diff --git a/hal/Android.mk b/hal/Android.mk
index f3d2798..c2801db 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -152,6 +152,8 @@
 
 LOCAL_MODULE_TAGS := optional
 
+LOCAL_CFLAGS += -Werror
+
 include $(BUILD_SHARED_LIBRARY)
 
 endif
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 0a5bccb..98821d2 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -2200,39 +2200,18 @@
 
 #ifdef NO_AUDIO_OUT
 static ssize_t out_write_for_no_output(struct audio_stream_out *stream,
-                                       const void *buffer, size_t bytes)
+                                       const void *buffer __unused, size_t bytes)
 {
     struct stream_out *out = (struct stream_out *)stream;
-    struct timespec t = { .tv_sec = 0, .tv_nsec = 0 };
-    int64_t now;
-    int64_t elapsed_time_since_last_write = 0;
-    int64_t sleep_time;
 
-    clock_gettime(CLOCK_MONOTONIC, &t);
-    now = (t.tv_sec * 1000000000LL + t.tv_nsec) / 1000;
-
+    /* No Output device supported other than BT for playback.
+     * Sleep for the amount of buffer duration
+     */
     lock_output_stream(out);
-    if (out->last_write_time_us)
-        elapsed_time_since_last_write = now - out->last_write_time_us;
-    sleep_time = bytes * 1000000LL / audio_stream_out_frame_size(stream) /
-               out_get_sample_rate(&stream->common) - elapsed_time_since_last_write;
-    if (sleep_time > 0) {
-        usleep(sleep_time);
-    } else {
-        // we don't sleep when we exit standby (this is typical for a real alsa buffer).
-        sleep_time = 0;
-    }
-    out->last_write_time_us = now + sleep_time;
+    usleep(bytes * 1000000 / audio_stream_out_frame_size(
+            (const struct audio_stream_out *)&out->stream) /
+            out_get_sample_rate(&out->stream.common));
     pthread_mutex_unlock(&out->lock);
-    // last_write_time_us is an approximation of when the (simulated) alsa
-    // buffer is believed completely full. The usleep above waits for more space
-    // in the buffer, but by the end of the sleep the buffer is considered
-    // topped-off.
-    //
-    // On the subsequent out_write(), we measure the elapsed time spent in
-    // the mixer. This is subtracted from the sleep estimate based on frames,
-    // thereby accounting for drain in the alsa buffer during mixing.
-    // This is a crude approximation; we don't handle underruns precisely.
     return bytes;
 }
 #endif
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index 2cf72e1..798d954 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -210,8 +210,6 @@
     card_status_t card_status;
 
     struct error_log error_log;
-
-    int64_t last_write_time_us;
 };
 
 struct stream_in {
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 4995bd8..f7953cc 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -969,7 +969,7 @@
             ALOGE("%s: Could not find the symbol acdb_loader_deallocate_ACDB from %s",
                   __func__, LIB_ACDB_LOADER);
 
-        my_data->acdb_send_audio_cal_v3 = (acdb_send_audio_cal_t)dlsym(my_data->acdb_handle,
+        my_data->acdb_send_audio_cal_v3 = (acdb_send_audio_cal_v3_t)dlsym(my_data->acdb_handle,
                                                     "acdb_loader_send_audio_cal_v3");
         if (!my_data->acdb_send_audio_cal_v3)
             ALOGE("%s: Could not find the symbol acdb_send_audio_cal from %s",
@@ -1073,7 +1073,7 @@
 {
     struct platform_data *my_data = (struct platform_data *)platform;
 
-    if (platform == NULL || device_name == NULL) {
+    if (platform == NULL) {
         ALOGW("%s: something wrong, use legacy get_snd_device name", __func__);
         strlcpy(device_name, platform_get_snd_device_name(snd_device),
                  DEVICE_NAME_MAX_SIZE);
@@ -1095,7 +1095,7 @@
 
 bool platform_check_and_set_capture_backend_cfg(struct audio_device* adev __unused,
                                               struct audio_usecase *usecase __unused,
-                                              snd_device_t snd_device)
+                                              snd_device_t snd_device __unused)
 {
     return false;
 }
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index 7b76df3..eafb64a 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -1099,7 +1099,8 @@
                                       snd_device_t snd_device,
                                       char *device_name)
 {
-    device_name = platform_get_snd_device_name(snd_device);
+    strlcpy(device_name, platform_get_snd_device_name(snd_device),
+            DEVICE_NAME_MAX_SIZE);
     return 0;
 }
 
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 9d4abec..26d78c1 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -1366,9 +1366,10 @@
 {
     struct platform_data *my_data = (struct platform_data *)platform;
 
