diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index 3602e94..55f74e1 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -322,6 +322,18 @@
     return NO_ERROR;
 }
 
+status_t AudioSystem::setPortsVolume(
+        const std::vector<audio_port_handle_t>& portIds, float volume, audio_io_handle_t output) {
+    const sp<IAudioFlinger> af = get_audio_flinger();
+    if (af == 0) return PERMISSION_DENIED;
+    std::vector<int32_t> portIdsAidl = VALUE_OR_RETURN_STATUS(
+            convertContainer<std::vector<int32_t>>(
+                    portIds, legacy2aidl_audio_port_handle_t_int32_t));
+    int32_t outputAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
+    af->setPortsVolume(portIdsAidl, volume, outputAidl);
+    return NO_ERROR;
+}
+
 status_t AudioSystem::setMode(audio_mode_t mode) {
     if (uint32_t(mode) >= AUDIO_MODE_CNT) return BAD_VALUE;
     const sp<IAudioFlinger> af = get_audio_flinger();
@@ -1081,7 +1093,8 @@
                                        audio_port_handle_t* portId,
                                        std::vector<audio_io_handle_t>* secondaryOutputs,
                                        bool *isSpatialized,
-                                       bool *isBitPerfect) {
+                                       bool *isBitPerfect,
+                                       float *volume) {
     if (attr == nullptr) {
         ALOGE("%s NULL audio attributes", __func__);
         return BAD_VALUE;
@@ -1147,6 +1160,7 @@
     *isBitPerfect = responseAidl.isBitPerfect;
     *attr = VALUE_OR_RETURN_STATUS(
             aidl2legacy_AudioAttributes_audio_attributes_t(responseAidl.attr));
+    *volume = responseAidl.volume;
 
     return OK;
 }
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index e0dca2d..9241973 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -350,6 +350,15 @@
     return statusTFromBinderStatus(mDelegate->setStreamMute(streamAidl, muted));
 }
 
+status_t AudioFlingerClientAdapter::setPortsVolume(
+        const std::vector<audio_port_handle_t>& portIds, float volume, audio_io_handle_t output) {
+    std::vector<int32_t> portIdsAidl = VALUE_OR_RETURN_STATUS(
+            convertContainer<std::vector<int32_t>>(
+                    portIds, legacy2aidl_audio_port_handle_t_int32_t));
+    int32_t outputAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
+    return statusTFromBinderStatus(mDelegate->setPortsVolume(portIdsAidl, volume, outputAidl));
+}
+
 status_t AudioFlingerClientAdapter::setMode(audio_mode_t mode) {
     AudioMode modeAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_mode_t_AudioMode(mode));
     return statusTFromBinderStatus(mDelegate->setMode(modeAidl));
@@ -1012,6 +1021,16 @@
     return Status::fromStatusT(mDelegate->setStreamMute(streamLegacy, muted));
 }
 
+Status AudioFlingerServerAdapter::setPortsVolume(
+        const std::vector<int32_t>& portIds, float volume, int32_t output) {
+    std::vector<audio_port_handle_t> portIdsLegacy = VALUE_OR_RETURN_BINDER(
+            convertContainer<std::vector<audio_port_handle_t>>(
+                    portIds, aidl2legacy_int32_t_audio_port_handle_t));
+    audio_io_handle_t outputLegacy = VALUE_OR_RETURN_BINDER(
+            aidl2legacy_int32_t_audio_io_handle_t(output));
+    return Status::fromStatusT(mDelegate->setPortsVolume(portIdsLegacy, volume, outputLegacy));
+}
+
 Status AudioFlingerServerAdapter::setMode(AudioMode mode) {
     audio_mode_t modeLegacy = VALUE_OR_RETURN_BINDER(aidl2legacy_AudioMode_audio_mode_t(mode));
     return Status::fromStatusT(mDelegate->setMode(modeLegacy));
diff --git a/media/libaudioclient/aidl/android/media/GetOutputForAttrResponse.aidl b/media/libaudioclient/aidl/android/media/GetOutputForAttrResponse.aidl
index b814b85..4b26d5b 100644
--- a/media/libaudioclient/aidl/android/media/GetOutputForAttrResponse.aidl
+++ b/media/libaudioclient/aidl/android/media/GetOutputForAttrResponse.aidl
@@ -39,4 +39,6 @@
     boolean isBitPerfect;
     /** The corrected audio attributes. **/
     AudioAttributes attr;
+    /** initial port volume for the new audio track */
+    float volume;
 }
diff --git a/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl b/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
index 29de9c2..1c825bc 100644
--- a/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
+++ b/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
@@ -100,6 +100,13 @@
     void setStreamVolume(AudioStreamType stream, float value, int /* audio_io_handle_t */ output);
     void setStreamMute(AudioStreamType stream, boolean muted);
 
+    /*
+     * Set AudioTrack port ids volume attribute. This is the new way of controlling volume from
+     * AudioPolicyManager to AudioFlinger.
+     */
+    void setPortsVolume(in int[] /* audio_port_handle_t[] */ portIds, float volume,
+            int /* audio_io_handle_t */ output);
+
     // set audio mode.
     void setMode(AudioMode mode);
 
diff --git a/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp b/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
index 4c94974..710a656 100644
--- a/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
+++ b/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
@@ -26,6 +26,7 @@
 #include <android/content/AttributionSourceState.h>
 #include <binder/IServiceManager.h>
 #include <binder/MemoryDealer.h>
+#include <com_android_media_audioserver.h>
 #include <media/AidlConversion.h>
 #include <media/AudioEffect.h>
 #include <media/AudioRecord.h>
@@ -41,6 +42,8 @@
 constexpr int32_t kMaxSampleRateHz = 192000;
 constexpr int32_t kSampleRateUnspecified = 0;
 
+namespace audioserver_flags = com::android::media::audioserver;
+
 using namespace std;
 using namespace android;
 
@@ -501,13 +504,19 @@
     AudioSystem::getMasterMute(&state);
     AudioSystem::isMicrophoneMuted(&state);
 
-    audio_stream_type_t stream = getValue(&mFdp, kStreamtypes);
-    AudioSystem::setStreamMute(getValue(&mFdp, kStreamtypes), mFdp.ConsumeBool());
+    audio_stream_type_t stream ;
+    if (!audioserver_flags::portid_volume_management()) {
+        stream = getValue(&mFdp, kStreamtypes);
+        AudioSystem::setStreamMute(getValue(&mFdp, kStreamtypes), mFdp.ConsumeBool());
 
-    stream = getValue(&mFdp, kStreamtypes);
-    AudioSystem::setStreamVolume(stream, mFdp.ConsumeFloatingPoint<float>(),
-                                 mFdp.ConsumeIntegral<int32_t>());
-
+        stream = getValue(&mFdp, kStreamtypes);
+        AudioSystem::setStreamVolume(stream, mFdp.ConsumeFloatingPoint<float>(),
+                                     mFdp.ConsumeIntegral<int32_t>());
+    } else {
+        std::vector <audio_port_handle_t> portsForVolumeChange{};
+        AudioSystem::setPortsVolume(portsForVolumeChange, mFdp.ConsumeFloatingPoint<float>(),
+                                    mFdp.ConsumeIntegral<int32_t>());
+    }
     audio_mode_t mode = getValue(&mFdp, kModes);
     AudioSystem::setMode(mode);
 
diff --git a/media/libaudioclient/include/media/AudioSystem.h b/media/libaudioclient/include/media/AudioSystem.h
index 67b3dcd..40e5673 100644
--- a/media/libaudioclient/include/media/AudioSystem.h
+++ b/media/libaudioclient/include/media/AudioSystem.h
@@ -131,6 +131,16 @@
     // mute/unmute stream
     static status_t setStreamMute(audio_stream_type_t stream, bool mute);
 
