Merge "configs: apq8098_latv: Add a flag to guard dolby libs compilation"
diff --git a/configs/sdm670/audio_output_policy.conf b/configs/sdm670/audio_output_policy.conf
index 52ef791..d66782e 100644
--- a/configs/sdm670/audio_output_policy.conf
+++ b/configs/sdm670/audio_output_policy.conf
@@ -82,3 +82,37 @@
     app_type 69940
   }
 }
+
+inputs {
+  primary {
+    flags AUDIO_INPUT_FLAG_NONE
+    formats AUDIO_FORMAT_PCM_16_BIT
+    sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000
+    bit_width 16
+    app_type 69938
+  }
+  record_24bit {
+    profile none
+    flags AUDIO_INPUT_FLAG_NONE
+    formats AUDIO_FORMAT_PCM_24_BIT_PACKED
+    sampling_rates 16000|48000
+    bit_width 24
+    app_type 69945
+  }
+  record_fluence1 {
+    flags AUDIO_INPUT_FLAG_COMPRESS
+    profile record_fluence
+    formats AUDIO_FORMAT_PCM_16_BIT
+    sampling_rates 48000
+    bit_width 16
+    app_type 69944
+  }
+  record_fluence {
+    flags AUDIO_INPUT_FLAG_TIMESTAMP
+    profile record_fluence
+    formats AUDIO_FORMAT_PCM_16_BIT
+    sampling_rates 48000
+    bit_width 16
+    app_type 69944
+  }
+}
diff --git a/configs/sdm670/audio_platform_info_intcodec.xml b/configs/sdm670/audio_platform_info_intcodec.xml
index b249102..87ebeb5 100644
--- a/configs/sdm670/audio_platform_info_intcodec.xml
+++ b/configs/sdm670/audio_platform_info_intcodec.xml
@@ -63,6 +63,11 @@
         <usecase name="USECASE_AUDIO_HFP_SCO_WB" type="in" id="12" />
         <usecase name="USECASE_AUDIO_PLAYBACK_VOIP" type="out" id="16" />
         <usecase name="USECASE_AUDIO_RECORD_VOIP" type="in" id="16" />
+        <usecase name="USECASE_AUDIO_RECORD_COMPRESS2" type="in" id="41"/>
+        <usecase name="USECASE_AUDIO_RECORD_COMPRESS3" type="in" id="42"/>
+        <usecase name="USECASE_AUDIO_RECORD_COMPRESS4" type="in" id="43"/>
+        <usecase name="USECASE_AUDIO_RECORD_COMPRESS5" type="in" id="44"/>
+        <usecase name="USECASE_AUDIO_RECORD_COMPRESS6" type="in" id="45"/>
     </pcm_ids>
     <config_params>
         <!-- In the below value string, the value indicates default mono -->
@@ -97,6 +102,7 @@
         <device name="SND_DEVICE_IN_UNPROCESSED_THREE_MIC" acdb_id="145"/>
         <device name="SND_DEVICE_IN_UNPROCESSED_QUAD_MIC" acdb_id="146"/>
         <device name="SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC" acdb_id="147"/>
+        <device name="SND_DEVICE_IN_HANDSET_GENERIC_QMIC" acdb_id="157"/>
     </acdb_ids>
     <backend_names>
         <device name="SND_DEVICE_OUT_HEADPHONES" backend="headphones" interface="INT0_MI2S_RX"/>
@@ -122,6 +128,7 @@
         <device name="SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_1" interface="INT4_MI2S_RX-and-INT0_MI2S_RX"/>
         <device name="SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2" interface="INT4_MI2S_RX-and-INT0_MI2S_RX"/>
         <device name="SND_DEVICE_OUT_VOICE_HANDSET" interface="INT4_MI2S_RX"/>
+        <device name="SND_DEVICE_IN_HANDSET_GENERIC_QMIC" interface="INT3_MI2S_TX"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER" interface="INT4_MI2S_RX"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER_VBAT" interface="INT4_MI2S_RX"/>
         <device name="SND_DEVICE_OUT_VOICE_SPEAKER_2" interface="INT4_MI2S_RX"/>
diff --git a/configs/sdm670/mixer_paths_mtp.xml b/configs/sdm670/mixer_paths_mtp.xml
index c5d2cf2..efb37b1 100644
--- a/configs/sdm670/mixer_paths_mtp.xml
+++ b/configs/sdm670/mixer_paths_mtp.xml
@@ -373,6 +373,11 @@
     <ctl name="IIR1 INP3 Volume" value="53" />
     <ctl name="IIR1 INP4 Volume" value="53" />
     <ctl name="IIR1 INP1 MUX" value="ZERO" />
+    <ctl name="MultiMedia17 Mixer INT3_MI2S_TX" value="0" />
+    <ctl name="MultiMedia18 Mixer INT3_MI2S_TX" value="0" />
+    <ctl name="MultiMedia19 Mixer INT3_MI2S_TX" value="0" />
+    <ctl name="MultiMedia28 Mixer INT3_MI2S_TX" value="0" />
+    <ctl name="MultiMedia29 Mixer INT3_MI2S_TX" value="0" />
 
