void audio_extn_spkr_prot_stop_processing(snd_device_t snd_device)
+{
+    struct audio_usecase *uc_info_tx;
+    struct audio_device *adev = handle.adev_handle;
+
+    ALOGV("%s: Entry", __func__);
+    snd_device = audio_extn_get_spkr_prot_snd_device(snd_device);
+    spkr_prot_set_spkrstatus(false);
+    pthread_mutex_lock(&handle.mutex_spkr_prot);
+    if (adev && handle.spkr_processing_state == SPKR_PROCESSING_IN_PROGRESS) {
+        uc_info_tx = get_usecase_from_list(adev, USECASE_AUDIO_SPKR_CALIB_TX);
+        if (handle.pcm_tx)
+            pcm_close(handle.pcm_tx);
+        handle.pcm_tx = NULL;
+        disable_snd_device(adev, SND_DEVICE_IN_CAPTURE_VI_FEEDBACK);
+        if (uc_info_tx) {
+            list_remove(&uc_info_tx->list);
+            disable_audio_route(adev, uc_info_tx);
+            free(uc_info_tx);
+        }
+    }
+    handle.spkr_processing_state = SPKR_PROCESSING_IN_IDLE;
+    pthread_mutex_unlock(&handle.mutex_spkr_prot);
+    if (adev)
+        audio_route_reset_and_update_path(adev->audio_route,
+                                      platform_get_snd_device_name(snd_device));
+    ALOGV("%s: Exit", __func__);
+}
+
+bool audio_extn_spkr_prot_is_enabled()
+{
+    return handle.spkr_prot_enable;
+}
+#endif /*SPKR_PROT_ENABLED*/

android/hardware/qcom/audio/hal/audio_extn/tfa_98xx.c

@@ -0,0 +1,545 @@
+/*
+ * Copyright (C) 2013-2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "tfa_98xx"
+/*#define LOG_NDEBUG 0*/
+#include <cutils/log.h>
+
+#include <stdlib.h>
+#include <audio_hw.h>
+#include <dlfcn.h>
+#include "audio_extn.h"
+#include <platform.h>
+#include <math.h>
+
+#define LIB_SPEAKER_BUNDLE "/system/lib/libexTfa98xx.so"
+
+
+enum exTfa98xx_Audio_Mode
+{
+    Audio_Mode_None = -1,
+    Audio_Mode_Music_Normal,
+    Audio_Mode_Hfp_Client,
+    Audio_Mode_Voice,
+    Audio_Mode_Hs_Hfp,
+    Audio_Mode_Max
+};
+typedef enum exTfa98xx_Audio_Mode exTfa98xx_audio_mode_t;
+
+enum exTfa98xx_Func_Mode
+{
+    Func_Mode_None = -1,
+    Func_Mode_Speaker,
+    Func_Mode_BT
+};
+typedef enum exTfa98xx_Func_Mode exTfa98xx_func_mode_t;
+
+#define I2S_CLOCK_ENABLE        1
+#define I2S_CLOCK_DISABLE       0
+#define HFP_MAX_VOLUME          (15.000000)
+#define TFA_98XX_HFP_VSETPS     (5.0)
+
+exTfa98xx_audio_mode_t current_audio_mode = Audio_Mode_None;
+
+typedef int (*set_speaker_on_t)(exTfa98xx_audio_mode_t);
+typedef int (*set_speaker_off_t)(void);
+typedef int (*set_speaker_calibration_t)(int);
+typedef void (*set_speaker_volume_step_t)(int, int);
+
+
+struct speaker_data {
+    struct audio_device *adev;
+    void *speaker_bundle;
+    set_speaker_on_t set_speaker_on;
+    set_speaker_off_t set_speaker_off;
+    set_speaker_calibration_t set_speaker_calibration;
+    set_speaker_volume_step_t set_speaker_volume_step;
+    int ref_cnt[Audio_Mode_Max];
+    int route_cnt[Audio_Mode_Max];
+    bool update_ref_cnt;
+};
+
+struct speaker_data *tfa98xx_speaker_data = NULL;
+
+static struct speaker_data* open_speaker_bundle()
+{
+    struct speaker_data *sd = calloc(1, sizeof(struct speaker_data));
+
+    sd->speaker_bundle = dlopen(LIB_SPEAKER_BUNDLE, RTLD_NOW);
+    if (sd->speaker_bundle == NULL) {
+        ALOGE("%s: DLOPEN failed for %s", __func__, LIB_SPEAKER_BUNDLE);
+        goto error;
+    } else {
+        ALOGV("%s: DLOPEN successful for %s", __func__, LIB_SPEAKER_BUNDLE);
+
+        sd->set_speaker_on = (set_speaker_on_t)dlsym(sd->speaker_bundle,
+                                             "exTfa98xx_speakeron");
+        if (sd->set_speaker_on == NULL) {
+            ALOGE("%s: dlsym error %s for exTfa98xx_speakeron", __func__,
+                  dlerror());
+            goto error;
+        }
+        sd->set_speaker_off = (set_speaker_off_t)dlsym(sd->speaker_bundle,
+                                             "exTfa98xx_speakeroff");
+        if (sd->set_speaker_off == NULL) {
+            ALOGE("%s: dlsym error %s for exTfa98xx_speakeroff", __func__,
+                  dlerror());
+            goto error;
+        }
+        sd->set_speaker_volume_step = (set_speaker_volume_step_t)dlsym(sd->speaker_bundle,
+                                             "exTfa98xx_setvolumestep");
+        if (sd->set_speaker_volume_step == NULL) {
+            ALOGE("%s: dlsym error %s for exTfa98xx_setvolumestep",
+                  __func__, dlerror());
+            goto error;
+        }
+        sd->set_speaker_calibration = (set_speaker_calibration_t)dlsym(sd->speaker_bundle,
+                                             "exTfa98xx_calibration");
+        if (sd->set_speaker_calibration == NULL) {
