diff --git a/include/hardware/sensors-base.h b/include/hardware/sensors-base.h
index ef7eead..b88a8c2 100644
--- a/include/hardware/sensors-base.h
+++ b/include/hardware/sensors-base.h
@@ -52,6 +52,7 @@
     SENSOR_TYPE_LOW_LATENCY_OFFBODY_DETECT = 34,
     SENSOR_TYPE_ACCELEROMETER_UNCALIBRATED = 35,
     SENSOR_TYPE_HINGE_ANGLE = 36,
+    SENSOR_TYPE_HEAD_TRACKER = 37,
     SENSOR_TYPE_DEVICE_PRIVATE_BASE = 65536 /* 0x10000 */,
 };
 
diff --git a/include/hardware/sensors.h b/include/hardware/sensors.h
index a03a409..cef5dd6 100644
--- a/include/hardware/sensors.h
+++ b/include/hardware/sensors.h
@@ -186,6 +186,7 @@
 #define SENSOR_STRING_TYPE_LOW_LATENCY_OFFBODY_DETECT   "android.sensor.low_latency_offbody_detect"
 #define SENSOR_STRING_TYPE_ACCELEROMETER_UNCALIBRATED   "android.sensor.accelerometer_uncalibrated"
 #define SENSOR_STRING_TYPE_HINGE_ANGLE                  "android.sensor.hinge_angle"
+#define SENSOR_STRING_TYPE_HEAD_TRACKER                 "android.sensor.head_tracker"
 
 /**
  * Values returned by the accelerometer in various locations in the universe.
diff --git a/modules/audio_remote_submix/audio_hw.cpp b/modules/audio_remote_submix/audio_hw.cpp
index b43a44d..42d3b98 100644
--- a/modules/audio_remote_submix/audio_hw.cpp
+++ b/modules/audio_remote_submix/audio_hw.cpp
@@ -83,10 +83,7 @@
 // multiple input streams from this device.  If this option is enabled, each input stream returned
 // is *the same stream* which means that readers will race to read data from these streams.
 #define ENABLE_LEGACY_INPUT_OPEN     1
-// Whether channel conversion (16-bit signed PCM mono->stereo, stereo->mono) is enabled.
-#define ENABLE_CHANNEL_CONVERSION    1
-// Whether resampling is enabled.
-#define ENABLE_RESAMPLING            1
+
 #if LOG_STREAMS_TO_FILES
 // Folder to save stream log files to.
 #define LOG_STREAM_FOLDER "/data/misc/audioserver"
@@ -130,11 +127,6 @@
     // channel bitfields are not equivalent.
     audio_channel_mask_t input_channel_mask;
     audio_channel_mask_t output_channel_mask;
-#if ENABLE_RESAMPLING
-    // Input stream and output stream sample rates.
-    uint32_t input_sample_rate;
-    uint32_t output_sample_rate;
-#endif // ENABLE_RESAMPLING
     size_t pipe_frame_size;  // Number of bytes in each audio frame in the pipe.
     size_t buffer_size_frames; // Size of the audio pipe in frames.
     // Maximum number of frames buffered by the input and output streams.
@@ -159,11 +151,6 @@
     // destroyed if both and input and output streams are destroyed.
     struct submix_stream_out *output;
     struct submix_stream_in *input;
-#if ENABLE_RESAMPLING
-    // Buffer used as temporary storage for resampled data prior to returning data to the output
-    // stream.
-    int16_t resampler_buffer[DEFAULT_PIPE_SIZE_IN_FRAMES];
-#endif // ENABLE_RESAMPLING
 } route_config_t;
 
