diff --git a/audio/7.0/IDevice.hal b/audio/7.0/IDevice.hal
index 7082d6b..eecd92e 100644
--- a/audio/7.0/IDevice.hal
+++ b/audio/7.0/IDevice.hal
@@ -315,7 +315,6 @@
      *                INVALID_STATE if the device was already closed
      *                or there are streams currently opened.
      */
-    @exit
     close() generates (Result retval);
 
     /**
diff --git a/audio/7.0/IStream.hal b/audio/7.0/IStream.hal
index dacd3fd..789cb1d 100644
--- a/audio/7.0/IStream.hal
+++ b/audio/7.0/IStream.hal
@@ -44,13 +44,6 @@
     getBufferSize() generates (uint64_t bufferSize);
 
     /**
-     * Return the sampling rate in Hz.
-     *
-     * @return sampleRateHz sample rate in Hz.
-     */
-    getSampleRate() generates (uint32_t sampleRateHz);
-
-    /**
      * Return supported native sampling rates of the stream for a given format.
      * A supported native sample rate is a sample rate that can be efficiently
      * played by the hardware (typically without sample-rate conversions).
@@ -72,23 +65,6 @@
             generates (Result retval, vec<uint32_t> sampleRates);
 
     /**
-     * Sets the sampling rate of the stream. Calling this method is equivalent
-     * to setting AUDIO_PARAMETER_STREAM_SAMPLING_RATE on the legacy HAL.
-     * Optional method. If implemented, only called on a stopped stream.
-     *
-     * @param sampleRateHz sample rate in Hz.
-     * @return retval operation completion status.
-     */
-    setSampleRate(uint32_t sampleRateHz) generates (Result retval);
-
-    /**
-     * Return the channel mask of the stream.
-     *
-     * @return mask channel mask.
-     */
-    getChannelMask() generates (bitfield<AudioChannelMask> mask);
-
-    /**
      * Return supported channel masks of the stream. Calling this method is
      * equivalent to getting AUDIO_PARAMETER_STREAM_SUP_CHANNELS on the legacy
      * HAL.
@@ -99,24 +75,7 @@
      * @return masks supported audio masks.
      */
     getSupportedChannelMasks(AudioFormat format)
-            generates (Result retval, vec<bitfield<AudioChannelMask>> masks);
-
-    /**
-     * Sets the channel mask of the stream. Calling this method is equivalent to
-     * setting AUDIO_PARAMETER_STREAM_CHANNELS on the legacy HAL.
-     * Optional method
-     *
-     * @param format audio format.
-     * @return retval operation completion status.
-     */
-    setChannelMask(bitfield<AudioChannelMask> mask) generates (Result retval);
-
-    /**
-     * Return the audio format of the stream.
-     *
-     * @return format audio format.
-     */
-    getFormat() generates (AudioFormat format);
+            generates (Result retval, vec<vec<AudioChannelMask>> masks);
 
     /**
      * Return supported audio formats of the stream. Calling this method is
@@ -130,25 +89,23 @@
     getSupportedFormats() generates (Result retval, vec<AudioFormat> formats);
 
     /**
-     * Sets the audio format of the stream. Calling this method is equivalent to
-     * setting AUDIO_PARAMETER_STREAM_FORMAT on the legacy HAL.
-     * Optional method
+     * Retrieves basic stream configuration: sample rate, audio format,
+     * channel mask.
      *
-     * @param format audio format.
-     * @return retval operation completion status.
+     * @return config basic stream configuration.
      */
-    setFormat(AudioFormat format) generates (Result retval);
+    getAudioProperties() generates (AudioBasicConfig config);
 
     /**
-     * Convenience method for retrieving several stream parameters in
-     * one transaction.
+     * Sets stream parameters. Only sets parameters that are specified.
+     * See the description of AudioBasicConfig for the details.
      *
-     * @return sampleRateHz sample rate in Hz.
-     * @return mask channel mask.
-     * @return format audio format.
+     * Optional method. If implemented, only called on a stopped stream.
+     *
+     * @param config basic stream configuration.
+     * @return retval operation completion status.
      */
-    getAudioProperties() generates (
-            uint32_t sampleRateHz, bitfield<AudioChannelMask> mask, AudioFormat format);
+    setAudioProperties(AudioBasicConfig config) generates (Result retval);
 
     /**
      * Applies audio effect to the stream.
@@ -312,6 +269,5 @@
      *                              output stream interface.
      *                INVALID_STATE if the stream was already closed.
      */
-    @exit
     close() generates (Result retval);
 };
diff --git a/audio/7.0/IStreamIn.hal b/audio/7.0/IStreamIn.hal
index 15e4363..0a3f24b 100644
--- a/audio/7.0/IStreamIn.hal
+++ b/audio/7.0/IStreamIn.hal
@@ -100,7 +100,7 @@
      *
      * The driver operates on a dedicated thread. The client must ensure that
      * the thread is given an appropriate priority and assigned to correct
-     * scheduler and cgroup. For this purpose, the method returns identifiers
+     * scheduler and cgroup. For this purpose, the method returns the identifier
      * of the driver thread.
      *
      * @param frameSize the size of a single frame, in bytes.
@@ -115,7 +115,9 @@
      *                specified at the stream opening.
      * @return statusMQ a message queue used for passing status from the driver
      *                  using ReadStatus structures.
-     * @return threadInfo identifiers of the driver's dedicated thread.
+     * @return threadId identifier of the driver's dedicated thread; the caller
+     *                  may adjust the thread priority to match the priority
+     *                  of the thread that provides audio data.
      */
     prepareForReading(uint32_t frameSize, uint32_t framesCount)
     generates (
@@ -123,7 +125,7 @@
             fmq_sync<ReadParameters> commandMQ,
             fmq_sync<uint8_t> dataMQ,
             fmq_sync<ReadStatus> statusMQ,
-            ThreadInfo threadInfo);
+            int32_t threadId);
 
     /**
      * Return the amount of input frames lost in the audio driver since the last
diff --git a/audio/7.0/IStreamOut.hal b/audio/7.0/IStreamOut.hal
index 208beb6..38d750f 100644
--- a/audio/7.0/IStreamOut.hal
+++ b/audio/7.0/IStreamOut.hal
@@ -95,7 +95,7 @@
      *
      * The driver operates on a dedicated thread. The client must ensure that
      * the thread is given an appropriate priority and assigned to correct
-     * scheduler and cgroup. For this purpose, the method returns identifiers
+     * scheduler and cgroup. For this purpose, the method returns the identifier
      * of the driver thread.
      *
      * @param frameSize the size of a single frame, in bytes.
@@ -109,7 +109,9 @@
      *                specified at the stream opening.
      * @return statusMQ a message queue used for passing status from the driver
      *                  using WriteStatus structures.
-     * @return threadInfo identifiers of the driver's dedicated thread.
+     * @return threadId identifier of the driver's dedicated thread; the caller
+     *                  may adjust the thread priority to match the priority
+     *                  of the thread that provides audio data.
      */
     prepareForWriting(uint32_t frameSize, uint32_t framesCount)
     generates (
@@ -117,7 +119,7 @@
             fmq_sync<WriteCommand> commandMQ,
             fmq_sync<uint8_t> dataMQ,
             fmq_sync<WriteStatus> statusMQ,
-            ThreadInfo threadInfo);
+            int32_t threadId);
 
     /**
      * Return the number of audio frames written by the audio DSP to DAC since
diff --git a/audio/7.0/config/api/current.txt b/audio/7.0/config/api/current.txt
index 98c5eac..fd9a8ef 100644
--- a/audio/7.0/config/api/current.txt
+++ b/audio/7.0/config/api/current.txt
@@ -6,6 +6,81 @@
     method public java.util.List<java.lang.String> getItem();
   }
 
+  public enum AudioChannelMask {
+    method public String getRawName();
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_1;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_10;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_11;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_12;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_13;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_14;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_15;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_16;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_17;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_18;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_19;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_2;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_20;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_21;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_22;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_23;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_24;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_3;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_4;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_5;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_6;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_7;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_8;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_INDEX_MASK_9;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_IN_2POINT0POINT2;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_IN_2POINT1POINT2;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_IN_3POINT0POINT2;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_IN_3POINT1POINT2;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_IN_5POINT1;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_IN_6;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_IN_FRONT_BACK;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_IN_MONO;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_IN_STEREO;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_IN_VOICE_CALL_MONO;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_IN_VOICE_DNLINK_MONO;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_IN_VOICE_UPLINK_MONO;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_2POINT0POINT2;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_2POINT1;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_2POINT1POINT2;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_3POINT0POINT2;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_3POINT1POINT2;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_5POINT1;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_5POINT1POINT2;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_5POINT1POINT4;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_5POINT1_BACK;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_5POINT1_SIDE;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_6POINT1;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_7POINT1;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_7POINT1POINT2;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_7POINT1POINT4;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_HAPTIC_AB;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_MONO;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_MONO_HAPTIC_A;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_MONO_HAPTIC_AB;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_PENTA;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_QUAD;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_QUAD_BACK;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_QUAD_SIDE;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_STEREO;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_STEREO_HAPTIC_A;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_STEREO_HAPTIC_AB;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioChannelMask AUDIO_CHANNEL_OUT_SURROUND;
+  }
+
+  public enum AudioContentType {
+    method public String getRawName();
+    enum_constant public static final audio.policy.configuration.V7_0.AudioContentType AUDIO_CONTENT_TYPE_MOVIE;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioContentType AUDIO_CONTENT_TYPE_MUSIC;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioContentType AUDIO_CONTENT_TYPE_SONIFICATION;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioContentType AUDIO_CONTENT_TYPE_SPEECH;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioContentType AUDIO_CONTENT_TYPE_UNKNOWN;
+  }
+
   public enum AudioDevice {
     method public String getRawName();
     enum_constant public static final audio.policy.configuration.V7_0.AudioDevice AUDIO_DEVICE_IN_AMBIENT;
@@ -116,6 +191,7 @@
     enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_APTX_HD;
     enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_APTX_TWSP;
     enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_CELT;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_DEFAULT;
     enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_DOLBY_TRUEHD;
     enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_DSD;
     enum_constant public static final audio.policy.configuration.V7_0.AudioFormat AUDIO_FORMAT_DTS;
@@ -164,18 +240,59 @@
     method public void setVersion(audio.policy.configuration.V7_0.Version);
   }
 
+  public enum AudioSource {
+    method public String getRawName();
+    enum_constant public static final audio.policy.configuration.V7_0.AudioSource AUDIO_SOURCE_CAMCORDER;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioSource AUDIO_SOURCE_DEFAULT;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioSource AUDIO_SOURCE_ECHO_REFERENCE;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioSource AUDIO_SOURCE_FM_TUNER;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioSource AUDIO_SOURCE_HOTWORD;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioSource AUDIO_SOURCE_MIC;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioSource AUDIO_SOURCE_REMOTE_SUBMIX;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioSource AUDIO_SOURCE_UNPROCESSED;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioSource AUDIO_SOURCE_VOICE_CALL;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioSource AUDIO_SOURCE_VOICE_COMMUNICATION;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioSource AUDIO_SOURCE_VOICE_DOWNLINK;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioSource AUDIO_SOURCE_VOICE_PERFORMANCE;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioSource AUDIO_SOURCE_VOICE_RECOGNITION;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioSource AUDIO_SOURCE_VOICE_UPLINK;
+  }
+
+  public enum AudioStreamType {
+    method public String getRawName();
+    enum_constant public static final audio.policy.configuration.V7_0.AudioStreamType AUDIO_STREAM_ACCESSIBILITY;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioStreamType AUDIO_STREAM_ALARM;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioStreamType AUDIO_STREAM_ASSISTANT;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioStreamType AUDIO_STREAM_BLUETOOTH_SCO;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioStreamType AUDIO_STREAM_DTMF;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioStreamType AUDIO_STREAM_ENFORCED_AUDIBLE;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioStreamType AUDIO_STREAM_MUSIC;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioStreamType AUDIO_STREAM_NOTIFICATION;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioStreamType AUDIO_STREAM_PATCH;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioStreamType AUDIO_STREAM_REROUTING;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioStreamType AUDIO_STREAM_RING;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioStreamType AUDIO_STREAM_SYSTEM;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioStreamType AUDIO_STREAM_TTS;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioStreamType AUDIO_STREAM_VOICE_CALL;
+  }
+
   public enum AudioUsage {
     method public String getRawName();
     enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_ALARM;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_ANNOUNCEMENT;
     enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY;
     enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE;
     enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_ASSISTANCE_SONIFICATION;
     enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_ASSISTANT;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_CALL_ASSISTANT;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_EMERGENCY;
     enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_GAME;
     enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_MEDIA;
     enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_NOTIFICATION;
     enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_SAFETY;
     enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_UNKNOWN;
+    enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_VEHICLE_STATUS;
     enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_VIRTUAL_SOURCE;
     enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_VOICE_COMMUNICATION;
     enum_constant public static final audio.policy.configuration.V7_0.AudioUsage AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING;
@@ -234,7 +351,7 @@
 
