diff --git a/audio/aidl/Android.bp b/audio/aidl/Android.bp
index 42a4ac1..340cdb5 100644
--- a/audio/aidl/Android.bp
+++ b/audio/aidl/Android.bp
@@ -119,6 +119,7 @@
         "android/hardware/audio/core/IStreamCommon.aidl",
         "android/hardware/audio/core/IStreamIn.aidl",
         "android/hardware/audio/core/IStreamOut.aidl",
+        "android/hardware/audio/core/IStreamOutEventCallback.aidl",
         "android/hardware/audio/core/ITelephony.aidl",
         "android/hardware/audio/core/MicrophoneDynamicInfo.aidl",
         "android/hardware/audio/core/MicrophoneInfo.aidl",
diff --git a/audio/aidl/aidl_api/android.hardware.audio.core/current/android/hardware/audio/core/IModule.aidl b/audio/aidl/aidl_api/android.hardware.audio.core/current/android/hardware/audio/core/IModule.aidl
index dd2279d..6a10295 100644
--- a/audio/aidl/aidl_api/android.hardware.audio.core/current/android/hardware/audio/core/IModule.aidl
+++ b/audio/aidl/aidl_api/android.hardware.audio.core/current/android/hardware/audio/core/IModule.aidl
@@ -46,6 +46,7 @@
   android.hardware.audio.core.AudioRoute[] getAudioRoutesForAudioPort(int portId);
   android.hardware.audio.core.IModule.OpenInputStreamReturn openInputStream(in android.hardware.audio.core.IModule.OpenInputStreamArguments args);
   android.hardware.audio.core.IModule.OpenOutputStreamReturn openOutputStream(in android.hardware.audio.core.IModule.OpenOutputStreamArguments args);
+  android.hardware.audio.core.IModule.SupportedPlaybackRateFactors getSupportedPlaybackRateFactors();
   android.hardware.audio.core.AudioPatch setAudioPatch(in android.hardware.audio.core.AudioPatch requested);
   boolean setAudioPortConfig(in android.media.audio.common.AudioPortConfig requested, out android.media.audio.common.AudioPortConfig suggested);
   void resetAudioPatch(int patchId);
@@ -84,12 +85,20 @@
     @nullable android.media.audio.common.AudioOffloadInfo offloadInfo;
     long bufferSizeFrames;
     @nullable android.hardware.audio.core.IStreamCallback callback;
+    @nullable android.hardware.audio.core.IStreamOutEventCallback eventCallback;
   }
   @VintfStability
   parcelable OpenOutputStreamReturn {
     android.hardware.audio.core.IStreamOut stream;
     android.hardware.audio.core.StreamDescriptor desc;
   }
+  @VintfStability
+  parcelable SupportedPlaybackRateFactors {
+    float minSpeed;
+    float maxSpeed;
+    float minPitch;
+    float maxPitch;
+  }
   @Backing(type="int") @VintfStability
   enum ScreenRotation {
     DEG_0 = 0,
diff --git a/audio/aidl/aidl_api/android.hardware.audio.core/current/android/hardware/audio/core/IStreamOut.aidl b/audio/aidl/aidl_api/android.hardware.audio.core/current/android/hardware/audio/core/IStreamOut.aidl
index 092b801..46acc11 100644
--- a/audio/aidl/aidl_api/android.hardware.audio.core/current/android/hardware/audio/core/IStreamOut.aidl
+++ b/audio/aidl/aidl_api/android.hardware.audio.core/current/android/hardware/audio/core/IStreamOut.aidl
@@ -38,6 +38,16 @@
   void updateMetadata(in android.hardware.audio.common.SourceMetadata sourceMetadata);
   float[] getHwVolume();
   void setHwVolume(in float[] channelVolumes);
+  float getAudioDescriptionMixLevel();
+  void setAudioDescriptionMixLevel(float leveldB);
+  android.media.audio.common.AudioDualMonoMode getDualMonoMode();
+  void setDualMonoMode(android.media.audio.common.AudioDualMonoMode mode);
+  android.media.audio.common.AudioLatencyMode[] getRecommendedLatencyModes();
+  void setLatencyMode(android.media.audio.common.AudioLatencyMode mode);
+  android.media.audio.common.AudioPlaybackRate getPlaybackRateParameters();
+  void setPlaybackRateParameters(in android.media.audio.common.AudioPlaybackRate playbackRate);
+  void selectPresentation(int presentationId, int programId);
   const int HW_VOLUME_MIN = 0;
   const int HW_VOLUME_MAX = 1;
+  const int AUDIO_DESCRIPTION_MIX_LEVEL_MAX = 48;
 }
diff --git a/audio/aidl/aidl_api/android.hardware.audio.core/current/android/hardware/audio/core/IStreamOutEventCallback.aidl b/audio/aidl/aidl_api/android.hardware.audio.core/current/android/hardware/audio/core/IStreamOutEventCallback.aidl
new file mode 100644
index 0000000..31cf0b7
--- /dev/null
+++ b/audio/aidl/aidl_api/android.hardware.audio.core/current/android/hardware/audio/core/IStreamOutEventCallback.aidl
@@ -0,0 +1,39 @@
+/*
+ * Copyright (C) 2022 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 FILE IS IMMUTABLE. DO NOT EDIT IN ANY CASE.                          //
+///////////////////////////////////////////////////////////////////////////////
+
+// This file is a snapshot of an AIDL file. Do not edit it manually. There are
+// two cases:
+// 1). this is a frozen version file - do not edit this in any case.
+// 2). this is a 'current' file. If you make a backwards compatible change to
+//     the interface (from the latest frozen version), the build system will
+//     prompt you to update this file with `m <name>-update-api`.