 struct submix_audio_device {
@@ -325,7 +312,6 @@
 static bool audio_config_compare(const audio_config * const input_config,
         const audio_config * const output_config)
 {
-#if !ENABLE_CHANNEL_CONVERSION
     const uint32_t input_channels = audio_channel_count_from_in_mask(input_config->channel_mask);
     const uint32_t output_channels = audio_channel_count_from_out_mask(output_config->channel_mask);
     if (input_channels != output_channels) {
@@ -333,13 +319,8 @@
               input_channels, output_channels);
         return false;
     }
-#endif // !ENABLE_CHANNEL_CONVERSION
-#if ENABLE_RESAMPLING
-    if (input_config->sample_rate != output_config->sample_rate &&
-            audio_channel_count_from_in_mask(input_config->channel_mask) != 1) {
-#else
+
     if (input_config->sample_rate != output_config->sample_rate) {
-#endif // ENABLE_RESAMPLING
         ALOGE("audio_config_compare() sample rate mismatch %ul vs. %ul",
               input_config->sample_rate, output_config->sample_rate);
         return false;
@@ -376,24 +357,11 @@
         in->route_handle = route_idx;
         rsxadev->routes[route_idx].input = in;
         rsxadev->routes[route_idx].config.input_channel_mask = config->channel_mask;
-#if ENABLE_RESAMPLING
-        rsxadev->routes[route_idx].config.input_sample_rate = config->sample_rate;
-        // If the output isn't configured yet, set the output sample rate to the maximum supported
-        // sample rate such that the smallest possible input buffer is created, and put a default
-        // value for channel count
-        if (!rsxadev->routes[route_idx].output) {
-            rsxadev->routes[route_idx].config.output_sample_rate = 48000;
-            rsxadev->routes[route_idx].config.output_channel_mask = AUDIO_CHANNEL_OUT_STEREO;
-        }
-#endif // ENABLE_RESAMPLING
     }
     if (out) {
         out->route_handle = route_idx;
         rsxadev->routes[route_idx].output = out;
         rsxadev->routes[route_idx].config.output_channel_mask = config->channel_mask;
-#if ENABLE_RESAMPLING
-        rsxadev->routes[route_idx].config.output_sample_rate = config->sample_rate;
-#endif // ENABLE_RESAMPLING
     }
     // Save the address
     strncpy(rsxadev->routes[route_idx].address, address, AUDIO_DEVICE_MAX_ADDRESS_LEN);
@@ -403,18 +371,14 @@
     {
         struct submix_config * const device_config = &rsxadev->routes[route_idx].config;
         uint32_t channel_count;
-        if (out)
+        if (out) {
             channel_count = audio_channel_count_from_out_mask(config->channel_mask);
-        else
+        } else {
             channel_count = audio_channel_count_from_in_mask(config->channel_mask);
-#if ENABLE_CHANNEL_CONVERSION
-        // If channel conversion is enabled, allocate enough space for the maximum number of
-        // possible channels stored in the pipe for the situation when the number of channels in
-        // the output stream don't match the number in the input stream.
-        const uint32_t pipe_channel_count = max(channel_count, 2);
-#else
+        }
+
         const uint32_t pipe_channel_count = channel_count;
-#endif // ENABLE_CHANNEL_CONVERSION
+
         const NBAIO_Format format = Format_from_SR_C(config->sample_rate, pipe_channel_count,
             config->format);
         const NBAIO_Format offers[1] = {format};
@@ -444,11 +408,7 @@
                 buffer_period_count;
         if (in) device_config->pipe_frame_size = audio_stream_in_frame_size(&in->stream);
         if (out) device_config->pipe_frame_size = audio_stream_out_frame_size(&out->stream);
-#if ENABLE_CHANNEL_CONVERSION
-        // Calculate the pipe frame size based upon the number of channels.
-        device_config->pipe_frame_size = (device_config->pipe_frame_size * pipe_channel_count) /
-                channel_count;
-#endif // ENABLE_CHANNEL_CONVERSION
+
         SUBMIX_ALOGV("submix_audio_device_create_pipe_l(): pipe frame size %zd, pipe size %zd, "
                      "period size %zd", device_config->pipe_frame_size,
                      device_config->buffer_size_frames, device_config->buffer_period_size_frames);
@@ -473,10 +433,6 @@
         rsxadev->routes[route_idx].rsxSource.clear();
     }
     memset(rsxadev->routes[route_idx].address, 0, AUDIO_DEVICE_MAX_ADDRESS_LEN);
-#if ENABLE_RESAMPLING
-    memset(rsxadev->routes[route_idx].resampler_buffer, 0,
-            sizeof(int16_t) * DEFAULT_PIPE_SIZE_IN_FRAMES);
-#endif
 }
 