+    /**
+     * Set volume for given AudioTrack port ids on specified output
+     * @param portIds to consider
+     * @param volume to set
+     * @param output to consider
+     * @return NO_ERROR if successful
+     */
+    static status_t setPortsVolume(const std::vector<audio_port_handle_t>& portIds,
+                                   float volume, audio_io_handle_t output);
+
     // set audio mode in audio hardware
     static status_t setMode(audio_mode_t mode);
 
@@ -334,7 +344,8 @@
                                      audio_port_handle_t *portId,
                                      std::vector<audio_io_handle_t> *secondaryOutputs,
                                      bool *isSpatialized,
-                                     bool *isBitPerfect);
+                                     bool *isBitPerfect,
+                                     float *volume);
     static status_t startOutput(audio_port_handle_t portId);
     static status_t stopOutput(audio_port_handle_t portId);
     static void releaseOutput(audio_port_handle_t portId);
diff --git a/media/libaudioclient/include/media/IAudioFlinger.h b/media/libaudioclient/include/media/IAudioFlinger.h
index 667e9ae..a5f3217 100644
--- a/media/libaudioclient/include/media/IAudioFlinger.h
+++ b/media/libaudioclient/include/media/IAudioFlinger.h
@@ -229,6 +229,16 @@
                                     audio_io_handle_t output) = 0;
     virtual     status_t    setStreamMute(audio_stream_type_t stream, bool muted) = 0;
 
+    /**
+     * Set volume for given AudioTrack port ids on specified output
+     * @param portIds to consider
+     * @param volume to set
+     * @param output to consider
+     * @return NO_ERROR if successful
+     */
+    virtual status_t setPortsVolume(const std::vector<audio_port_handle_t>& portIds, float volume,
+            audio_io_handle_t output) = 0;
+
     // set audio mode
     virtual     status_t    setMode(audio_mode_t mode) = 0;
 
@@ -420,6 +430,8 @@
     status_t setStreamVolume(audio_stream_type_t stream, float value,
                              audio_io_handle_t output) override;
     status_t setStreamMute(audio_stream_type_t stream, bool muted) override;
+    status_t setPortsVolume(const std::vector<audio_port_handle_t>& portIds, float volume,
+            audio_io_handle_t output) override;
     status_t setMode(audio_mode_t mode) override;
     status_t setMicMute(bool state) override;
     bool getMicMute() const override;
@@ -542,6 +554,7 @@
             MASTER_MUTE = media::BnAudioFlingerService::TRANSACTION_masterMute,
             SET_STREAM_VOLUME = media::BnAudioFlingerService::TRANSACTION_setStreamVolume,
             SET_STREAM_MUTE = media::BnAudioFlingerService::TRANSACTION_setStreamMute,
+            SET_PORTS_VOLUME = media::BnAudioFlingerService::TRANSACTION_setPortsVolume,
             SET_MODE = media::BnAudioFlingerService::TRANSACTION_setMode,
             SET_MIC_MUTE = media::BnAudioFlingerService::TRANSACTION_setMicMute,
             GET_MIC_MUTE = media::BnAudioFlingerService::TRANSACTION_getMicMute,
@@ -664,6 +677,8 @@
     Status setStreamVolume(media::audio::common::AudioStreamType stream,
                            float value, int32_t output) override;
     Status setStreamMute(media::audio::common::AudioStreamType stream, bool muted) override;
+    Status setPortsVolume(const std::vector<int32_t>& portIds, float volume, int32_t output)
+            override;
     Status setMode(media::audio::common::AudioMode mode) override;
     Status setMicMute(bool state) override;
     Status getMicMute(bool* _aidl_return) override;
diff --git a/services/audioflinger/Android.bp b/services/audioflinger/Android.bp
index 2abf682..e5ec5d8 100644
--- a/services/audioflinger/Android.bp
+++ b/services/audioflinger/Android.bp
@@ -146,6 +146,7 @@
         "audioflinger-aidl-cpp",
         "av-types-aidl-cpp",
         "com.android.media.audio-aconfig-cc",
+        "com.android.media.audioserver-aconfig-cc",
         "effect-aidl-cpp",
         "libactivitymanager_aidl",
         "libaudioclient",
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 20cd40c..b2c9e32 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -187,6 +187,7 @@
 BINDER_METHOD_ENTRY(masterMute) \
 BINDER_METHOD_ENTRY(setStreamVolume) \
 BINDER_METHOD_ENTRY(setStreamMute) \
+BINDER_METHOD_ENTRY(setPortsVolume) \
 BINDER_METHOD_ENTRY(setMode) \
 BINDER_METHOD_ENTRY(setMicMute) \
 BINDER_METHOD_ENTRY(getMicMute) \
@@ -617,6 +618,7 @@
         std::vector<audio_io_handle_t> secondaryOutputs;
         bool isSpatialized;
         bool isBitPerfect;
+        float volume;
         ret = AudioSystem::getOutputForAttr(&localAttr, &io,
                                             actualSessionId,
                                             &streamType, adjAttributionSource,
@@ -624,7 +626,8 @@
                                             (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_MMAP_NOIRQ |
                                                     AUDIO_OUTPUT_FLAG_DIRECT),
                                             deviceId, &portId, &secondaryOutputs, &isSpatialized,
-                                            &isBitPerfect);
+                                            &isBitPerfect,
+                                            &volume);
         if (ret != NO_ERROR) {
             config->sample_rate = fullConfig.sample_rate;
             config->channel_mask = fullConfig.channel_mask;
@@ -1061,6 +1064,7 @@
     std::vector<audio_io_handle_t> secondaryOutputs;
     bool isSpatialized = false;
     bool isBitPerfect = false;
+    float volume;
 
     audio_io_handle_t effectThreadId = AUDIO_IO_HANDLE_NONE;
     std::vector<int> effectIds;
@@ -1121,7 +1125,7 @@
     lStatus = AudioSystem::getOutputForAttr(&localAttr, &output.outputId, sessionId, &streamType,
                                             adjAttributionSource, &input.config, input.flags,
                                             &output.selectedDeviceId, &portId, &secondaryOutputs,
-                                            &isSpatialized, &isBitPerfect);
+                                            &isSpatialized, &isBitPerfect, &volume);
 
     if (lStatus != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
         ALOGE("createTrack() getOutputForAttr() return error %d or invalid output handle", lStatus);
@@ -1178,7 +1182,7 @@
         if (effectThread == nullptr) {
             effectChain = getOrphanEffectChain_l(sessionId);
         }
-        ALOGV("createTrack() sessionId: %d", sessionId);
+        ALOGV("createTrack() sessionId: %d volume: %f", sessionId, volume);
 
         output.sampleRate = input.config.sample_rate;
         output.frameCount = input.frameCount;
@@ -1193,7 +1197,7 @@
                                       input.sharedBuffer, sessionId, &output.flags,
                                       callingPid, adjAttributionSource, input.clientInfo.clientTid,
                                       &lStatus, portId, input.audioTrackCallback, isSpatialized,
-                                      isBitPerfect, &output.afTrackFlags);
+                                      isBitPerfect, &output.afTrackFlags, volume);
         LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
         // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
 
@@ -1644,6 +1648,33 @@
     return NO_ERROR;
 }
 
+status_t AudioFlinger::setPortsVolume(
+        const std::vector<audio_port_handle_t>& ports, float volume, audio_io_handle_t output)
+{
+    for (const auto& port : ports) {
+        if (port == AUDIO_PORT_HANDLE_NONE) {
+            return BAD_VALUE;
+        }
+    }
+    if (isnan(volume) || volume > 1.0f || volume < 0.0f) {
+        return BAD_VALUE;
+    }
+    if (output == AUDIO_IO_HANDLE_NONE) {
+        return BAD_VALUE;
+    }
+    audio_utils::lock_guard lock(mutex());
+    IAfPlaybackThread *thread = checkPlaybackThread_l(output);
+    if (thread != nullptr) {
+        return thread->setPortsVolume(ports, volume);
+    }
+    const sp<IAfMmapThread> mmapThread = checkMmapThread_l(output);
+    if (mmapThread != nullptr && mmapThread->isOutput()) {
+        IAfMmapPlaybackThread *mmapPlaybackThread = mmapThread->asIAfMmapPlaybackThread().get();
+        return mmapPlaybackThread->setPortsVolume(ports, volume);
+    }
+    return BAD_VALUE;
+}
+
 status_t AudioFlinger::setRequestedLatencyMode(
         audio_io_handle_t output, audio_latency_mode_t mode) {
     if (output == AUDIO_IO_HANDLE_NONE) {
@@ -3824,8 +3855,7 @@
 