+//
+// You must not make a backward incompatible change to any AIDL file built
+// with the aidl_interface module type with versions property set. The module
+// type is used to build AIDL files in a way that they can be used across
+// independently updatable components of the system. If a device is shipped
+// with such a backward incompatible change, it has a high risk of breaking
+// later when a module using the interface is updated, e.g., Mainline modules.
+
+package android.hardware.audio.core;
+@VintfStability
+interface IStreamOutEventCallback {
+  oneway void onCodecFormatChanged(in byte[] audioMetadata);
+  oneway void onRecommendedLatencyModeChanged(in android.media.audio.common.AudioLatencyMode[] modes);
+}
diff --git a/audio/aidl/android/hardware/audio/core/IModule.aidl b/audio/aidl/android/hardware/audio/core/IModule.aidl
index b278ac4..f4ee1f3 100644
--- a/audio/aidl/android/hardware/audio/core/IModule.aidl
+++ b/audio/aidl/android/hardware/audio/core/IModule.aidl
@@ -25,6 +25,7 @@
 import android.hardware.audio.core.IStreamCallback;
 import android.hardware.audio.core.IStreamIn;
 import android.hardware.audio.core.IStreamOut;
+import android.hardware.audio.core.IStreamOutEventCallback;
 import android.hardware.audio.core.ITelephony;
 import android.hardware.audio.core.MicrophoneInfo;
 import android.hardware.audio.core.ModuleDebug;
@@ -34,6 +35,7 @@
 import android.media.audio.common.AudioOffloadInfo;
 import android.media.audio.common.AudioPort;
 import android.media.audio.common.AudioPortConfig;
+import android.media.audio.common.Float;
 
 /**
  * Each instance of IModule corresponds to a separate audio module. The system
@@ -389,6 +391,8 @@
         long bufferSizeFrames;
         /** Client callback interface for the non-blocking output mode. */
         @nullable IStreamCallback callback;
+        /** Optional callback to notify client about stream events. */
+        @nullable IStreamOutEventCallback eventCallback;
     }
     @VintfStability
     parcelable OpenOutputStreamReturn {
@@ -398,6 +402,33 @@
     OpenOutputStreamReturn openOutputStream(in OpenOutputStreamArguments args);
 
     /**
+     * Get supported ranges of playback rate factors.
+     *
+     * See 'PlaybackRate' for the information on the playback rate parameters.
+     * This method provides supported ranges (inclusive) for the speed factor
+     * and the pitch factor.
+     *
+     * If the HAL module supports setting the playback rate, it is recommended
+     * to support speed and pitch factor values at least in the range from 0.5f
+     * to 2.0f.
+     *
+     * @throws EX_UNSUPPORTED_OPERATION If setting of playback rate parameters
+     *                                  is not supported by the module.
+     */
+    @VintfStability
+    parcelable SupportedPlaybackRateFactors {
+        /** The minimum allowed speed factor. */
+        float minSpeed;
+        /** The maximum allowed speed factor. */
+        float maxSpeed;
+        /** The minimum allowed pitch factor. */
+        float minPitch;
+        /** The maximum allowed pitch factor. */
+        float maxPitch;
+    }
+    SupportedPlaybackRateFactors getSupportedPlaybackRateFactors();
+
+    /**
      * Set an audio patch.
      *
      * This method creates new or updates an existing audio patch. If the
diff --git a/audio/aidl/android/hardware/audio/core/IStreamOut.aidl b/audio/aidl/android/hardware/audio/core/IStreamOut.aidl
index 85da00d..b60b0fd 100644
--- a/audio/aidl/android/hardware/audio/core/IStreamOut.aidl
+++ b/audio/aidl/android/hardware/audio/core/IStreamOut.aidl
@@ -18,6 +18,9 @@
 
 import android.hardware.audio.common.SourceMetadata;
 import android.hardware.audio.core.IStreamCommon;
+import android.media.audio.common.AudioDualMonoMode;
+import android.media.audio.common.AudioLatencyMode;
+import android.media.audio.common.AudioPlaybackRate;
 
 /**
  * This interface provides means for sending audio data to output devices.
@@ -86,4 +89,140 @@
      * @throws EX_UNSUPPORTED_OPERATION If hardware volume control is not supported.
      */
     void setHwVolume(in float[] channelVolumes);
+
+    // aidl: Constant of type float is not supported (b/251286924).
+    // const float AUDIO_DESCRIPTION_MIX_LEVEL_MIN = -Inf;
+    const int AUDIO_DESCRIPTION_MIX_LEVEL_MAX = 48;
+    /**
+     * Returns the Audio Description Mix level in dB.
+     *
+     * The level is applied to streams incorporating a secondary Audio
+     * Description stream. It specifies the relative level of mixing for
+     * the Audio Description with a reference to the Main Audio.
+     *
+     * The value of the relative level is in the range from negative infinity
+     * to +48, see AUDIO_DESCRIPTION_MIX_LEVEL_* constants.
+     *
+     * @return The current Audio Description Mix Level in dB.
+     * @throws EX_ILLEGAL_STATE If the stream is closed.
+     * @throws EX_UNSUPPORTED_OPERATION If the information is unavailable.
+     */
+    float getAudioDescriptionMixLevel();
+    /**
+     * Sets the Audio Description Mix level in dB.
+     *
+     * For streams incorporating a secondary Audio Description stream the
+     * relative level of mixing of the Audio Description to the Main Audio is
+     * controlled by this method.
+     *
+     * The value of the relative level must be in the range from negative
+     * infinity to +48, see AUDIO_DESCRIPTION_MIX_LEVEL_* constants.
+     *
+     * @param leveldB Audio Description Mix Level in dB.