 // Remove references to the specified input and output streams.  When the device no longer
@@ -624,11 +580,7 @@
 {
     const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(
             const_cast<struct audio_stream *>(stream));
-#if ENABLE_RESAMPLING
-    const uint32_t out_rate = out->dev->routes[out->route_handle].config.output_sample_rate;
-#else
     const uint32_t out_rate = out->dev->routes[out->route_handle].config.common.sample_rate;
-#endif // ENABLE_RESAMPLING
     SUBMIX_ALOGV("out_get_sample_rate() returns %u for addr %s",
             out_rate, out->dev->routes[out->route_handle].address);
     return out_rate;
@@ -637,17 +589,6 @@
 static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
 {
     struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream);
-#if ENABLE_RESAMPLING
-    // The sample rate of the stream can't be changed once it's set since this would change the
-    // output buffer size and hence break playback to the shared pipe.
-    if (rate != out->dev->routes[out->route_handle].config.output_sample_rate) {
-        ALOGE("out_set_sample_rate() resampling enabled can't change sample rate from "
-              "%u to %u for addr %s",
-              out->dev->routes[out->route_handle].config.output_sample_rate, rate,
-              out->dev->routes[out->route_handle].address);
-        return -ENOSYS;
-    }
-#endif // ENABLE_RESAMPLING
     if (!sample_rate_supported(rate)) {
         ALOGE("out_set_sample_rate(rate=%u) rate unsupported", rate);
         return -ENOSYS;
@@ -994,11 +935,7 @@
 {
     const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(
         const_cast<struct audio_stream*>(stream));
-#if ENABLE_RESAMPLING
-    const uint32_t rate = in->dev->routes[in->route_handle].config.input_sample_rate;
-#else
     const uint32_t rate = in->dev->routes[in->route_handle].config.common.sample_rate;
-#endif // ENABLE_RESAMPLING
     SUBMIX_ALOGV("in_get_sample_rate() returns %u", rate);
     return rate;
 }
@@ -1006,15 +943,6 @@
 static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
 {
     const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream);
-#if ENABLE_RESAMPLING
-    // The sample rate of the stream can't be changed once it's set since this would change the
-    // input buffer size and hence break recording from the shared pipe.
-    if (rate != in->dev->routes[in->route_handle].config.input_sample_rate) {
-        ALOGE("in_set_sample_rate() resampling enabled can't change sample rate from "
-              "%u to %u", in->dev->routes[in->route_handle].config.input_sample_rate, rate);
-        return -ENOSYS;
-    }
-#endif // ENABLE_RESAMPLING
     if (!sample_rate_supported(rate)) {
         ALOGE("in_set_sample_rate(rate=%u) rate unsupported", rate);
         return -ENOSYS;
@@ -1033,13 +961,6 @@
                             audio_stream_in_frame_size((const struct audio_stream_in *)stream);
     size_t buffer_size_frames = calculate_stream_pipe_size_in_frames(
         stream, config, config->buffer_period_size_frames, stream_frame_size);
-#if ENABLE_RESAMPLING
-    // Scale the size of the buffer based upon the maximum number of frames that could be returned
-    // given the ratio of output to input sample rate.
-    buffer_size_frames = (size_t)(((float)buffer_size_frames *
-                                   (float)config->input_sample_rate) /
-                                  (float)config->output_sample_rate);
-#endif // ENABLE_RESAMPLING
     const size_t buffer_size_bytes = buffer_size_frames * stream_frame_size;
     SUBMIX_ALOGV("in_get_buffer_size() returns %zu bytes, %zu frames", buffer_size_bytes,
                  buffer_size_frames);
@@ -1168,65 +1089,10 @@
         // read the data from the pipe (it's non blocking)
         int attempts = 0;
         char* buff = (char*)buffer;
-#if ENABLE_CHANNEL_CONVERSION
-        // Determine whether channel conversion is required.
-        const uint32_t input_channels = audio_channel_count_from_in_mask(
-            rsxadev->routes[in->route_handle].config.input_channel_mask);
-        const uint32_t output_channels = audio_channel_count_from_out_mask(
-            rsxadev->routes[in->route_handle].config.output_channel_mask);
-        if (input_channels != output_channels) {
-            SUBMIX_ALOGV("in_read(): %d output channels will be converted to %d "
-                         "input channels", output_channels, input_channels);
-            // Only support 16-bit PCM channel conversion from mono to stereo or stereo to mono.
-            ALOG_ASSERT(rsxadev->routes[in->route_handle].config.common.format ==
-                    AUDIO_FORMAT_PCM_16_BIT);
-            ALOG_ASSERT((input_channels == 1 && output_channels == 2) ||
-                        (input_channels == 2 && output_channels == 1));
-        }
-#endif // ENABLE_CHANNEL_CONVERSION
-
-#if ENABLE_RESAMPLING
-        const uint32_t input_sample_rate = in_get_sample_rate(&stream->common);
-        const uint32_t output_sample_rate =
-                rsxadev->routes[in->route_handle].config.output_sample_rate;
-        const size_t resampler_buffer_size_frames =
-            sizeof(rsxadev->routes[in->route_handle].resampler_buffer) /
-                sizeof(rsxadev->routes[in->route_handle].resampler_buffer[0]);
-        float resampler_ratio = 1.0f;
-        // Determine whether resampling is required.
-        if (input_sample_rate != output_sample_rate) {
-            resampler_ratio = (float)output_sample_rate / (float)input_sample_rate;
-            // Only support 16-bit PCM mono resampling.
-            // NOTE: Resampling is performed after the channel conversion step.
-            ALOG_ASSERT(rsxadev->routes[in->route_handle].config.common.format ==
-                    AUDIO_FORMAT_PCM_16_BIT);
-            ALOG_ASSERT(audio_channel_count_from_in_mask(
-                    rsxadev->routes[in->route_handle].config.input_channel_mask) == 1);
-        }
-#endif // ENABLE_RESAMPLING
 