 
 // checkPlaybackThread_l() must be called with AudioFlinger::mutex() held
-sp<VolumeInterface> AudioFlinger::getVolumeInterface_l(audio_io_handle_t output) const
-{
+sp<VolumeInterface> AudioFlinger::getVolumeInterface_l(audio_io_handle_t output) const {
     sp<VolumeInterface> volumeInterface = mPlaybackThreads.valueFor(output).get();
     if (volumeInterface == nullptr) {
         IAfMmapThread* const mmapThread = mMmapThreads.valueFor(output).get();
@@ -4020,7 +4050,8 @@
                                                        outputFlags,
                                                        0ns /* timeout */,
                                                        frameCountToBeReady,
-                                                       track->getSpeed());
+                                                       track->getSpeed(),
+                                                       track->getPortVolume());
         status = patchTrack->initCheck();
         if (status != NO_ERROR) {
             ALOGE("Secondary output patchTrack init failed: %d", status);
@@ -5119,6 +5150,7 @@
         case TransactionCode::GET_AUDIO_MIX_PORT:
         case TransactionCode::SET_TRACKS_INTERNAL_MUTE:
         case TransactionCode::RESET_REFERENCES_FOR_TEST:
+        case TransactionCode::SET_PORTS_VOLUME:
             ALOGW("%s: transaction %d received from PID %d",
                   __func__, static_cast<int>(code), IPCThreadState::self()->getCallingPid());
             // return status only for non void methods
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index adec4aa..902df0a 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -96,6 +96,9 @@
     status_t setStreamMute(audio_stream_type_t stream, bool muted) final
             EXCLUDES_AudioFlinger_Mutex;
 
+    status_t setPortsVolume(const std::vector<audio_port_handle_t>& portIds, float volume,
+            audio_io_handle_t output) final EXCLUDES_AudioFlinger_Mutex;
+
     status_t setMode(audio_mode_t mode) final EXCLUDES_AudioFlinger_Mutex;
 
     status_t setMicMute(bool state) final EXCLUDES_AudioFlinger_Mutex;
@@ -551,6 +554,7 @@
     IAfPlaybackThread* checkMixerThread_l(audio_io_handle_t output) const REQUIRES(mutex());
 
     sp<VolumeInterface> getVolumeInterface_l(audio_io_handle_t output) const REQUIRES(mutex());
+
     std::vector<sp<VolumeInterface>> getAllVolumeInterfaces_l() const REQUIRES(mutex());
 
 
diff --git a/services/audioflinger/IAfThread.h b/services/audioflinger/IAfThread.h
index 4d26aa0..8596acb 100644
--- a/services/audioflinger/IAfThread.h
+++ b/services/audioflinger/IAfThread.h
@@ -26,6 +26,7 @@
 #include <datapath/AudioStreamIn.h>
 #include <datapath/AudioStreamOut.h>
 #include <datapath/VolumeInterface.h>
+#include <datapath/VolumePortInterface.h>
 #include <fastpath/FastMixerDumpState.h>
 #include <media/DeviceDescriptorBase.h>
 #include <media/MmapStreamInterface.h>
@@ -479,7 +480,8 @@
             const sp<media::IAudioTrackCallback>& callback,
             bool isSpatialized,
             bool isBitPerfect,
-            audio_output_flags_t* afTrackFlags)
+            audio_output_flags_t* afTrackFlags,
+            float volume)
             REQUIRES(audio_utils::AudioFlinger_Mutex) = 0;
 
     virtual status_t addTrack_l(const sp<IAfTrack>& track) REQUIRES(mutex()) = 0;
@@ -555,6 +557,9 @@
 
     virtual void setTracksInternalMute(std::map<audio_port_handle_t, bool>* tracksInternalMute)
             EXCLUDES_ThreadBase_Mutex = 0;
+
+    virtual status_t setPortsVolume(const std::vector<audio_port_handle_t>& portIds, float volume)
+            EXCLUDES_ThreadBase_Mutex = 0;
 };
 
 class IAfDirectOutputThread : public virtual IAfPlaybackThread {
@@ -694,6 +699,9 @@
             AudioHwDevice* hwDev, AudioStreamOut* output, bool systemReady);
 
     virtual AudioStreamOut* clearOutput() EXCLUDES_ThreadBase_Mutex = 0;
+
+    virtual status_t setPortsVolume(const std::vector<audio_port_handle_t>& portIds, float volume)
+            EXCLUDES_ThreadBase_Mutex = 0;
 };
 
 class IAfMmapCaptureThread : public virtual IAfMmapThread {
diff --git a/services/audioflinger/IAfTrack.h b/services/audioflinger/IAfTrack.h
index a9c87ad..ee834d6 100644
--- a/services/audioflinger/IAfTrack.h
+++ b/services/audioflinger/IAfTrack.h
@@ -21,6 +21,7 @@
 #include <audio_utils/mutex.h>
 #include <audiomanager/IAudioManager.h>
 #include <binder/IMemory.h>
+#include <datapath/VolumePortInterface.h>
 #include <fastpath/FastMixerDumpState.h>
 #include <media/AudioSystem.h>
 #include <media/VolumeShaper.h>
@@ -254,7 +255,7 @@
 };
 
 // Common interface for Playback tracks.
-class IAfTrack : public virtual IAfTrackBase {
+class IAfTrack : public virtual IAfTrackBase, public virtual VolumePortInterface {
 public:
     // FillingStatus is used for suppressing volume ramp at begin of playing
     enum FillingStatus { FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE };
@@ -289,7 +290,8 @@
             size_t frameCountToBeReady = SIZE_MAX,
             float speed = 1.0f,
             bool isSpatialized = false,
-            bool isBitPerfect = false);
+            bool isBitPerfect = false,
+            float volume = 0.0f);
 
     virtual void pause() = 0;
     virtual void flush() = 0;
@@ -452,7 +454,7 @@
     virtual ExtendedTimestamp getClientProxyTimestamp() const = 0;
 };
 
-class IAfMmapTrack : public virtual IAfTrackBase {
+class IAfMmapTrack : public virtual IAfTrackBase, public virtual VolumePortInterface {
 public:
     static sp<IAfMmapTrack> create(IAfThreadBase* thread,
             const audio_attributes_t& attr,
@@ -463,7 +465,8 @@
             bool isOut,
             const android::content::AttributionSourceState& attributionSource,
             pid_t creatorPid,
-            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
+            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE,
+            float volume = 0.0f);
 