+     * @throws EX_ILLEGAL_ARGUMENT If the provided value is out of range.
+     * @throws EX_ILLEGAL_STATE If the stream is closed.
+     * @throws EX_UNSUPPORTED_OPERATION If setting of this parameter is not supported.
+     */
+    void setAudioDescriptionMixLevel(float leveldB);
+
+    /**
+     * Returns the Dual Mono mode presentation setting.
+     *
+     * @return The current setting of Dual Mono mode.
+     * @throws EX_ILLEGAL_STATE If the stream is closed.
+     * @throws EX_UNSUPPORTED_OPERATION If the information is unavailable.
+     */
+    AudioDualMonoMode getDualMonoMode();
+    /**
+     * Sets the Dual Mono mode presentation on the output device.
+     *
+     * The Dual Mono mode is generally applied to stereo audio streams
+     * where the left and right channels come from separate sources.
+     *
+     * @param mode Selected Dual Mono mode.
+     * @throws EX_ILLEGAL_STATE If the stream is closed.
+     * @throws EX_UNSUPPORTED_OPERATION If setting of this parameter is not supported.
+     */
+    void setDualMonoMode(AudioDualMonoMode mode);
+
+    /**
+     * Retrieve supported latency modes.
+     *
+     * Indicates which latency modes are currently supported on this output
+     * stream. If the transport protocol (for example, Bluetooth A2DP) used by
+     * this output stream to reach the output device supports variable latency
+     * modes, the HAL indicates which modes are currently supported. The client
+     * can then call setLatencyMode() with one of the supported modes to select
+     * the desired operation mode.
+     *
+     * Implementation for this method is mandatory only on specific spatial
+     * audio streams indicated by AUDIO_OUTPUT_FLAG_SPATIALIZER flag if they can
+     * be routed to a BT classic sink.
+     *
+     * @return Currently supported latency modes.
+     * @throws EX_ILLEGAL_STATE If the stream is closed.
+     * @throws EX_UNSUPPORTED_OPERATION If the information is unavailable.
+     */
+    AudioLatencyMode[] getRecommendedLatencyModes();
+    /**
+     * Sets the latency mode.
+     *
+     * The requested mode must be one of the modes returned by the
+     * 'getRecommendedLatencyModes()' method.
+     *
+     * Implementation for this method is mandatory only on specific spatial
+     * audio streams indicated by AUDIO_OUTPUT_FLAG_SPATIALIZER flag if they can
+     * be routed to a BT classic sink.
+     *
+     * @throws EX_ILLEGAL_ARGUMENT If the specified mode is not supported.
+     * @throws EX_ILLEGAL_STATE If the stream is closed.
+     * @throws EX_UNSUPPORTED_OPERATION If setting of this parameter is not supported.
+     */
+    void setLatencyMode(AudioLatencyMode mode);
+
+    /**
+     * Retrieve current playback rate parameters.
+     *
+     * @return Current playback parameters.
+     * @throws EX_ILLEGAL_STATE If the stream is closed.
+     * @throws EX_UNSUPPORTED_OPERATION If the information is unavailable.
+     */
+    AudioPlaybackRate getPlaybackRateParameters();
+    /**
+     * Set playback rate parameters.
+     *
+     * Sets the playback rate parameters that control playback behavior. This
+     * is normally used when playing encoded content and decoding is performed
+     * in hardware. Otherwise, the client can apply necessary transformations
+     * itself.
+     *
+     * The range of supported values for speed and pitch factors is provided by
+     * the 'IModule.getSupportedPlaybackRateFactors' method. Out of range speed
+     * and pitch values must not be rejected if the fallback mode is 'MUTE'.
+     *
+     * @param playbackRate Playback parameters to set.
+     * @throws EX_ILLEGAL_ARGUMENT If provided parameters are out of acceptable range.
+     * @throws EX_ILLEGAL_STATE If the stream is closed.
+     * @throws EX_UNSUPPORTED_OPERATION If setting playback rate parameters
+     *                                  is not supported.
+     */
+    void setPlaybackRateParameters(in AudioPlaybackRate playbackRate);
+
+    /**
+     * Select presentation and program from for decoding.
+     *
+     * Selects a presentation for decoding from a next generation media stream
+     * (as defined per ETSI TS 103 190-2) and a program within the presentation.
+     * The client must obtain valid presentation and program IDs from the media
+     * stream on its own.
+     *
+     * @param presentationId Selected audio presentation.
+     * @param programId Refinement for the presentation.
+     * @throws EX_ILLEGAL_ARGUMENT If the HAL module is unable to locate
+     *                             the specified presentation or program in
+     *                             the media stream.
+     * @throws EX_ILLEGAL_STATE If the stream is closed.
+     * @throws EX_UNSUPPORTED_OPERATION If presentation selection is not supported.
+     */
+    void selectPresentation(int presentationId, int programId);
 }
diff --git a/audio/aidl/android/hardware/audio/core/IStreamOutEventCallback.aidl b/audio/aidl/android/hardware/audio/core/IStreamOutEventCallback.aidl
new file mode 100644
index 0000000..75d7385
--- /dev/null
+++ b/audio/aidl/android/hardware/audio/core/IStreamOutEventCallback.aidl
@@ -0,0 +1,168 @@
+/*
+ * Copyright (C) 2022 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.
+ */
+
+package android.hardware.audio.core;
+
+import android.media.audio.common.AudioLatencyMode;
+
+/**
+ * This interface provides means for asynchronous notification of the client
+ * by an output stream.
+ */
+@VintfStability
+oneway interface IStreamOutEventCallback {
+    /**
+     * Codec format changed notification.