+            ALOGE("%s: dlsym error %s for exTfa98xx_calibration",
+                  __func__, dlerror());
+            goto error;
+        }
+    }
+    return sd;
+
+error:
+    free(sd);
+    return 0;
+}
+
+static void close_speaker_bundle(struct speaker_data *sd)
+{
+    if (sd != NULL) {
+        dlclose(sd->speaker_bundle);
+        free(sd);
+        sd = NULL;
+    }
+}
+
+static int adev_i2s_clock_operation(int enable, struct audio_device *adev, char *paths)
+{
+    int ret = -1;
+
+    ALOGD("%s: mixer paths is: %s, enable: %d\n", __func__, paths, enable);
+    if(I2S_CLOCK_ENABLE == enable) {
+        ret = audio_route_apply_and_update_path(adev->audio_route, paths);
+        if(ret) {
+            ALOGE("%s: audio_route_apply_and_update_path return %d\n", __func__, ret);
+            return ret;
+        }
+    } else {
+        ret = audio_route_reset_and_update_path(adev->audio_route, paths);
+        if(ret) {
+            ALOGE("%s: audio_route_reset_and_update_path return %d\n", __func__, ret);
+            return ret;
+        }
+    }
+    return 0;
+}
+
+static int tfa_98xx_set_audio_mode(int enable, struct audio_device *adev, exTfa98xx_audio_mode_t audio_mode)
+{
+    char paths[32] = "init_smart_pa";
+
+    switch(audio_mode) {
+        case Audio_Mode_Music_Normal:
+            strcat(paths, " music");
+            break;
+        case Audio_Mode_Voice:
+        case Audio_Mode_Hfp_Client:
+        case Audio_Mode_Hs_Hfp:
+            strcat(paths, " voice");
+            break;
+        default:
+            ALOGE("%s: function %d not support!\n",__func__, audio_mode);
+            return -EINVAL;
+    }
+
+    ALOGV("%s: mixer paths is: %s, enable: %d\n", __func__, paths, enable);
+    adev_i2s_clock_operation(enable, adev, paths);
+    return 0;
+
+}
+
+static exTfa98xx_audio_mode_t tfa_98xx_get_audio_mode(struct speaker_data *data)
+{
+    exTfa98xx_audio_mode_t tfa_98xx_audio_mode = Audio_Mode_None;
+    struct listnode *node;
+    struct audio_usecase *usecase;
+    audio_mode_t mode = data->adev->mode;
+    int i = 0;
+
+    ALOGV("%s: enter\n", __func__);
+
+    for (i = 0; i < Audio_Mode_Max; i++)
+        data->route_cnt[i] = 0;
+
+    list_for_each(node, &data->adev->usecase_list) {
+        usecase = node_to_item(node, struct audio_usecase, list);
+        if (usecase->devices & AUDIO_DEVICE_OUT_ALL_SCO) {
+            if(data->adev->snd_dev_ref_cnt[usecase->out_snd_device] != 0) {
+                tfa_98xx_audio_mode = Audio_Mode_Hs_Hfp;
+                data->route_cnt[tfa_98xx_audio_mode]++;
+                ALOGV("%s: audio_mode hs_hfp\n", __func__);
+            }
+        } else if (usecase->devices & AUDIO_DEVICE_OUT_SPEAKER) {
+            if ((mode == AUDIO_MODE_IN_CALL) || audio_extn_hfp_is_active(data->adev)) {
+                if (audio_extn_hfp_is_active(data->adev)) {
+                    if(data->adev->snd_dev_ref_cnt[usecase->out_snd_device] != 0) {
+                        tfa_98xx_audio_mode = Audio_Mode_Hfp_Client;
+                        data->route_cnt[tfa_98xx_audio_mode]++;
+                        ALOGV("%s: audio_mode hfp client\n", __func__);
+                    }
+                } else {
+                    if(data->adev->snd_dev_ref_cnt[usecase->out_snd_device] != 0) {
+                        tfa_98xx_audio_mode = Audio_Mode_Voice;
+                        data->route_cnt[tfa_98xx_audio_mode]++;
+                        ALOGV("%s: audio_mode voice\n", __func__);
+                    }
+                }
+            } else {
+                if (data->adev->snd_dev_ref_cnt[usecase->out_snd_device] != 0) {
+                    tfa_98xx_audio_mode = Audio_Mode_Music_Normal;
+                    data->route_cnt[tfa_98xx_audio_mode]++;
+                    ALOGV("%s: tfa_98xx_audio_mode music\n", __func__);
+                }
+            }
+        } else {
+            ALOGE("%s: no device match \n", __func__);
+        }
+    }
+    ALOGV("%s: tfa_98xx_audio_mode %d exit\n", __func__, tfa_98xx_audio_mode);
+
+    return tfa_98xx_audio_mode;
+}
+
+static int tfa_98xx_set_func_mode(int enable, struct audio_device *adev, exTfa98xx_func_mode_t func_mode)
+{
+    struct speaker_data *data = tfa98xx_speaker_data;
+    char paths[32] = "init_smart_pa";
+
+    if (data) {
+        switch(func_mode) {
+            case Func_Mode_Speaker:
+                strcat(paths, " func_speaker");
+                break;
+            case Func_Mode_BT:
+                strcat(paths, " func_bt");
+                break;
+            default:
+                ALOGE("%s: function %d not support!\n",__func__, func_mode);