   public static class Gains.Gain {
     ctor public Gains.Gain();
-    method public String getChannel_mask();
+    method public audio.policy.configuration.V7_0.AudioChannelMask getChannel_mask();
     method public int getDefaultValueMB();
     method public int getMaxRampMs();
     method public int getMaxValueMB();
@@ -244,7 +361,7 @@
     method public String getName();
     method public int getStepValueMB();
     method public boolean getUseForVolume();
-    method public void setChannel_mask(String);
+    method public void setChannel_mask(audio.policy.configuration.V7_0.AudioChannelMask);
     method public void setDefaultValueMB(int);
     method public void setMaxRampMs(int);
     method public void setMaxValueMB(int);
@@ -327,14 +444,14 @@
 
   public class Profile {
     ctor public Profile();
-    method public String getChannelMasks();
+    method public java.util.List<audio.policy.configuration.V7_0.AudioChannelMask> getChannelMasks();
     method public String getFormat();
     method public String getName();
-    method public String getSamplingRates();
-    method public void setChannelMasks(String);
+    method public java.util.List<java.math.BigInteger> getSamplingRates();
+    method public void setChannelMasks(java.util.List<audio.policy.configuration.V7_0.AudioChannelMask>);
     method public void setFormat(String);
     method public void setName(String);
-    method public void setSamplingRates(String);
+    method public void setSamplingRates(java.util.List<java.math.BigInteger>);
   }
 
   public class Reference {
@@ -365,24 +482,6 @@
     method public void setType(audio.policy.configuration.V7_0.MixType);
   }
 
-  public enum Stream {
-    method public String getRawName();
-    enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_ACCESSIBILITY;
-    enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_ALARM;
-    enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_ASSISTANT;
-    enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_BLUETOOTH_SCO;
-    enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_DTMF;
-    enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_ENFORCED_AUDIBLE;
-    enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_MUSIC;
-    enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_NOTIFICATION;
-    enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_PATCH;
-    enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_REROUTING;
-    enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_RING;
-    enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_SYSTEM;
-    enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_TTS;
-    enum_constant public static final audio.policy.configuration.V7_0.Stream AUDIO_STREAM_VOICE_CALL;
-  }
-
   public class SurroundFormats {
     ctor public SurroundFormats();
     method public java.util.List<audio.policy.configuration.V7_0.SurroundFormats.Format> getFormat();
@@ -412,10 +511,10 @@
     method public audio.policy.configuration.V7_0.DeviceCategory getDeviceCategory();
     method public java.util.List<java.lang.String> getPoint();
     method public String getRef();
-    method public audio.policy.configuration.V7_0.Stream getStream();
+    method public audio.policy.configuration.V7_0.AudioStreamType getStream();
     method public void setDeviceCategory(audio.policy.configuration.V7_0.DeviceCategory);
     method public void setRef(String);
-    method public void setStream(audio.policy.configuration.V7_0.Stream);
+    method public void setStream(audio.policy.configuration.V7_0.AudioStreamType);
   }
 
   public class Volumes {
diff --git a/audio/7.0/config/audio_policy_configuration.xsd b/audio/7.0/config/audio_policy_configuration.xsd
index 19c6f70..4555a88 100644
--- a/audio/7.0/config/audio_policy_configuration.xsd
+++ b/audio/7.0/config/audio_policy_configuration.xsd
@@ -13,7 +13,6 @@
          See the License for the specific language governing permissions and
          limitations under the License.
 -->
-<!-- TODO: define a targetNamespace. Note that it will break retrocompatibility -->
 <xs:schema version="2.0"
            elementFormDefault="qualified"
            attributeFormDefault="unqualified"
@@ -27,7 +26,9 @@
     <xs:simpleType name="halVersion">
         <xs:annotation>
             <xs:documentation xml:lang="en">
-                Version of the interface the hal implements.
+                Version of the interface the hal implements. Note that this
+                relates to legacy HAL API versions since HIDL APIs are versioned
+                using other mechanisms.
             </xs:documentation>
         </xs:annotation>
         <xs:restriction base="xs:decimal">
@@ -154,7 +155,6 @@
             <xs:element name="item" type="xs:token" minOccurs="0" maxOccurs="unbounded"/>
         </xs:sequence>
     </xs:complexType>
-    <!-- TODO: separate values by space for better xsd validations. -->
     <xs:simpleType name="audioInOutFlags">
         <xs:annotation>
             <xs:documentation xml:lang="en">
@@ -212,9 +212,6 @@
             </xs:element>
         </xs:sequence>
     </xs:complexType>
-    <!-- Enum values of audio_device_t in audio.h
-         TODO: generate from hidl to avoid manual sync.
-         TODO: separate source and sink in the xml for better xsd validations. -->
     <xs:simpleType name="audioDevice">
         <xs:restriction base="xs:string">
             <xs:enumeration value="AUDIO_DEVICE_NONE"/>
@@ -252,7 +249,6 @@
             <xs:enumeration value="AUDIO_DEVICE_OUT_DEFAULT"/>
             <xs:enumeration value="AUDIO_DEVICE_OUT_STUB"/>
 
