diff --git a/configs/lahaina/lahaina.mk b/configs/lahaina/lahaina.mk
index aaa7ad2..36022af 100644
--- a/configs/lahaina/lahaina.mk
+++ b/configs/lahaina/lahaina.mk
@@ -449,6 +449,10 @@
 PRODUCT_PROPERTY_OVERRIDES += \
 persist.vendor.bt.aac_frm_ctl.enabled=true
 
+#Set speaker protection cal tx path sampling rate to 48k
+PRODUCT_PROPERTY_OVERRIDES += \
+vendor.audio.spkr_prot.tx.sampling_rate=48000
+
 #enable VBR frame ctl
 PRODUCT_PROPERTY_OVERRIDES += \
 persist.vendor.bt.aac_vbr_frm_ctl.enabled=true
diff --git a/configs/lahaina/yupik_overlay/mixer_paths_yupikqrd_overlay.xml b/configs/lahaina/yupik_overlay/mixer_paths_yupikqrd_overlay.xml
index 94ad89d..d57fdad 100644
--- a/configs/lahaina/yupik_overlay/mixer_paths_yupikqrd_overlay.xml
+++ b/configs/lahaina/yupik_overlay/mixer_paths_yupikqrd_overlay.xml
@@ -173,33 +173,23 @@
     </path>
     <path name="speaker-dmic-endfire">
         <ctl name="TX_CDC_DMA_TX_3 Channels" value="Two" />
-        <ctl name="TX DEC1 MUX" value="SWR_MIC" />
-        <ctl name="TX SMIC MUX1" value="SWR_MIC1" />
-        <ctl name="TX_AIF1_CAP Mixer DEC1" value="1" />
-        <ctl name="ADC2 ChMap" value="SWRM_TX1_CH2" />
-        <ctl name="ADC2_MIXER Switch" value="1" />
-        <ctl name="ADC2 MUX" value="INP3" />
-        <ctl name="HDR12 MUX" value="NO_HDR12" />
-        <ctl name="TX DEC2 MUX" value="SWR_MIC" />
-        <ctl name="TX SMIC MUX2" value="SWR_MIC0" />
-        <ctl name="TX_AIF1_CAP Mixer DEC2" value="1" />
-        <ctl name="ADC1 ChMap" value="SWRM_TX1_CH1" />
-        <ctl name="ADC1_MIXER Switch" value="1" />
+        <ctl name="TX DEC1 MUX" value="SWR_MIC"/>
+        <ctl name="TX SMIC MUX1" value="SWR_MIC4"/>
+        <ctl name="TX_AIF1_CAP Mixer DEC1" value="1"/>
+        <ctl name="ADC3 ChMap" value="SWRM_TX2_CH1"/>
+        <ctl name="ADC3 MUX" value="INP4"/>
+        <ctl name="HDR34 MUX" value="NO_HDR34"/>
+        <ctl name="ADC3_MIXER Switch" value="1"/>
+        <ctl name="TX DEC2 MUX" value="SWR_MIC"/>
+        <ctl name="TX SMIC MUX2" value="SWR_MIC1"/>
+        <ctl name="TX_AIF1_CAP Mixer DEC2" value="1"/>
+        <ctl name="ADC2 ChMap" value="SWRM_TX1_CH2"/>
+        <ctl name="ADC2 MUX" value="INP3"/>
+        <ctl name="HDR12 MUX" value="NO_HDR12"/>
+        <ctl name="ADC2_MIXER Switch" value="1"/>
     </path>
     <path name="speaker-dmic-broadside">
-        <ctl name="TX_CDC_DMA_TX_3 Channels" value="Two" />
-        <ctl name="TX DEC1 MUX" value="SWR_MIC" />
-        <ctl name="TX SMIC MUX1" value="SWR_MIC1" />
-        <ctl name="TX_AIF1_CAP Mixer DEC1" value="1" />
-        <ctl name="ADC2 ChMap" value="SWRM_TX1_CH2" />
-        <ctl name="ADC2_MIXER Switch" value="1" />
-        <ctl name="ADC2 MUX" value="INP3" />
-        <ctl name="HDR12 MUX" value="NO_HDR12" />
-        <ctl name="TX DEC2 MUX" value="SWR_MIC" />
-        <ctl name="TX SMIC MUX2" value="SWR_MIC0" />
-        <ctl name="TX_AIF1_CAP Mixer DEC2" value="1" />
-        <ctl name="ADC1 ChMap" value="SWRM_TX1_CH1" />
-        <ctl name="ADC1_MIXER Switch" value="1" />
+          <path name="speaker-dmic-endfire" />
     </path>
     <path name="three-mic">
         <ctl name="TX_CDC_DMA_TX_3 Channels" value="Three" />
diff --git a/configs/msm8937/audio_platform_info_sdm429w_dvt2.xml b/configs/msm8937/audio_platform_info_sdm429w_dvt2.xml
index f179369..3a5c477 100644
--- a/configs/msm8937/audio_platform_info_sdm429w_dvt2.xml
+++ b/configs/msm8937/audio_platform_info_sdm429w_dvt2.xml
@@ -56,6 +56,7 @@
         <param key="spkr_1_tz_name" value="wsa881x.0f"/>
         <param key="spkr_2_tz_name" value=""/>
         <param key="hfp_vol_mixer_ctl" value="SLIMBUS_7 LOOPBACK Volume"/>
+        <param key="hfp_pcm_dev_id" value="18"/>
     </config_params>
     <acdb_ids>
         <device name="SND_DEVICE_OUT_SPEAKER_PROTECTED" acdb_id="136"/>
@@ -63,6 +64,8 @@
         <device name="SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT" acdb_id="136"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT" acdb_id="136"/>
         <device name="SND_DEVICE_IN_CAPTURE_VI_FEEDBACK" acdb_id="137"/>
+        <device name="SND_DEVICE_OUT_VOICE_SPEAKER_HFP" acdb_id="140"/>
+        <device name="SND_DEVICE_IN_VOICE_SPEAKER_MIC_HFP" acdb_id="141"/>
     </acdb_ids>
 