+                return -EINVAL;
+        }
+
+        ALOGV("%s: mixer paths is: %s, enable: %d\n", __func__, paths, enable);
+        adev_i2s_clock_operation(enable, adev, paths);
+    }
+    return 0;
+}
+
+static exTfa98xx_func_mode_t tfa_98xx_get_func_mode(exTfa98xx_audio_mode_t audio_mode)
+{
+    exTfa98xx_func_mode_t func_mode = Func_Mode_None;
+
+    switch(audio_mode) {
+        case Audio_Mode_Music_Normal:
+        case Audio_Mode_Voice:
+            ALOGV("%s: tfa_98xx_func_mode speaker \n", __func__);
+            func_mode = Func_Mode_Speaker;
+            break;
+        case Audio_Mode_Hfp_Client:
+        case Audio_Mode_Hs_Hfp:
+            ALOGV("%s: tfa_98xx_func_mode bt \n", __func__);
+            func_mode = Func_Mode_BT;
+            break;
+        default:
+            break;
+    }
+    return func_mode;
+}
+
+static void tfa_98xx_disable_speaker(void)
+{
+    struct speaker_data *data = tfa98xx_speaker_data;
+    int ret = 0;
+
+    ret = data->set_speaker_off();
+    if (ret) {
+        ALOGE("%s: exTfa98xx_speakeroff failed result = %d\n", __func__, ret);
+        goto on_error;
+    }
+
+    ret = tfa_98xx_set_audio_mode(I2S_CLOCK_DISABLE, data->adev, current_audio_mode);
+    if (ret) {
+        ALOGE("%s: tfa_98xx_set_audio_mode disable failed return %d\n", __func__, ret);
+        goto on_error;
+    }
+    current_audio_mode = Audio_Mode_None;
+on_error:
+    return;
+
+}
+
+
+void audio_extn_tfa_98xx_disable_speaker(snd_device_t snd_device)
+{
+    struct speaker_data *data = tfa98xx_speaker_data;
+    int i = 0;
+    exTfa98xx_audio_mode_t new_audio_mode = Audio_Mode_None;
+
+    ALOGV("%s: enter\n", __func__);
+
+    if (data) {
+        if ((current_audio_mode == Audio_Mode_None) || (snd_device > SND_DEVICE_OUT_END))
+            goto on_exit;
+
+        switch(snd_device) {
+            case SND_DEVICE_OUT_SPEAKER:
+                new_audio_mode = Audio_Mode_Music_Normal;
+                break;
+            case SND_DEVICE_OUT_VOICE_SPEAKER:
+                new_audio_mode = Audio_Mode_Voice;
+                break;
+            case SND_DEVICE_OUT_VOICE_SPEAKER_HFP:
+                new_audio_mode = Audio_Mode_Hfp_Client;
+                break;
+            case SND_DEVICE_OUT_BT_SCO:
+                new_audio_mode = Audio_Mode_Hs_Hfp;
+                break;
+            default:
+                break;
+        }
+
+        if ((new_audio_mode == Audio_Mode_None) || (data->ref_cnt[new_audio_mode] <= 0)) {
+            ALOGE("%s: device ref cnt is already 0", __func__);
+            goto on_exit;
+        }
+
+        data->ref_cnt[new_audio_mode]--;
+
+        for (i = 0; i < Audio_Mode_Max; i++) {
+            if (data->ref_cnt[i] > 0) {
+                ALOGD("%s: exTfa98xx_speaker still in use\n", __func__);
+                goto on_exit;
+            }
+        }
+
+        if (data->adev->enable_hfp)
+            data->set_speaker_volume_step(0, 0);
+
+        tfa_98xx_disable_speaker();
+    }
+
+    ALOGV("%s: exit\n", __func__);
+on_exit:
+    return;
+}
+
+int audio_extn_tfa_98xx_enable_speaker(void)
+{
+    struct speaker_data *data = tfa98xx_speaker_data;
+    exTfa98xx_audio_mode_t new_audio_mode = Audio_Mode_Music_Normal;
+    int ret = 0;
+    int i = 0;
+
+    ALOGV("%s: enter\n", __func__);
+
+    if (data) {
+
+        new_audio_mode = tfa_98xx_get_audio_mode(data);
+        if ((new_audio_mode != Audio_Mode_None) && (data->ref_cnt[new_audio_mode] >= 1)) {
+            ALOGD("%s, mode %d already active!", __func__, new_audio_mode);
+            data->ref_cnt[new_audio_mode]++;
+            goto on_exit;
+        }
+
+        ret = tfa_98xx_set_audio_mode(I2S_CLOCK_ENABLE, data->adev, new_audio_mode);
+        if (ret) {
+            ALOGE("%s: tfa_98xx_set_audio_mode enable failed return %d\n", __func__, ret);
+            goto on_exit;
+        }
+
+        ret = data->set_speaker_on(new_audio_mode);
+        if (ret) {
+            ALOGE("%s: exTfa98xx_speakeron failed result = %d\n", __func__, ret);
+            goto on_exit;
+        }
+
+        current_audio_mode = new_audio_mode;
+        for (i = 0; i < Audio_Mode_Max; i++) {
+            data->ref_cnt[i] = data->route_cnt[i];
+        }
+        data->update_ref_cnt = false;
+    }
+
+    ALOGV("%s: exit\n", __func__);
+
+on_exit:
+    return ret;
+
+}
+
+void audio_extn_tfa_98xx_set_mode(void)
+{
+    int ret = 0;
+    struct speaker_data *data = tfa98xx_speaker_data;
+    exTfa98xx_audio_mode_t new_audio_mode = Audio_Mode_None;
+    exTfa98xx_func_mode_t new_func_mode = Func_Mode_None;
+