-    if (platform == NULL || device_name == NULL) {
+    if (platform == NULL) {
         ALOGW("%s: something wrong, use legacy get_snd_device name", __func__);
-        device_name = platform_get_snd_device_name(snd_device);
+        strlcpy(device_name, platform_get_snd_device_name(snd_device),
+                DEVICE_NAME_MAX_SIZE);
     } else if (snd_device >= SND_DEVICE_MIN && snd_device < SND_DEVICE_MAX) {
         if (operator_specific_device_table[snd_device] != NULL) {
             strlcpy(device_name, get_operator_specific_device_mixer_path(snd_device),
@@ -1527,8 +1528,8 @@
     return ret;
 }
 
-int platform_get_default_app_type_v2(void *platform, usecase_type_t type __unused,
-                                     int *app_type)
+int platform_get_default_app_type_v2(void *platform __unused, usecase_type_t type __unused,
+                                     int *app_type __unused)
 {
     ALOGE("%s: Not implemented", __func__);
     return -ENOSYS;
@@ -2160,10 +2161,10 @@
             } else if ((channel_mask == AUDIO_CHANNEL_IN_FRONT_BACK) &&
                        (my_data->source_mic_type & SOURCE_DUAL_MIC)) {
                 snd_device = SND_DEVICE_IN_VOICE_REC_DMIC_STEREO;
-            } else if (((int)channel_mask == AUDIO_CHANNEL_INDEX_MASK_3) &&
+            } else if ((channel_mask == AUDIO_CHANNEL_INDEX_MASK_3) &&
                        (my_data->source_mic_type & SOURCE_THREE_MIC)) {
                 snd_device = SND_DEVICE_IN_THREE_MIC;
-            } else if (((int)channel_mask == AUDIO_CHANNEL_INDEX_MASK_4) &&
+            } else if ((channel_mask == AUDIO_CHANNEL_INDEX_MASK_4) &&
                        (my_data->source_mic_type & SOURCE_QUAD_MIC)) {
                 snd_device = SND_DEVICE_IN_QUAD_MIC;
             }
@@ -2190,10 +2191,10 @@
                  (channel_mask == AUDIO_CHANNEL_IN_STEREO)) &&
                        (my_data->source_mic_type & SOURCE_DUAL_MIC)) {
                 snd_device = SND_DEVICE_IN_UNPROCESSED_STEREO_MIC;
-            } else if (((int)channel_mask == AUDIO_CHANNEL_INDEX_MASK_3) &&
+            } else if ((channel_mask == AUDIO_CHANNEL_INDEX_MASK_3) &&
                        (my_data->source_mic_type & SOURCE_THREE_MIC)) {
                 snd_device = SND_DEVICE_IN_UNPROCESSED_THREE_MIC;
-            } else if (((int)channel_mask == AUDIO_CHANNEL_INDEX_MASK_4) &&
+            } else if ((channel_mask == AUDIO_CHANNEL_INDEX_MASK_4) &&
                        (my_data->source_mic_type & SOURCE_QUAD_MIC)) {
                 snd_device = SND_DEVICE_IN_UNPROCESSED_QUAD_MIC;
             } else {
@@ -2281,10 +2282,10 @@
             !(in_device & AUDIO_DEVICE_IN_COMMUNICATION)) {
         if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC) {
             if ((my_data->source_mic_type & SOURCE_QUAD_MIC) &&
-                (int)channel_mask == AUDIO_CHANNEL_INDEX_MASK_4) {
+                channel_mask == AUDIO_CHANNEL_INDEX_MASK_4) {
                 snd_device = SND_DEVICE_IN_QUAD_MIC;
             } else if ((my_data->source_mic_type & SOURCE_THREE_MIC) &&
-                       (int)channel_mask == AUDIO_CHANNEL_INDEX_MASK_3) {
+                       channel_mask == AUDIO_CHANNEL_INDEX_MASK_3) {
                 snd_device = SND_DEVICE_IN_THREE_MIC;
             } else if ((my_data->source_mic_type & SOURCE_DUAL_MIC) &&
                        channel_count == 2) {
@@ -2626,7 +2627,8 @@
 }
 
 bool platform_check_and_set_capture_backend_cfg(struct audio_device* adev,
-         struct audio_usecase *usecase, snd_device_t snd_device)
+         struct audio_usecase *usecase __unused,
+         snd_device_t snd_device __unused)
 {
     enum pcm_format  in_pcm_format = PCM_FORMAT_S16_LE;
 
diff --git a/hal/platform_api.h b/hal/platform_api.h
index ebbe9d2..83554b0 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -28,6 +28,7 @@
 };
 
 enum card_status_t;
+struct audio_usecase;
 enum usecase_type_t;
 
 void *platform_init(struct audio_device *adev);
diff --git a/post_proc/bundle.c b/post_proc/bundle.c
index df327ab..73b9ae5 100644
--- a/post_proc/bundle.c
+++ b/post_proc/bundle.c
@@ -244,7 +244,7 @@
 __attribute__ ((visibility ("default")))
 int offload_effects_bundle_hal_stop_output(audio_io_handle_t output, int pcm_id)
 {
-    int ret;
+    int ret = 0;
     struct listnode *node;
     struct listnode *fx_node;
     output_context_t *out_ctxt;
@@ -750,11 +750,11 @@
 