     // protected by MMapThread::mLock
     virtual void setSilenced_l(bool silenced) = 0;
@@ -583,7 +586,8 @@
                                              *  as soon as possible to have
                                              *  the lowest possible latency
                                              *  even if it might glitch. */
-            float speed = 1.0f);
+            float speed = 1.0f,
+            float volume = 1.0f);
 };
 
 class IAfPatchRecord : public virtual IAfRecordTrack, public virtual IAfPatchTrackBase {
diff --git a/services/audioflinger/MmapTracks.h b/services/audioflinger/MmapTracks.h
index 85ce142..8758bd0 100644
--- a/services/audioflinger/MmapTracks.h
+++ b/services/audioflinger/MmapTracks.h
@@ -35,7 +35,8 @@
                             bool isOut,
                             const android::content::AttributionSourceState& attributionSource,
                             pid_t creatorPid,
-                            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
+                            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE,
+                            float volume = 0.0f);
     ~MmapTrack() override;
 
     status_t initCheck() const final;
@@ -65,6 +66,13 @@
     void processMuteEvent_l(const sp<IAudioManager>& audioManager,
                             mute_state_t muteState)
                             /* REQUIRES(MmapPlaybackThread::mLock) */ final;
+
+    // VolumePortInterface implementation
+    void setPortVolume(float volume) override {
+        mVolume = volume;
+    }
+    float getPortVolume() const override { return mVolume; }
+
 private:
     DISALLOW_COPY_AND_ASSIGN(MmapTrack);
 
@@ -87,6 +95,8 @@
             /* GUARDED_BY(MmapPlaybackThread::mLock) */;
     mute_state_t mMuteState
             /* GUARDED_BY(MmapPlaybackThread::mLock) */;
+
+    float mVolume = 0.0f;
 };  // end of Track
 
 } // namespace android
\ No newline at end of file
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index f57470f..06f7887 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -647,7 +647,8 @@
                                            outputFlags,
                                            {} /*timeout*/,
                                            frameCountToBeReady,
-                                           1.0f);
+                                           1.0f /*speed*/,
+                                           1.0f /*volume*/);
     status = mPlayback.checkTrack(tempPatchTrack.get());
     if (status != NO_ERROR) {
         return status;
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 2cc6236..3edeee3 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -96,7 +96,8 @@
                                 size_t frameCountToBeReady = SIZE_MAX,
                                 float speed = 1.0f,
                                 bool isSpatialized = false,
-                                bool isBitPerfect = false);
+                                bool isBitPerfect = false,
+                                float volume = 0.0f);
     ~Track() override;
     status_t initCheck() const final;
     void appendDumpHeader(String8& result) const final;
@@ -222,6 +223,13 @@
 
     bool getInternalMute() const final { return mInternalMute; }
     void setInternalMute(bool muted) final { mInternalMute = muted; }
+
+    // VolumePortInterface implementation
+    void setPortVolume(float volume) override {
+        mVolume = volume;
+    }
+    float getPortVolume() const override { return mVolume; }
+
 protected:
 
     DISALLOW_COPY_AND_ASSIGN(Track);
@@ -362,6 +370,8 @@
         for (auto& tp : mTeePatches) { f(tp.patchTrack); }
     };
 
+    void                populateUsageAndContentTypeFromStreamType();
+
     size_t              mPresentationCompleteFrames = 0; // (Used for Mixed tracks)
                                     // The number of frames written to the
                                     // audio HAL when this track is considered fully rendered.
@@ -403,8 +413,8 @@
     // access these two variables only when holding player thread lock.
     std::unique_ptr<os::PersistableBundle> mMuteEventExtras;
     mute_state_t        mMuteState;
-
     bool                mInternalMute = false;
+    float mVolume = 0.0f;
 };  // end of Track
 
 
@@ -501,7 +511,8 @@
                                                                     *  as soon as possible to have
                                                                     *  the lowest possible latency
                                                                     *  even if it might glitch. */
-                                   float speed = 1.0f);
+                                   float speed = 1.0f,
+                                   float volume = 1.0f);
     ~PatchTrack() override;
 
     size_t framesReady() const final;
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 7c7d812..6359846 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -49,6 +49,7 @@
 #include <binder/IServiceManager.h>
 #include <binder/PersistableBundle.h>
 #include <com_android_media_audio.h>
+#include <com_android_media_audioserver.h>
 #include <cutils/bitops.h>
 #include <cutils/properties.h>
 #include <fastpath/AutoPark.h>
@@ -122,6 +123,7 @@
 }
 
 using com::android::media::permission::ValidatedAttributionSourceState;
+namespace audioserver_flags = com::android::media::audioserver;
 
 namespace android {
 
@@ -2217,17 +2219,18 @@
                 (int64_t)(mIsMsdDevice ? AUDIO_DEVICE_OUT_BUS // turn on by default for MSD
                                        : AUDIO_DEVICE_NONE));
     }
-
-    for (int i = AUDIO_STREAM_MIN; i < AUDIO_STREAM_FOR_POLICY_CNT; ++i) {
-        const audio_stream_type_t stream{static_cast<audio_stream_type_t>(i)};
-        mStreamTypes[stream].volume = 0.0f;
-        mStreamTypes[stream].mute = mAfThreadCallback->streamMute_l(stream);
+    if (!audioserver_flags::portid_volume_management()) {
+        for (int i = AUDIO_STREAM_MIN; i < AUDIO_STREAM_FOR_POLICY_CNT; ++i) {
+            const audio_stream_type_t stream{static_cast<audio_stream_type_t>(i)};
+            mStreamTypes[stream].volume = 0.0f;
+            mStreamTypes[stream].mute = mAfThreadCallback->streamMute_l(stream);
+        }
+        // Audio patch and call assistant volume are always max
+        mStreamTypes[AUDIO_STREAM_PATCH].volume = 1.0f;
+        mStreamTypes[AUDIO_STREAM_PATCH].mute = false;
+        mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].volume = 1.0f;
+        mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].mute = false;
     }
-    // Audio patch and call assistant volume are always max
-    mStreamTypes[AUDIO_STREAM_PATCH].volume = 1.0f;
-    mStreamTypes[AUDIO_STREAM_PATCH].mute = false;
-    mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].volume = 1.0f;
-    mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].mute = false;
 }
 
 PlaybackThread::~PlaybackThread()
@@ -2278,16 +2281,17 @@
 void PlaybackThread::dumpTracks_l(int fd, const Vector<String16>& /* args */)
 {
     String8 result;
-
-    result.appendFormat("  Stream volumes in dB: ");
-    for (int i = 0; i < AUDIO_STREAM_CNT; ++i) {
-        const stream_type_t *st = &mStreamTypes[i];
-        if (i > 0) {
-            result.appendFormat(", ");
-        }
-        result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume));
-        if (st->mute) {
-            result.append("M");
+    if (!audioserver_flags::portid_volume_management()) {
+        result.appendFormat("  Stream volumes in dB: ");
+        for (int i = 0; i < AUDIO_STREAM_CNT; ++i) {
+            const stream_type_t *st = &mStreamTypes[i];
+            if (i > 0) {
+                result.appendFormat(", ");
+            }
+            result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume));
+            if (st->mute) {
+                result.append("M");
+            }
         }
     }
     result.append("\n");
@@ -2395,7 +2399,8 @@
         const sp<media::IAudioTrackCallback>& callback,
         bool isSpatialized,
         bool isBitPerfect,
-        audio_output_flags_t *afTrackFlags)
+        audio_output_flags_t *afTrackFlags,
+        float volume)
 {
     size_t frameCount = *pFrameCount;
     size_t notificationFrameCount = *pNotificationFrameCount;
@@ -2724,7 +2729,7 @@
                           nullptr /* buffer */, (size_t)0 /* bufferSize */, sharedBuffer,
                           sessionId, creatorPid, attributionSource, trackFlags,
                           IAfTrackBase::TYPE_DEFAULT, portId, SIZE_MAX /*frameCountToBeReady*/,
-                          speed, isSpatialized, isBitPerfect);
+                          speed, isSpatialized, isBitPerfect, volume);
 
         lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
         if (lStatus != NO_ERROR) {
@@ -2852,6 +2857,22 @@
     return mStreamTypes[stream].volume;
 }
 