+    ALOGV("%s: enter\n", __func__);
+
+    if (data) {
+        new_audio_mode = tfa_98xx_get_audio_mode(data);
+
+        new_func_mode = tfa_98xx_get_func_mode(new_audio_mode);
+        if (new_func_mode == Func_Mode_None)
+            return;
+
+        ret = tfa_98xx_set_func_mode(I2S_CLOCK_ENABLE, data->adev, new_func_mode);
+        if (ret) {
+            ALOGE("%s: tfa_98xx_set_func_mode enable return %d\n", __func__, ret);
+        }
+        data->update_ref_cnt = true;
+    }
+
+    ALOGV("%s: exit\n", __func__);
+}
+
+void audio_extn_tfa_98xx_set_mode_bt(void)
+{
+    struct speaker_data *data = tfa98xx_speaker_data;
+    int ret = 0;
+
+    if (data) {
+        ret = tfa_98xx_set_func_mode(I2S_CLOCK_ENABLE, data->adev, Func_Mode_BT);
+        if (ret) {
+            ALOGE("%s: tfa_98xx_set_func_mode enable return %d\n", __func__, ret);
+        }
+    }
+}
+
+void audio_extn_tfa_98xx_update(void)
+{
+    struct speaker_data *data = tfa98xx_speaker_data;
+    exTfa98xx_audio_mode_t new_audio_mode = Audio_Mode_Music_Normal;
+
+    ALOGD("%s: enter\n", __func__);
+
+    if (data) {
+
+        new_audio_mode = tfa_98xx_get_audio_mode(data);
+        if (new_audio_mode <= current_audio_mode) {
+            ALOGE("%s: audio_extn_tfa_98xx_update same mode\n", __func__);
+            if (data->update_ref_cnt == true) {
+                data->ref_cnt[new_audio_mode]++;
+                data->update_ref_cnt = false;
+            }
+            goto on_error;
+        }
+
+        if (current_audio_mode != Audio_Mode_None) {
+            tfa_98xx_disable_speaker();
+        }
+
+        audio_extn_tfa_98xx_enable_speaker();
+
+    }
+
+    ALOGV("%s: exit\n", __func__);
+on_error:
+    return;
+
+}
+
+void audio_extn_tfa_98xx_set_voice_vol(float vol)
+{
+    struct speaker_data *data = tfa98xx_speaker_data;
+    int vsteps = 0;
+
+    if (data) {
+        if (data->adev->enable_hfp) {
+            if (vol < 0.0) {
+                vol = 0.0;
+            } else {
+                vol = ((vol > HFP_MAX_VOLUME) ? 1.0 : (vol / HFP_MAX_VOLUME));
+            }
+            vsteps = (int)floorf((1.0 - vol) * TFA_98XX_HFP_VSETPS);
+        } else {
+            return;
+        }
+        ALOGD("%s: vsteps %d\n", __func__, vsteps);
+        data->set_speaker_volume_step(vsteps, vsteps);
+    }
+}
+
+bool audio_extn_tfa_98xx_is_supported(void)
+{
+    struct speaker_data *data = tfa98xx_speaker_data;
+    if (data)
+        return true;
+    else
+        return false;
+}
+
+int audio_extn_tfa_98xx_init(struct audio_device *adev)
+{
+    int ret = 0;
+    struct speaker_data *data = open_speaker_bundle();
+
+    ALOGV("%s: enter\n", __func__);
+
+    if (data) {
+        ret = tfa_98xx_set_audio_mode(I2S_CLOCK_ENABLE, adev, Audio_Mode_Music_Normal);
+        if (ret) {
+            ALOGE("%s: tfa_98xx_set_audio_mode enable return %d\n", __func__, ret);
+            goto err_init;
+        }
+
+        ret = data->set_speaker_calibration(0);
+        if (ret) {
+            ALOGE("%s: exTfa98xx_calibration return %d\n", __func__, ret);
+        }
+
+        ret = tfa_98xx_set_audio_mode(I2S_CLOCK_DISABLE, adev, Audio_Mode_Music_Normal);
+        if (ret) {
+            ALOGE("%s: tfa_98xx_set_audio_mode disable return %d\n", __func__, ret);
+            goto err_init;
+        }
+
+        data->adev = adev;
+        tfa98xx_speaker_data = data;
+        ALOGV("%s: exit\n", __func__);
+        return 0;
+
+    }
+
+err_init:
+    close_speaker_bundle(data);
+    return -EINVAL;
+}
+
+void audio_extn_tfa_98xx_deinit(void)
+{
+    struct speaker_data *data = tfa98xx_speaker_data;
+
+    if (data) {
+        data->set_speaker_off();
+        close_speaker_bundle(data);
+        tfa98xx_speaker_data = NULL;
+    }
+}
+

android/hardware/qcom/audio/hal/audio_extn/tfa_98xx.h

@@ -0,0 +1,42 @@
+/*
+ * Copyright (C) 2013-2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef TFA_98XX_H
+#define TFA_98XX_H
+
+#ifdef SMART_PA_TFA_98XX_SUPPORTED
+int audio_extn_tfa_98xx_enable_speaker(void);
+void audio_extn_tfa_98xx_disable_speaker(snd_device_t snd_device);
+void audio_extn_tfa_98xx_set_mode();
+void audio_extn_tfa_98xx_set_mode_bt(void);
+void audio_extn_tfa_98xx_update(void);
+void audio_extn_tfa_98xx_set_voice_vol(float vol);
+int audio_extn_tfa_98xx_init(struct audio_device *adev);
+void audio_extn_tfa_98xx_deinit(void);
+bool audio_extn_tfa_98xx_is_supported(void);
+#else
+#define audio_extn_tfa_98xx_enable_speaker(void)                (0)
+#define audio_extn_tfa_98xx_disable_speaker(snd_device)         (0)
+#define audio_extn_tfa_98xx_set_mode()                          (0)