-            <!-- Due to the xml format, IN types can not be a separated from OUT types -->
             <xs:enumeration value="AUDIO_DEVICE_IN_COMMUNICATION"/>
             <xs:enumeration value="AUDIO_DEVICE_IN_AMBIENT"/>
             <xs:enumeration value="AUDIO_DEVICE_IN_BUILTIN_MIC"/>
@@ -298,10 +294,9 @@
     <xs:simpleType name="extendableAudioDevice">
         <xs:union memberTypes="audioDevice vendorExtension"/>
     </xs:simpleType>
-    <!-- Enum values of audio_format_t in audio.h
-         TODO: generate from hidl to avoid manual sync. -->
     <xs:simpleType name="audioFormat">
         <xs:restriction base="xs:string">
+            <xs:enumeration value="AUDIO_FORMAT_DEFAULT" />
             <xs:enumeration value="AUDIO_FORMAT_PCM_16_BIT" />
             <xs:enumeration value="AUDIO_FORMAT_PCM_8_BIT"/>
             <xs:enumeration value="AUDIO_FORMAT_PCM_32_BIT"/>
@@ -382,9 +377,14 @@
     <xs:simpleType name="extendableAudioFormat">
         <xs:union memberTypes="audioFormat vendorExtension"/>
     </xs:simpleType>
-    <!-- Enum values of audio::common::4_0::AudioUsage
-         TODO: generate from HIDL to avoid manual sync. -->
     <xs:simpleType name="audioUsage">
+        <xs:annotation>
+            <xs:documentation xml:lang="en">
+                Audio usage specifies the intended use case for the sound being played.
+                Please consult frameworks/base/media/java/android/media/AudioAttributes.java
+                for the description of each value.
+            </xs:documentation>
+        </xs:annotation>
         <xs:restriction base="xs:string">
             <xs:enumeration value="AUDIO_USAGE_UNKNOWN" />
             <xs:enumeration value="AUDIO_USAGE_MEDIA" />
@@ -399,34 +399,119 @@
             <xs:enumeration value="AUDIO_USAGE_GAME" />
             <xs:enumeration value="AUDIO_USAGE_VIRTUAL_SOURCE" />
             <xs:enumeration value="AUDIO_USAGE_ASSISTANT" />
+            <xs:enumeration value="AUDIO_USAGE_CALL_ASSISTANT" />
+            <xs:enumeration value="AUDIO_USAGE_EMERGENCY" />
+            <xs:enumeration value="AUDIO_USAGE_SAFETY" />
+            <xs:enumeration value="AUDIO_USAGE_VEHICLE_STATUS" />
+            <xs:enumeration value="AUDIO_USAGE_ANNOUNCEMENT" />
         </xs:restriction>
     </xs:simpleType>
     <xs:simpleType name="audioUsageList">
         <xs:list itemType="audioUsage"/>
     </xs:simpleType>
-    <!-- TODO: Change to a space separated list to xsd enforce correctness. -->
-    <xs:simpleType name="samplingRates">
-        <xs:restriction base="xs:string">
-            <xs:pattern value="[0-9]+(,[0-9]+)*"/>
-        </xs:restriction>
-    </xs:simpleType>
-    <!-- TODO: Change to a space separated list to xsd enforce correctness. -->
-    <xs:simpleType name="channelMask">
+    <xs:simpleType name="audioContentType">
         <xs:annotation>
             <xs:documentation xml:lang="en">
-                Comma (",") separated list of channel flags
-                from audio_channel_mask_t.
+                Audio content type expresses the general category of the content.
+                Please consult frameworks/base/media/java/android/media/AudioAttributes.java
+                for the description of each value.
             </xs:documentation>
         </xs:annotation>
         <xs:restriction base="xs:string">
-            <xs:pattern value="[_A-Z][_A-Z0-9]*(,[_A-Z][_A-Z0-9]*)*"/>
+            <xs:enumeration value="AUDIO_CONTENT_TYPE_UNKNOWN"/>
+            <xs:enumeration value="AUDIO_CONTENT_TYPE_SPEECH"/>
+            <xs:enumeration value="AUDIO_CONTENT_TYPE_MUSIC"/>
+            <xs:enumeration value="AUDIO_CONTENT_TYPE_MOVIE"/>
+            <xs:enumeration value="AUDIO_CONTENT_TYPE_SONIFICATION"/>
         </xs:restriction>
     </xs:simpleType>
+    <xs:simpleType name="samplingRates">
+        <xs:list itemType="xs:nonNegativeInteger" />
+    </xs:simpleType>
+    <xs:simpleType name="audioChannelMask">
+        <xs:annotation>
+            <xs:documentation xml:lang="en">
+                Audio channel mask specifies presence of particular channels.
+                There are two representations:
+                - representation position (traditional discrete channel specification,
+                  e.g. "left", "right");
+                - indexed (this is similar to "tracks" in audio mixing, channels
+                  are represented using numbers).
+            </xs:documentation>
+        </xs:annotation>
+        <xs:restriction base="xs:string">
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_MONO"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_STEREO"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_2POINT1"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_2POINT0POINT2"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_2POINT1POINT2"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_3POINT0POINT2"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_3POINT1POINT2"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_QUAD"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_QUAD_BACK"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_QUAD_SIDE"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_SURROUND"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_PENTA"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_5POINT1"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_5POINT1_BACK"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_5POINT1_SIDE"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_5POINT1POINT2"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_5POINT1POINT4"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_6POINT1"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_7POINT1"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_7POINT1POINT2"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_7POINT1POINT4"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_MONO_HAPTIC_A"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_STEREO_HAPTIC_A"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_HAPTIC_AB"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_MONO_HAPTIC_AB"/>
+            <xs:enumeration value="AUDIO_CHANNEL_OUT_STEREO_HAPTIC_AB"/>
+            <xs:enumeration value="AUDIO_CHANNEL_IN_MONO"/>
+            <xs:enumeration value="AUDIO_CHANNEL_IN_STEREO"/>
+            <xs:enumeration value="AUDIO_CHANNEL_IN_FRONT_BACK"/>
+            <xs:enumeration value="AUDIO_CHANNEL_IN_6"/>
+            <xs:enumeration value="AUDIO_CHANNEL_IN_2POINT0POINT2"/>
+            <xs:enumeration value="AUDIO_CHANNEL_IN_2POINT1POINT2"/>
+            <xs:enumeration value="AUDIO_CHANNEL_IN_3POINT0POINT2"/>
+            <xs:enumeration value="AUDIO_CHANNEL_IN_3POINT1POINT2"/>
+            <xs:enumeration value="AUDIO_CHANNEL_IN_5POINT1"/>
+            <xs:enumeration value="AUDIO_CHANNEL_IN_VOICE_UPLINK_MONO"/>
+            <xs:enumeration value="AUDIO_CHANNEL_IN_VOICE_DNLINK_MONO"/>
+            <xs:enumeration value="AUDIO_CHANNEL_IN_VOICE_CALL_MONO"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_1"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_2"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_3"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_4"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_5"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_6"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_7"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_8"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_9"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_10"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_11"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_12"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_13"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_14"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_15"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_16"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_17"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_18"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_19"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_20"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_21"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_22"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_23"/>
+            <xs:enumeration value="AUDIO_CHANNEL_INDEX_MASK_24"/>
+        </xs:restriction>
+    </xs:simpleType>
+    <xs:simpleType name="channelMasks">
+        <xs:list itemType="audioChannelMask" />
+    </xs:simpleType>
     <xs:complexType name="profile">
         <xs:attribute name="name" type="xs:token" use="optional"/>
         <xs:attribute name="format" type="extendableAudioFormat" use="optional"/>
         <xs:attribute name="samplingRates" type="samplingRates" use="optional"/>
-        <xs:attribute name="channelMasks" type="channelMask" use="optional"/>
+        <xs:attribute name="channelMasks" type="channelMasks" use="optional"/>
     </xs:complexType>
     <xs:simpleType name="gainMode">
         <xs:restriction base="xs:string">
@@ -441,7 +526,7 @@
                 <xs:complexType>
                     <xs:attribute name="name" type="xs:token" use="required"/>
                     <xs:attribute name="mode" type="gainMode" use="required"/>
-                    <xs:attribute name="channel_mask" type="channelMask" use="optional"/>
+                    <xs:attribute name="channel_mask" type="audioChannelMask" use="optional"/>
                     <xs:attribute name="minValueMB" type="xs:int" use="optional"/>
                     <xs:attribute name="maxValueMB" type="xs:int" use="optional"/>
                     <xs:attribute name="defaultValueMB" type="xs:int" use="optional"/>
@@ -537,9 +622,14 @@
             <xs:pattern value="([0-9]{1,2}|100),-?[0-9]+"/>
         </xs:restriction>
     </xs:simpleType>
-    <!-- Enum values of audio_stream_type_t in audio-base.h
-         TODO: generate from hidl to avoid manual sync. -->
-    <xs:simpleType name="stream">
+    <xs:simpleType name="audioStreamType">
+        <xs:annotation>
+            <xs:documentation xml:lang="en">
+                Audio stream type describing the intended use case of a stream.
+                Please consult frameworks/base/media/java/android/media/AudioSystem.java
+                for the description of each value.
+            </xs:documentation>
+        </xs:annotation>
         <xs:restriction base="xs:string">
             <xs:enumeration value="AUDIO_STREAM_VOICE_CALL"/>
             <xs:enumeration value="AUDIO_STREAM_SYSTEM"/>
@@ -557,8 +647,32 @@
             <xs:enumeration value="AUDIO_STREAM_PATCH"/>
         </xs:restriction>
     </xs:simpleType>
-    <!-- Enum values of device_category from Volume.h.
-         TODO: generate from hidl to avoid manual sync. -->
+    <xs:simpleType name="audioSource">
+        <xs:annotation>
+            <xs:documentation xml:lang="en">
+                An audio source defines the intended use case for the sound being recorded.
+                Please consult frameworks/base/media/java/android/media/MediaRecorder.java
+                for the description of each value.
+            </xs:documentation>
+        </xs:annotation>
+        <xs:restriction base="xs:string">
+            <xs:enumeration value="AUDIO_SOURCE_DEFAULT"/>
+            <xs:enumeration value="AUDIO_SOURCE_MIC"/>
+            <xs:enumeration value="AUDIO_SOURCE_VOICE_UPLINK"/>
+            <xs:enumeration value="AUDIO_SOURCE_VOICE_DOWNLINK"/>
+            <xs:enumeration value="AUDIO_SOURCE_VOICE_CALL"/>
+            <xs:enumeration value="AUDIO_SOURCE_CAMCORDER"/>
+            <xs:enumeration value="AUDIO_SOURCE_VOICE_RECOGNITION"/>
+            <xs:enumeration value="AUDIO_SOURCE_VOICE_COMMUNICATION"/>
+            <xs:enumeration value="AUDIO_SOURCE_REMOTE_SUBMIX"/>
+            <xs:enumeration value="AUDIO_SOURCE_UNPROCESSED"/>
+            <xs:enumeration value="AUDIO_SOURCE_VOICE_PERFORMANCE"/>
+            <xs:enumeration value="AUDIO_SOURCE_ECHO_REFERENCE"/>
+            <xs:enumeration value="AUDIO_SOURCE_FM_TUNER"/>
+            <xs:enumeration value="AUDIO_SOURCE_HOTWORD"/>
+        </xs:restriction>
+    </xs:simpleType>
+    <!-- Enum values of device_category from Volume.h. -->
     <xs:simpleType name="deviceCategory">
         <xs:restriction base="xs:string">
             <xs:enumeration value="DEVICE_CATEGORY_HEADSET"/>
@@ -591,7 +705,7 @@
         <xs:sequence>
             <xs:element name="point" type="volumePoint" minOccurs="0" maxOccurs="unbounded"/>
         </xs:sequence>
-        <xs:attribute name="stream" type="stream"/>
+        <xs:attribute name="stream" type="audioStreamType"/>
         <xs:attribute name="deviceCategory" type="deviceCategory"/>
         <xs:attribute name="ref" type="xs:token" use="optional"/>
     </xs:complexType>
diff --git a/audio/7.0/config/update_audio_policy_config.sh b/audio/7.0/config/update_audio_policy_config.sh
new file mode 100755
index 0000000..8714b5f
--- /dev/null
+++ b/audio/7.0/config/update_audio_policy_config.sh
@@ -0,0 +1,159 @@
+#!/bin/bash
+
+# Copyright (C) 2020 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.
+
+# This script is used to update audio policy configuration files
+# to comply with the updated audio_policy_configuration.xsd from V7.0.
+#
+# The main difference is the separator used in lists for attributes.
+# Since the XML Schema Definition standard only allows space to be
+# used as a separator (see https://www.w3.org/TR/xmlschema11-2/#list-datatypes)
+# the previous versions used a regular expression to validate lists
+# in attribute values. E.g. the channel masks were validated using
+# the following regexp: [_A-Z][_A-Z0-9]*(,[_A-Z][_A-Z0-9]*)*
+# This has an obvious drawback of missing typos in the config file.
+#
+# The V7.0 has shifted to defining most of the frequently changed
+# types in the XSD schema only. This allows for verifying all the values
+# in lists, but in order to comply with XML Schema requirements
+# list elements must be separated by space.
+#
+# Since the APM config files typically use include directives,
+# the script must be pointed to the main APM config file and will
+# take care all the included files automatically.
+# If the included file is a shared version from 'frameworks/av',
+# instead of updating it the script checks if there is a newer
+# version with the corresponding name suffix (e.g.
+# 'a2dp_audio_policy_configuration_7_0.xml') and updates the include
+# path instead.
+
+set -euo pipefail
+
+if (echo "$@" | grep -qe -h); then
+    echo "This script will update Audio Policy Manager config file"
+    echo "to the format required by V7.0 XSD schema from a previous"
+    echo "version."
+    echo
+    echo "USAGE: $0 [APM_XML_FILE] [OLD_VERSION]"
+    echo "       APM_XML_FILE specifies the path to audio_policy_configuration.xml"
+    echo "                    relative to Android repository root"
+    echo "       OLD_VERSION specifies the version of schema currently used"
+    echo
+    echo "Example: $0 device/generic/goldfish/audio/policy/audio_policy_configuration.xml 6.0"
+    exit
+fi
+readonly HAL_DIRECTORY=hardware/interfaces/audio
+readonly SHARED_CONFIGS_DIRECTORY=frameworks/av/services/audiopolicy/config
+readonly OLD_VERSION=${2:-$(ls ${ANDROID_BUILD_TOP}/${HAL_DIRECTORY} | grep -E '[0-9]+\.[0-9]+' |
+                                sort -n | tail -n1)}
+readonly NEW_VERSION=7.0
+readonly NEW_VERSION_UNDERSCORE=7_0
+
+readonly SOURCE_CONFIG=${ANDROID_BUILD_TOP}/$1
+
+# First, validate the input using the schema of the current version
+
+echo Validating the source against the $OLD_VERSION schema
+xmllint --noout --xinclude \
+        --nofixup-base-uris --path "$ANDROID_BUILD_TOP/$SHARED_CONFIGS_DIRECTORY" \
+        --schema ${ANDROID_BUILD_TOP}/${HAL_DIRECTORY}/${OLD_VERSION}/config/audio_policy_configuration.xsd \
+        ${SOURCE_CONFIG}
+if [ $? -ne 0 ]; then
+    echo
+    echo "Config file fails validation for the specified version $OLD_VERSION--unsafe to update"
+    exit 1
+fi
+
+# Find all the source files recursively
+
+SOURCE_FILES=${SOURCE_CONFIG}
+SHARED_FILES=
+findIncludes() {
+    local FILES_TO_CHECK=
+    for F in $1; do
+        local FOUND_INCLUDES=$(grep -Po '<xi:include href="\K[^"]+(?="\/>)' ${F})
+        for I in ${FOUND_INCLUDES}; do
+            SOURCE_FULL_PATH=$(dirname ${F})/${I}
+            SHARED_FULL_PATH=${ANDROID_BUILD_TOP}/${SHARED_CONFIGS_DIRECTORY}/${I}
+            if [ -f "$SOURCE_FULL_PATH" ]; then
+                # Device-specific file.
+                SOURCE_FILES+=$'\n'${SOURCE_FULL_PATH}
+                FILES_TO_CHECK+=$'\n'${SOURCE_FULL_PATH}
+            elif [ -f "$SHARED_FULL_PATH" ]; then
+                # Shared file from the frameworks repo.
+                SHARED_FILES+=$'\n'${I}
+                FILES_TO_CHECK+=$'\n'${SHARED_FULL_PATH}
+            else
+                echo
+                echo "Include file not found: $I"
+                exit 1
+            fi
+        done
+    done
+    if [ "$FILES_TO_CHECK" ]; then
+        findIncludes "$FILES_TO_CHECK"
+    fi
+}
+findIncludes ${SOURCE_FILES}
+
+echo "Will update $1 and included device-specific files in place."
+echo "Will update paths to shared included files."
+echo "Press Ctrl-C to cancel, Enter to continue"
+read
+
+updateFile() {
+    FILE=$1
+    ATTR=$2
+    SEPARATOR=$3
+    SRC_LINES=$(grep -nPo "$ATTR=\"[^\"]+\"" ${FILE} || true)
+    for S in $SRC_LINES; do
+        # Prepare instruction for 'sed' for in-place editing of specified line
+        R=$(echo ${S} | sed -e 's/^[0-9]\+:/\//' | sed -e "s/$SEPARATOR/ /g")
+        S=$(echo ${S} | sed -e 's/:/s\//')${R}/
+        echo ${S} | sed -i -f - ${FILE}
+    done
+}
+for F in $SOURCE_FILES; do
+    updateFile ${F} "channelMasks" ","
+    updateFile ${F} "samplingRates" ","
+done;
+
+updateIncludes() {
+    FILE=$1
+    for I in $SHARED_FILES; do
+        NEW_VERSION_I=${I%.*}_${NEW_VERSION_UNDERSCORE}.${I##*.}
+        if [ -e "$ANDROID_BUILD_TOP/$SHARED_CONFIGS_DIRECTORY/$NEW_VERSION_I" ]; then
+            echo "s/$I/$NEW_VERSION_I/g" | sed -i -f - ${FILE}
+        fi
+    done
+}
+for F in $SOURCE_FILES; do
+    updateIncludes ${F}
+done
+
+# Validate the results against the new schema
+
+echo Validating the result against the $NEW_VERSION schema
+xmllint --noout --xinclude \
+        --nofixup-base-uris --path "$ANDROID_BUILD_TOP/$SHARED_CONFIGS_DIRECTORY" \
+        --schema ${ANDROID_BUILD_TOP}/${HAL_DIRECTORY}/${NEW_VERSION}/config/audio_policy_configuration.xsd \
+        ${SOURCE_CONFIG}
+if [ $? -ne 0 ]; then
+    echo
+    echo "Config file fails validation for the specified version $NEW_VERSION--please check the changes"
+    exit 1
+fi
+echo
+echo "Please check the diff and update path to APM shared files in the device makefile!"
diff --git a/audio/7.0/types.hal b/audio/7.0/types.hal
index b0b0843..15ca492 100644
--- a/audio/7.0/types.hal
+++ b/audio/7.0/types.hal
@@ -355,3 +355,58 @@
      */
     TimestretchFallbackMode fallbackMode;
 };
+
+/**
+ * The audio output flags serve two purposes:
+ *
+ *  - when an output stream is created they indicate its attributes;
+ *
+ *  - when present in an output profile descriptor listed for a particular audio
+ *    hardware module, they indicate that an output stream can be opened that
+ *    supports the attributes indicated by the flags.
+ */
+@export(name="audio_output_flags_t", value_prefix="AUDIO_OUTPUT_FLAG_")
+enum AudioOutputFlag : int32_t {
+    NONE    = 0x0, // no attributes
+    DIRECT  = 0x1, // this output directly connects a track
+                   // to one output stream: no software mixer
+    PRIMARY = 0x2, // this output is the primary output of the device. It is
+                   // unique and must be present. It is opened by default and
+                   // receives routing, audio mode and volume controls related
+                   // to voice calls.
+    FAST    = 0x4,    // output supports "fast tracks", defined elsewhere
+    DEEP_BUFFER      = 0x8,   // use deep audio buffers
+    COMPRESS_OFFLOAD = 0x10,  // offload playback of compressed streams to
+                              // hardware codec
+    NON_BLOCKING     = 0x20,  // use non-blocking write
+    HW_AV_SYNC = 0x40,   // output uses a hardware A/V sync
+    TTS        = 0x80,   // output for streams transmitted through speaker at a
+                         // sample rate high enough to accommodate lower-range
+                         // ultrasonic p/b
+    RAW        = 0x100,  // minimize signal processing
+    SYNC       = 0x200,  // synchronize I/O streams
+    IEC958_NONAUDIO = 0x400, // Audio stream contains compressed audio in SPDIF
+                             // data bursts, not PCM.
+    DIRECT_PCM = 0x2000,     // Audio stream containing PCM data that needs
+                             // to pass through compress path for DSP post proc.
+    MMAP_NOIRQ = 0x4000, // output operates in MMAP no IRQ mode.
+    VOIP_RX = 0x8000,    // preferred output for VoIP calls.
+    /** preferred output for call music */
+    INCALL_MUSIC = 0x10000,
+};
+
+/**
+ * The audio input flags are analogous to audio output flags.
+ */
+@export(name="audio_input_flags_t", value_prefix="AUDIO_INPUT_FLAG_")
+enum AudioInputFlag : int32_t {
+    NONE         = 0x0,  // no attributes
+    FAST         = 0x1,  // prefer an input that supports "fast tracks"
+    HW_HOTWORD   = 0x2,  // prefer an input that captures from hw hotword source
+    RAW          = 0x4,  // minimize signal processing
+    SYNC         = 0x8,  // synchronize I/O streams
+    MMAP_NOIRQ   = 0x10, // input operates in MMAP no IRQ mode.
+    VOIP_TX      = 0x20, // preferred input for VoIP calls.
+    HW_AV_SYNC   = 0x40, // input connected to an output that uses a hardware A/V sync
+    DIRECT       = 0x80, // for acquiring encoded streams
+};
diff --git a/audio/common/7.0/types.hal b/audio/common/7.0/types.hal
index 2288eb1..94d0af7 100644
--- a/audio/common/7.0/types.hal
+++ b/audio/common/7.0/types.hal
@@ -18,34 +18,18 @@
 