     <!-- anc related -->
 
@@ -1270,6 +1275,26 @@
         <ctl name="MultiMedia8 Mixer INT3_MI2S_TX" value="1" />
     </path>
 
+    <path name="audio-record-compress2">
+        <ctl name="MultiMedia17 Mixer INT3_MI2S_TX" value="1" />
+    </path>
+
+    <path name="audio-record-compress3">
+        <ctl name="MultiMedia18 Mixer INT3_MI2S_TX" value="1" />
+    </path>
+
+    <path name="audio-record-compress4">
+        <ctl name="MultiMedia19 Mixer INT3_MI2S_TX" value="1" />
+    </path>
+
+    <path name="audio-record-compress5">
+        <ctl name="MultiMedia28 Mixer INT3_MI2S_TX" value="1" />
+    </path>
+
+    <path name="audio-record-compress6">
+        <ctl name="MultiMedia29 Mixer INT3_MI2S_TX" value="1" />
+    </path>
+
     <path name="audio-record-compress bt-sco">
         <ctl name="MultiMedia8 Mixer SLIM_7_TX" value="1" />
     </path>
diff --git a/configs/sdm670/sdm670.mk b/configs/sdm670/sdm670.mk
index d34ff01..b939b09 100644
--- a/configs/sdm670/sdm670.mk
+++ b/configs/sdm670/sdm670.mk
@@ -6,6 +6,7 @@
 ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
 USE_CUSTOM_AUDIO_POLICY := 1
 AUDIO_FEATURE_ENABLED_COMPRESS_CAPTURE := false
+AUDIO_FEATURE_ENABLED_COMPRESS_INPUT := true
 AUDIO_FEATURE_ENABLED_COMPRESS_VOIP := false
 AUDIO_FEATURE_ENABLED_DYNAMIC_ECNS := false
 AUDIO_FEATURE_ENABLED_EXTN_FORMATS := true
@@ -140,8 +141,14 @@
 persist.vendor.audio.fluence.voicecall=true\
 persist.vendor.audio.fluence.voicerec=false\
 persist.vendor.audio.fluence.speaker=true\
+persist.vendor.audio.fluence.audiorec=false\
 persist.vendor.audio.fluence.tmic.enabled=false
 
+# Mutlirec Apptype
+PRODUCT_PROPERTY_OVERRIDES += \
+    vendor.audio.apptype.multirec.enabled=false \
+    vendor.audio.record.multiple.enabled=false
+
 ##speaker protection v3 switch and ADSP AFE API version
 PRODUCT_PROPERTY_OVERRIDES += \
 persist.vendor.audio.spv3.enable=true\
diff --git a/configs/sdm670/sound_trigger_platform_info.xml b/configs/sdm670/sound_trigger_platform_info.xml
index b9e36f5..1b8f45a 100644
--- a/configs/sdm670/sound_trigger_platform_info.xml
+++ b/configs/sdm670/sound_trigger_platform_info.xml
@@ -38,8 +38,10 @@
         <!-- Below backend params must match with port used in mixer path file -->
         <!-- param used to configure backend sample rate, format and channels -->
         <param backend_port_name="SLIM_0_TX" />
+        <param backend_port_name="INT3_MI2S_TX" />
         <!-- Param used to match and obtain device backend index -->
         <param backend_dai_name="SLIMBUS_0_TX" />
+        <param backend_dai_name="INT3_MI2S_TX" />
     </common_config>
     <acdb_ids>
         <param DEVICE_HANDSET_MIC_APE="100" />
diff --git a/configs/sdm845/mixer_paths_i2s.xml b/configs/sdm845/mixer_paths_i2s.xml
index 10d5ec5..75d984f 100644
--- a/configs/sdm845/mixer_paths_i2s.xml
+++ b/configs/sdm845/mixer_paths_i2s.xml
@@ -144,11 +144,11 @@
     </path>
 