+     *
+     * onCodecFormatChanged returns an AudioMetadata object in read-only
+     * ByteString format.  It represents the most recent codec format decoded by
+     * a HW audio decoder.
+     *
+     * Codec format is an optional message from HW audio decoders. It serves to
+     * notify the application about the codec format and audio objects contained
+     * within the compressed audio stream for control, informational,
+     * and display purposes.
+     *
+     * audioMetadata ByteString is convertible to an AudioMetadata object
+     * through both a C++ and a C API present in Metadata.h [1], or through a
+     * Java API present in AudioMetadata.java [2].
+     *
+     * The ByteString format is a stable format used for parcelling
+     * (marshalling) across JNI, AIDL, and HIDL interfaces.  The test for R
+     * compatibility for native marshalling is TEST(metadata_tests,
+     * compatibility_R) [3]. The test for R compatibility for JNI marshalling
+     * is android.media.cts.AudioMetadataTest#testCompatibilityR [4].
+     *
+     * Android R defined keys are as follows [2]:
+     * "bitrate", int32
+     * "channel-mask", int32
+     * "mime", string
+     * "sample-rate", int32
+     * "bit-width", int32
+     * "has-atmos", int32
+     * "audio-encoding", int32
+     *
+     * Android S in addition adds the following keys:
+     * "presentation-id", int32
+     * "program-id", int32
+     * "presentation-content-classifier", int32
+     *    presentation-content-classifier key values can be referenced from
+     *    frameworks/base/media/java/android/media/AudioPresentation.java
+     *    i.e. AudioPresentation.ContentClassifier
+     *    It can contain any of the below values
+     *    CONTENT_UNKNOWN   = -1,
+     *    CONTENT_MAIN      =  0,
+     *    CONTENT_MUSIC_AND_EFFECTS = 1,
+     *    CONTENT_VISUALLY_IMPAIRED = 2,
+     *    CONTENT_HEARING_IMPAIRED  = 3,
+     *    CONTENT_DIALOG = 4,
+     *    CONTENT_COMMENTARY = 5,
+     *    CONTENT_EMERGENCY = 6,
+     *    CONTENT_VOICEOVER = 7
+     * "presentation-language", string  // represents ISO 639-2 (three letter code)
+     *
+     * Parceling Format:
+     * All values are native endian order. [1]
+     *
+     * using type_size_t = uint32_t;
+     * using index_size_t = uint32_t;
+     * using datum_size_t = uint32_t;
+     *
+     * Permitted type indexes are
+     * TYPE_NONE = 0, // Reserved
+     * TYPE_INT32 = 1,
+     * TYPE_INT64 = 2,
+     * TYPE_FLOAT = 3,
+     * TYPE_DOUBLE = 4,
+     * TYPE_STRING = 5,
+     * TYPE_DATA = 6,  // A data table of <String, Datum>
+     *
+     * Datum = {
+     *           (type_size_t)  Type (the type index from type_as_value<T>.)
+     *           (datum_size_t) Size (size of the Payload)
+     *           (byte string)  Payload<Type>
+     *         }
+     *
+     * The data is specified in native endian order. Since the size of the
+     * Payload is always present, unknown types may be skipped.
+     *
+     * Payload<Fixed-size Primitive_Value>
+     * [ sizeof(Primitive_Value) in raw bytes ]
+     *
+     * Example of Payload<Int32> of 123:
+     * Payload<Int32>
+     * [ value of 123                   ] =  0x7b 0x00 0x00 0x00       123
+     *
+     * Payload<String>
+     * [ (index_size_t) length, not including zero terminator.]
+     * [ (length) raw bytes ]
+     *
+     * Example of Payload<String> of std::string("hi"):
+     * [ (index_size_t) length          ] = 0x02 0x00 0x00 0x00        2 strlen("hi")
+     * [ raw bytes "hi"                 ] = 0x68 0x69                  "hi"
+     *
+     * Payload<Data>
+     * [ (index_size_t) entries ]
+     * [ raw bytes   (entry 1) Key   (Payload<String>)
+     *                         Value (Datum)
+     *                ...  (until #entries) ]
+     *
+     * Example of Payload<Data> of {{"hello", "world"},
+     *                              {"value", (int32_t)1000}};
+     * [ (index_size_t) #entries        ] = 0x02 0x00 0x00 0x00        2 entries
+     *    Key (Payload<String>)
+     *    [ index_size_t length         ] = 0x05 0x00 0x00 0x00        5 strlen("hello")
+     *    [ raw bytes "hello"           ] = 0x68 0x65 0x6c 0x6c 0x6f   "hello"
+     *    Value (Datum)
+     *    [ (type_size_t) type          ] = 0x05 0x00 0x00 0x00        5 (TYPE_STRING)
+     *    [ (datum_size_t) size         ] = 0x09 0x00 0x00 0x00        sizeof(index_size_t) +
+     *                                                                 strlen("world")
+     *       Payload<String>
+     *       [ (index_size_t) length    ] = 0x05 0x00 0x00 0x00        5 strlen("world")
+     *       [ raw bytes "world"        ] = 0x77 0x6f 0x72 0x6c 0x64   "world"
+     *    Key (Payload<String>)
+     *    [ index_size_t length         ] = 0x05 0x00 0x00 0x00        5 strlen("value")
+     *    [ raw bytes "value"           ] = 0x76 0x61 0x6c 0x75 0x65   "value"
+     *    Value (Datum)
+     *    [ (type_size_t) type          ] = 0x01 0x00 0x00 0x00        1 (TYPE_INT32)
+     *    [ (datum_size_t) size         ] = 0x04 0x00 0x00 0x00        4 sizeof(int32_t)
+     *        Payload<Int32>
+     *        [ raw bytes 1000          ] = 0xe8 0x03 0x00 0x00        1000
+     *
+     * The contents of audioMetadata is a Payload<Data>.