         while ((remaining_frames > 0) && (attempts < MAX_READ_ATTEMPTS)) {
             ssize_t frames_read = -1977;
             size_t read_frames = remaining_frames;
-#if ENABLE_RESAMPLING
-            char* const saved_buff = buff;
-            if (resampler_ratio != 1.0f) {
-                // Calculate the number of frames from the pipe that need to be read to generate
-                // the data for the input stream read.
-                const size_t frames_required_for_resampler = (size_t)(
-                    (float)read_frames * (float)resampler_ratio);
-                read_frames = min(frames_required_for_resampler, resampler_buffer_size_frames);
-                // Read into the resampler buffer.
-                buff = (char*)rsxadev->routes[in->route_handle].resampler_buffer;
-            }
-#endif // ENABLE_RESAMPLING
-#if ENABLE_CHANNEL_CONVERSION
-            if (output_channels == 1 && input_channels == 2) {
-                // Need to read half the requested frames since the converted output
-                // data will take twice the space (mono->stereo).
-                read_frames /= 2;
-            }
-#endif // ENABLE_CHANNEL_CONVERSION
 
             SUBMIX_ALOGV("in_read(): frames available to read %zd", source->availableToRead());
 
@@ -1234,56 +1100,6 @@
 
             SUBMIX_ALOGV("in_read(): frames read %zd", frames_read);
 
-#if ENABLE_CHANNEL_CONVERSION
-            // Perform in-place channel conversion.
-            // NOTE: In the following "input stream" refers to the data returned by this function
-            // and "output stream" refers to the data read from the pipe.
-            if (input_channels != output_channels && frames_read > 0) {
-                int16_t *data = (int16_t*)buff;
-                if (output_channels == 2 && input_channels == 1) {
-                    // Offset into the output stream data in samples.
-                    ssize_t output_stream_offset = 0;
-                    for (ssize_t input_stream_frame = 0; input_stream_frame < frames_read;
-                         input_stream_frame++, output_stream_offset += 2) {
-                        // Average the content from both channels.
-                        data[input_stream_frame] = ((int32_t)data[output_stream_offset] +
-                                                    (int32_t)data[output_stream_offset + 1]) / 2;
-                    }
-                } else if (output_channels == 1 && input_channels == 2) {
-                    // Offset into the input stream data in samples.
-                    ssize_t input_stream_offset = (frames_read - 1) * 2;
-                    for (ssize_t output_stream_frame = frames_read - 1; output_stream_frame >= 0;
-                         output_stream_frame--, input_stream_offset -= 2) {
-                        const short sample = data[output_stream_frame];
-                        data[input_stream_offset] = sample;
-                        data[input_stream_offset + 1] = sample;
-                    }
-                }
-            }
-#endif // ENABLE_CHANNEL_CONVERSION
-
-#if ENABLE_RESAMPLING
-            if (resampler_ratio != 1.0f) {
-                SUBMIX_ALOGV("in_read(): resampling %zd frames", frames_read);
-                const int16_t * const data = (int16_t*)buff;
-                int16_t * const resampled_buffer = (int16_t*)saved_buff;
-                // Resample with *no* filtering - if the data from the ouptut stream was really
-                // sampled at a different rate this will result in very nasty aliasing.
-                const float output_stream_frames = (float)frames_read;
-                size_t input_stream_frame = 0;
-                for (float output_stream_frame = 0.0f;
-                     output_stream_frame < output_stream_frames &&
-                     input_stream_frame < remaining_frames;
-                     output_stream_frame += resampler_ratio, input_stream_frame++) {
-                    resampled_buffer[input_stream_frame] = data[(size_t)output_stream_frame];
-                }
-                ALOG_ASSERT(input_stream_frame <= (ssize_t)resampler_buffer_size_frames);
-                SUBMIX_ALOGV("in_read(): resampler produced %zd frames", input_stream_frame);
-                frames_read = input_stream_frame;
-                buff = saved_buff;
-            }
-#endif // ENABLE_RESAMPLING
-
             if (frames_read > 0) {
 #if LOG_STREAMS_TO_FILES
                 if (in->log_fd >= 0) write(in->log_fd, buff, frames_read * frame_size);
@@ -1411,7 +1227,6 @@
     struct submix_audio_device * const rsxadev = audio_hw_device_get_submix_audio_device(dev);
     ALOGD("adev_open_output_stream(address=%s)", address);
     struct submix_stream_out *out;
-    bool force_pipe_creation = false;
     (void)handle;
     (void)devices;
     (void)flags;
@@ -1464,17 +1279,10 @@
     out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
     out->stream.get_presentation_position = out_get_presentation_position;
 