     <path name="adc2">
+        <ctl name="AIF1_CAP Mixer SLIM TX6" value="1"/>
         <ctl name="MI2S_TX Channels" value="One" />
-        <ctl name="AIF1_CAP Mixer SLIM TX8" value="1"/>
-        <ctl name="SLIM TX8 MUX" value="DEC8" />
-        <ctl name="ADC MUX8" value="AMIC" />
-        <ctl name="AMIC MUX8" value="ADC2" />
+        <ctl name="SLIM TX6 MUX" value="DEC6" />
+        <ctl name="ADC MUX6" value="AMIC" />
+        <ctl name="AMIC MUX6" value="ADC2" />
     </path>
 
     <path name="dmic1">
@@ -159,14 +159,15 @@
     </path>
 
     <path name="speaker">
-        <ctl name="SLIM RX3 MUX" value="AIF1_PB" />
-        <ctl name="SLIM RX4 MUX" value="AIF1_PB" />
-        <ctl name="QUAT_MI2S_RX Channels" value="Two" />
-        <ctl name="RX1 MIX1 INP1" value="RX3" />
-        <ctl name="RX2 MIX1 INP1" value="RX4" />
-        <ctl name="CLASS_H_DSM MUX" value="DSM_HPHL_RX1" />
-        <ctl name="HPHL DAC Switch" value="1" />
-        <ctl name="COMP1 Switch" value="1" />
+        <ctl name="I2S RX0 MUX" value="AIF1_PB" />
+        <ctl name="MI2S_RX Channels" value="One" />
+        <ctl name="CDC_IF RX0 MUX" value="I2S RX0" />
+        <ctl name="RX INT8_1 MIX1 INP0" value="RX0" />
+        <ctl name="COMP8 Switch" value="1" />
+        <ctl name="SpkrRight COMP Switch" value="1" />
+        <ctl name="SpkrRight BOOST Switch" value="1" />
+        <ctl name="SpkrRight VISENSE Switch" value="1" />
+        <ctl name="SpkrRight SWR DAC_Port Switch" value="1" />
     </path>
 
    <path name="sidetone-iir">
@@ -210,11 +211,13 @@
     </path>
 
     <path name="headphones">
-        <ctl name="SLIM RX2 MUX" value="AIF1_PB" />
-        <ctl name="SLIM RX3 MUX" value="AIF1_PB" />
+        <ctl name="I2S RX0 MUX" value="AIF1_PB" />
+        <ctl name="I2S RX1 MUX" value="AIF1_PB" />
+        <ctl name="CDC_IF RX0 MUX" value="I2S RX0" />
+        <ctl name="CDC_IF RX1 MUX" value="I2S RX1" />
         <ctl name="MI2S_RX Channels" value="Two" />
-        <ctl name="RX INT1_1 MIX1 INP0" value="RX2" />
-        <ctl name="RX INT2_1 MIX1 INP0" value="RX3" />
+        <ctl name="RX INT1_2 MUX" value="RX0" />
+        <ctl name="RX INT2_2 MUX" value="RX1" />
         <ctl name="RX INT1 DEM MUX" value="CLSH_DSM_OUT" />
         <ctl name="RX INT2 DEM MUX" value="CLSH_DSM_OUT" />
         <ctl name="COMP1 Switch" value="1" />
diff --git a/configs/sdm845/sound_trigger_mixer_paths_wcd9340.xml b/configs/sdm845/sound_trigger_mixer_paths_wcd9340.xml
index 676ae95..55dd42f 100644
--- a/configs/sdm845/sound_trigger_mixer_paths_wcd9340.xml
+++ b/configs/sdm845/sound_trigger_mixer_paths_wcd9340.xml
@@ -1,5 +1,5 @@
 <?xml version="1.0" encoding="ISO-8859-1"?>
-<!--- Copyright (c) 2014-2017, The Linux Foundation. All rights reserved.       -->
+<!--- Copyright (c) 2014-2018, 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    -->
@@ -193,6 +193,21 @@
         <ctl name="MAD_CPE1 Switch" value="1" />
     </path>
 