+     * An implementation dependent detail is that the Keys are always
+     * stored sorted, so the byte string representation generated is unique.
+     *
+     * Vendor keys are allowed for informational and debugging purposes.
+     * Vendor keys should consist of the vendor company name followed
+     * by a dot; for example, "vendorCompany.someVolume" [2].
+     *
+     * [1] system/media/audio_utils/include/audio_utils/Metadata.h
+     * [2] frameworks/base/media/java/android/media/AudioMetadata.java
+     * [3] system/media/audio_utils/tests/metadata_tests.cpp
+     * [4] cts/tests/tests/media/src/android/media/cts/AudioMetadataTest.java
+     *
+     * @param audioMetadata A buffer containing decoded format changes
+     *     reported by codec. The buffer contains data that can be transformed
+     *     to audio metadata, which is a C++ object based map.
+     */
+    void onCodecFormatChanged(in byte[] audioMetadata);
+
+    /**
+     * Called with the new list of supported latency modes when a change occurs.
+     */
+    void onRecommendedLatencyModeChanged(in AudioLatencyMode[] modes);
+}
diff --git a/audio/aidl/default/Module.cpp b/audio/aidl/default/Module.cpp
index 4a424bd..6a833c4 100644
--- a/audio/aidl/default/Module.cpp
+++ b/audio/aidl/default/Module.cpp
@@ -603,6 +603,13 @@
     return ndk::ScopedAStatus::ok();
 }
 
+ndk::ScopedAStatus Module::getSupportedPlaybackRateFactors(
+        SupportedPlaybackRateFactors* _aidl_return) {
+    LOG(DEBUG) << __func__;
+    (void)_aidl_return;
+    return ndk::ScopedAStatus::fromExceptionCode(EX_UNSUPPORTED_OPERATION);
+}
+
 ndk::ScopedAStatus Module::setAudioPatch(const AudioPatch& in_requested, AudioPatch* _aidl_return) {
     LOG(DEBUG) << __func__ << ": requested patch " << in_requested.toString();
     if (in_requested.sourcePortConfigIds.empty()) {
diff --git a/audio/aidl/default/Stream.cpp b/audio/aidl/default/Stream.cpp
index bb123a2..a490a2a 100644
--- a/audio/aidl/default/Stream.cpp
+++ b/audio/aidl/default/Stream.cpp
@@ -27,7 +27,10 @@
 using aidl::android::hardware::audio::common::SinkMetadata;
 using aidl::android::hardware::audio::common::SourceMetadata;
 using aidl::android::media::audio::common::AudioDevice;
+using aidl::android::media::audio::common::AudioDualMonoMode;
+using aidl::android::media::audio::common::AudioLatencyMode;
 using aidl::android::media::audio::common::AudioOffloadInfo;
+using aidl::android::media::audio::common::AudioPlaybackRate;
 using android::hardware::audio::common::getChannelCount;
 using android::hardware::audio::common::getFrameSizeInBytes;
 
@@ -724,4 +727,55 @@
     return ndk::ScopedAStatus::fromExceptionCode(EX_UNSUPPORTED_OPERATION);
 }
 
+ndk::ScopedAStatus StreamOut::getAudioDescriptionMixLevel(float* _aidl_return) {
+    LOG(DEBUG) << __func__;
+    (void)_aidl_return;
+    return ndk::ScopedAStatus::fromExceptionCode(EX_UNSUPPORTED_OPERATION);
+}
+
+ndk::ScopedAStatus StreamOut::setAudioDescriptionMixLevel(float in_leveldB) {
+    LOG(DEBUG) << __func__ << ": description mix level " << in_leveldB;
+    return ndk::ScopedAStatus::fromExceptionCode(EX_UNSUPPORTED_OPERATION);
+}
+
+ndk::ScopedAStatus StreamOut::getDualMonoMode(AudioDualMonoMode* _aidl_return) {
+    LOG(DEBUG) << __func__;
+    (void)_aidl_return;
+    return ndk::ScopedAStatus::fromExceptionCode(EX_UNSUPPORTED_OPERATION);
+}
+
+ndk::ScopedAStatus StreamOut::setDualMonoMode(AudioDualMonoMode in_mode) {
+    LOG(DEBUG) << __func__ << ": dual mono mode " << toString(in_mode);
+    return ndk::ScopedAStatus::fromExceptionCode(EX_UNSUPPORTED_OPERATION);
+}
+
+ndk::ScopedAStatus StreamOut::getRecommendedLatencyModes(
+        std::vector<AudioLatencyMode>* _aidl_return) {
+    LOG(DEBUG) << __func__;
+    (void)_aidl_return;
+    return ndk::ScopedAStatus::fromExceptionCode(EX_UNSUPPORTED_OPERATION);
+}
+
+ndk::ScopedAStatus StreamOut::setLatencyMode(AudioLatencyMode in_mode) {
+    LOG(DEBUG) << __func__ << ": latency mode " << toString(in_mode);
+    return ndk::ScopedAStatus::fromExceptionCode(EX_UNSUPPORTED_OPERATION);
+}
+
+ndk::ScopedAStatus StreamOut::getPlaybackRateParameters(AudioPlaybackRate* _aidl_return) {
+    LOG(DEBUG) << __func__;
+    (void)_aidl_return;
+    return ndk::ScopedAStatus::fromExceptionCode(EX_UNSUPPORTED_OPERATION);
+}
+
+ndk::ScopedAStatus StreamOut::setPlaybackRateParameters(const AudioPlaybackRate& in_playbackRate) {
+    LOG(DEBUG) << __func__ << ": " << in_playbackRate.toString();
+    return ndk::ScopedAStatus::fromExceptionCode(EX_UNSUPPORTED_OPERATION);
+}
+
+ndk::ScopedAStatus StreamOut::selectPresentation(int32_t in_presentationId, int32_t in_programId) {
+    LOG(DEBUG) << __func__ << ": presentationId " << in_presentationId << ", programId "