 import android.hidl.safe_union@1.0;
 
-/*
- *
- *  IDs and Handles
- *
- */
-
 /**
- * Handle type for identifying audio sources and sinks.
+ * Handle type for identifying audio resources. Handles are allocated by the framework.
  */
 typedef int32_t AudioIoHandle;
 
 /**
- * Audio hw module handle functions or structures referencing a module.
- */
-typedef int32_t AudioModuleHandle;
-
-/**
  * Each port has a unique ID or handle allocated by policy manager.
  */
 typedef int32_t AudioPortHandle;
 
 /**
- * Each patch is identified by a handle at the interface used to create that
- * patch. For instance, when a patch is created by the audio HAL, the HAL
- * allocates and returns a handle.  This handle is unique to a given audio HAL
- * hardware module.  But the same patch receives another system wide unique
- * handle allocated by the framework.  This unique handle is used for all
- * transactions inside the framework.
+ * Each patch is identified by a handle allocated by the HAL.
  */
 typedef int32_t AudioPatchHandle;
 
@@ -55,17 +39,6 @@
 typedef uint32_t AudioHwSync;
 
 /**
- * Each port has a unique ID or handle allocated by policy manager.
- */
-@export(name="")
-enum AudioHandleConsts : int32_t {
-    AUDIO_IO_HANDLE_NONE = 0,
-    AUDIO_MODULE_HANDLE_NONE = 0,
-    AUDIO_PORT_HANDLE_NONE = 0,
-    AUDIO_PATCH_HANDLE_NONE = 0,
-};
-
-/**
  * Commonly used structure for passing unique identifieds (UUID).
  * For the definition of UUID, refer to ITU-T X.667 spec.
  */
@@ -86,116 +59,25 @@
 
 /**
  * Audio stream type describing the intended use case of a stream.
+ * See 'audioStreamType' in audio_policy_configuration.xsd for the
+ * list of allowed values.
  */
-@export(name="audio_stream_type_t", value_prefix="AUDIO_STREAM_")
-enum AudioStreamType : int32_t {
-    // These values must kept in sync with
-    //  frameworks/base/media/java/android/media/AudioSystem.java
-    /** Used to identify the default audio stream volume. */
-    DEFAULT          = -1,
-    /** Specifies the minimum value for use in checks and loops. */
-    MIN              = 0,
-    /** Used to identify the volume of audio streams for phone calls. */
-    VOICE_CALL       = 0,
-    /** Used to identify the volume of audio streams for system sounds. */
-    SYSTEM           = 1,
-    /**
-     * Used to identify the volume of audio streams for the phone ring
-     * and message alerts.
-     */
-    RING             = 2,
-    /** Used to identify the volume of audio streams for music playback. */
-    MUSIC            = 3,
-    /** Used to identify the volume of audio streams for alarms. */
-    ALARM            = 4,
-    /** Used to identify the volume of audio streams for notifications. */
-    NOTIFICATION     = 5,
-    /**
-     * Used to identify the volume of audio streams for phone calls
-     * when connected on bluetooth.
-     */
-    BLUETOOTH_SCO    = 6,
-    /**
-     * Used to identify the volume of audio streams for enforced system
-     * sounds in certain countries (e.g camera in Japan). */
-    ENFORCED_AUDIBLE = 7,
-    /** Used to identify the volume of audio streams for DTMF tones. */
-    DTMF             = 8,
-    /**
-     * Used to identify the volume of audio streams exclusively transmitted
-     * through the speaker (TTS) of the device.
-     */
-    TTS              = 9,
-    /**
-     * Used to identify the volume of audio streams for accessibility prompts.
-     */
-    ACCESSIBILITY    = 10,
-    /** Used to identify the volume of audio streams for virtual assistant. */
-    ASSISTANT        = 11,
-};
+typedef string AudioStreamType;
 
-@export(name="audio_source_t", value_prefix="AUDIO_SOURCE_")
-enum AudioSource : int32_t {
-    // These values must kept in sync with
-    //  frameworks/base/media/java/android/media/MediaRecorder.java,
-    //  system/media/audio_effects/include/audio_effects/audio_effects_conf.h
-    /** Default audio source. */
-    DEFAULT             = 0,
-    /** Microphone audio source. */
-    MIC                 = 1,
-    /** Voice call uplink (Tx) audio source. */
-    VOICE_UPLINK        = 2,
-    /** Voice call downlink (Rx) audio source. */
-    VOICE_DOWNLINK      = 3,
-    /** Voice call uplink + downlink audio source. */
-    VOICE_CALL          = 4,
-    /**
-     * Microphone audio source tuned for video recording, with the same
-     * orientation as the camera if available.
-     */
-    CAMCORDER           = 5,
-    /** Microphone audio source tuned for voice recognition. */
-    VOICE_RECOGNITION   = 6,
-    /**
-     * Microphone audio source tuned for voice communications such as VoIP. It
-     * will for instance take advantage of echo cancellation or automatic gain
-     * control if available.
-     */
-    VOICE_COMMUNICATION = 7,
-    /**
-     * Source for the mix to be presented remotely. An example of remote
-     * presentation is Wifi Display where a dongle attached to a TV can be used
-     * to play the mix captured by this audio source.
-     */
-    REMOTE_SUBMIX       = 8,
-    /**
-     * Source for unprocessed sound. Usage examples include level measurement
-     * and raw signal analysis.
-     */
-    UNPROCESSED         = 9,
-    /**
-     * Source for capturing audio meant to be processed in real time and played back for live
-     * performance (e.g karaoke). The capture path will minimize latency and coupling with
-     * playback path.
-     */
-    VOICE_PERFORMANCE    = 10,
-    /**
-     * Source for an echo canceller to capture the reference signal to be cancelled.
-     * The echo reference signal will be captured as close as possible to the DAC in order
-     * to include all post processing applied to the playback path.
-     */
-    ECHO_REFERENCE      = 1997,
-    /** Virtual source for the built-in FM tuner. */
-    FM_TUNER            = 1998,
-    /** Virtual source for the last captured hotword. */
-    HOTWORD             = 1999,
-};
-
-typedef int32_t AudioSession;
 /**
- * Special audio session values.
+ * An audio source defines the intended use case for the sound being recorded.
+ * See 'audioSource' in audio_policy_configuration.xsd for the
+ * list of allowed values.
  */
-@export(name="audio_session_t", value_prefix="AUDIO_SESSION_")
+typedef string AudioSource;
+
+/**
+ * An audio session identifier is used to designate the particular
+ * playback or recording session (e.g. playback performed by a certain
+ * application).
+ */
+typedef int32_t AudioSession;
+
 enum AudioSessionConsts : int32_t {
     /**
      * Session for effects attached to a particular sink or source audio device
@@ -213,382 +95,29 @@
      * (value must be 0)
      */
     OUTPUT_MIX = 0,
-    /**
-     * Application does not specify an explicit session ID to be used, and
-     * requests a new session ID to be allocated. Corresponds to
-     * AudioManager.AUDIO_SESSION_ID_GENERATE and
-     * AudioSystem.AUDIO_SESSION_ALLOCATE.
-     */
-    ALLOCATE = 0,
-    /**
-     * For use with AudioRecord::start(), this indicates no trigger session.
-     * It is also used with output tracks and patch tracks, which never have a
-     * session.
-     */
-    NONE = 0
 };
 