+#define audio_extn_tfa_98xx_set_mode_bt()                       (0)
+#define audio_extn_tfa_98xx_update(void)                        (0)
+#define audio_extn_tfa_98xx_set_voice_vol(vol)                  (0)
+#define audio_extn_tfa_98xx_init(adev)                          (0)
+#define audio_extn_tfa_98xx_deinit(void)                        (0)
+#define audio_extn_tfa_98xx_is_supported(void)                  (false)
+#endif
+
+#endif /* TFA_98XX_H */

android/hardware/qcom/audio/hal/audio_extn/utils.c

@@ -0,0 +1,608 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "audio_hw_utils"
+//#define LOG_NDEBUG 0
+
+#include <errno.h>
+#include <cutils/properties.h>
+#include <cutils/config_utils.h>
+#include <stdlib.h>
+#include <dlfcn.h>
+#include <unistd.h>
+#include <cutils/str_parms.h>
+#include <cutils/log.h>
+#include <cutils/misc.h>
+
+#include "acdb.h"
+#include "audio_hw.h"
+#include "platform.h"
+#include "platform_api.h"
+#include "audio_extn.h"
+
+#define MAX_LENGTH_MIXER_CONTROL_IN_INT 128
+
+static int set_stream_app_type_mixer_ctrl(struct audio_device *adev,
+                                          int pcm_device_id, int app_type,
+                                          int acdb_dev_id, int sample_rate,
+                                          int stream_type,
+                                          snd_device_t snd_device)
+{
+
+    char mixer_ctl_name[MAX_LENGTH_MIXER_CONTROL_IN_INT];
+    struct mixer_ctl *ctl;
+    int app_type_cfg[MAX_LENGTH_MIXER_CONTROL_IN_INT], len = 0, rc = 0;
+    int snd_device_be_idx = -1;
+
+    if (stream_type == PCM_PLAYBACK) {
+        snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
+             "Audio Stream %d App Type Cfg", pcm_device_id);
+    } else if (stream_type == PCM_CAPTURE) {
+        snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
+             "Audio Stream Capture %d App Type Cfg", 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);
+        rc = -EINVAL;
+        goto exit;
+    }
+    app_type_cfg[len++] = app_type;
+    app_type_cfg[len++] = acdb_dev_id;
+    app_type_cfg[len++] = sample_rate;
+
+    snd_device_be_idx = platform_get_snd_device_backend_index(snd_device);
+    if (snd_device_be_idx > 0)
+        app_type_cfg[len++] = snd_device_be_idx;
+    ALOGV("%s: stream type %d app_type %d, acdb_dev_id %d "
+          "sample rate %d, snd_device_be_idx %d",
+          __func__, stream_type, app_type, acdb_dev_id, sample_rate,
+          snd_device_be_idx);
+    mixer_ctl_set_array(ctl, app_type_cfg, len);
+
+exit:
+    return rc;
+}
+
+void audio_extn_utils_send_default_app_type_cfg(void *platform, struct mixer *mixer)
+{
+    int app_type_cfg[MAX_LENGTH_MIXER_CONTROL_IN_INT] = {-1};
+    int length = 0, app_type = 0,rc = 0;
+    struct mixer_ctl *ctl = NULL;
+    const char *mixer_ctl_name = "App Type Config";
+
+    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;
+    }
+    rc = platform_get_default_app_type_v2(platform, PCM_PLAYBACK, &app_type);
+    if (rc == 0) {
+        app_type_cfg[length++] = 1;
+        app_type_cfg[length++] = app_type;
+        app_type_cfg[length++] = 48000;
+        app_type_cfg[length++] = 16;
+        mixer_ctl_set_array(ctl, app_type_cfg, length);
+    }
+    return;
+}
+
+static const char *flags_to_mode(int dir, uint32_t flags)
+{
+    if (dir == 0) {
+        if (flags & AUDIO_OUTPUT_FLAG_VOIP_RX) {
+            return "voip";
+        }
+    } else if (dir == 1) {
+        if (flags & AUDIO_INPUT_FLAG_VOIP_TX) {
+            return "voip";
+        }
+    }
+    return "default";
+}
+
+static int audio_extn_utils_send_app_type_cfg_hfp(struct audio_device *adev,
+                                       struct audio_usecase *usecase)
+{
+    struct mixer_ctl *ctl;
+    int pcm_device_id, acdb_dev_id = 0, snd_device = usecase->out_snd_device;
+    int32_t sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+    int app_type = 0, rc = 0;
+
+    ALOGV("%s", __func__);
+
+    if (usecase->type != PCM_HFP_CALL) {
+        ALOGV("%s: not a playback or HFP path, no need to cfg app type", __func__);
+        rc = 0;
+        goto exit_send_app_type_cfg;
+    }
+    if ((usecase->id != USECASE_AUDIO_HFP_SCO) &&
+        (usecase->id != USECASE_AUDIO_HFP_SCO_WB)) {
+        ALOGV("%s: a playback path where app type cfg is not required", __func__);
+        rc = 0;
+        goto exit_send_app_type_cfg;
+    }
+
+    snd_device = usecase->out_snd_device;
+    pcm_device_id = platform_get_pcm_device_id(usecase->id, PCM_PLAYBACK);
+
+    snd_device = (snd_device == SND_DEVICE_OUT_SPEAKER) ?