     <backend_names>
diff --git a/configs/msm8937/audio_policy_configuration_sdm429w.xml b/configs/msm8937/audio_policy_configuration_sdm429w.xml
index 67b930e..b210fbc 100644
--- a/configs/msm8937/audio_policy_configuration_sdm429w.xml
+++ b/configs/msm8937/audio_policy_configuration_sdm429w.xml
@@ -120,13 +120,34 @@
                 <devicePort tagName="Speaker Safe" type="AUDIO_DEVICE_OUT_SPEAKER_SAFE" role="sink">
                 </devicePort>
                 <devicePort tagName="BT SCO" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO" role="sink">
+                       <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                                samplingRates="8000,16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
                 </devicePort>
                 <devicePort tagName="BT SCO Headset" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET" role="sink">
+                      <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                               samplingRates="8000,16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
                 </devicePort>
                 <devicePort tagName="BT SCO Car Kit" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT" role="sink">
+                      <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                               samplingRates="8000,16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
                 </devicePort>
                 <devicePort tagName="Telephony Tx" type="AUDIO_DEVICE_OUT_TELEPHONY_TX" role="sink">
                 </devicePort>
+                <devicePort tagName="BT A2DP Out" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP" role="sink"
+                            encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE">
+                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </devicePort>
+                <devicePort tagName="BT A2DP Headphones" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES" role="sink"
+                            encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE">
+                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </devicePort>
+                <devicePort tagName="BT A2DP Speaker" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER" role="sink"
+                            encodedFormats="AUDIO_FORMAT_SBC AUDIO_FORMAT_AAC AUDIO_FORMAT_APTX AUDIO_FORMAT_APTX_HD AUDIO_FORMAT_LDAC AUDIO_FORMAT_CELT AUDIO_FORMAT_APTX_ADAPTIVE">
+                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                              samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </devicePort>
                 <devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source">
                 </devicePort>
                 <devicePort tagName="Built-In Back Mic" type="AUDIO_DEVICE_IN_BACK_MIC" role="source">
@@ -156,11 +177,36 @@
                        sources="Built-In Mic,Built-In Back Mic,BT SCO Headset Mic"/>
                 <route type="mix" sink="voice_rx"
                        sources="Telephony Rx"/>
+                <route type="mix" sink="BT A2DP Out"
+                       sources="primary output,deep_buffer,compressed_offload"/>
+                <route type="mix" sink="BT A2DP Headphones"
+                       sources="primary output,deep_buffer,compressed_offload"/>
+                <route type="mix" sink="BT A2DP Speaker"
+                       sources="primary output,deep_buffer,compressed_offload"/>
             </routes>
         </module>
 