+               << in_programId;
+    return ndk::ScopedAStatus::fromExceptionCode(EX_UNSUPPORTED_OPERATION);
+}
+
 }  // namespace aidl::android::hardware::audio::core
diff --git a/audio/aidl/default/include/core-impl/Module.h b/audio/aidl/default/include/core-impl/Module.h
index aa05d2a..ff45f81 100644
--- a/audio/aidl/default/include/core-impl/Module.h
+++ b/audio/aidl/default/include/core-impl/Module.h
@@ -66,6 +66,8 @@
                     in_args,
             ::aidl::android::hardware::audio::core::IModule::OpenOutputStreamReturn* _aidl_return)
             override;
+    ndk::ScopedAStatus getSupportedPlaybackRateFactors(
+            SupportedPlaybackRateFactors* _aidl_return) override;
     ndk::ScopedAStatus setAudioPatch(const AudioPatch& in_requested,
                                      AudioPatch* _aidl_return) override;
     ndk::ScopedAStatus setAudioPortConfig(
diff --git a/audio/aidl/default/include/core-impl/Stream.h b/audio/aidl/default/include/core-impl/Stream.h
index a5d240f..5abd4de 100644
--- a/audio/aidl/default/include/core-impl/Stream.h
+++ b/audio/aidl/default/include/core-impl/Stream.h
@@ -367,6 +367,23 @@
     }
     ndk::ScopedAStatus getHwVolume(std::vector<float>* _aidl_return) override;
     ndk::ScopedAStatus setHwVolume(const std::vector<float>& in_channelVolumes) override;
+    ndk::ScopedAStatus getAudioDescriptionMixLevel(float* _aidl_return) override;
+    ndk::ScopedAStatus setAudioDescriptionMixLevel(float in_leveldB) override;
+    ndk::ScopedAStatus getDualMonoMode(
+            ::aidl::android::media::audio::common::AudioDualMonoMode* _aidl_return) override;
+    ndk::ScopedAStatus setDualMonoMode(
+            ::aidl::android::media::audio::common::AudioDualMonoMode in_mode) override;
+    ndk::ScopedAStatus getRecommendedLatencyModes(
+            std::vector<::aidl::android::media::audio::common::AudioLatencyMode>* _aidl_return)
+            override;
+    ndk::ScopedAStatus setLatencyMode(
+            ::aidl::android::media::audio::common::AudioLatencyMode in_mode) override;
+    ndk::ScopedAStatus getPlaybackRateParameters(
+            ::aidl::android::media::audio::common::AudioPlaybackRate* _aidl_return) override;
+    ndk::ScopedAStatus setPlaybackRateParameters(
+            const ::aidl::android::media::audio::common::AudioPlaybackRate& in_playbackRate)
+            override;
+    ndk::ScopedAStatus selectPresentation(int32_t in_presentationId, int32_t in_programId) override;
 
   public:
     static ndk::ScopedAStatus createInstance(
diff --git a/audio/aidl/vts/VtsHalAudioCoreModuleTargetTest.cpp b/audio/aidl/vts/VtsHalAudioCoreModuleTargetTest.cpp
index 2508afd..1b46498 100644
--- a/audio/aidl/vts/VtsHalAudioCoreModuleTargetTest.cpp
+++ b/audio/aidl/vts/VtsHalAudioCoreModuleTargetTest.cpp
@@ -72,9 +72,12 @@
 using aidl::android::media::audio::common::AudioDevice;
 using aidl::android::media::audio::common::AudioDeviceAddress;
 using aidl::android::media::audio::common::AudioDeviceType;
+using aidl::android::media::audio::common::AudioDualMonoMode;
 using aidl::android::media::audio::common::AudioFormatType;
 using aidl::android::media::audio::common::AudioIoFlags;
+using aidl::android::media::audio::common::AudioLatencyMode;
 using aidl::android::media::audio::common::AudioOutputFlags;
+using aidl::android::media::audio::common::AudioPlaybackRate;
 using aidl::android::media::audio::common::AudioPort;
 using aidl::android::media::audio::common::AudioPortConfig;
 using aidl::android::media::audio::common::AudioPortDeviceExt;
@@ -2392,6 +2395,192 @@
             << "when no async callback is provided for a non-blocking mix port";
 }
 
+TEST_P(AudioStreamOut, AudioDescriptionMixLevel) {
+    const auto ports = moduleConfig->getOutputMixPorts(false /*attachedOnly*/);
+    if (ports.empty()) {
+        GTEST_SKIP() << "No output mix ports";
+    }
+    bool atLeastOneSupports = false;
+    for (const auto& port : ports) {
+        const auto portConfig = moduleConfig->getSingleConfigForMixPort(false, port);
+        ASSERT_TRUE(portConfig.has_value()) << "No profiles specified for output mix port";
+        WithStream<IStreamOut> stream(portConfig.value());
+        ASSERT_NO_FATAL_FAILURE(stream.SetUp(module.get(), kDefaultBufferSizeFrames));
+        bool isSupported = false;
+        EXPECT_NO_FATAL_FAILURE(
+                TestAccessors<float>(stream.get(), &IStreamOut::getAudioDescriptionMixLevel,
+                                     &IStreamOut::setAudioDescriptionMixLevel,
+                                     {IStreamOut::AUDIO_DESCRIPTION_MIX_LEVEL_MAX,
+                                      IStreamOut::AUDIO_DESCRIPTION_MIX_LEVEL_MAX - 1, 0,
+                                      -INFINITY /*IStreamOut::AUDIO_DESCRIPTION_MIX_LEVEL_MIN*/},