+                 audio_extn_get_spkr_prot_snd_device(snd_device) : snd_device;
+    acdb_dev_id = platform_get_snd_device_acdb_id(snd_device);
+    if (acdb_dev_id < 0) {
+        ALOGE("%s: Couldn't get the acdb dev id", __func__);
+        rc = -EINVAL;
+        goto exit_send_app_type_cfg;
+    }
+
+    if (usecase->type == PCM_HFP_CALL) {
+
+        /* config HFP session:1 playback path */
+        rc = platform_get_default_app_type_v2(adev->platform, PCM_PLAYBACK, &app_type);
+        if (rc < 0)
+            goto exit_send_app_type_cfg;
+
+        sample_rate= CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+        rc = set_stream_app_type_mixer_ctrl(adev, pcm_device_id, app_type,
+                                            acdb_dev_id, sample_rate,
+                                            PCM_PLAYBACK,
+                                            SND_DEVICE_NONE); // use legacy behavior
+        if (rc < 0)
+            goto exit_send_app_type_cfg;
+        /* config HFP session:1 capture path */
+        rc = platform_get_default_app_type_v2(adev->platform, PCM_CAPTURE, &app_type);
+
+        if (rc == 0) {
+            rc = set_stream_app_type_mixer_ctrl(adev, pcm_device_id, app_type,
+                                                acdb_dev_id, sample_rate,
+                                                PCM_CAPTURE,
+                                                SND_DEVICE_NONE);
+            if (rc < 0)
+                goto exit_send_app_type_cfg;
+        }
+        /* config HFP session:2 capture path */
+        pcm_device_id = HFP_ASM_RX_TX;
+        snd_device = usecase->in_snd_device;
+        acdb_dev_id = platform_get_snd_device_acdb_id(snd_device);
+        if (acdb_dev_id <= 0) {
+            ALOGE("%s: Couldn't get the acdb dev id", __func__);
+            rc = -EINVAL;
+            goto exit_send_app_type_cfg;
+        }
+        rc = platform_get_default_app_type_v2(adev->platform, PCM_CAPTURE, &app_type);
+        if (rc == 0) {
+            rc = set_stream_app_type_mixer_ctrl(adev, pcm_device_id, app_type,
+                                                acdb_dev_id, sample_rate, PCM_CAPTURE,
+                                                SND_DEVICE_NONE);
+            if (rc < 0)
+                goto exit_send_app_type_cfg;
+        }
+
+        /* config HFP session:2 playback path */
+        rc = platform_get_default_app_type_v2(adev->platform, PCM_PLAYBACK, &app_type);
+        if (rc == 0) {
+            rc = set_stream_app_type_mixer_ctrl(adev, pcm_device_id, app_type,
+                                acdb_dev_id, sample_rate,
+                                PCM_PLAYBACK, SND_DEVICE_NONE);
+            if (rc < 0)
+                goto exit_send_app_type_cfg;
+        }
+    }
+
+    rc = 0;
+exit_send_app_type_cfg:
+    return rc;
+}
+
+
+static int derive_capture_app_type_cfg(struct audio_device *adev,
+                                       struct audio_usecase *usecase,
+                                       int *app_type,
+                                       int *sample_rate)
+{
+    if (usecase->stream.in == NULL) {
+        return -1;
+    }
+    struct stream_in *in = usecase->stream.in;
+    struct stream_app_type_cfg *app_type_cfg = &in->app_type_cfg;
+
+    *sample_rate = DEFAULT_INPUT_SAMPLING_RATE;
+    if (audio_is_usb_in_device(in->device)) {
+        platform_check_and_update_copp_sample_rate(adev->platform,
+                                                   usecase->in_snd_device,
+                                                   in->sample_rate,
+                                                   sample_rate);
+    }
+
+    app_type_cfg->mode = flags_to_mode(1 /*capture*/, in->flags);
+    ALOGV("%s mode %s", __func__, app_type_cfg->mode);
+    if (in->format == AUDIO_FORMAT_PCM_16_BIT) {
+        platform_get_app_type_v2(adev->platform,
+                                 PCM_CAPTURE,
+                                 app_type_cfg->mode,
+                                 16,
+                                 app_type_cfg->sample_rate,
+                                 app_type);
+    } else if (in->format == AUDIO_FORMAT_PCM_24_BIT_PACKED ||
+               in->format == AUDIO_FORMAT_PCM_8_24_BIT) {
+        platform_get_app_type_v2(adev->platform,
+                                 PCM_CAPTURE,
+                                 app_type_cfg->mode,
+                                 24,
+                                 app_type_cfg->sample_rate,
+                                 app_type);
+    } else if (in->format == AUDIO_FORMAT_PCM_32_BIT) {
+        platform_get_app_type_v2(adev->platform,
+                                 PCM_CAPTURE,
+                                 app_type_cfg->mode,
+                                 32,
+                                 app_type_cfg->sample_rate,
+                                 app_type);
+    } else {
+        ALOGE("%s bad format\n", __func__);
+        return -1;
+    }
+
+    app_type_cfg->app_type = *app_type;
+    app_type_cfg->sample_rate = *sample_rate;
+    return 0;
+}
+
+static int derive_playback_app_type_cfg(struct audio_device *adev,
+                                        struct audio_usecase *usecase,
+                                        int *app_type,
+                                        int *sample_rate)
+{
+    if (usecase->stream.out == NULL) {
+        return -1;
+    }
+    struct stream_out *out = usecase->stream.out;
+    struct stream_app_type_cfg *app_type_cfg = &out->app_type_cfg;
+
+    *sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+
+    // add speaker prot changes if needed
+    // and use that to check for device
+    if (audio_is_usb_out_device(out->devices)) {
+        platform_check_and_update_copp_sample_rate(adev->platform,
+                                                   usecase->out_snd_device,
+                                                   out->sample_rate,
+                                                   sample_rate);