+status_t PlaybackThread::setPortsVolume(
+        const std::vector<audio_port_handle_t>& portIds, float volume) {
+    audio_utils::lock_guard _l(mutex());
+    for (const auto& portId : portIds) {
+        for (size_t i = 0; i < mTracks.size(); i++) {
+            sp<IAfTrack> track = mTracks[i].get();
+            if (portId == track->portId()) {
+                track->setPortVolume(volume);
+                break;
+            }
+        }
+    }
+    broadcast_l();
+    return NO_ERROR;
+}
+
 void PlaybackThread::setVolumeForOutput_l(float left, float right) const
 {
     mOutput->stream->setVolume(left, right);
@@ -5783,12 +5804,19 @@
                 }
                 sp<AudioTrackServerProxy> proxy = track->audioTrackServerProxy();
                 float volume;
-                if (track->isPlaybackRestricted() || mStreamTypes[track->streamType()].mute) {
-                    volume = 0.f;
+                if (!audioserver_flags::portid_volume_management()) {
+                    if (track->isPlaybackRestricted() || mStreamTypes[track->streamType()].mute) {
+                        volume = 0.f;
+                    } else {
+                        volume = masterVolume * mStreamTypes[track->streamType()].volume;
+                    }
                 } else {
-                    volume = masterVolume * mStreamTypes[track->streamType()].volume;
+                    if (track->isPlaybackRestricted()) {
+                        volume = 0.f;
+                    } else {
+                        volume = masterVolume * track->getPortVolume();
+                    }
                 }
-
                 handleVoipVolume_l(&volume);
 
                 // cache the combined master volume and stream type volume for fast mixer; this
@@ -5800,15 +5828,23 @@
                 gain_minifloat_packed_t vlr = proxy->getVolumeLR();
                 float vlf = float_from_gain(gain_minifloat_unpack_left(vlr));
                 float vrf = float_from_gain(gain_minifloat_unpack_right(vlr));
-
-                track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
-                    /*muteState=*/{masterVolume == 0.f,
-                                   mStreamTypes[track->streamType()].volume == 0.f,
-                                   mStreamTypes[track->streamType()].mute,
-                                   track->isPlaybackRestricted(),
-                                   vlf == 0.f && vrf == 0.f,
-                                   vh == 0.f});
-
+                if (!audioserver_flags::portid_volume_management()) {
+                    track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
+                            /*muteState=*/{masterVolume == 0.f,
+                                           mStreamTypes[track->streamType()].volume == 0.f,
+                                           mStreamTypes[track->streamType()].mute,
+                                           track->isPlaybackRestricted(),
+                                           vlf == 0.f && vrf == 0.f,
+                                           vh == 0.f});
+                } else {
+                    track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
+                            /*muteState=*/{masterVolume == 0.f,
+                                           track->getPortVolume() == 0.f,
+                                           /* muteFromStreamMuted= */ false,
+                                           track->isPlaybackRestricted(),
+                                           vlf == 0.f && vrf == 0.f,
+                                           vh == 0.f});
+                }
                 vlf *= volume;
                 vrf *= volume;
 
@@ -5959,16 +5995,22 @@
             uint32_t vl, vr;       // in U8.24 integer format
             float vlf, vrf, vaf;   // in [0.0, 1.0] float format
             // read original volumes with volume control
-            float v = masterVolume * mStreamTypes[track->streamType()].volume;
             // Always fetch volumeshaper volume to ensure state is updated.
             const sp<AudioTrackServerProxy> proxy = track->audioTrackServerProxy();
             const float vh = track->getVolumeHandler()->getVolume(
                     track->audioTrackServerProxy()->framesReleased()).first;
-
-            if (mStreamTypes[track->streamType()].mute || track->isPlaybackRestricted()) {
-                v = 0;
+            float v;
+            if (!audioserver_flags::portid_volume_management()) {
+                v = masterVolume * mStreamTypes[track->streamType()].volume;
+                if (mStreamTypes[track->streamType()].mute || track->isPlaybackRestricted()) {
+                    v = 0;
+                }
+            } else {
+                v = masterVolume * track->getPortVolume();
+                if (track->isPlaybackRestricted()) {
+                    v = 0;
+                }
             }
-
             handleVoipVolume_l(&v);
 
             if (track->isPausing()) {
@@ -5988,15 +6030,23 @@
                     ALOGV("Track right volume out of range: %.3g", vrf);
                     vrf = GAIN_FLOAT_UNITY;
                 }
-
-                track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
-                    /*muteState=*/{masterVolume == 0.f,
-                                   mStreamTypes[track->streamType()].volume == 0.f,
-                                   mStreamTypes[track->streamType()].mute,
-                                   track->isPlaybackRestricted(),
-                                   vlf == 0.f && vrf == 0.f,
-                                   vh == 0.f});
-
+                if (!audioserver_flags::portid_volume_management()) {
+                    track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
+                            /*muteState=*/{masterVolume == 0.f,
+                                           mStreamTypes[track->streamType()].volume == 0.f,
+                                           mStreamTypes[track->streamType()].mute,
+                                           track->isPlaybackRestricted(),
+                                           vlf == 0.f && vrf == 0.f,
+                                           vh == 0.f});
+                } else {
+                    track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
+                            /*muteState=*/{masterVolume == 0.f,
+                                           track->getPortVolume() == 0.f,
+                                           /* muteFromStreamMuted= */ false,
+                                           track->isPlaybackRestricted(),
+                                           vlf == 0.f && vrf == 0.f,
+                                           vh == 0.f});
+                }
                 // now apply the master volume and stream type volume and shaper volume
                 vlf *= v * vh;
                 vrf *= v * vh;
@@ -6722,34 +6772,64 @@
 
     const bool clientVolumeMute = (left == 0.f && right == 0.f);
 
-    if (mMasterMute || mStreamTypes[track->streamType()].mute || track->isPlaybackRestricted()) {
-        left = right = 0;
-    } else {
-        float typeVolume = mStreamTypes[track->streamType()].volume;
-        const float v = mMasterVolume * typeVolume * shaperVolume;
+    if (!audioserver_flags::portid_volume_management()) {
+        if (mMasterMute || mStreamTypes[track->streamType()].mute ||
+            track->isPlaybackRestricted()) {
+            left = right = 0;
+        } else {
+            float typeVolume = mStreamTypes[track->streamType()].volume;
+            const float v = mMasterVolume * typeVolume * shaperVolume;
 
-        if (left > GAIN_FLOAT_UNITY) {
-            left = GAIN_FLOAT_UNITY;
-        }
-        if (right > GAIN_FLOAT_UNITY) {
-            right = GAIN_FLOAT_UNITY;
-        }
-        left *= v;
-        right *= v;
-        if (mAfThreadCallback->getMode() != AUDIO_MODE_IN_COMMUNICATION
+            if (left > GAIN_FLOAT_UNITY) {
+                left = GAIN_FLOAT_UNITY;
+            }
+            if (right > GAIN_FLOAT_UNITY) {
+                right = GAIN_FLOAT_UNITY;
+            }
+            left *= v;
+            right *= v;
+            if (mAfThreadCallback->getMode() != AUDIO_MODE_IN_COMMUNICATION
                 || audio_channel_count_from_out_mask(mChannelMask) > 1) {
-            left *= mMasterBalanceLeft; // DirectOutputThread balance applied as track volume
-            right *= mMasterBalanceRight;
+                left *= mMasterBalanceLeft; // DirectOutputThread balance applied as track volume
+                right *= mMasterBalanceRight;
+            }
         }
-    }
+        track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
+                /*muteState=*/{mMasterMute,
+                               mStreamTypes[track->streamType()].volume == 0.f,
+                               mStreamTypes[track->streamType()].mute,
+                               track->isPlaybackRestricted(),
+                               clientVolumeMute,
+                               shaperVolume == 0.f});
+    } else {
+        if (mMasterMute || track->isPlaybackRestricted()) {
+            left = right = 0;
+        } else {
+            float typeVolume = track->getPortVolume();
+            const float v = mMasterVolume * typeVolume * shaperVolume;
 