+                                     {IStreamOut::AUDIO_DESCRIPTION_MIX_LEVEL_MAX * 2,
+                                      IStreamOut::AUDIO_DESCRIPTION_MIX_LEVEL_MAX * 1.1f},
+                                     &isSupported));
+        if (isSupported) atLeastOneSupports = true;
+    }
+    if (!atLeastOneSupports) {
+        GTEST_SKIP() << "Audio description mix level is not supported";
+    }
+}
+
+TEST_P(AudioStreamOut, DualMonoMode) {
+    const auto ports = moduleConfig->getOutputMixPorts(false /*attachedOnly*/);
+    if (ports.empty()) {
+        GTEST_SKIP() << "No output mix ports";
+    }
+    bool atLeastOneSupports = false;
+    for (const auto& port : ports) {
+        const auto portConfig = moduleConfig->getSingleConfigForMixPort(false, port);
+        ASSERT_TRUE(portConfig.has_value()) << "No profiles specified for output mix port";
+        WithStream<IStreamOut> stream(portConfig.value());
+        ASSERT_NO_FATAL_FAILURE(stream.SetUp(module.get(), kDefaultBufferSizeFrames));
+        bool isSupported = false;
+        EXPECT_NO_FATAL_FAILURE(TestAccessors<AudioDualMonoMode>(
+                stream.get(), &IStreamOut::getDualMonoMode, &IStreamOut::setDualMonoMode,
+                std::vector<AudioDualMonoMode>(enum_range<AudioDualMonoMode>().begin(),
+                                               enum_range<AudioDualMonoMode>().end()),
+                {}, &isSupported));
+        if (isSupported) atLeastOneSupports = true;
+    }
+    if (!atLeastOneSupports) {
+        GTEST_SKIP() << "Audio dual mono mode is not supported";
+    }
+}
+
+TEST_P(AudioStreamOut, LatencyMode) {
+    const auto ports = moduleConfig->getOutputMixPorts(false /*attachedOnly*/);
+    if (ports.empty()) {
+        GTEST_SKIP() << "No output mix ports";
+    }
+    bool atLeastOneSupports = false;
+    for (const auto& port : ports) {
+        const auto portConfig = moduleConfig->getSingleConfigForMixPort(false, port);
+        ASSERT_TRUE(portConfig.has_value()) << "No profiles specified for output mix port";
+        WithStream<IStreamOut> stream(portConfig.value());
+        ASSERT_NO_FATAL_FAILURE(stream.SetUp(module.get(), kDefaultBufferSizeFrames));
+        bool isSupported = false;
+        std::vector<AudioLatencyMode> supportedModes;
+        ndk::ScopedAStatus status = stream.get()->getRecommendedLatencyModes(&supportedModes);
+        if (status.getExceptionCode() == EX_UNSUPPORTED_OPERATION) continue;
+        if (!status.isOk()) {
+            ADD_FAILURE() << "When latency modes are supported, getRecommendedLatencyModes "
+                          << "must succeed on a non-closed stream, but it failed with " << status;
+            continue;
+        }
+        std::set<AudioLatencyMode> unsupportedModes(enum_range<AudioLatencyMode>().begin(),
+                                                    enum_range<AudioLatencyMode>().end());
+        for (const auto mode : supportedModes) {
+            unsupportedModes.erase(mode);
+            ndk::ScopedAStatus status = stream.get()->setLatencyMode(mode);
+            if (status.getExceptionCode() == EX_UNSUPPORTED_OPERATION) {
+                ADD_FAILURE() << "When latency modes are supported, both getRecommendedLatencyModes"
+                              << " and setLatencyMode must be supported";
+            }
+            EXPECT_IS_OK(status) << "Setting of supported latency mode must succeed";
+        }
+        for (const auto mode : unsupportedModes) {
+            EXPECT_STATUS(EX_ILLEGAL_ARGUMENT, stream.get()->setLatencyMode(mode));
+        }
+        if (isSupported) atLeastOneSupports = true;
+    }
+    if (!atLeastOneSupports) {
+        GTEST_SKIP() << "Audio latency modes are not supported";
+    }
+}
+
+TEST_P(AudioStreamOut, PlaybackRate) {
+    static const auto kStatuses = {EX_NONE, EX_UNSUPPORTED_OPERATION};
+    const auto offloadMixPorts =
+            moduleConfig->getOffloadMixPorts(true /*attachedOnly*/, false /*singlePort*/);
+    if (offloadMixPorts.empty()) {
+        GTEST_SKIP()
+                << "No mix port for compressed offload that could be routed to attached devices";
+    }
+    ndk::ScopedAStatus status;
+    IModule::SupportedPlaybackRateFactors factors;
+    EXPECT_STATUS(kStatuses, status = module.get()->getSupportedPlaybackRateFactors(&factors));
+    if (status.getExceptionCode() == EX_UNSUPPORTED_OPERATION) {
+        GTEST_SKIP() << "Audio playback rate configuration is not supported";
+    }
+    EXPECT_LE(factors.minSpeed, factors.maxSpeed);
+    EXPECT_LE(factors.minPitch, factors.maxPitch);
+    EXPECT_LE(factors.minSpeed, 1.0f);
+    EXPECT_GE(factors.maxSpeed, 1.0f);
+    EXPECT_LE(factors.minPitch, 1.0f);