-#if ENABLE_RESAMPLING
-    // Recreate the pipe with the correct sample rate so that MonoPipe.write() rate limits
-    // writes correctly.
-    force_pipe_creation = rsxadev->routes[route_idx].config.common.sample_rate
-            != config->sample_rate;
-#endif // ENABLE_RESAMPLING
-
     // If the sink has been shutdown or pipe recreation is forced (see above), delete the pipe so
     // that it's recreated.
     if ((rsxadev->routes[route_idx].rsxSink != NULL
-            && rsxadev->routes[route_idx].rsxSink->isShutdown()) || force_pipe_creation) {
+            && rsxadev->routes[route_idx].rsxSink->isShutdown())) {
         submix_audio_device_release_pipe_l(rsxadev, route_idx);
     }
 
@@ -1779,16 +1587,9 @@
     int n = snprintf(msg, sizeof(msg), "\nReroute submix audio module:\n");
     write(fd, &msg, n);
     for (int i=0 ; i < MAX_ROUTES ; i++) {
-#if ENABLE_RESAMPLING
-        n = snprintf(msg, sizeof(msg), " route[%d] rate in=%d out=%d, addr=[%s]\n", i,
-                rsxadev->routes[i].config.input_sample_rate,
-                rsxadev->routes[i].config.output_sample_rate,
-                rsxadev->routes[i].address);
-#else
         n = snprintf(msg, sizeof(msg), " route[%d], rate=%d addr=[%s]\n", i,
                 rsxadev->routes[i].config.common.sample_rate,
                 rsxadev->routes[i].address);
-#endif
         write(fd, &msg, n);
     }
     return 0;
diff --git a/modules/sensors/dynamic_sensor/HidRawSensor.cpp b/modules/sensors/dynamic_sensor/HidRawSensor.cpp
index 8aaf2d4..6654228 100644
--- a/modules/sensors/dynamic_sensor/HidRawSensor.cpp
+++ b/modules/sensors/dynamic_sensor/HidRawSensor.cpp
@@ -439,6 +439,7 @@
 
     featureValue->reportModeFlag = SENSOR_FLAG_SPECIAL_REPORTING_MODE;
     featureValue->isWakeUp = false;
+    featureValue->useUniqueIdForUuid = false;
     memset(featureValue->uuid, 0, sizeof(featureValue->uuid));
     featureValue->isAndroidCustom = false;
 }
@@ -465,28 +466,16 @@
         for (const auto & r : packet.reports) {
             switch (r.usage) {
                 case FRIENDLY_NAME:
-                    if (!r.isByteAligned() || r.bitSize != 16 || r.count < 1) {
-                        // invalid friendly name
-                        break;
-                    }
                     if (decodeString(r, buffer, &str) && !str.empty()) {
                         featureValue->name = str;
                     }
                     break;
                 case SENSOR_MANUFACTURER:
-                    if (!r.isByteAligned() || r.bitSize != 16 || r.count < 1) {
-                        // invalid manufacturer
-                        break;
-                    }
                     if (decodeString(r, buffer, &str) && !str.empty()) {
                         featureValue->vendor = str;
                     }
                     break;
                 case PERSISTENT_UNIQUE_ID:
-                    if (!r.isByteAligned() || r.bitSize != 16 || r.count < 1) {
-                        // invalid unique id string
-                        break;
-                    }
                     if (decodeString(r, buffer, &str) && !str.empty()) {
                         featureValue->uniqueId = str;
                     }
@@ -541,10 +530,19 @@
     }
 