-    track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
-        /*muteState=*/{mMasterMute,
-                       mStreamTypes[track->streamType()].volume == 0.f,
-                       mStreamTypes[track->streamType()].mute,
-                       track->isPlaybackRestricted(),
-                       clientVolumeMute,
-                       shaperVolume == 0.f});
+            if (left > GAIN_FLOAT_UNITY) {
+                left = GAIN_FLOAT_UNITY;
+            }
+            if (right > GAIN_FLOAT_UNITY) {
+                right = GAIN_FLOAT_UNITY;
+            }
+            left *= v;
+            right *= v;
+            if (mAfThreadCallback->getMode() != AUDIO_MODE_IN_COMMUNICATION
+                || audio_channel_count_from_out_mask(mChannelMask) > 1) {
+                left *= mMasterBalanceLeft; // DirectOutputThread balance applied as track volume
+                right *= mMasterBalanceRight;
+            }
+        }
+        track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
+                /*muteState=*/{mMasterMute,
+                               track->getPortVolume() == 0.f,
+                               /* muteFromStreamMuted= */ false,
+                               track->isPlaybackRestricted(),
+                               clientVolumeMute,
+                               shaperVolume == 0.f});
+    }
 
     if (lastTrack) {
         track->setFinalVolume(left, right);
@@ -7843,7 +7923,9 @@
         ALOGE("addOutputTrack() initCheck failed %d", status);
         return;
     }
-    thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f);
+    if (!audioserver_flags::portid_volume_management()) {
+        thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f);
+    }
     mOutputTracks.add(outputTrack);
     ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread);
     updateWaitTime_l();
@@ -10330,6 +10412,7 @@
 
     const auto localSessionId = mSessionId;
     auto localAttr = mAttr;
+    float volume = 0.0f;
     if (isOutput()) {
         audio_config_t config = AUDIO_CONFIG_INITIALIZER;
         config.sample_rate = mSampleRate;
@@ -10353,7 +10436,8 @@
                                             &portId,
                                             &secondaryOutputs,
                                             &isSpatialized,
-                                            &isBitPerfect);
+                                            &isBitPerfect,
+                                            &volume);
         mutex().lock();
         mAttr = localAttr;
         ALOGD_IF(!secondaryOutputs.empty(),
@@ -10422,7 +10506,8 @@
             this, attr == nullptr ? mAttr : *attr, mSampleRate, mFormat,
                                         mChannelMask, mSessionId, isOutput(),
                                         client.attributionSource,
-                                        IPCThreadState::self()->getCallingPid(), portId);
+                                        IPCThreadState::self()->getCallingPid(), portId,
+                                        volume);
     if (!isOutput()) {
         track->setSilenced_l(isClientSilenced_l(portId));
     }
@@ -11007,18 +11092,18 @@
     mChannelCount = audio_channel_count_from_out_mask(mChannelMask);
     mMasterVolume = afThreadCallback->masterVolume_l();
     mMasterMute = afThreadCallback->masterMute_l();
-
-    for (int i = AUDIO_STREAM_MIN; i < AUDIO_STREAM_FOR_POLICY_CNT; ++i) {
-        const audio_stream_type_t stream{static_cast<audio_stream_type_t>(i)};
-        mStreamTypes[stream].volume = 0.0f;
-        mStreamTypes[stream].mute = mAfThreadCallback->streamMute_l(stream);
+    if (!audioserver_flags::portid_volume_management()) {
+        for (int i = AUDIO_STREAM_MIN; i < AUDIO_STREAM_FOR_POLICY_CNT; ++i) {
+            const audio_stream_type_t stream{static_cast<audio_stream_type_t>(i)};
+            mStreamTypes[stream].volume = 0.0f;
+            mStreamTypes[stream].mute = mAfThreadCallback->streamMute_l(stream);
+        }
+        // Audio patch and call assistant volume are always max
+        mStreamTypes[AUDIO_STREAM_PATCH].volume = 1.0f;
+        mStreamTypes[AUDIO_STREAM_PATCH].mute = false;
+        mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].volume = 1.0f;
+        mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].mute = false;
     }
-    // Audio patch and call assistant volume are always max
-    mStreamTypes[AUDIO_STREAM_PATCH].volume = 1.0f;
-    mStreamTypes[AUDIO_STREAM_PATCH].mute = false;
-    mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].volume = 1.0f;
-    mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].mute = false;
-
     if (mAudioHwDev) {
         if (mAudioHwDev->canSetMasterVolume()) {
             mMasterVolume = 1.0;
@@ -11097,6 +11182,21 @@
     }
 }
 
+status_t MmapPlaybackThread::setPortsVolume(
+        const std::vector<audio_port_handle_t>& portIds, float volume) {
+    audio_utils::lock_guard _l(mutex());
+    for (const auto& portId : portIds) {
+        for (const sp<IAfMmapTrack>& track : mActiveTracks) {
+            if (portId == track->portId()) {
+                track->setPortVolume(volume);
+                break;
+            }
+        }
+    }
+    broadcast_l();
+    return NO_ERROR;
+}
+
 void MmapPlaybackThread::invalidateTracks(audio_stream_type_t streamType)
 {
     audio_utils::lock_guard _l(mutex());
@@ -11130,14 +11230,26 @@
 void MmapPlaybackThread::processVolume_l()
 NO_THREAD_SAFETY_ANALYSIS // access of track->processMuteEvent_l
 {
-    float volume;
-
-    if (mMasterMute || streamMuted_l()) {
-        volume = 0;
+    float volume = 0;
+    if (!audioserver_flags::portid_volume_management()) {
+        if (mMasterMute || streamMuted_l()) {
+            volume = 0;
+        } else {
+            volume = mMasterVolume * streamVolume_l();
+        }
     } else {
-        volume = mMasterVolume * streamVolume_l();
+        if (mMasterMute) {
+            volume = 0;
+        } else {
+            // All mmap tracks are declared with the same audio attributes to the audio policy
+            // manager. Hence, they follow the same routing / volume group. Any change of volume
+            // will be broadcasted to all tracks. Thus, take arbitrarily first track volume.
+            size_t numtracks = mActiveTracks.size();
+            if (numtracks) {
+                volume = mMasterVolume * mActiveTracks[0]->getPortVolume();
+            }
+        }
     }
-
     if (volume != mHalVolFloat) {
         // Convert volumes from float to 8.24
         uint32_t vol = (uint32_t)(volume * (1 << 24));
@@ -11170,14 +11282,25 @@
         }
         for (const sp<IAfMmapTrack>& track : mActiveTracks) {
             track->setMetadataHasChanged();
-            track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
-                /*muteState=*/{mMasterMute,
-                               streamVolume_l() == 0.f,
-                               streamMuted_l(),
-                               // TODO(b/241533526): adjust logic to include mute from AppOps
-                               false /*muteFromPlaybackRestricted*/,
-                               false /*muteFromClientVolume*/,
-                               false /*muteFromVolumeShaper*/});
+            if (!audioserver_flags::portid_volume_management()) {
+                track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
+                        /*muteState=*/{mMasterMute,
+                        streamVolume_l() == 0.f,
+                        streamMuted_l(),
+                        // TODO(b/241533526): adjust logic to include mute from AppOps
+                        false /*muteFromPlaybackRestricted*/,
+                        false /*muteFromClientVolume*/,
+                        false /*muteFromVolumeShaper*/});
+            } else {
+                track->processMuteEvent_l(mAfThreadCallback->getOrCreateAudioManager(),
+                    /*muteState=*/{mMasterMute,
+                                   track->getPortVolume() == 0.f,
+                                   /* muteFromStreamMuted= */ false,
+                                   // TODO(b/241533526): adjust logic to include mute from AppOps
+                                   false /*muteFromPlaybackRestricted*/,
+                                   false /*muteFromClientVolume*/,
+                                   false /*muteFromVolumeShaper*/});
+                }
         }
     }
 }
@@ -11284,9 +11407,13 @@
 void MmapPlaybackThread::dumpInternals_l(int fd, const Vector<String16>& args)
 {
     MmapThread::dumpInternals_l(fd, args);
-
-    dprintf(fd, "  Stream type: %d Stream volume: %f HAL volume: %f Stream mute %d\n",
-            mStreamType, streamVolume_l(), mHalVolFloat, streamMuted_l());
+    if (!audioserver_flags::portid_volume_management()) {
+        dprintf(fd, "  Stream type: %d Stream volume: %f HAL volume: %f Stream mute %d",
+                mStreamType, streamVolume_l(), mHalVolFloat, streamMuted_l());
+    } else {
+        dprintf(fd, "  HAL volume: %f", mHalVolFloat);
+    }
+    dprintf(fd, "\n");
     dprintf(fd, "  Master volume: %f Master mute %d\n", mMasterVolume, mMasterMute);
 }
 