 /**
- * Audio format  is a 32-bit word that consists of:
- *   main format field (upper 8 bits)
- *   sub format field (lower 24 bits).
- *
- * The main format indicates the main codec type. The sub format field indicates
- * options and parameters for each format. The sub format is mainly used for
- * record to indicate for instance the requested bitrate or profile.  It can
- * also be used for certain formats to give informations not present in the
- * encoded audio stream (e.g. octet alignement for AMR).
+ * Audio format indicates audio codec type.
+ * See 'audioFormat' in audio_policy_configuration.xsd for the
+ * list of allowed values.
  */
-@export(name="audio_format_t", value_prefix="AUDIO_FORMAT_")
-enum AudioFormat : uint32_t {
-    INVALID             = 0xFFFFFFFFUL,
-    DEFAULT             = 0,
-    PCM                 = 0x00000000UL,
-    MP3                 = 0x01000000UL,
-    AMR_NB              = 0x02000000UL,
-    AMR_WB              = 0x03000000UL,
-    AAC                 = 0x04000000UL,
-    /** Deprecated, Use AAC_HE_V1 */
-    HE_AAC_V1           = 0x05000000UL,
-    /** Deprecated, Use AAC_HE_V2 */
-    HE_AAC_V2           = 0x06000000UL,
-    VORBIS              = 0x07000000UL,
-    OPUS                = 0x08000000UL,
-    AC3                 = 0x09000000UL,
-    E_AC3               = 0x0A000000UL,
-    DTS                 = 0x0B000000UL,
-    DTS_HD              = 0x0C000000UL,
-    /** IEC61937 is encoded audio wrapped in 16-bit PCM. */
-    IEC61937            = 0x0D000000UL,
-    DOLBY_TRUEHD        = 0x0E000000UL,
-    EVRC                = 0x10000000UL,
-    EVRCB               = 0x11000000UL,
-    EVRCWB              = 0x12000000UL,
-    EVRCNW              = 0x13000000UL,
-    AAC_ADIF            = 0x14000000UL,
-    WMA                 = 0x15000000UL,
-    WMA_PRO             = 0x16000000UL,
-    AMR_WB_PLUS         = 0x17000000UL,
-    MP2                 = 0x18000000UL,
-    QCELP               = 0x19000000UL,
-    DSD                 = 0x1A000000UL,
-    FLAC                = 0x1B000000UL,
-    ALAC                = 0x1C000000UL,
-    APE                 = 0x1D000000UL,
-    AAC_ADTS            = 0x1E000000UL,
-    SBC                 = 0x1F000000UL,
-    APTX                = 0x20000000UL,
-    APTX_HD             = 0x21000000UL,
-    AC4                 = 0x22000000UL,
-    LDAC                = 0x23000000UL,
-    /** Dolby Metadata-enhanced Audio Transmission */
-    MAT                 = 0x24000000UL,
-    AAC_LATM            = 0x25000000UL,
-    CELT                = 0x26000000UL,
-    APTX_ADAPTIVE       = 0x27000000UL,
-    LHDC                = 0x28000000UL,
-    LHDC_LL             = 0x29000000UL,
-    APTX_TWSP           = 0x2A000000UL,
-
-    /** Deprecated */
-    MAIN_MASK           = 0xFF000000UL,
-    SUB_MASK            = 0x00FFFFFFUL,
-
-    /* Subformats */
-    PCM_SUB_16_BIT        = 0x1, // PCM signed 16 bits
-    PCM_SUB_8_BIT         = 0x2, // PCM unsigned 8 bits
-    PCM_SUB_32_BIT        = 0x3, // PCM signed .31 fixed point
-    PCM_SUB_8_24_BIT      = 0x4, // PCM signed 8.23 fixed point
-    PCM_SUB_FLOAT         = 0x5, // PCM single-precision float pt
-    PCM_SUB_24_BIT_PACKED = 0x6, // PCM signed .23 fix pt (3 bytes)
-
-    MP3_SUB_NONE          = 0x0,
-
-    AMR_SUB_NONE          = 0x0,
-
-    AAC_SUB_MAIN          = 0x1,
-    AAC_SUB_LC            = 0x2,
-    AAC_SUB_SSR           = 0x4,
-    AAC_SUB_LTP           = 0x8,
-    AAC_SUB_HE_V1         = 0x10,
-    AAC_SUB_SCALABLE      = 0x20,
-    AAC_SUB_ERLC          = 0x40,
-    AAC_SUB_LD            = 0x80,
-    AAC_SUB_HE_V2         = 0x100,
-    AAC_SUB_ELD           = 0x200,
-    AAC_SUB_XHE           = 0x300,
-
-    VORBIS_SUB_NONE       = 0x0,
-
-    E_AC3_SUB_JOC         = 0x1,
-
-    MAT_SUB_1_0           = 0x1,
-    MAT_SUB_2_0           = 0x2,
-    MAT_SUB_2_1           = 0x3,
-
-    /* Aliases */
-    /** note != AudioFormat.ENCODING_PCM_16BIT */
-    PCM_16_BIT          = (PCM | PCM_SUB_16_BIT),
-    /** note != AudioFormat.ENCODING_PCM_8BIT */
-    PCM_8_BIT           = (PCM | PCM_SUB_8_BIT),
-    PCM_32_BIT          = (PCM | PCM_SUB_32_BIT),
-    PCM_8_24_BIT        = (PCM | PCM_SUB_8_24_BIT),
-    PCM_FLOAT           = (PCM | PCM_SUB_FLOAT),
-    PCM_24_BIT_PACKED   = (PCM | PCM_SUB_24_BIT_PACKED),
-    AAC_MAIN            = (AAC | AAC_SUB_MAIN),
-    AAC_LC              = (AAC | AAC_SUB_LC),
-    AAC_SSR             = (AAC | AAC_SUB_SSR),
-    AAC_LTP             = (AAC | AAC_SUB_LTP),
-    AAC_HE_V1           = (AAC | AAC_SUB_HE_V1),
-    AAC_SCALABLE        = (AAC | AAC_SUB_SCALABLE),
-    AAC_ERLC            = (AAC | AAC_SUB_ERLC),
-    AAC_LD              = (AAC | AAC_SUB_LD),
-    AAC_HE_V2           = (AAC | AAC_SUB_HE_V2),
-    AAC_ELD             = (AAC | AAC_SUB_ELD),
-    AAC_XHE             = (AAC | AAC_SUB_XHE),
-    AAC_ADTS_MAIN       = (AAC_ADTS | AAC_SUB_MAIN),
-    AAC_ADTS_LC         = (AAC_ADTS | AAC_SUB_LC),
-    AAC_ADTS_SSR        = (AAC_ADTS | AAC_SUB_SSR),
-    AAC_ADTS_LTP        = (AAC_ADTS | AAC_SUB_LTP),
-    AAC_ADTS_HE_V1      = (AAC_ADTS | AAC_SUB_HE_V1),
-    AAC_ADTS_SCALABLE   = (AAC_ADTS | AAC_SUB_SCALABLE),
-    AAC_ADTS_ERLC       = (AAC_ADTS | AAC_SUB_ERLC),
-    AAC_ADTS_LD         = (AAC_ADTS | AAC_SUB_LD),
-    AAC_ADTS_HE_V2      = (AAC_ADTS | AAC_SUB_HE_V2),
-    AAC_ADTS_ELD        = (AAC_ADTS | AAC_SUB_ELD),
-    AAC_ADTS_XHE        = (AAC_ADTS | AAC_SUB_XHE),
-    E_AC3_JOC           = (E_AC3 | E_AC3_SUB_JOC),
-    MAT_1_0             = (MAT | MAT_SUB_1_0),
-    MAT_2_0             = (MAT | MAT_SUB_2_0),
-    MAT_2_1             = (MAT | MAT_SUB_2_1),
-    AAC_LATM_LC         = (AAC_LATM | AAC_SUB_LC),
-    AAC_LATM_HE_V1      = (AAC_LATM | AAC_SUB_HE_V1),
-    AAC_LATM_HE_V2      = (AAC_LATM | AAC_SUB_HE_V2),
-};
+typedef string AudioFormat;
 
 /**
- * Usage of these values highlights places in the code that use 2- or 8- channel
- * assumptions.
+ * Audio channel mask indicates presence of particular channels.
+ * See 'audioChannelMask' in audio_policy_configuration.xsd for the
+ * list of allowed values.
  */
-@export(name="")
-enum FixedChannelCount : int32_t {
-    FCC_2 = 2, // This is typically due to legacy implementation of stereo I/O
-    FCC_8 = 8  // This is typically due to audio mixer and resampler limitations
-};
+typedef string AudioChannelMask;
 