     // initialize uuid field, use name, vendor and uniqueId
-    if (mFeatureInfo.name.size() >= 4
-            && mFeatureInfo.vendor.size() >= 4
-            && mFeatureInfo.typeString.size() >= 4
-            && mFeatureInfo.uniqueId.size() >= 4) {
+    // initialize uuid field using one of the following methods:
+    //
+    // 1. use uniqueId
+    // 2. use name, vendor and uniqueId
+    if (mFeatureInfo.useUniqueIdForUuid) {
+        if (mFeatureInfo.uniqueId.size() == sizeof(mFeatureInfo.uuid)) {
+            memcpy(mFeatureInfo.uuid, mFeatureInfo.uniqueId.c_str(),
+                   sizeof(mFeatureInfo.uuid));
+        }
+    } else if (mFeatureInfo.name.size() >= 4
+                   && mFeatureInfo.vendor.size() >= 4
+                   && mFeatureInfo.typeString.size() >= 4
+                   && mFeatureInfo.uniqueId.size() >= 4) {
         uint32_t tmp[4], h;
         std::hash<std::string> stringHash;
         h = stringHash(mFeatureInfo.uniqueId);
@@ -643,6 +641,11 @@
     mFeatureInfo.permission = "";
     mFeatureInfo.isWakeUp = false;
 
+    // HID head tracker sensors must use the HID unique ID for the sensor UUID
+    // to permit association between the sensor and audio device (see
+    // specification for HEAD_TRACKER in SensorType).
+    mFeatureInfo.useUniqueIdForUuid = true;
+
     return true;
 }
 
@@ -1055,11 +1058,15 @@
           << "  fifoSize: " << mFeatureInfo.fifoSize << LOG_ENDL
           << "  fifoMaxSize: " << mFeatureInfo.fifoMaxSize << LOG_ENDL
           << "  reportModeFlag: " << mFeatureInfo.reportModeFlag << LOG_ENDL
-          << "  isWakeUp: " << (mFeatureInfo.isWakeUp ? "true" : "false") << LOG_ENDL
-          << "  uniqueId: " << mFeatureInfo.uniqueId << LOG_ENDL
-          << "  uuid: ";
+          << "  isWakeUp: " << (mFeatureInfo.isWakeUp ? "true" : "false") << LOG_ENDL;
 
-    ss << std::hex << std::setfill('0');
+    ss << "  uniqueId: " << std::hex << std::setfill('0');
+    for (auto d : mFeatureInfo.uniqueId) {
+          ss << std::setw(2) << static_cast<int>(d) << " ";
+    }
+    ss << std::dec << std::setfill(' ') << LOG_ENDL;
+
+    ss << "  uuid: " << std::hex << std::setfill('0');
     for (auto d : mFeatureInfo.uuid) {
           ss << std::setw(2) << static_cast<int>(d) << " ";
     }
diff --git a/modules/sensors/dynamic_sensor/HidRawSensor.h b/modules/sensors/dynamic_sensor/HidRawSensor.h
index 0989651..f6d13b5 100644
--- a/modules/sensors/dynamic_sensor/HidRawSensor.h
+++ b/modules/sensors/dynamic_sensor/HidRawSensor.h
@@ -86,6 +86,7 @@
         size_t fifoMaxSize;
         uint32_t reportModeFlag;
         bool isWakeUp;
+        bool useUniqueIdForUuid;
 
         // dynamic sensor specific
         std::string uniqueId;
diff --git a/modules/sensors/dynamic_sensor/HidUtils/HidParser.cpp b/modules/sensors/dynamic_sensor/HidUtils/HidParser.cpp
index 28d87d9..19aa429 100644
--- a/modules/sensors/dynamic_sensor/HidUtils/HidParser.cpp
+++ b/modules/sensors/dynamic_sensor/HidUtils/HidParser.cpp
@@ -240,10 +240,12 @@
                 auto logical = r.getLogicalRange();
                 auto physical = r.getPhysicalRange();
 
-                int64_t offset = physical.first - logical.first;
                 double scale = static_cast<double>((physical.second - physical.first))
                         / (logical.second - logical.first);
                 scale *= r.getExponentValue();
+                int64_t offset =
+                        (physical.first * r.getExponentValue() / scale) -
+                        logical.first;
 