         <!-- A2dp Audio HAL -->
-        <xi:include href="/vendor/etc/a2dp_audio_policy_configuration.xml"/>
+        <module name="a2dp" halVersion="2.0">
+             <mixPorts>
+                 <mixPort name="a2dp input" role="sink">
+                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                              samplingRates="44100,48000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
+                 </mixPort>
+             </mixPorts>
+
+             <devicePorts>
+                 <devicePort tagName="BT A2DP In" type="AUDIO_DEVICE_IN_BLUETOOTH_A2DP" role="source">
+                     <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                              samplingRates="44100,48000" channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
+                 </devicePort>
+             </devicePorts>
+
+             <routes>
+                 <route type="mix" sink="a2dp input"
+                        sources="BT A2DP In"/>
+             </routes>
+        </module>
 
         <!-- Usb Audio HAL -->
         <xi:include href="/vendor/etc/usb_audio_policy_configuration.xml"/>
diff --git a/hal/Android.mk b/hal/Android.mk
index 4cc91a2..e57b859 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -126,6 +126,10 @@
   LOCAL_CFLAGS += -DPLATFORM_AUTO
 endif
 
+ifeq ($(TARGET_SUPPORTS_WEARABLES),true)
+   LOCAL_CFLAGS += -DENABLE_HFP_CALIBRATION
+endif
+
 ifeq ($(strip $(AUDIO_FEATURE_ENABLED_DAEMON_SUPPORT)), true)
   LOCAL_CFLAGS += -DDAEMON_SUPPORT_AUTO
 endif
diff --git a/hal/audio_extn/hfp.c b/hal/audio_extn/hfp.c
index cb85e1a..ca5d3bb 100644
--- a/hal/audio_extn/hfp.c
+++ b/hal/audio_extn/hfp.c
@@ -1,4 +1,4 @@
-/* Copyright (c) 2012-2020, The Linux Foundation. All rights reserved.
+/* Copyright (c) 2012-2021, The Linux Foundation. All rights reserved.
 
 Redistribution and use in source and binary forms, with or without
 modification, are permitted provided that the following conditions are
@@ -50,6 +50,8 @@
 #define AUDIO_PARAMETER_HFP_SET_SAMPLING_RATE "hfp_set_sampling_rate"
 #define AUDIO_PARAMETER_KEY_HFP_VOLUME "hfp_volume"
 #define AUDIO_PARAMETER_HFP_PCM_DEV_ID "hfp_pcm_dev_id"
+#define AUDIO_PARAMETER_HFP_VOL_MIXER_CTL "hfp_vol_mixer_ctl"
+#define AUDIO_PARAMETER_HFP_VALUE_MAX   128
 
 #define AUDIO_PARAMETER_KEY_HFP_MIC_VOLUME "hfp_mic_volume"
 #define PLAYBACK_VOLUME_MAX 0x2000
@@ -85,6 +87,7 @@
     struct pcm *hfp_pcm_tx;
     bool is_hfp_running;
     float hfp_volume;
+    char  hfp_vol_mixer_ctl[AUDIO_PARAMETER_HFP_VALUE_MAX];
     int32_t hfp_pcm_dev_id;
     audio_usecase_t ucid;
     float mic_volume;
@@ -98,6 +101,7 @@
     .hfp_pcm_tx = NULL,
     .is_hfp_running = 0,
     .hfp_volume = 0,
+    .hfp_vol_mixer_ctl = {0, },
     .hfp_pcm_dev_id = HFP_ASM_RX_TX,
     .ucid = USECASE_AUDIO_HFP_SCO,
     .mic_volume = CAPTURE_VOLUME_DEFAULT,
@@ -155,8 +159,13 @@
     }
 
     ALOGD("%s: Setting HFP volume to %d \n", __func__, vol);
-    ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
-    if (!ctl) {
+
+    if (0 == hfpmod.hfp_vol_mixer_ctl[0])
+        ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+    else
+        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__, mixer_ctl_name);
         return -EINVAL;
@@ -584,6 +593,15 @@
     }
 