+    <path name="listen-cpe-handset-dmic">
+        <ctl name="CLK MODE" value="INTERNAL" />
+        <ctl name= "ADC MUX0" value="DMIC" />
+        <ctl name= "DMIC MUX0" value="DMIC2" />
+        <ctl name= "DEC0 Volume" value="84" />
+        <ctl name= "ADC MUX1" value="DMIC" />
+        <ctl name= "DMIC MUX1" value="DMIC0" />
+        <ctl name= "DEC1 Volume" value="84" />
+        <ctl name= "WDMA3 PORT0 MUX" value="DEC0" />
+        <ctl name= "WDMA3 PORT1 MUX" value="DEC1" />
+        <ctl name= "WDMA3 CH0 MUX" value="PORT_0" />
+        <ctl name= "WDMA3 CH1 MUX" value="PORT_1" />
+        <ctl name= "WDMA3_ON_OFF Switch" value="1" />
+    </path>
+
     <path name="listen-cpe-handset-tmic">
         <ctl name="CLK MODE" value="INTERNAL" />
         <ctl name= "ADC MUX0" value="DMIC" />
diff --git a/configs/sdm845/sound_trigger_platform_info.xml b/configs/sdm845/sound_trigger_platform_info.xml
index e1f21a7..b017cc4 100644
--- a/configs/sdm845/sound_trigger_platform_info.xml
+++ b/configs/sdm845/sound_trigger_platform_info.xml
@@ -1,5 +1,5 @@
 <?xml version="1.0" encoding="ISO-8859-1"?>
-<!--- Copyright (c) 2013-2017, The Linux Foundation. All rights reserved.       -->
+<!--- Copyright (c) 2013-2018, 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    -->
@@ -26,7 +26,11 @@
 <!--- OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN    -->
 <!--- IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.                             -->
 <sound_trigger_platform_info>
-    <param version="0x0101" /> <!-- this must be the first param -->
+    <param version="0x0102" /> <!-- this must be the first param -->
+<!--- Version History:                                                          -->
+<!--- 0x0101: Legacy version.                                                   -->
+<!--- 0x0102: Includes acdb_ids param with the gcs_usecase tag. This matches    -->
+<!--- the gcs_usecase with the acdb device that uses it.                        -->
     <common_config>
         <param max_cpe_sessions="1" />
         <param max_wdsp_sessions="2" />
@@ -50,6 +54,7 @@
         <param DEVICE_HANDSET_QMIC_APE="137" />
         <param DEVICE_HEADSET_MIC_CPE="139" />
         <param DEVICE_HANDSET_DMIC_APE="149" />
+        <param DEVICE_HANDSET_DMIC_CPE="153" />
     </acdb_ids>
     <!-- Multiple sound_model_config tags can be listed, each with unique   -->
     <!-- vendor_uuid. The below tag represents QTI SVA engine sound model   -->
@@ -72,12 +77,13 @@
         <param adm_cfg_profile="NONE" />
         <!-- fluence_type: "FLUENCE", "FLUENCE_DMIC", "FLUENCE_TMIC", -->
         <!-- "FLUENCE_QMIC". param value is valid when adm_cfg_profile="FLUENCE"-->
-        <param fluence_type="FLUENCE_QMIC" />
+        <param fluence_type="FLUENCE_DMIC" />
         <!-- wdsp_fluence_type: fluence disabled: "NONE" -->
         <!-- fluence enabled: "FLUENCE_DMIC", "FLUENCE_TMIC", "FLUENCE_QMIC" -->
         <param wdsp_fluence_type="NONE" />
         <gcs_usecase>
             <param uid="0x1" />
+            <param acdb_devices="DEVICE_HANDSET_MIC_CPE, DEVICE_HANDSET_TMIC_CPE, DEVICE_HEADSET_MIC_CPE" />
             <!-- module_id, instance_id, param_id -->
             <param load_sound_model_ids="0x00012C0D, 0x2, 0x00012C14" />
             <param confidence_levels_ids="0x00012C0D, 0x2, 0x00012C28" />
@@ -88,6 +94,7 @@
         </gcs_usecase>
         <gcs_usecase>
             <param uid="0x2" />
+            <param acdb_devices="DEVICE_HANDSET_MIC_CPE, DEVICE_HANDSET_TMIC_CPE, DEVICE_HEADSET_MIC_CPE" />
             <param load_sound_model_ids="0x00012C0D, 0x3, 0x00012C14" />
             <param confidence_levels_ids="0x00012C0D, 0x3, 0x00012C28" />
             <param detection_event_ids="0x00012C0D, 0x3, 0x00012C29" />
@@ -145,12 +152,13 @@
         <param adm_cfg_profile="NONE" />
         <!-- fluence_type: "FLUENCE", "FLUENCE_DMIC", "FLUENCE_TMIC", -->
         <!-- "FLUENCE_QMIC". param value is valid when adm_cfg_profile="FLUENCE"-->
-        <param fluence_type="FLUENCE_QMIC" />
+        <param fluence_type="FLUENCE_DMIC" />
         <!-- wdsp_fluence_type: fluence disabled: "NONE" -->
         <!-- fluence enabled: "FLUENCE_DMIC", "FLUENCE_TMIC", "FLUENCE_QMIC" -->
         <param wdsp_fluence_type="NONE" />
         <gcs_usecase>
             <param uid="0x3" />
+            <param acdb_devices="DEVICE_HANDSET_MIC_CPE, DEVICE_HANDSET_TMIC_CPE" />
             <param load_sound_model_ids="0x18000001, 0x4, 0x00012C14" />
             <param confidence_levels_ids="0x18000001, 0x4, 0x00012C28" />
             <param detection_event_ids="0x18000001, 0x4, 0x00012C29" />
@@ -158,6 +166,16 @@
             <param read_rsp_ids="0x00020013, 0x4, 0x00020016" />
             <param custom_config_ids="0x18000001, 0x4, 0x00012C20" />
         </gcs_usecase>
+        <gcs_usecase>
+            <param uid="0x4" />
+            <param acdb_devices="DEVICE_HANDSET_DMIC_CPE" />
+            <param load_sound_model_ids="0x18000001, 0x5, 0x00012C14" />
+            <param confidence_levels_ids="0x18000001, 0x5, 0x00012C28" />
+            <param detection_event_ids="0x18000001, 0x5, 0x00012C29" />
+            <param read_cmd_ids="0x00020013, 0x5, 0x00020015" />
+            <param read_rsp_ids="0x00020013, 0x5, 0x00020016" />
+            <param custom_config_ids="0x18000001, 0x5, 0x00012C20" />
+        </gcs_usecase>
         <!-- Module and param ids with which the algorithm is integrated
             in non-graphite firmware (note these must come after gcs params)
             Extends flexibility to have different ids based on execution type.
@@ -205,6 +223,13 @@
         <param bit_width="16" />
     </adm_config>
 