+    }
+
+    app_type_cfg->mode = flags_to_mode(0 /*playback*/, out->flags);
+    if (!audio_is_linear_pcm(out->format)) {
+        platform_get_app_type_v2(adev->platform,
+                                 PCM_PLAYBACK,
+                                 app_type_cfg->mode,
+                                 24,
+                                 *sample_rate,
+                                 app_type);
+        ALOGV("Non pcm got app type %d", *app_type);
+    } else if (out->format == AUDIO_FORMAT_PCM_16_BIT) {
+        platform_get_app_type_v2(adev->platform,
+                                 PCM_PLAYBACK,
+                                 app_type_cfg->mode,
+                                 16,
+                                 *sample_rate,
+                                 app_type);
+    } else if (out->format == AUDIO_FORMAT_PCM_24_BIT_PACKED ||
+               out->format == AUDIO_FORMAT_PCM_8_24_BIT) {
+        platform_get_app_type_v2(adev->platform,
+                                 PCM_PLAYBACK,
+                                 app_type_cfg->mode,
+                                 24,
+                                 *sample_rate,
+                                 app_type);
+    } else if (out->format == AUDIO_FORMAT_PCM_32_BIT) {
+        platform_get_app_type_v2(adev->platform,
+                                 PCM_PLAYBACK,
+                                 app_type_cfg->mode,
+                                 32,
+                                 *sample_rate,
+                                 app_type);
+    } else {
+        ALOGE("%s bad format\n", __func__);
+        return -1;
+    }
+
+    app_type_cfg->app_type = *app_type;
+    app_type_cfg->sample_rate = *sample_rate;
+    return 0;
+}
+
+static int derive_acdb_dev_id(struct audio_device *adev __unused,
+                              struct audio_usecase *usecase)
+{
+    struct stream_out *out;
+    struct stream_in *in;
+
+    if (usecase->type == PCM_PLAYBACK) {
+        return platform_get_snd_device_acdb_id(usecase->out_snd_device);
+    } else if(usecase->type == PCM_CAPTURE) {
+        return platform_get_snd_device_acdb_id(usecase->in_snd_device);
+    }
+    return -1;
+}
+
+int audio_extn_utils_send_app_type_cfg(struct audio_device *adev,
+                                       struct audio_usecase *usecase)
+{
+    int len = 0;
+    int sample_rate;
+    int app_type;
+    int acdb_dev_id;
+    size_t app_type_cfg[MAX_LENGTH_MIXER_CONTROL_IN_INT] = {0};
+    char mixer_ctl_name[MAX_LENGTH_MIXER_CONTROL_IN_INT] = {0};
+    int pcm_device_id;
+    struct mixer_ctl *ctl;
+    int ret;
+
+    if (usecase->type == PCM_HFP_CALL) {
+        return audio_extn_utils_send_app_type_cfg_hfp(adev, usecase);
+    }
+
+    if (!platform_supports_app_type_cfg())
+        return -1;
+
+    if (usecase->type == PCM_PLAYBACK) {
+        ret = derive_playback_app_type_cfg(adev,
+                                           usecase,
+                                           &app_type,
+                                           &sample_rate);
+    } else if (usecase->type == PCM_CAPTURE) {
+        ret = derive_capture_app_type_cfg(adev,
+                                          usecase,
+                                          &app_type,
+                                          &sample_rate);
+    } else {
+        ALOGE("%s: Invalid uc type : 0x%x", __func__, usecase->type);
+        return -1;
+    }
+
+    if (ret < 0) {
+        ALOGE("%s: Failed to derive app_type for uc type : 0x%x", __func__,
+              usecase->type);
+        return -1;
+    }
+
+    acdb_dev_id = derive_acdb_dev_id(adev, usecase);
+    if (acdb_dev_id <= 0) {
+        ALOGE("%s: Couldn't get the acdb dev id", __func__);
+        return -1;
+    }
+
+    pcm_device_id = platform_get_pcm_device_id(usecase->id, usecase->type);
+    set_stream_app_type_mixer_ctrl(adev, pcm_device_id, app_type, acdb_dev_id,
+                                   sample_rate,
+                                   usecase->type,
+                                   usecase->type == PCM_PLAYBACK ? usecase->out_snd_device :
+                                                                   usecase->in_snd_device);
+    return 0;
+}
+
+int audio_extn_utils_send_app_type_gain(struct audio_device *adev,
+                                        int app_type,
+                                        int *gain)
+{
+    int gain_cfg[4];
+    const char *mixer_ctl_name = "App Type Gain";
+    struct mixer_ctl *ctl;
+    ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+    if (!ctl) {
+        ALOGE("%s: Could not get volume ctl mixer %s", __func__,
+              mixer_ctl_name);
+        return -EINVAL;
+    }
+    gain_cfg[0] = 0;
+    gain_cfg[1] = app_type;
+    gain_cfg[2] = gain[0];
+    gain_cfg[3] = gain[1];
+    ALOGV("%s app_type %d l(%d) r(%d)", __func__,  app_type, gain[0], gain[1]);
+    return mixer_ctl_set_array(ctl, gain_cfg,
+                               sizeof(gain_cfg)/sizeof(gain_cfg[0]));
+}
+
+// this assumes correct app_type and sample_rate fields
+// have been set for the stream using audio_extn_utils_send_app_type_cfg
+void audio_extn_utils_send_audio_calibration(struct audio_device *adev,
+                                             struct audio_usecase *usecase)
+{
+    int type = usecase->type;
+    int app_type = 0;
+
+    if (type == PCM_PLAYBACK && usecase->stream.out != NULL) {
+        struct stream_out *out = usecase->stream.out;
+        ALOGV("%s send cal for app_type %d, rate %d", __func__,
+              out->app_type_cfg.app_type,
+              out->app_type_cfg.sample_rate);
+        platform_send_audio_calibration_v2(adev->platform, usecase,
+                                           out->app_type_cfg.app_type,
+                                           out->app_type_cfg.sample_rate);
+    } else if (type == PCM_CAPTURE && usecase->stream.in != NULL) {
+        struct stream_in *in = usecase->stream.in;