     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);
+        strlcpy(hfpmod.hfp_vol_mixer_ctl, value, sizeof(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) {
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index 9cd7416..a4d17b8 100755
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (c) 2014-2020, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2014-2021, The Linux Foundation. All rights reserved.
  * Not a Contribution.
  *
  * Copyright (C) 2014 The Android Open Source Project
@@ -1310,9 +1310,6 @@
 
         if (usecase->id == USECASE_AUDIO_PLAYBACK_VOIP) {
             usecase->stream.out->app_type_cfg.sample_rate = usecase->stream.out->sample_rate;
-        } else if (compare_device_type(&usecase->stream.out->device_list,
-                                       AUDIO_DEVICE_OUT_SPEAKER)) {
-            usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
         } else if ((snd_device == SND_DEVICE_OUT_HDMI ||
                     snd_device == SND_DEVICE_OUT_USB_HEADSET ||
                     snd_device == SND_DEVICE_OUT_DISPLAY_PORT) &&
@@ -1331,9 +1328,7 @@
             (usecase->stream.out->sample_rate < OUTPUT_SAMPLING_RATE_44100)) {
             /* Reset to default if no native stream is active*/
             usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
-        } else if (usecase->out_snd_device == SND_DEVICE_OUT_BT_A2DP ||
-                   usecase->out_snd_device == SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP ||
-                   usecase->out_snd_device == SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_A2DP) {
+        } else if (snd_device == SND_DEVICE_OUT_BT_A2DP) {
                  /*
                   * For a2dp playback get encoder sampling rate and set copp sampling rate,
                   * for bit width use the stream param only.
@@ -1341,6 +1336,9 @@
                    audio_extn_a2dp_get_enc_sample_rate(&usecase->stream.out->app_type_cfg.sample_rate);
                    ALOGI("%s using %d sample rate rate for A2DP CoPP",
                         __func__, usecase->stream.out->app_type_cfg.sample_rate);
+        } else if (compare_device_type(&usecase->stream.out->device_list,
+                                       AUDIO_DEVICE_OUT_SPEAKER)) {
+            usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
         }
         audio_extn_btsco_get_sample_rate(snd_device, &usecase->stream.out->app_type_cfg.sample_rate);
         sample_rate = usecase->stream.out->app_type_cfg.sample_rate;
@@ -1669,6 +1667,9 @@
             num_devices = 1;
         }
         break;
+    case PCM_HFP_CALL:
+        rc = audio_extn_utils_send_app_type_cfg_hfp(adev,usecase);
+        return rc;
     default:
         ALOGI("%s: not a playback/capture path, no need to cfg app type", __func__);
         rc = 0;
@@ -2040,7 +2041,15 @@
     } else if (type == PCM_CAPTURE && usecase->stream.in != NULL) {
         platform_send_audio_calibration(adev->platform, usecase,
                          usecase->stream.in->app_type_cfg.app_type);
-    } else if ((type == PCM_HFP_CALL) || (type == PCM_CAPTURE) ||
+    } else if (type == PCM_HFP_CALL) {
+        /* when app type is default. the sample rate is not used to send cal */
+#ifdef ENABLE_HFP_CALIBRATION
+        platform_send_audio_calibration_hfp(adev->platform, usecase->in_snd_device);
+#else
+        platform_send_audio_calibration(adev->platform, usecase,
+                         platform_get_default_app_type_v2(adev->platform, usecase->type));
+#endif
+    } else if ((type == PCM_CAPTURE) ||
                (type == TRANSCODE_LOOPBACK_RX && usecase->stream.inout != NULL) ||
                (type == ICC_CALL) || (type == SYNTH_LOOPBACK)) {
         platform_send_audio_calibration(adev->platform, usecase,
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 071c594..062386c 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -8797,7 +8797,30 @@
         /* When set to false, HAL should disable EC and NS */
         if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0){
             adev->bt_sco_on = true;
-        } else {
+            /*
+             * When ever BT_SCO=ON arrives, make sure to route
+             * all use cases to SCO device, otherwise due to delay in
+             * BT_SCO=ON and lack of synchronization with create audio patch
+             * request for SCO device, some times use case not routed properly to
+             * SCO device
+             */
+            struct audio_usecase *usecase;
+            struct listnode *node;
+            list_for_each(node, &adev->usecase_list) {
+                usecase = node_to_item(node, struct audio_usecase, list);
+                if (usecase->stream.in && (usecase->type == PCM_CAPTURE) &&
+                    (!is_btsco_device(SND_DEVICE_NONE, usecase->in_snd_device))) {
+                    ALOGD("BT_SCO ON, switch all in use case to it");
+                    select_devices(adev, usecase->id);
+                    }
+                if (usecase->stream.out && (usecase->type == PCM_PLAYBACK) &&
+                    (!is_btsco_device(usecase->out_snd_device, SND_DEVICE_NONE))) {
+                     ALOGD("BT_SCO ON, switch all out use case to it");
+                     select_devices(adev, usecase->id);
+                    }
+            }
+         }
+         else {
             adev->bt_sco_on = false;
             audio_extn_sco_reset_configuration();
         }
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 98122f2..2a335f0 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -3780,6 +3780,12 @@
     return 0;
 }
 