 __attribute__ ((visibility ("default")))
 audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = {
-    tag : AUDIO_EFFECT_LIBRARY_TAG,
-    version : EFFECT_LIBRARY_API_VERSION,
-    name : "Offload Effects Bundle Library",
-    implementor : "The Android Open Source Project",
-    create_effect : effect_lib_create,
-    release_effect : effect_lib_release,
-    get_descriptor : effect_lib_get_descriptor,
+    .tag = AUDIO_EFFECT_LIBRARY_TAG,
+    .version = EFFECT_LIBRARY_API_VERSION,
+    .name = "Offload Effects Bundle Library",
+    .implementor = "The Android Open Source Project",
+    .create_effect = effect_lib_create,
+    .release_effect = effect_lib_release,
+    .get_descriptor = effect_lib_get_descriptor,
 };
diff --git a/post_proc/equalizer.c b/post_proc/equalizer.c
index 1cd40d2..c3a438f 100644
--- a/post_proc/equalizer.c
+++ b/post_proc/equalizer.c
@@ -183,7 +183,7 @@
 
 int equalizer_get_num_presets(equalizer_context_t *context __unused)
 {
-    ALOGV("%s: presets_num: %d", __func__,
+    ALOGV("%s: presets_num: %zu", __func__,
            sizeof(equalizer_preset_names)/sizeof(char *));
     return sizeof(equalizer_preset_names)/sizeof(char *);
 }
diff --git a/post_proc/volume_listener.c b/post_proc/volume_listener.c
index 1402ae6..2858e04 100644
--- a/post_proc/volume_listener.c
+++ b/post_proc/volume_listener.c
@@ -366,7 +366,9 @@
             ALOGE("%s: EFFECT_CMD_INIT: %s, sending -EINVAL", __func__,
                   (p_reply_data == NULL) ? "p_reply_data is NULL" :
                   "*reply_size != sizeof(int)");
-            return -EINVAL;
+            android_errorWriteLog(0x534e4554, "32669549");
+            status = -EINVAL;
+            goto exit;
         }
         *(int *)p_reply_data = 0;
         break;
@@ -375,7 +377,9 @@
         ALOGV("%s :: cmd called EFFECT_CMD_SET_CONFIG", __func__);
         if (p_cmd_data == NULL || cmd_size != sizeof(effect_config_t)
                 || p_reply_data == NULL || reply_size == NULL || *reply_size != sizeof(int)) {
-            return -EINVAL;
+            android_errorWriteLog(0x534e4554, "32669549");
+            status = -EINVAL;
+            goto exit;
         }
         context->config = *(effect_config_t *)p_cmd_data;
         *(int *)p_reply_data = 0;
@@ -399,7 +403,9 @@
             ALOGE("%s: EFFECT_CMD_OFFLOAD: %s, sending -EINVAL", __func__,
                   (p_reply_data == NULL) ? "p_reply_data is NULL" :
                   "*reply_size != sizeof(int)");
-            return -EINVAL;
+            android_errorWriteLog(0x534e4554, "32669549");
+            status = -EINVAL;
+            goto exit;
         }
         *(int *)p_reply_data = 0;
         break;
diff --git a/visualizer/offload_visualizer.c b/visualizer/offload_visualizer.c
index c56ebff..4d1373f 100644
--- a/visualizer/offload_visualizer.c
+++ b/visualizer/offload_visualizer.c
@@ -81,11 +81,11 @@
     effect_ops_t ops;
 };
 
-typedef struct output_context_s {
+struct output_context_s {
     struct listnode outputs_list_node;  /* node in active_outputs_list */
     audio_io_handle_t handle; /* io handle */
     struct listnode effects_list; /* list of effects attached to this output */
-} output_context_t;
+};
 
 
 /* maximum time since last capture buffer update before resetting capture buffer. This means
@@ -427,7 +427,7 @@
 
 __attribute__ ((visibility ("default")))
 int visualizer_hal_start_output(audio_io_handle_t output, int pcm_id) {
-    int ret;
+    int ret = 0;
     struct listnode *node;
 
     ALOGV("%s output %d pcm_id %d", __func__, output, pcm_id);
@@ -473,7 +473,7 @@
 
 __attribute__ ((visibility ("default")))
 int visualizer_hal_stop_output(audio_io_handle_t output, int pcm_id) {
-    int ret;
+    int ret = 0;
     struct listnode *node;
     struct listnode *fx_node;
     output_context_t *out_ctxt;