+        ALOGV("%s send cal for capture app_type %d, rate %d", __func__,
+              in->app_type_cfg.app_type,
+              in->app_type_cfg.sample_rate);
+        platform_send_audio_calibration_v2(adev->platform, usecase,
+                                           in->app_type_cfg.app_type,
+                                           in->app_type_cfg.sample_rate);
+    } else {
+        /* when app type is default. the sample rate is not used to send cal */
+        platform_get_default_app_type_v2(adev->platform, type, &app_type);
+        platform_send_audio_calibration_v2(adev->platform, usecase, app_type,
+                                           48000);
+    }
+}
+
+#define MAX_SND_CARD 8
+#define RETRY_US 500000
+#define RETRY_NUMBER 10
+
+#define min(a, b) ((a) < (b) ? (a) : (b))
+
+static const char *kConfigLocationList[] =
+        {"/odm/etc", "/vendor/etc", "/system/etc"};
+static const int kConfigLocationListSize =
+        (sizeof(kConfigLocationList) / sizeof(kConfigLocationList[0]));
+
+bool audio_extn_utils_resolve_config_file(char file_name[MIXER_PATH_MAX_LENGTH])
+{
+    char full_config_path[MIXER_PATH_MAX_LENGTH];
+    for (int i = 0; i < kConfigLocationListSize; i++) {
+        snprintf(full_config_path,
+                 MIXER_PATH_MAX_LENGTH,
+                 "%s/%s",
+                 kConfigLocationList[i],
+                 file_name);
+        if (F_OK == access(full_config_path, 0)) {
+            strcpy(file_name, full_config_path);
+            return true;
+        }
+    }
+    return false;
+}
+
+/* platform_info_file should be size 'MIXER_PATH_MAX_LENGTH' */
+int audio_extn_utils_get_platform_info(const char* snd_card_name, char* platform_info_file)
+{
+    if (NULL == snd_card_name) {
+        return -1;
+    }
+
+    struct snd_card_split *snd_split_handle = NULL;
+    int ret = 0;
+    audio_extn_set_snd_card_split(snd_card_name);
+    snd_split_handle = audio_extn_get_snd_card_split();
+
+    snprintf(platform_info_file, MIXER_PATH_MAX_LENGTH, "%s_%s_%s.xml",
+                     PLATFORM_INFO_XML_BASE_STRING, snd_split_handle->snd_card,
+                     snd_split_handle->form_factor);
+
+    if (!audio_extn_utils_resolve_config_file(platform_info_file)) {
+        memset(platform_info_file, 0, MIXER_PATH_MAX_LENGTH);
+        snprintf(platform_info_file, MIXER_PATH_MAX_LENGTH, "%s_%s.xml",
+                     PLATFORM_INFO_XML_BASE_STRING, snd_split_handle->snd_card);
+
+        if (!audio_extn_utils_resolve_config_file(platform_info_file)) {
+            memset(platform_info_file, 0, MIXER_PATH_MAX_LENGTH);
+            strlcpy(platform_info_file, PLATFORM_INFO_XML_PATH, MIXER_PATH_MAX_LENGTH);
+            ret = audio_extn_utils_resolve_config_file(platform_info_file) ? 0 : -1;
+        }
+    }
+
+    return ret;
+}
+
+int audio_extn_utils_get_snd_card_num()
+{
+
+    void *hw_info = NULL;
+    struct mixer *mixer = NULL;
+    int retry_num = 0;
+    int snd_card_num = 0;
+    const char* snd_card_name = NULL;
+    char platform_info_file[MIXER_PATH_MAX_LENGTH]= {0};
+
+    struct acdb_platform_data *my_data = calloc(1, sizeof(struct acdb_platform_data));
+
+    bool card_verifed[MAX_SND_CARD] = {0};
+    const int retry_limit = property_get_int32("audio.snd_card.open.retries", RETRY_NUMBER);
+
+    for (;;) {
+        if (snd_card_num >= MAX_SND_CARD) {
+            if (retry_num++ >= retry_limit) {
+                ALOGE("%s: Unable to find correct sound card, aborting.", __func__);
+                snd_card_num = -1;
+                goto done;
+            }
+
+            snd_card_num = 0;
+            usleep(RETRY_US);
+            continue;
+        }
+
+        if (card_verifed[snd_card_num]) {
+            ++snd_card_num;
+            continue;
+        }
+
+        mixer = mixer_open(snd_card_num);
+
+        if (!mixer) {
+            ALOGE("%s: Unable to open the mixer card: %d", __func__,
+               snd_card_num);
+            ++snd_card_num;
+            continue;
+        }
+
+        card_verifed[snd_card_num] = true;
+
+        snd_card_name = mixer_get_name(mixer);
+        hw_info = hw_info_init(snd_card_name);
+
+        if (audio_extn_utils_get_platform_info(snd_card_name, platform_info_file) < 0) {
+            ALOGE("Failed to find platform_info_file");
+            goto cleanup;
+        }
+
+        /* Initialize snd card name specific ids and/or backends*/
+        if (snd_card_info_init(platform_info_file, my_data,
+                               &acdb_set_parameters) < 0) {
+            ALOGE("Failed to find platform_info_file");
+            goto cleanup;
+        }
+
+        /* validate the sound card name
+         * my_data->snd_card_name can contain
+         *     <a> complete sound card name, i.e. <device>-<codec>-<form_factor>-snd-card
+         *         example: msm8994-tomtom-mtp-snd-card
+         *     <b> or sub string of the card name, i.e. <device>-<codec>
+         *         example: msm8994-tomtom
+         * snd_card_name is truncated to 32 charaters as per mixer_get_name() implementation
+         * so use min of my_data->snd_card_name and snd_card_name length for comparison
+         */
+
+        if (my_data->snd_card_name != NULL &&
+                strncmp(snd_card_name, my_data->snd_card_name,
+                        min(strlen(snd_card_name), strlen(my_data->snd_card_name))) != 0) {
+            ALOGI("%s: found valid sound card %s, but not primary sound card %s",
+                   __func__, snd_card_name, my_data->snd_card_name);
+            goto cleanup;
+        }
+
+        ALOGI("%s: found sound card %s, primary sound card expected is %s",
+              __func__, snd_card_name, my_data->snd_card_name);
+        break;
+  cleanup:
+        ++snd_card_num;
+        mixer_close(mixer);
+        mixer = NULL;
+        hw_info_deinit(hw_info);
+        hw_info = NULL;
+    }
+
+done:
+    mixer_close(mixer);
+    hw_info_deinit(hw_info);
+
+    if (my_data)
+        free(my_data);
+
+    return snd_card_num;
+}