 /**
- * A channel mask per se only defines the presence or absence of a channel, not
- * the order.
- *
- * The channel order convention is that channels are interleaved in order from
- * least significant channel mask bit to most significant channel mask bit,
- * with unused bits skipped. For example for stereo, LEFT would be first,
- * followed by RIGHT.
- * Any exceptions to this convention are noted at the appropriate API.
- *
- * AudioChannelMask is an opaque type and its internal layout should not be
- * assumed as it may change in the future.  Instead, always use functions
- * to examine it.
- *
- * These are the current representations:
- *
- *   REPRESENTATION_POSITION
- *     is a channel mask representation for position assignment.  Each low-order
- *     bit corresponds to the spatial position of a transducer (output), or
- *     interpretation of channel (input).  The user of a channel mask needs to
- *     know the context of whether it is for output or input.  The constants
- *     OUT_* or IN_* apply to the bits portion.  It is not permitted for no bits
- *     to be set.
- *
- *   REPRESENTATION_INDEX
- *     is a channel mask representation for index assignment.  Each low-order
- *     bit corresponds to a selected channel.  There is no platform
- *     interpretation of the various bits.  There is no concept of output or
- *     input.  It is not permitted for no bits to be set.
- *
- * All other representations are reserved for future use.
- *
- * Warning: current representation distinguishes between input and output, but
- * this will not the be case in future revisions of the platform. Wherever there
- * is an ambiguity between input and output that is currently resolved by
- * checking the channel mask, the implementer should look for ways to fix it
- * with additional information outside of the mask.
+ * Basic configuration applicable to any stream of audio.
  */
-@export(name="", value_prefix="AUDIO_CHANNEL_")
-enum AudioChannelMask : uint32_t {
-    /** must be 0 for compatibility */
-    REPRESENTATION_POSITION = 0,
-    /** 1 is reserved for future use */
-    REPRESENTATION_INDEX    = 2,
-    /* 3 is reserved for future use */
-
-    /** These can be a complete value of AudioChannelMask */
-    NONE                      = 0x0,
-    INVALID                   = 0xC0000000,
-
-   /*
-    * These can be the bits portion of an AudioChannelMask
-    * with representation REPRESENTATION_POSITION.
-    */
-
-    /** output channels */
-    OUT_FRONT_LEFT            = 0x1,
-    OUT_FRONT_RIGHT           = 0x2,
-    OUT_FRONT_CENTER          = 0x4,
-    OUT_LOW_FREQUENCY         = 0x8,
-    OUT_BACK_LEFT             = 0x10,
-    OUT_BACK_RIGHT            = 0x20,
-    OUT_FRONT_LEFT_OF_CENTER  = 0x40,
-    OUT_FRONT_RIGHT_OF_CENTER = 0x80,
-    OUT_BACK_CENTER           = 0x100,
-    OUT_SIDE_LEFT             = 0x200,
-    OUT_SIDE_RIGHT            = 0x400,
-    OUT_TOP_CENTER            = 0x800,
-    OUT_TOP_FRONT_LEFT        = 0x1000,
-    OUT_TOP_FRONT_CENTER      = 0x2000,
-    OUT_TOP_FRONT_RIGHT       = 0x4000,
-    OUT_TOP_BACK_LEFT         = 0x8000,
-    OUT_TOP_BACK_CENTER       = 0x10000,
-    OUT_TOP_BACK_RIGHT        = 0x20000,
-    OUT_TOP_SIDE_LEFT         = 0x40000,
-    OUT_TOP_SIDE_RIGHT        = 0x80000,
-
-    /**
-     * Haptic channel characteristics are specific to a device and
-     * only used to play device specific resources (eg: ringtones).
-     * The HAL can freely map A and B to haptic controllers, the
-     * framework shall not interpret those values and forward them
-     * from the device audio assets.
-     */
-    OUT_HAPTIC_A              = 0x20000000,
-    OUT_HAPTIC_B              = 0x10000000,
-
-    OUT_MONO     = OUT_FRONT_LEFT,
-    OUT_STEREO   = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT),
-    OUT_2POINT1  = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT | OUT_LOW_FREQUENCY),
-    OUT_2POINT0POINT2 = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
-                         OUT_TOP_SIDE_LEFT | OUT_TOP_SIDE_RIGHT),
-    OUT_2POINT1POINT2 = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
-                         OUT_TOP_SIDE_LEFT | OUT_TOP_SIDE_RIGHT |
-                         OUT_LOW_FREQUENCY),
-    OUT_3POINT0POINT2 = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT | OUT_FRONT_CENTER |
-                         OUT_TOP_SIDE_LEFT | OUT_TOP_SIDE_RIGHT),
-    OUT_3POINT1POINT2 = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT | OUT_FRONT_CENTER |
-                        OUT_TOP_SIDE_LEFT | OUT_TOP_SIDE_RIGHT |
-                        OUT_LOW_FREQUENCY),
-    OUT_QUAD     = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
-            OUT_BACK_LEFT | OUT_BACK_RIGHT),
-    OUT_QUAD_BACK = OUT_QUAD,
-    /** like OUT_QUAD_BACK with *_SIDE_* instead of *_BACK_* */
-    OUT_QUAD_SIDE = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
-            OUT_SIDE_LEFT | OUT_SIDE_RIGHT),
-    OUT_SURROUND = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
-            OUT_FRONT_CENTER | OUT_BACK_CENTER),
-    OUT_PENTA = (OUT_QUAD | OUT_FRONT_CENTER),
-    OUT_5POINT1   = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
-            OUT_FRONT_CENTER | OUT_LOW_FREQUENCY |
-            OUT_BACK_LEFT | OUT_BACK_RIGHT),
-    OUT_5POINT1_BACK = OUT_5POINT1,
-    /** like OUT_5POINT1_BACK with *_SIDE_* instead of *_BACK_* */
-    OUT_5POINT1_SIDE = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
-            OUT_FRONT_CENTER | OUT_LOW_FREQUENCY |
-            OUT_SIDE_LEFT | OUT_SIDE_RIGHT),
-    OUT_5POINT1POINT2  = (OUT_5POINT1 | OUT_TOP_SIDE_LEFT | OUT_TOP_SIDE_RIGHT),
-    OUT_5POINT1POINT4  = (OUT_5POINT1 |
-            OUT_TOP_FRONT_LEFT | OUT_TOP_FRONT_RIGHT |
-            OUT_TOP_BACK_LEFT | OUT_TOP_BACK_RIGHT),
-    OUT_6POINT1 = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
-            OUT_FRONT_CENTER | OUT_LOW_FREQUENCY |
-            OUT_BACK_LEFT | OUT_BACK_RIGHT |
-            OUT_BACK_CENTER),
-    /** matches the correct AudioFormat.CHANNEL_OUT_7POINT1_SURROUND */
-    OUT_7POINT1  = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
-            OUT_FRONT_CENTER | OUT_LOW_FREQUENCY |
-            OUT_BACK_LEFT | OUT_BACK_RIGHT |
-            OUT_SIDE_LEFT | OUT_SIDE_RIGHT),
-    OUT_7POINT1POINT2  = (OUT_7POINT1 | OUT_TOP_SIDE_LEFT | OUT_TOP_SIDE_RIGHT),
-    OUT_7POINT1POINT4  = (OUT_7POINT1 |
-            OUT_TOP_FRONT_LEFT | OUT_TOP_FRONT_RIGHT |
-            OUT_TOP_BACK_LEFT | OUT_TOP_BACK_RIGHT),
-    OUT_MONO_HAPTIC_A = (OUT_FRONT_LEFT | OUT_HAPTIC_A),
-    OUT_STEREO_HAPTIC_A = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT | OUT_HAPTIC_A),
-    OUT_HAPTIC_AB = (OUT_HAPTIC_A | OUT_HAPTIC_B),
-    OUT_MONO_HAPTIC_AB = (OUT_FRONT_LEFT | OUT_HAPTIC_A | OUT_HAPTIC_B),
-    OUT_STEREO_HAPTIC_AB = (OUT_FRONT_LEFT | OUT_FRONT_RIGHT |
-            OUT_HAPTIC_A | OUT_HAPTIC_B),
-    // Note that the 2.0 OUT_ALL* have been moved to helper functions
-
-    /* These are bits only, not complete values */
-
-    /** input channels */
-    IN_LEFT            = 0x4,
-    IN_RIGHT           = 0x8,
-    IN_FRONT           = 0x10,
-    IN_BACK            = 0x20,
-    IN_LEFT_PROCESSED  = 0x40,
-    IN_RIGHT_PROCESSED = 0x80,
-    IN_FRONT_PROCESSED = 0x100,
-    IN_BACK_PROCESSED  = 0x200,
-    IN_PRESSURE        = 0x400,
-    IN_X_AXIS          = 0x800,
-    IN_Y_AXIS          = 0x1000,
-    IN_Z_AXIS          = 0x2000,
-    IN_BACK_LEFT       = 0x10000,
-    IN_BACK_RIGHT      = 0x20000,
-    IN_CENTER          = 0x40000,
-    IN_LOW_FREQUENCY   = 0x100000,
-    IN_TOP_LEFT        = 0x200000,
-    IN_TOP_RIGHT       = 0x400000,
-
-    IN_VOICE_UPLINK    = 0x4000,
-    IN_VOICE_DNLINK    = 0x8000,
-
-    IN_MONO   = IN_FRONT,
-    IN_STEREO = (IN_LEFT | IN_RIGHT),
-    IN_FRONT_BACK = (IN_FRONT | IN_BACK),
-    IN_6 = (IN_LEFT | IN_RIGHT |
-            IN_FRONT | IN_BACK |
-            IN_LEFT_PROCESSED | IN_RIGHT_PROCESSED),
-    IN_2POINT0POINT2 = (IN_LEFT | IN_RIGHT | IN_TOP_LEFT | IN_TOP_RIGHT),
-    IN_2POINT1POINT2 = (IN_LEFT | IN_RIGHT | IN_TOP_LEFT | IN_TOP_RIGHT |
-            IN_LOW_FREQUENCY),
-    IN_3POINT0POINT2 = (IN_LEFT | IN_CENTER | IN_RIGHT | IN_TOP_LEFT | IN_TOP_RIGHT),
-    IN_3POINT1POINT2 = (IN_LEFT | IN_CENTER | IN_RIGHT |
-            IN_TOP_LEFT | IN_TOP_RIGHT | IN_LOW_FREQUENCY),
-    IN_5POINT1 = (IN_LEFT | IN_CENTER | IN_RIGHT |
-                  IN_BACK_LEFT | IN_BACK_RIGHT | IN_LOW_FREQUENCY),
-    IN_VOICE_UPLINK_MONO = (IN_VOICE_UPLINK | IN_MONO),
-    IN_VOICE_DNLINK_MONO = (IN_VOICE_DNLINK | IN_MONO),
-    IN_VOICE_CALL_MONO   = (IN_VOICE_UPLINK_MONO |
-            IN_VOICE_DNLINK_MONO),
-    // Note that the 2.0 IN_ALL* have been moved to helper functions
-
-    COUNT_MAX    = 30,
-    INDEX_HDR    = REPRESENTATION_INDEX << COUNT_MAX,
-    INDEX_MASK_1 = INDEX_HDR | ((1 << 1) - 1),
-    INDEX_MASK_2 = INDEX_HDR | ((1 << 2) - 1),
-    INDEX_MASK_3 = INDEX_HDR | ((1 << 3) - 1),
-    INDEX_MASK_4 = INDEX_HDR | ((1 << 4) - 1),
-    INDEX_MASK_5 = INDEX_HDR | ((1 << 5) - 1),
-    INDEX_MASK_6 = INDEX_HDR | ((1 << 6) - 1),
-    INDEX_MASK_7 = INDEX_HDR | ((1 << 7) - 1),
-    INDEX_MASK_8 = INDEX_HDR | ((1 << 8) - 1),
-    INDEX_MASK_9 = INDEX_HDR | ((1 << 9) - 1),
-    INDEX_MASK_10 = INDEX_HDR | ((1 << 10) - 1),
-    INDEX_MASK_11 = INDEX_HDR | ((1 << 11) - 1),
-    INDEX_MASK_12 = INDEX_HDR | ((1 << 12) - 1),
-    INDEX_MASK_13 = INDEX_HDR | ((1 << 13) - 1),
-    INDEX_MASK_14 = INDEX_HDR | ((1 << 14) - 1),
-    INDEX_MASK_15 = INDEX_HDR | ((1 << 15) - 1),
-    INDEX_MASK_16 = INDEX_HDR | ((1 << 16) - 1),
-    INDEX_MASK_17 = INDEX_HDR | ((1 << 17) - 1),
-    INDEX_MASK_18 = INDEX_HDR | ((1 << 18) - 1),
-    INDEX_MASK_19 = INDEX_HDR | ((1 << 19) - 1),
-    INDEX_MASK_20 = INDEX_HDR | ((1 << 20) - 1),
-    INDEX_MASK_21 = INDEX_HDR | ((1 << 21) - 1),
-    INDEX_MASK_22 = INDEX_HDR | ((1 << 22) - 1),
-    INDEX_MASK_23 = INDEX_HDR | ((1 << 23) - 1),
-    INDEX_MASK_24 = INDEX_HDR | ((1 << 24) - 1),
+struct AudioBasicConfig {
+    uint32_t sampleRateHz;              // 0 means 'unspecified'
+    vec<AudioChannelMask> channelMask;  // empty means 'unspecified'
+    AudioFormat format;                 // 'DEFAULT' means 'unspecified'
 };
 
 /**
@@ -607,301 +136,55 @@
     CALL_SCREEN      = 4,
 };
 