                 ReportItem digest = {
                     .usage = r.getFullUsage(),
diff --git a/modules/sensors/dynamic_sensor/README.md b/modules/sensors/dynamic_sensor/README.md
new file mode 100644
index 0000000..49e541e
--- /dev/null
+++ b/modules/sensors/dynamic_sensor/README.md
@@ -0,0 +1,152 @@
+# Dynamic Sensors
+
+[TOC]
+
+## Links
+
+*   [Sensor HAL dynamic sensor support](https://source.android.com/devices/sensors/sensors-hal2#dynamic-sensors)
+*   [Sensors Multi-HAL](https://source.android.com/devices/sensors/sensors-multihal)
+
+## Adding dynamic sensor support to a device
+
+A few files need to be modified to add dynamic sensor support to a device. The
+dynamic sensor HAL must be enabled in the device product makefile and in the
+sensor sub-HAL configuration file, raw HID devices must be configured, and raw
+HID device and dynamic sensor property permissions must be set up in the SELinux
+policy files.
+
+```shell
+acme-co$ git -C device/acme/rocket-phone diff
+diff --git a/sensor_hal/hals.conf b/sensor_hal/hals.conf
+index a1f4b8b..d112546 100644
+--- a/sensor_hal/hals.conf
++++ b/sensor_hal/hals.conf
+@@ -1 +1,2 @@
++sensors.dynamic_sensor_hal.so
+ sensors.rocket-phone.so
+diff --git a/rocket-phone.mk b/rocket-phone.mk
+index 3fc8538..b1bd8a1 100644
+--- a/rocket-phone.mk
++++ b/rocket-phone.mk
+@@ -73,6 +73,9 @@
+ PRODUCT_PACKAGES += sensors.rocket-phone
+ PRODUCT_PACKAGES += thruster_stats
+
++# Add the dynamic sensor HAL.
++PRODUCT_PACKAGES += sensors.dynamic_sensor_hal
++
+ # Only install test tools in debug build or eng build.
+ ifneq ($(filter userdebug eng,$(TARGET_BUILD_VARIANT)),)
+   PRODUCT_PACKAGES += thruster_test
+diff --git a/conf/ueventd.rc b/conf/ueventd.rc
+index 88ee00b..2f03009 100644
+--- a/conf/ueventd.rc
++++ b/conf/ueventd.rc
+@@ -209,3 +209,7 @@
+
+ # Thrusters
+ /dev/thruster*                         0600   system     system
++
++# Raw HID devices
++/dev/hidraw*                           0660   system     system
++
+diff --git a/sepolicy/sensor_hal.te b/sepolicy/sensor_hal.te
+index 0797253..22a4208 100644
+--- a/sepolicy/sensor_hal.te
++++ b/sepolicy/sensor_hal.te
+@@ -52,6 +52,9 @@
+ # Allow sensor HAL to read thruster state.
+ allow hal_sensors_default thruster_state:file r_file_perms;
+
++# Allow access for dynamic sensor properties.
++get_prop(hal_sensors_default, vendor_dynamic_sensor_prop)
++
++# Allow access to raw HID devices for dynamic sensors.
++allow hal_sensors_default hidraw_device:chr_file rw_file_perms;
++
+ #
+ # Thruster sensor enforcements.
+ #
+diff --git a/sepolicy/device.te b/sepolicy/device.te
+index bc3c947..bad0be0 100644
+--- a/sepolicy/device.te
++++ b/sepolicy/device.te
+@@ -55,3 +55,7 @@
+
+ # Thruster
+ type thruster_device, dev_type;
++
++# Raw HID device
++type hidraw_device, dev_type;
++
+diff --git a/sepolicy/property.te b/sepolicy/property.te
+index 4b671a4..bb0894f 100644
+--- a/sepolicy/property.te
++++ b/sepolicy/property.te
+@@ -49,3 +49,7 @@
+
+ # Thruster
+ vendor_internal_prop(vendor_thruster_debug_prop)
++
++# Dynamic sensor
++vendor_internal_prop(vendor_dynamic_sensor_prop)
++
+diff --git a/sepolicy/file_contexts b/sepolicy/file_contexts
+index bc03a78..ff401dc 100644
+--- a/sepolicy/file_contexts
++++ b/sepolicy/file_contexts
+@@ -441,3 +441,7 @@
+ /dev/thruster-fuel                  u:object_r:thruster_device:s0
+ /dev/thruster-output                u:object_r:thruster_device:s0
+ /dev/thruster-telemetry             u:object_r:thruster_device:s0
++
++# Raw HID device
++/dev/hidraw[0-9]*                  u:object_r:hidraw_device:s0
++
+diff --git a/sepolicy/property_contexts b/sepolicy/property_contexts
+index 5d2f018..18a6059 100644
+--- a/sepolicy/property_contexts
++++ b/sepolicy/property_contexts
+@@ -104,3 +104,7 @@
+
+ # Thruster
+ vendor.thruster.debug                           u:object_r:vendor_thruster_debug_prop:s0
++
++# Dynamic sensor
++vendor.dynamic_sensor.                          u:object_r:vendor_dynamic_sensor_prop:s0
++
+acme-co$
+```
+
+Once the file modifications are made, rebuild and flash. The dynamic sensor HAL
+should be initialized and appear in the sensor service.
+
+```shell
+acme-co$ make -j28 && fastboot flashall
+.
+.
+.
+acme-co$ adb logcat -d | grep DynamicSensorHal
+12-15 18:18:45.735   791   791 D DynamicSensorHal: DynamicSensorsSubHal::getSensorsList_2_1 invoked.
+12-15 18:18:47.474   791   791 D DynamicSensorHal: DynamicSensorsSubHal::initialize invoked.
+acme-co$ adb shell dumpsys sensorservice | grep Dynamic
+0000000000) Dynamic Sensor Manager    | Google          | ver: 1 | type: android.sensor.dynamic_sensor_meta(32) | perm: n/a | flags: 0x00000007
+Dynamic Sensor Manager (handle=0x00000000, connections=1)
+         Dynamic Sensor Manager 0x00000000 | status: active | pending flush events 0
+acme-co$ adb logcat -c
+acme-co$
+```
+
+When a dynamic sensor is paired with the device (e.g., Bluetooth rocket buds),
+it will appear in the sensor service.
+
+```shell
+acme-co$ adb logcat -d | grep "DynamicSensorHal\|hidraw\|Rocket"
+12-15 18:19:55.268   157   157 I hid-generic 0003: 1234:5678.0001: hidraw0: BLUETOOTH HID v0.00 Device [RocketBuds] on
+12-15 18:19:55.235   791   809 E DynamicSensorHal: return 1 sensors
+12-15 18:19:56.239  1629  1787 I SensorService: Dynamic sensor handle 0x1 connected, type 65536, name RocketBuds
+acme-co$ adb shell dumpsys sensorservice | grep Rocket
+0x00000001) RocketBuds                | BLUETOOTH 1234:1234   | ver: 1 | type: com.google.hardware.sensor.hid_dynamic.headtracker(65536) | perm: n/a | flags: 0x00000020
+acme-co$
+```
+
diff --git a/modules/usbaudio/audio_hal.c b/modules/usbaudio/audio_hal.c
index 39c0fb5..fe921d6 100644
--- a/modules/usbaudio/audio_hal.c
+++ b/modules/usbaudio/audio_hal.c
@@ -460,6 +460,41 @@
     return num_sample_rates;
 }
 