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 654b841..89e41c8 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -836,6 +836,12 @@
                     typename SortedVector<sp<T>>::iterator end() {
                         return mActiveTracks.end();
                     }
+                    typename SortedVector<const sp<T>>::iterator begin() const {
+                        return mActiveTracks.begin();
+                    }
+                    typename SortedVector<const sp<T>>::iterator end() const {
+                        return mActiveTracks.end();
+                    }
 
                     // Due to Binder recursion optimization, clear() and updatePowerState()
                     // cannot be called from a Binder thread because they may call back into
@@ -1011,6 +1017,9 @@
     void setStreamVolume(audio_stream_type_t stream, float value) final EXCLUDES_ThreadBase_Mutex;
     void setStreamMute(audio_stream_type_t stream, bool muted) final EXCLUDES_ThreadBase_Mutex;
     float streamVolume(audio_stream_type_t stream) const final EXCLUDES_ThreadBase_Mutex;
+    status_t setPortsVolume(const std::vector<audio_port_handle_t>& portIds, float volume)
+            final EXCLUDES_ThreadBase_Mutex;
+
     void setVolumeForOutput_l(float left, float right) const final;
 
     sp<IAfTrack> createTrack_l(
@@ -1035,7 +1044,8 @@
                                 const sp<media::IAudioTrackCallback>& callback,
                                 bool isSpatialized,
                                 bool isBitPerfect,
-                                audio_output_flags_t* afTrackFlags) final
+                                audio_output_flags_t* afTrackFlags,
+                                float volume) final
             REQUIRES(audio_utils::AudioFlinger_Mutex);
 
     bool isTrackActive(const sp<IAfTrack>& track) const final {
@@ -2385,6 +2395,8 @@
     void setStreamVolume(audio_stream_type_t stream, float value) final EXCLUDES_ThreadBase_Mutex;
     void setStreamMute(audio_stream_type_t stream, bool muted) final EXCLUDES_ThreadBase_Mutex;
     float streamVolume(audio_stream_type_t stream) const final EXCLUDES_ThreadBase_Mutex;
+    status_t setPortsVolume(const std::vector<audio_port_handle_t>& portIds, float volume)
+            final EXCLUDES_ThreadBase_Mutex;
 
     void setMasterMute_l(bool muted) REQUIRES(mutex()) { mMasterMute = muted; }
 
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index a0b85f7..1342b7b 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -347,7 +347,7 @@
     size_t              mBufferSize; // size of mBuffer in bytes
     // we don't really need a lock for these
     MirroredVariable<track_state>  mState;
-    const audio_attributes_t mAttr;
+    audio_attributes_t  mAttr;
     const uint32_t      mSampleRate;    // initial sample rate only; for tracks which
                         // support dynamic rates, the current value is in control block
     const audio_format_t mFormat;
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index f5f11cc..51e140d 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -715,7 +715,8 @@
         size_t frameCountToBeReady,
         float speed,
         bool isSpatialized,
-        bool isBitPerfect) {
+        bool isBitPerfect,
+        float volume) {
     return sp<Track>::make(thread,
             client,
             streamType,
@@ -736,7 +737,8 @@
             frameCountToBeReady,
             speed,
             isSpatialized,
-            isBitPerfect);
+            isBitPerfect,
+            volume);
 }
 
 // Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
@@ -761,7 +763,8 @@
             size_t frameCountToBeReady,
             float speed,
             bool isSpatialized,
-            bool isBitPerfect)
+            bool isBitPerfect,
+            float volume)
     :   TrackBase(thread, client, attr, sampleRate, format, channelMask, frameCount,
                   // TODO: Using unsecurePointer() has some associated security pitfalls
                   //       (see declaration for details).
@@ -797,7 +800,8 @@
     mFlags(flags),
     mSpeed(speed),
     mIsSpatialized(isSpatialized),
-    mIsBitPerfect(isBitPerfect)
+    mIsBitPerfect(isBitPerfect),
+    mVolume(volume)
 {
     // client == 0 implies sharedBuffer == 0
     ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
@@ -843,6 +847,14 @@
         thread->fastTrackAvailMask_l() &= ~(1 << i);
     }
 
+    populateUsageAndContentTypeFromStreamType();
+
+    // Audio patch and call assistant volume are always max
+    if (mAttr.usage == AUDIO_USAGE_CALL_ASSISTANT
+            || mAttr.usage == AUDIO_USAGE_VIRTUAL_SOURCE) {
+        mVolume = 1.0f;
+    }
+
     mServerLatencySupported = checkServerLatencySupported(format, flags);
 #ifdef TEE_SINK
     mTee.setId(std::string("_") + std::to_string(mThreadIoHandle)
@@ -865,6 +877,62 @@
     mTrackMetrics.logConstructor(creatorPid, uid, id(), traits, streamType);
 }
 
+// When attributes are undefined, derive default values from stream type.
+// See AudioAttributes.java, usageForStreamType() and Builder.setInternalLegacyStreamType()
+void Track::populateUsageAndContentTypeFromStreamType() {
+    if (mAttr.usage == AUDIO_USAGE_UNKNOWN) {
+        switch (mStreamType) {
+        case AUDIO_STREAM_VOICE_CALL:
+            mAttr.usage = AUDIO_USAGE_VOICE_COMMUNICATION;
+            mAttr.content_type = AUDIO_CONTENT_TYPE_SPEECH;
+            break;
+        case AUDIO_STREAM_SYSTEM:
+            mAttr.usage = AUDIO_USAGE_ASSISTANCE_SONIFICATION;
+            mAttr.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+            break;
+        case AUDIO_STREAM_RING:
+            mAttr.usage = AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE;
+            mAttr.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+            break;
+        case AUDIO_STREAM_MUSIC:
+            mAttr.usage = AUDIO_USAGE_MEDIA;
+            mAttr.content_type = AUDIO_CONTENT_TYPE_MUSIC;
+            break;
+        case AUDIO_STREAM_ALARM:
+            mAttr.usage = AUDIO_USAGE_ALARM;
+            mAttr.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+            break;
+        case AUDIO_STREAM_NOTIFICATION:
+            mAttr.usage = AUDIO_USAGE_NOTIFICATION;
+            mAttr.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+            break;
+        case AUDIO_STREAM_DTMF:
+            mAttr.usage = AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING;
+            mAttr.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+            break;
+        case AUDIO_STREAM_ACCESSIBILITY:
+            mAttr.usage = AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY;
+            mAttr.content_type = AUDIO_CONTENT_TYPE_SPEECH;
+            break;
+        case AUDIO_STREAM_ASSISTANT:
+            mAttr.usage = AUDIO_USAGE_ASSISTANT;
+            mAttr.content_type = AUDIO_CONTENT_TYPE_SPEECH;
+            break;
+        case AUDIO_STREAM_REROUTING:
+        case AUDIO_STREAM_PATCH:
+            mAttr.usage = AUDIO_USAGE_VIRTUAL_SOURCE;
+            // unknown content type
+            break;
+        case AUDIO_STREAM_CALL_ASSISTANT:
+            mAttr.usage = AUDIO_USAGE_CALL_ASSISTANT;
+            mAttr.content_type = AUDIO_CONTENT_TYPE_SPEECH;
+            break;
+        default:
+            break;
+        }
+    }
+}
+
 Track::~Track()
 {
     ALOGV("%s(%d)", __func__, mId);
@@ -923,7 +991,7 @@
     result.appendFormat("Type     Id Active Client Session Port Id S  Flags "
                         "  Format Chn mask  SRate "
                         "ST Usg CT "
-                        " G db  L dB  R dB  VS dB "
+                        " G db  L dB  R dB  VS dB  PortVol dB "
                         "  Server FrmCnt  FrmRdy F Underruns  Flushed BitPerfect InternalMute"
                         "%s\n",
                         isServerLatencySupported() ? "   Latency" : "");
@@ -1009,7 +1077,7 @@
     result.appendFormat("%7s %6u %7u %7u %2s 0x%03X "
                         "%08X %08X %6u "
                         "%2u %3x %2x "
-                        "%5.2g %5.2g %5.2g %5.2g%c "
+                        "%5.2g %5.2g %5.2g %5.2g%c %11.2g "
                         "%08X %6zu%c %6zu %c %9u%c %7u %10s %12s",
             active ? "yes" : "no",
             (mClient == 0) ? getpid() : mClient->pid(),
@@ -1031,6 +1099,7 @@
             20.0 * log10(float_from_gain(gain_minifloat_unpack_right(vlr))),
             20.0 * log10(vsVolume.first), // VolumeShaper(s) total volume
             vsVolume.second ? 'A' : ' ',  // if any VolumeShapers active
+            20.0 * log10(mVolume),
 