+    <adm_config>
+        <param adm_cfg_profile="FLUENCE_STEREO" />
+        <param app_type="69948" />
+        <param sample_rate="16000" />
+        <param bit_width="16" />
+    </adm_config>
+
     <!-- backend_type tag defines backend type for each device -->
     <!-- Default value is assumed for devices that are not listed here -->
     <backend_type>
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 9422704..0f82a38 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -3421,7 +3421,6 @@
     struct stream_out *out = (struct stream_out *)stream;
 
     lock_output_stream(out);
-
     // always send CMD_ERROR for offload streams, this
     // is needed e.g. when SSR happens within compress_open
     // since the stream is active, offload_callback_thread is also active.
@@ -3429,18 +3428,9 @@
         stop_compressed_output_l(out);
         send_offload_cmd_l(out, OFFLOAD_CMD_ERROR);
     }
-
-    // for compress streams , if the stream is not in standby
-    // it will be triggered eventually from AF.
-    bool do_standby = !out->standby &&
-                      !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
-
     pthread_mutex_unlock(&out->lock);
 
-    if (do_standby)
-        return out_standby(&out->stream.common);
-
-    return 0;
+    return out_standby(&out->stream.common);
 }
 
 /*
diff --git a/hal/msm8974/hw_info.c b/hal/msm8974/hw_info.c
index bddaf97..bdc41a5 100644
--- a/hal/msm8974/hw_info.c
+++ b/hal/msm8974/hw_info.c
@@ -558,7 +558,7 @@
     } else if (strstr(snd_card_name, "sdm660") || strstr(snd_card_name, "sdm670")) {
         ALOGV("Bear - variant soundcard");
         update_hardware_info_bear(hw_info, snd_card_name);
-    } else if (strncmp(snd_card_name, "sdx", sizeof("sdx"))) {
+    } else if (strstr(snd_card_name, "sdx")) {
         ALOGV("SDX - variant soundcard");
         update_hardware_info_sdx(hw_info, snd_card_name);
     } else {
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 39f1af7..b41784e 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (c) 2013-2017, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2018, The Linux Foundation. All rights reserved.
  * Not a Contribution.
  *
  * Copyright (C) 2013 The Android Open Source Project
@@ -2061,6 +2061,7 @@
     my_data->ext_disp_type = EXT_DISPLAY_TYPE_NONE;
     my_data->hw_dep_fd = -1;
     my_data->mono_speaker = SPKR_1;
+    my_data->speaker_lr_swap = false;
 