+static bool are_all_devices_found(unsigned int num_devices_to_find,
+                                  const int cards_to_find[],
+                                  const int devices_to_find[],
+                                  unsigned int num_devices,
+                                  const int cards[],
+                                  const int devices[]) {
+    for (unsigned int i = 0; i < num_devices_to_find; ++i) {
+        unsigned int j = 0;
+        for (; j < num_devices; ++j) {
+            if (cards_to_find[i] == cards[j] && devices_to_find[i] == devices[j]) {
+                break;
+            }
+        }
+        if (j >= num_devices) {
+            return false;
+        }
+    }
+    return true;
+}
+
+static bool are_devices_the_same(unsigned int left_num_devices,
+                                 const int left_cards[],
+                                 const int left_devices[],
+                                 unsigned int right_num_devices,
+                                 const int right_cards[],
+                                 const int right_devices[]) {
+    if (left_num_devices != right_num_devices) {
+        return false;
+    }
+    return are_all_devices_found(left_num_devices, left_cards, left_devices,
+                                 right_num_devices, right_cards, right_devices) &&
+           are_all_devices_found(right_num_devices, right_cards, right_devices,
+                                 left_num_devices, left_cards, left_devices);
+}
+
 /*
  * HAl Functions
  */
@@ -548,10 +583,11 @@
     list_for_each(node, alsa_devices) {
         struct alsa_device_info *device_info =
                 node_to_item(node, struct alsa_device_info, list_node);
-        dprintf(fd, "Output Profile %zu:\n", i);
+        const char* direction = device_info->profile.direction == PCM_OUT ? "Output" : "Input";
+        dprintf(fd, "%s Profile %zu:\n", direction, i);
         profile_dump(&device_info->profile, fd);
 
-        dprintf(fd, "Output Proxy %zu:\n", i);
+        dprintf(fd, "%s Proxy %zu:\n", direction, i);
         proxy_dump(&device_info->proxy, fd);
     }
 }
@@ -1648,6 +1684,13 @@
         saved_devices[num_saved_devices++] = device_info->profile.device;
     }
 
+    if (are_devices_the_same(
+                num_configs, cards, devices, num_saved_devices, saved_cards, saved_devices)) {
+        // The new devices are the same as original ones. No need to update.
+        stream_unlock(lock);
+        return 0;
+    }
+
     device_lock(adev);
     stream_standby_l(alsa_devices, out == NULL ? &in->standby : &out->standby);
     device_unlock(adev);