-@export(name="", value_prefix="AUDIO_DEVICE_")
-enum AudioDevice : uint32_t {
-    NONE                          = 0x0,
-    /** reserved bits */
-    BIT_IN                        = 0x80000000,
-    BIT_DEFAULT                   = 0x40000000,
-    /** output devices */
-    OUT_EARPIECE                  = 0x1,
-    OUT_SPEAKER                   = 0x2,
-    OUT_WIRED_HEADSET             = 0x4,
-    OUT_WIRED_HEADPHONE           = 0x8,
-    OUT_BLUETOOTH_SCO             = 0x10,
-    OUT_BLUETOOTH_SCO_HEADSET     = 0x20,
-    OUT_BLUETOOTH_SCO_CARKIT      = 0x40,
-    OUT_BLUETOOTH_A2DP            = 0x80,
-    OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100,
-    OUT_BLUETOOTH_A2DP_SPEAKER    = 0x200,
-    OUT_AUX_DIGITAL               = 0x400,
-    OUT_HDMI                      = OUT_AUX_DIGITAL,
-    /** uses an analog connection (multiplexed over the USB pins for instance) */
-    OUT_ANLG_DOCK_HEADSET         = 0x800,
-    OUT_DGTL_DOCK_HEADSET         = 0x1000,
-    /** USB accessory mode: Android device is USB device and dock is USB host */
-    OUT_USB_ACCESSORY             = 0x2000,
-    /** USB host mode: Android device is USB host and dock is USB device */
-    OUT_USB_DEVICE                = 0x4000,
-    OUT_REMOTE_SUBMIX             = 0x8000,
-    /** Telephony voice TX path */
-    OUT_TELEPHONY_TX              = 0x10000,
-    /** Analog jack with line impedance detected */
-    OUT_LINE                      = 0x20000,
-    /** HDMI Audio Return Channel */
-    OUT_HDMI_ARC                  = 0x40000,
-    /** S/PDIF out */
-    OUT_SPDIF                     = 0x80000,
-    /** FM transmitter out */
-    OUT_FM                        = 0x100000,
-    /** Line out for av devices */
-    OUT_AUX_LINE                  = 0x200000,
-    /** limited-output speaker device for acoustic safety */
-    OUT_SPEAKER_SAFE              = 0x400000,
-    OUT_IP                        = 0x800000,
-    /** audio bus implemented by the audio system (e.g an MOST stereo channel) */
-    OUT_BUS                       = 0x1000000,
-    OUT_PROXY                     = 0x2000000,
-    OUT_USB_HEADSET               = 0x4000000,
-    OUT_HEARING_AID               = 0x8000000,
-    OUT_ECHO_CANCELLER            = 0x10000000,
-    OUT_DEFAULT                   = BIT_DEFAULT,
-    // Note that the 2.0 OUT_ALL* have been moved to helper functions
-
-    /** input devices */
-    IN_COMMUNICATION         = BIT_IN | 0x1,
-    IN_AMBIENT               = BIT_IN | 0x2,
-    IN_BUILTIN_MIC           = BIT_IN | 0x4,
-    IN_BLUETOOTH_SCO_HEADSET = BIT_IN | 0x8,
-    IN_WIRED_HEADSET         = BIT_IN | 0x10,
-    IN_AUX_DIGITAL           = BIT_IN | 0x20,
-    IN_HDMI                  = IN_AUX_DIGITAL,
-    /** Telephony voice RX path */
-    IN_VOICE_CALL            = BIT_IN | 0x40,
-    IN_TELEPHONY_RX          = IN_VOICE_CALL,
-    IN_BACK_MIC              = BIT_IN | 0x80,
-    IN_REMOTE_SUBMIX         = BIT_IN | 0x100,
-    IN_ANLG_DOCK_HEADSET     = BIT_IN | 0x200,
-    IN_DGTL_DOCK_HEADSET     = BIT_IN | 0x400,
-    IN_USB_ACCESSORY         = BIT_IN | 0x800,
-    IN_USB_DEVICE            = BIT_IN | 0x1000,
-    /** FM tuner input */
-    IN_FM_TUNER              = BIT_IN | 0x2000,
-    /** TV tuner input */
-    IN_TV_TUNER              = BIT_IN | 0x4000,
-    /** Analog jack with line impedance detected */
-    IN_LINE                  = BIT_IN | 0x8000,
-    /** S/PDIF in */
-    IN_SPDIF                 = BIT_IN | 0x10000,
-    IN_BLUETOOTH_A2DP        = BIT_IN | 0x20000,
-    IN_LOOPBACK              = BIT_IN | 0x40000,
-    IN_IP                    = BIT_IN | 0x80000,
-    /** audio bus implemented by the audio system (e.g an MOST stereo channel) */
-    IN_BUS                   = BIT_IN | 0x100000,
-    IN_PROXY                 = BIT_IN | 0x1000000,
-    IN_USB_HEADSET           = BIT_IN | 0x2000000,
-    IN_BLUETOOTH_BLE         = BIT_IN | 0x4000000,
-    IN_ECHO_REFERENCE        = BIT_IN | 0x10000000,
-    IN_DEFAULT               = BIT_IN | BIT_DEFAULT,
-
-    // Note that the 2.0 IN_ALL* have been moved to helper functions
-};
-
-/**
- * IEEE 802 MAC address.
- */
-typedef uint8_t[6] MacAddress;
-
 /**
  * Specifies a device address in case when several devices of the same type
  * can be connected (e.g. BT A2DP, USB).
  */
 struct DeviceAddress {
-    AudioDevice device;  // discriminator
-    union Address {
-        MacAddress mac;     // used for BLUETOOTH_A2DP_*
-        uint8_t[4] ipv4;    // used for IP
+    /**
+     * Audio device specifies type (or category) of audio I/O device
+     * (e.g. speaker or headphones).
+     * See 'audioDevice' in audio_policy_configuration.xsd for the
+     * list of allowed values.
+     */
+    string deviceType;
+    safe_union Address {
+        /**
+         * The address may be left unspecified if 'device' specifies
+         * a physical device unambiguously.
+         */
+        Monostate unspecified;
+        /** IEEE 802 MAC address. Set for Bluetooth devices. */
+        uint8_t[6] mac;
+        /** IPv4 Address. Set for IPv4 devices. */
+        uint8_t[4] ipv4;
+        /** IPv6 Address. Set for IPv6 devices. */
+        uint16_t[8] ipv6;
+        /** PCI bus Address. Set for USB devices. */
         struct Alsa {
             int32_t card;
             int32_t device;
-        } alsa;             // used for USB_*
+        } alsa;
+        /** Arbitrary BUS device unique address. Not interpreted by the framework. */
+        string bus;
+        /** Arbitrary REMOTE_SUBMIX device unique address. Not interpreted by the HAL. */
+        string rSubmix;
     } address;
-    /** Arbitrary BUS device unique address. Should not be interpreted by the framework. */
-    string busAddress;
-    /** Arbitrary REMOTE_SUBMIX device unique address. Should not be interpreted by the HAL. */
-    string rSubmixAddress;
 };
 
 /**
- * The audio output flags serve two purposes:
- *
- *  - when an AudioTrack is created they indicate a "wish" to be connected to an
- *    output stream with attributes corresponding to the specified flags;
- *
- *  - when present in an output profile descriptor listed for a particular audio
- *    hardware module, they indicate that an output stream can be opened that
- *    supports the attributes indicated by the flags.
- *
- * The audio policy manager will try to match the flags in the request
- * (when getOuput() is called) to an available output stream.
+ * Audio usage specifies the intended use case for the sound being played.
+ * See 'audioUsage' in audio_policy_configuration.xsd for the
+ * list of allowed values.
  */
-@export(name="audio_output_flags_t", value_prefix="AUDIO_OUTPUT_FLAG_")
-enum AudioOutputFlag : int32_t {
-    NONE    = 0x0, // no attributes
-    DIRECT  = 0x1, // this output directly connects a track
-                   // to one output stream: no software mixer
-    PRIMARY = 0x2, // this output is the primary output of the device. It is
-                   // unique and must be present. It is opened by default and
-                   // receives routing, audio mode and volume controls related
-                   // to voice calls.
-    FAST    = 0x4,    // output supports "fast tracks", defined elsewhere
-    DEEP_BUFFER      = 0x8,   // use deep audio buffers
-    COMPRESS_OFFLOAD = 0x10,  // offload playback of compressed streams to
-                              // hardware codec
-    NON_BLOCKING     = 0x20,  // use non-blocking write
-    HW_AV_SYNC = 0x40,   // output uses a hardware A/V sync
-    TTS        = 0x80,   // output for streams transmitted through speaker at a
-                         // sample rate high enough to accommodate lower-range
-                         // ultrasonic p/b
-    RAW        = 0x100,  // minimize signal processing
-    SYNC       = 0x200,  // synchronize I/O streams
-    IEC958_NONAUDIO = 0x400, // Audio stream contains compressed audio in SPDIF
-                             // data bursts, not PCM.
-    DIRECT_PCM = 0x2000,     // Audio stream containing PCM data that needs
-                             // to pass through compress path for DSP post proc.
-    MMAP_NOIRQ = 0x4000, // output operates in MMAP no IRQ mode.
-    VOIP_RX = 0x8000,    // preferred output for VoIP calls.
-    /** preferred output for call music */
-    INCALL_MUSIC = 0x10000,
-};
+typedef string AudioUsage;
 
 /**
- * The audio input flags are analogous to audio output flags.
- * Currently they are used only when an AudioRecord is created,
- * to indicate a preference to be connected to an input stream with
- * attributes corresponding to the specified flags.
+ * Audio content type expresses the general category of the content.
+ * See 'audioContentType' in audio_policy_configuration.xsd for the
+ * list of allowed values.
  */
-@export(name="audio_input_flags_t", value_prefix="AUDIO_INPUT_FLAG_")
-enum AudioInputFlag : int32_t {
-    NONE         = 0x0,  // no attributes
-    FAST         = 0x1,  // prefer an input that supports "fast tracks"
-    HW_HOTWORD   = 0x2,  // prefer an input that captures from hw hotword source
-    RAW          = 0x4,  // minimize signal processing
-    SYNC         = 0x8,  // synchronize I/O streams
-    MMAP_NOIRQ   = 0x10, // input operates in MMAP no IRQ mode.
-    VOIP_TX      = 0x20, // preferred input for VoIP calls.
-    HW_AV_SYNC   = 0x40, // input connected to an output that uses a hardware A/V sync
-    DIRECT       = 0x80, // for acquiring encoded streams
-};
-
-@export(name="audio_usage_t", value_prefix="AUDIO_USAGE_")
-enum AudioUsage : int32_t {
-    // These values must kept in sync with
-    //  frameworks/base/media/java/android/media/AudioAttributes.java
-    // Note that not all framework values are exposed
-    /**
-     * Usage value to use when the usage is unknown.
-     */
-    UNKNOWN                            = 0,
-    /**
-     * Usage value to use when the usage is media, such as music, or movie
-     * soundtracks.
-     */
-    MEDIA                              = 1,
-    /**
-     * Usage value to use when the usage is voice communications, such as
-     * telephony or VoIP.
-     */
-    VOICE_COMMUNICATION                = 2,
-    /**
-     * Usage value to use when the usage is in-call signalling, such as with
-     * a "busy" beep, or DTMF tones.
-     */
-    VOICE_COMMUNICATION_SIGNALLING     = 3,
-    /**
-     * Usage value to use when the usage is an alarm (e.g. wake-up alarm).
-     */
-    ALARM                              = 4,
-    /**
-     * Usage value to use when the usage is a generic notification.
-     */
-    NOTIFICATION                       = 5,
-    /**
-     * Usage value to use when the usage is telephony ringtone.
-     */
-    NOTIFICATION_TELEPHONY_RINGTONE    = 6,
-    /**
-     * Usage value to use when the usage is for accessibility, such as with
-     * a screen reader.
-     */
-    ASSISTANCE_ACCESSIBILITY           = 11,
-    /**
-     * Usage value to use when the usage is driving or navigation directions.
-     */
-    ASSISTANCE_NAVIGATION_GUIDANCE     = 12,
-    /**
-     * Usage value to use when the usage is sonification, such as  with user
-     * interface sounds.
-     */
-    ASSISTANCE_SONIFICATION            = 13,
-    /**
-     * Usage value to use when the usage is for game audio.
-     */
-    GAME                               = 14,
-    /**
-     * Usage value to use when feeding audio to the platform and replacing
-     * "traditional" audio source, such as audio capture devices.
-     */
-    VIRTUAL_SOURCE                     = 15,
-    /**
-     * Usage value to use for audio responses to user queries, audio
-     * instructions or help utterances.
-     */
-    ASSISTANT                          = 16,
-    /**
-     * Usage value to use for assistant voice interaction with remote caller
-     * on Cell and VoIP calls.
-     */
-    CALL_ASSISTANT                     = 17,
-    /**
-     * Usage value to use when the usage is an emergency.
-     */
-    EMERGENCY                          = 1000,
-    /**
-     * Usage value to use when the usage is a safety sound.
-     */
-    SAFETY                             = 1001,
-    /**
-     * Usage value to use when the usage is a vehicle status.
-     */
-    VEHICLE_STATUS                     = 1002,
-    /**
-     * Usage value to use when the usage is an announcement.
-     */
-    ANNOUNCEMENT                       = 1003,
-};
-
-/** Type of audio generated by an application. */
-@export(name="audio_content_type_t", value_prefix="AUDIO_CONTENT_TYPE_")
-enum AudioContentType : uint32_t {
-    // Do not change these values without updating their counterparts
-    // in frameworks/base/media/java/android/media/AudioAttributes.java
-    /**
-     * Content type value to use when the content type is unknown, or other than
-     * the ones defined.
-     */
-    UNKNOWN      = 0,
-    /**
-     * Content type value to use when the content type is speech.
-     */
-    SPEECH       = 1,
-    /**
-     * Content type value to use when the content type is music.
-     */
-    MUSIC        = 2,
-    /**
-     * Content type value to use when the content type is a soundtrack,
-     * typically accompanying a movie or TV program.
-     */
-    MOVIE        = 3,
-    /**
-     * Content type value to use when the content type is a sound used to
-     * accompany a user action, such as a beep or sound effect expressing a key
-     * click, or event, such as the type of a sound for a bonus being received
-     * in a game. These sounds are mostly synthesized or short Foley sounds.
-     */
-    SONIFICATION = 4,
-};
+typedef string AudioContentType;
 
 /** Encapsulation mode used for sending audio compressed data. */
 @export(name="audio_encapsulation_mode_t", value_prefix="AUDIO_ENCAPSULATION_MODE_")
@@ -926,9 +209,7 @@
  * Additional information about the stream passed to hardware decoders.
  */
 struct AudioOffloadInfo {
-    uint32_t sampleRateHz;
-    bitfield<AudioChannelMask> channelMask;
-    AudioFormat format;
+    AudioBasicConfig base;
     AudioStreamType streamType;
     uint32_t bitRatePerSecond;
     int64_t durationMicroseconds;  // -1 if unknown
@@ -946,9 +227,7 @@
  * Commonly used audio stream configuration parameters.
  */
 struct AudioConfig {
-    uint32_t sampleRateHz;
-    bitfield<AudioChannelMask> channelMask;
-    AudioFormat format;
+    AudioBasicConfig base;
     AudioOffloadInfo offloadInfo;
     uint64_t frameCount;
 };
@@ -985,8 +264,7 @@
     safe_union Destination {
         Monostate unspecified;
         DeviceAddress device;
-    };
-    Destination destination;
+    } destination;
 };
 