+    EXPECT_GE(factors.maxPitch, 1.0f);
+    constexpr auto tsDefault = AudioPlaybackRate::TimestretchMode::DEFAULT;
+    constexpr auto tsVoice = AudioPlaybackRate::TimestretchMode::VOICE;
+    constexpr auto fbFail = AudioPlaybackRate::TimestretchFallbackMode::FAIL;
+    constexpr auto fbMute = AudioPlaybackRate::TimestretchFallbackMode::MUTE;
+    const std::vector<AudioPlaybackRate> validValues = {
+            AudioPlaybackRate{1.0f, 1.0f, tsDefault, fbFail},
+            AudioPlaybackRate{1.0f, 1.0f, tsDefault, fbMute},
+            AudioPlaybackRate{factors.maxSpeed, factors.maxPitch, tsDefault, fbMute},
+            AudioPlaybackRate{factors.minSpeed, factors.minPitch, tsDefault, fbMute},
+            AudioPlaybackRate{1.0f, 1.0f, tsVoice, fbMute},
+            AudioPlaybackRate{1.0f, 1.0f, tsVoice, fbFail},
+            AudioPlaybackRate{factors.maxSpeed, factors.maxPitch, tsVoice, fbMute},
+            AudioPlaybackRate{factors.minSpeed, factors.minPitch, tsVoice, fbMute},
+            // Out of range speed / pitch values must not be rejected if the fallback mode is "mute"
+            AudioPlaybackRate{factors.maxSpeed * 2, factors.maxPitch * 2, tsDefault, fbMute},
+            AudioPlaybackRate{factors.minSpeed / 2, factors.minPitch / 2, tsDefault, fbMute},
+            AudioPlaybackRate{factors.maxSpeed * 2, factors.maxPitch * 2, tsVoice, fbMute},
+            AudioPlaybackRate{factors.minSpeed / 2, factors.minPitch / 2, tsVoice, fbMute},
+    };
+    const std::vector<AudioPlaybackRate> invalidValues = {
+            AudioPlaybackRate{factors.maxSpeed, factors.maxPitch * 2, tsDefault, fbFail},
+            AudioPlaybackRate{factors.maxSpeed * 2, factors.maxPitch, tsDefault, fbFail},
+            AudioPlaybackRate{factors.minSpeed, factors.minPitch / 2, tsDefault, fbFail},
+            AudioPlaybackRate{factors.minSpeed / 2, factors.minPitch, tsDefault, fbFail},
+            AudioPlaybackRate{factors.maxSpeed, factors.maxPitch * 2, tsVoice, fbFail},
+            AudioPlaybackRate{factors.maxSpeed * 2, factors.maxPitch, tsVoice, fbFail},
+            AudioPlaybackRate{factors.minSpeed, factors.minPitch / 2, tsVoice, fbFail},
+            AudioPlaybackRate{factors.minSpeed / 2, factors.minPitch, tsVoice, fbFail},
+            AudioPlaybackRate{1.0f, 1.0f, tsDefault,
+                              AudioPlaybackRate::TimestretchFallbackMode::SYS_RESERVED_CUT_REPEAT},
+            AudioPlaybackRate{1.0f, 1.0f, tsDefault,
+                              AudioPlaybackRate::TimestretchFallbackMode::SYS_RESERVED_DEFAULT},
+    };
+    bool atLeastOneSupports = false;
+    for (const auto& port : offloadMixPorts) {
+        const auto portConfig = moduleConfig->getSingleConfigForMixPort(false, port);
+        ASSERT_TRUE(portConfig.has_value()) << "No profiles specified for output mix port";
+        WithStream<IStreamOut> stream(portConfig.value());
+        ASSERT_NO_FATAL_FAILURE(stream.SetUp(module.get(), kDefaultBufferSizeFrames));
+        bool isSupported = false;
+        EXPECT_NO_FATAL_FAILURE(TestAccessors<AudioPlaybackRate>(
+                stream.get(), &IStreamOut::getPlaybackRateParameters,
+                &IStreamOut::setPlaybackRateParameters, validValues, invalidValues, &isSupported));
+        if (isSupported) atLeastOneSupports = true;
+    }
+    if (!atLeastOneSupports) {
+        GTEST_SKIP() << "Audio playback rate configuration is not supported";
+    }
+}
+
+TEST_P(AudioStreamOut, SelectPresentation) {
+    static const auto kStatuses = {EX_ILLEGAL_ARGUMENT, EX_UNSUPPORTED_OPERATION};
+    const auto offloadMixPorts =
+            moduleConfig->getOffloadMixPorts(true /*attachedOnly*/, false /*singlePort*/);
+    if (offloadMixPorts.empty()) {
+        GTEST_SKIP()
+                << "No mix port for compressed offload that could be routed to attached devices";
+    }
+    bool atLeastOneSupports = false;
+    for (const auto& port : offloadMixPorts) {
+        const auto portConfig = moduleConfig->getSingleConfigForMixPort(false, port);
+        ASSERT_TRUE(portConfig.has_value()) << "No profiles specified for output mix port";
+        WithStream<IStreamOut> stream(portConfig.value());
+        ASSERT_NO_FATAL_FAILURE(stream.SetUp(module.get(), kDefaultBufferSizeFrames));
+        ndk::ScopedAStatus status;
+        EXPECT_STATUS(kStatuses, status = stream.get()->selectPresentation(0, 0));
+        if (status.getExceptionCode() != EX_UNSUPPORTED_OPERATION) atLeastOneSupports = true;
+    }
+    if (!atLeastOneSupports) {
+        GTEST_SKIP() << "Presentation selection is not supported";
+    }
+}
+
 class StreamLogicDefaultDriver : public StreamLogicDriver {
   public:
     explicit StreamLogicDefaultDriver(std::shared_ptr<StateSequence> commands)