+int platform_send_audio_calibration_hfp(void *platform __unused,
+                                        snd_device_t snd_device __unused)
+{
+    return 0;
+}
+
 int platform_switch_voice_call_device_pre(void *platform)
 {
     struct platform_data *my_data = (struct platform_data *)platform;
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index 94c0a62..439b6d7 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -536,6 +536,12 @@
     return -ENOSYS;
 }
 
+int platform_send_audio_calibration_hfp(void *platform __unused,
+                                        snd_device_t snd_device __unused)
+{
+    return 0;
+}
+
 int platform_send_audio_calibration(void *platform, struct audio_usecase *usecase,
                                     int app_type __unused)
 {
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 4e7deba..c447bd8 100755
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -5609,6 +5609,52 @@
     return 0;
 }
 
+int platform_send_audio_calibration_hfp(void *platform, snd_device_t snd_device)
+{
+    struct platform_data *my_data = (struct platform_data *)platform;
+    int acdb_dev_id, acdb_dev_type;
+    int sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+    int app_type = 0;
+
+    acdb_dev_id = platform_get_snd_device_acdb_id(snd_device);
+    if (acdb_dev_id < 0) {
+        ALOGE("%s: Could not find acdb id for device(%d)",
+              __func__, snd_device);
+        return -EINVAL;
+    }
+
+    if ((snd_device >= SND_DEVICE_OUT_BEGIN) &&
+        (snd_device < SND_DEVICE_OUT_END)) {
+        acdb_dev_type = ACDB_DEV_TYPE_OUT;
+        app_type = DEFAULT_APP_TYPE_RX_PATH;
+    } else {
+        acdb_dev_type = ACDB_DEV_TYPE_IN;
+        app_type = DEFAULT_APP_TYPE_TX_PATH;
+    }
+
+    if ((my_data->acdb_send_audio_cal_v3) &&
+        ((snd_device == SND_DEVICE_IN_VOICE_SPEAKER_MIC_HFP) ||
+        (snd_device == SND_DEVICE_IN_VOICE_SPEAKER_DMIC))) {
+        /* TX path calibration */
+        my_data->acdb_send_audio_cal_v3(acdb_dev_id, ACDB_DEV_TYPE_IN,
+                                DEFAULT_APP_TYPE_TX_PATH, sample_rate, 0);
+        my_data->acdb_send_audio_cal_v3(acdb_dev_id, ACDB_DEV_TYPE_OUT,
+                                DEFAULT_APP_TYPE_RX_PATH, sample_rate, 0);
+    } else if ((my_data->acdb_send_audio_cal_v3) &&
+               (snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_HFP)) {
+        /* RX path calibration */
+        my_data->acdb_send_audio_cal_v3(acdb_dev_id, ACDB_DEV_TYPE_IN,
+                                DEFAULT_APP_TYPE_TX_PATH, sample_rate,0 );
+        my_data->acdb_send_audio_cal_v3(acdb_dev_id, ACDB_DEV_TYPE_OUT,
+                                DEFAULT_APP_TYPE_RX_PATH, sample_rate,0 );
+    } else if (my_data->acdb_send_audio_cal) {
+        my_data->acdb_send_audio_cal(acdb_dev_id, acdb_dev_type, app_type,
+                                     sample_rate);
+    }
+
+    return 0;
+}
+
 int platform_switch_voice_call_device_pre(void *platform)
 {
     struct platform_data *my_data = (struct platform_data *)platform;
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 9d08378..922d61e 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -182,6 +182,7 @@
 int platform_get_native_support();
 int platform_send_audio_calibration(void *platform, struct audio_usecase *usecase,
                                     int app_type);
+int platform_send_audio_calibration_hfp(void *platform, snd_device_t snd_device);
 int platform_get_default_app_type(void *platform);
 int platform_get_default_app_type_v2(void *platform, usecase_type_t  type);
 int platform_switch_voice_call_device_pre(void *platform);