 /** Metadatas of the sink of a StreamIn. */
@@ -994,7 +272,6 @@
     vec<RecordTrackMetadata> tracks;
 };
 
-
 /*
  *
  *  Volume control
@@ -1017,7 +294,7 @@
  */
 struct AudioGain {
     bitfield<AudioGainMode> mode;
-    bitfield<AudioChannelMask> channelMask; // channels which gain an be controlled
+    vec<AudioChannelMask> channelMask; // channels which gain an be controlled
     int32_t minValue;     // minimum gain value in millibels
     int32_t maxValue;     // maximum gain value in millibels
     int32_t defaultValue; // default gain value in millibels
@@ -1033,10 +310,8 @@
 struct AudioGainConfig {
     int32_t index;  // index of the corresponding AudioGain in AudioPort.gains
     AudioGainMode mode;
-    AudioChannelMask channelMask;  // channels which gain value follows
+    vec<AudioChannelMask> channelMask;  // channels which gain value follows
     /**
-     * 4 = sizeof(AudioChannelMask),
-     * 8 is not "FCC_8", so it won't need to be changed for > 8 channels.
      * Gain values in millibels for each channel ordered from LSb to MSb in
      * channel mask. The number of values is 1 in joint mode or
      * popcount(channel_mask).
@@ -1060,132 +335,85 @@
  * the interface.
  */
 
-/** Audio port role: either source or sink */
-@export(name="audio_port_role_t", value_prefix="AUDIO_PORT_ROLE_")
-enum AudioPortRole : int32_t {
-    NONE,
-    SOURCE,
-    SINK,
-};
-
 /**
- * Audio port type indicates if it is a session (e.g AudioTrack), a mix (e.g
- * PlaybackThread output) or a physical device (e.g OUT_SPEAKER)
+ * A helper aggregate structure providing parameters that depend on the
+ * port role.
  */
-@export(name="audio_port_type_t", value_prefix="AUDIO_PORT_TYPE_")
-enum AudioPortType : int32_t {
-    NONE,
-    DEVICE,
-    MIX,
-    SESSION,
-};
-
-/**
- * Extension for audio port configuration structure when the audio port is a
- * hardware device.
- */
-struct AudioPortConfigDeviceExt {
-    AudioModuleHandle hwModule;  // module the device is attached to
-    AudioDevice type;            // device type (e.g OUT_SPEAKER)
-    uint8_t[32] address;         // device address. "" if N/A
-};
-
-/**
- * Extension for audio port configuration structure when the audio port is an
- * audio session.
- */
-struct AudioPortConfigSessionExt {
+safe_union AudioPortExtendedInfo {
+    /** Set when no information is provided. */
+    Monostate unspecified;
+    /** Set when the audio port is an audio device. */
+    DeviceAddress device;
+    /** Set when the audio port is a mix. The handle is of a stream. */
+    struct AudioPortMixExt {
+        /** I/O handle of the input/output stream. */
+        AudioIoHandle ioHandle;
+        safe_union UseCase {
+            /** Specified when the port is in the SOURCE role. */
+            AudioStreamType stream;
+            /** Specified when the port is in the SINK role. */
+            AudioSource source;
+        } useCase;
+    } mix;
+    /** Set when the audio port is an audio session. */
     AudioSession session;
 };
 
 /**
- * Flags indicating which fields are to be considered in AudioPortConfig.
- */
-@export(name="", value_prefix="AUDIO_PORT_CONFIG_")
-enum AudioPortConfigMask : uint32_t {
-    SAMPLE_RATE = 0x1,
-    CHANNEL_MASK =  0x2,
-    FORMAT = 0x4,
-    GAIN = 0x8,
-};
-
-/**
  * Audio port configuration structure used to specify a particular configuration
  * of an audio port.
  */
 struct AudioPortConfig {
+    /**
+     * The 'id' field is set when it is needed to select the port and
+     * apply new configuration for it.
+     */
     AudioPortHandle id;
-    bitfield<AudioPortConfigMask> configMask;
-    uint32_t sampleRateHz;
-    bitfield<AudioChannelMask> channelMask;
-    AudioFormat format;
-    AudioGainConfig gain;
-    AudioPortType type;  // type is used as a discriminator for Ext union
-    AudioPortRole role;  // role is used as a discriminator for UseCase union
-    union Ext {
-        AudioPortConfigDeviceExt device;
-        struct AudioPortConfigMixExt {
-            AudioModuleHandle hwModule; // module the stream is attached to
-            AudioIoHandle ioHandle;     // I/O handle of the input/output stream
-            union UseCase {
-                AudioStreamType stream;
-                AudioSource source;
-            } useCase;
-        } mix;
-        AudioPortConfigSessionExt session;
-    } ext;
+    /**
+     * Basic parameters: sampling rate, format, channel mask. Only some of the
+     * parameters (or none) may be set. See the documentation of the
+     * AudioBasicConfig struct.
+     */
+    AudioBasicConfig config;
+    /** Associated gain control. */
+    safe_union OptionalGain {
+        Monostate unspecified;
+        AudioGainConfig config;
+    } gain;
+    /** Parameters that depend on the actual port role. */
+    AudioPortExtendedInfo ext;
 };
 
 /**
- * Extension for audio port structure when the audio port is a hardware device.
+ * Audio port structure describes the capabilities of an audio port
+ * as well as its current configuration.
  */
-struct AudioPortDeviceExt {
-    AudioModuleHandle hwModule;    // module the device is attached to
-    AudioDevice type;
-    /** 32 byte string identifying the port. */
-    uint8_t[32] address;
-};
-
-/**
- * Latency class of the audio mix.
- */
-@export(name="audio_mix_latency_class_t", value_prefix="AUDIO_LATENCY_")
-enum AudioMixLatencyClass : int32_t {
-    LOW,
-    NORMAL
-};
-
-struct AudioPortMixExt {
-    AudioModuleHandle hwModule;     // module the stream is attached to
-    AudioIoHandle ioHandle;         // I/O handle of the stream
-    AudioMixLatencyClass latencyClass;
-};
-
-/**
- * Extension for audio port structure when the audio port is an audio session.
- */
-struct AudioPortSessionExt {
-    AudioSession session;
-};
-
 struct AudioPort {
+    /**
+     * Unique identifier of the port within this HAL service. When calling
+     * from the client side functions like IDevice.getAudioPort is it allowed
+     * to only specify the 'id' and leave the other fields unspecified.
+     */
     AudioPortHandle id;
-    AudioPortRole role;
+    /**
+     * Human-readable name describing the function of the port.
+     * E.g. "telephony_tx" or "fm_tuner".
+     */
     string name;
-    vec<uint32_t> sampleRates;
-    vec<bitfield<AudioChannelMask>> channelMasks;
-    vec<AudioFormat> formats;
+    /** List of audio profiles supported by the port. */
+    struct AudioProfile {
+        AudioFormat format;
+        /** List of the sample rates supported by the profile. */
+        vec<uint32_t> sampleRates;
+        /** List of channel masks supported by the profile. */
+        vec<AudioChannelMask> channelMasks;
+    };
+    vec<AudioProfile> profiles;
+    /** List of gain controls attached to the port. */
     vec<AudioGain> gains;
-    AudioPortConfig activeConfig; // current audio port configuration
-    AudioPortType type;  // type is used as a discriminator
-    union Ext {
-        AudioPortDeviceExt device;
-        AudioPortMixExt mix;
-        AudioPortSessionExt session;
-    } ext;
-};
-
-struct ThreadInfo {
-    int64_t pid;
-    int64_t tid;
+    /**
+     * Current configuration of the audio port, may have all the fields left
+     * unspecified.
+     */
+    AudioPortConfig activeConfig;
 };
diff --git a/audio/effect/7.0/IEffect.hal b/audio/effect/7.0/IEffect.hal
index 5b176dc..aa94f6d 100644
--- a/audio/effect/7.0/IEffect.hal
+++ b/audio/effect/7.0/IEffect.hal
@@ -56,7 +56,6 @@
      *
      * @return retval operation completion status.
      */
-    @callflow(next={"prepareForProcessing"})
     enable() generates (Result retval);
 
     /**
@@ -64,7 +63,6 @@
      *
      * @return retval operation completion status.
      */
-    @callflow(next={"close"})
     disable() generates (Result retval);
 
     /**
@@ -78,7 +76,7 @@
      * @param device output device specification.
      * @return retval operation completion status.
      */
-    setDevice(bitfield<AudioDevice> device) generates (Result retval);
+    setDevice(DeviceAddress device) generates (Result retval);
 
     /**
      * Set and get volume. Used by audio framework to delegate volume control to
@@ -147,7 +145,7 @@
      * @param device input device specification.
      * @return retval operation completion status.
      */
-    setInputDevice(bitfield<AudioDevice> device) generates (Result retval);
+    setInputDevice(DeviceAddress device) generates (Result retval);
 
     /**
      * Read audio parameters configurations for input and output buffers.
@@ -251,7 +249,6 @@
      *                                  the queue.
      * @return statusMQ a message queue used for passing status from the effect.
      */
-    @callflow(next={"setProcessBuffers"})
     prepareForProcessing() generates (Result retval, fmq_sync<Result> statusMQ);
 
     /**
@@ -416,6 +413,5 @@
      * @return retval OK in case the success.
      *                INVALID_STATE if the effect was already closed.
      */
-    @exit
     close() generates (Result retval);
 };
diff --git a/audio/effect/7.0/IVirtualizerEffect.hal b/audio/effect/7.0/IVirtualizerEffect.hal
index 0e6ff54..141b4e6 100644
--- a/audio/effect/7.0/IVirtualizerEffect.hal
+++ b/audio/effect/7.0/IVirtualizerEffect.hal
@@ -48,7 +48,7 @@
 
     struct SpeakerAngle {
         /** Speaker channel mask */
-        bitfield<AudioChannelMask> mask;
+        vec<AudioChannelMask> mask;
         // all angles are expressed in degrees and
         // are relative to the listener.
         int16_t azimuth; // 0 is the direction the listener faces
@@ -61,17 +61,17 @@
      * Retrieves virtual speaker angles for the given channel mask on the
      * specified device.
      */
-    getVirtualSpeakerAngles(bitfield<AudioChannelMask> mask, AudioDevice device)
+    getVirtualSpeakerAngles(vec<AudioChannelMask> mask, DeviceAddress device)
             generates (Result retval, vec<SpeakerAngle> speakerAngles);
 
     /**
      * Forces the virtualizer effect for the given output device.
      */
-    forceVirtualizationMode(AudioDevice device) generates (Result retval);
+    forceVirtualizationMode(DeviceAddress device) generates (Result retval);
 
     /**
      * Returns audio device reflecting the current virtualization mode,
-     * AUDIO_DEVICE_NONE when not virtualizing.
+     * Device type can be empty when not virtualizing.
      */
-    getVirtualizationMode() generates (Result retval, AudioDevice device);
+    getVirtualizationMode() generates (Result retval, DeviceAddress device);
 };
diff --git a/audio/effect/7.0/types.hal b/audio/effect/7.0/types.hal
index 7f5a382..fe4ee51 100644
--- a/audio/effect/7.0/types.hal
+++ b/audio/effect/7.0/types.hal
@@ -257,7 +257,7 @@
 struct EffectBufferConfig {
     AudioBuffer buffer;
     uint32_t samplingRateHz;
-    bitfield<AudioChannelMask> channels;
+    AudioChannelMask channels;
     AudioFormat format;
     EffectBufferAccess accessMode;
     bitfield<EffectConfigParameters> mask;
@@ -276,8 +276,8 @@
 };
 
 struct EffectAuxChannelsConfig {
-    bitfield<AudioChannelMask> mainChannels;  // channel mask for main channels
-    bitfield<AudioChannelMask> auxChannels;   // channel mask for auxiliary channels
+    vec<AudioChannelMask> mainChannels;  // channel mask for main channels
+    vec<AudioChannelMask> auxChannels;   // channel mask for auxiliary channels
 };
 
 struct EffectOffloadParameter {