     be_dai_name_table = NULL;
 
diff --git a/mm-audio/aenc-aac/qdsp6/Android.mk b/mm-audio/aenc-aac/qdsp6/Android.mk
index b427233..00d7106 100644
--- a/mm-audio/aenc-aac/qdsp6/Android.mk
+++ b/mm-audio/aenc-aac/qdsp6/Android.mk
@@ -63,6 +63,7 @@
 LOCAL_PRELINK_MODULE    := false
 LOCAL_SHARED_LIBRARIES  := libmm-omxcore
 LOCAL_SHARED_LIBRARIES  += libOmxAacEnc
+LOCAL_VENDOR_MODULE     := true
 LOCAL_SRC_FILES         := test/omx_aac_enc_test.c
 
 include $(BUILD_EXECUTABLE)
diff --git a/mm-audio/aenc-amrnb/qdsp6/Android.mk b/mm-audio/aenc-amrnb/qdsp6/Android.mk
index 3fa619e..ee6b439 100644
--- a/mm-audio/aenc-amrnb/qdsp6/Android.mk
+++ b/mm-audio/aenc-amrnb/qdsp6/Android.mk
@@ -63,6 +63,7 @@
 LOCAL_PRELINK_MODULE    := false
 LOCAL_SHARED_LIBRARIES  := libmm-omxcore
 LOCAL_SHARED_LIBRARIES  += libOmxAmrEnc
+LOCAL_VENDOR_MODULE     := true
 LOCAL_SRC_FILES         := test/omx_amr_enc_test.c
 
 include $(BUILD_EXECUTABLE)
diff --git a/mm-audio/aenc-evrc/qdsp6/Android.mk b/mm-audio/aenc-evrc/qdsp6/Android.mk
index 03965cb..14a2b70 100644
--- a/mm-audio/aenc-evrc/qdsp6/Android.mk
+++ b/mm-audio/aenc-evrc/qdsp6/Android.mk
@@ -62,6 +62,7 @@
 LOCAL_PRELINK_MODULE    := false
 LOCAL_SHARED_LIBRARIES  := libmm-omxcore
 LOCAL_SHARED_LIBRARIES  += libOmxEvrcEnc
+LOCAL_VENDOR_MODULE     := true
 LOCAL_SRC_FILES         := test/omx_evrc_enc_test.c
 
 include $(BUILD_EXECUTABLE)
diff --git a/mm-audio/aenc-g711/qdsp6/Android.mk b/mm-audio/aenc-g711/qdsp6/Android.mk
index 6b2b453..d2dc9d1 100644
--- a/mm-audio/aenc-g711/qdsp6/Android.mk
+++ b/mm-audio/aenc-g711/qdsp6/Android.mk
@@ -66,6 +66,7 @@
 LOCAL_PRELINK_MODULE    := false
 LOCAL_SHARED_LIBRARIES  := libmm-omxcore
 LOCAL_SHARED_LIBRARIES  += libOmxG711Enc
+LOCAL_VENDOR_MODULE     := true
 LOCAL_SRC_FILES         := test/omx_g711_enc_test.c
 
 include $(BUILD_EXECUTABLE)
diff --git a/mm-audio/aenc-qcelp13/qdsp6/Android.mk b/mm-audio/aenc-qcelp13/qdsp6/Android.mk
index f4b904a..b88c348 100644
--- a/mm-audio/aenc-qcelp13/qdsp6/Android.mk
+++ b/mm-audio/aenc-qcelp13/qdsp6/Android.mk
@@ -65,6 +65,7 @@
 LOCAL_PRELINK_MODULE    := false
 LOCAL_SHARED_LIBRARIES  := libmm-omxcore
 LOCAL_SHARED_LIBRARIES  += libOmxQcelp13Enc
+LOCAL_VENDOR_MODULE     := true
 LOCAL_SRC_FILES         := test/omx_qcelp13_enc_test.c
 
 include $(BUILD_EXECUTABLE)