             mCblk->mServer,
             bufferSizeInFrames,
@@ -1587,59 +1656,6 @@
             .gain = mFinalVolume,
     };
 
-    // When attributes are undefined, derive default values from stream type.
-    // See AudioAttributes.java, usageForStreamType() and Builder.setInternalLegacyStreamType()
-    if (mAttr.usage == AUDIO_USAGE_UNKNOWN) {
-        switch (mStreamType) {
-        case AUDIO_STREAM_VOICE_CALL:
-            metadata.base.usage = AUDIO_USAGE_VOICE_COMMUNICATION;
-            metadata.base.content_type = AUDIO_CONTENT_TYPE_SPEECH;
-            break;
-        case AUDIO_STREAM_SYSTEM:
-            metadata.base.usage = AUDIO_USAGE_ASSISTANCE_SONIFICATION;
-            metadata.base.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
-            break;
-        case AUDIO_STREAM_RING:
-            metadata.base.usage = AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE;
-            metadata.base.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
-            break;
-        case AUDIO_STREAM_MUSIC:
-            metadata.base.usage = AUDIO_USAGE_MEDIA;
-            metadata.base.content_type = AUDIO_CONTENT_TYPE_MUSIC;
-            break;
-        case AUDIO_STREAM_ALARM:
-            metadata.base.usage = AUDIO_USAGE_ALARM;
-            metadata.base.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
-            break;
-        case AUDIO_STREAM_NOTIFICATION:
-            metadata.base.usage = AUDIO_USAGE_NOTIFICATION;
-            metadata.base.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
-            break;
-        case AUDIO_STREAM_DTMF:
-            metadata.base.usage = AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING;
-            metadata.base.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
-            break;
-        case AUDIO_STREAM_ACCESSIBILITY:
-            metadata.base.usage = AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY;
-            metadata.base.content_type = AUDIO_CONTENT_TYPE_SPEECH;
-            break;
-        case AUDIO_STREAM_ASSISTANT:
-            metadata.base.usage = AUDIO_USAGE_ASSISTANT;
-            metadata.base.content_type = AUDIO_CONTENT_TYPE_SPEECH;
-            break;
-        case AUDIO_STREAM_REROUTING:
-            metadata.base.usage = AUDIO_USAGE_VIRTUAL_SOURCE;
-            // unknown content type
-            break;
-        case AUDIO_STREAM_CALL_ASSISTANT:
-            metadata.base.usage = AUDIO_USAGE_CALL_ASSISTANT;
-            metadata.base.content_type = AUDIO_CONTENT_TYPE_SPEECH;
-            break;
-        default:
-            break;
-        }
-    }
-
     metadata.channel_mask = mChannelMask;
     strncpy(metadata.tags, mAttr.tags, AUDIO_ATTRIBUTES_TAGS_MAX_SIZE);
     *backInserter++ = metadata;
@@ -2191,14 +2207,13 @@
             size_t frameCount,
             const AttributionSourceState& attributionSource)
     :   Track(playbackThread, NULL, AUDIO_STREAM_PATCH,
-              audio_attributes_t{} /* currently unused for output track */,
+              AUDIO_ATTRIBUTES_INITIALIZER ,
               sampleRate, format, channelMask, frameCount,
               nullptr /* buffer */, (size_t)0 /* bufferSize */, nullptr /* sharedBuffer */,
               AUDIO_SESSION_NONE, getpid(), attributionSource, AUDIO_OUTPUT_FLAG_NONE,
               TYPE_OUTPUT),
     mActive(false), mSourceThread(sourceThread)
 {
-
     if (mCblk != NULL) {
         mOutBuffer.frameCount = 0;
         playbackThread->addOutputTrack_l(this);
@@ -2464,7 +2479,8 @@
                                          *  as soon as possible to have
                                          *  the lowest possible latency
                                          *  even if it might glitch. */
-        float speed)
+        float speed,
+        float volume)
 {
     return sp<PatchTrack>::make(
             playbackThread,
@@ -2478,7 +2494,8 @@
             flags,
             timeout,
             frameCountToBeReady,
-            speed);
+            speed,
+            volume);
 }
 
 PatchTrack::PatchTrack(IAfPlaybackThread* playbackThread,
@@ -2492,13 +2509,15 @@
                                                      audio_output_flags_t flags,
                                                      const Timeout& timeout,
                                                      size_t frameCountToBeReady,
-                                                     float speed)
+                                                     float speed,
+                                                     float volume)
     :   Track(playbackThread, NULL, streamType,
-              audio_attributes_t{} /* currently unused for patch track */,
+              AUDIO_ATTRIBUTES_INITIALIZER,
               sampleRate, format, channelMask, frameCount,
               buffer, bufferSize, nullptr /* sharedBuffer */,
               AUDIO_SESSION_NONE, getpid(), audioServerAttributionSource(getpid()), flags,
-              TYPE_PATCH, AUDIO_PORT_HANDLE_NONE, frameCountToBeReady, speed),
+              TYPE_PATCH, AUDIO_PORT_HANDLE_NONE, frameCountToBeReady, speed,
+              false /*isSpatialized*/, false /*isBitPerfect*/, volume),
         PatchTrackBase(mCblk ? new AudioTrackClientProxy(mCblk, mBuffer, frameCount, mFrameSize,
                         true /*clientInServer*/) : nullptr,
                        playbackThread, timeout)
@@ -3482,7 +3501,8 @@
           bool isOut,
           const android::content::AttributionSourceState& attributionSource,
           pid_t creatorPid,
-          audio_port_handle_t portId)
+          audio_port_handle_t portId,
+          float volume)
 {
     return sp<MmapTrack>::make(
             thread,
@@ -3494,7 +3514,8 @@
             isOut,
             attributionSource,
             creatorPid,
-            portId);
+            portId,
+            volume);
 }
 
 MmapTrack::MmapTrack(IAfThreadBase* thread,
@@ -3506,7 +3527,8 @@
         bool isOut,
         const AttributionSourceState& attributionSource,
         pid_t creatorPid,
-        audio_port_handle_t portId)
+        audio_port_handle_t portId,
+        float volume)
     :   TrackBase(thread, NULL, attr, sampleRate, format,
                   channelMask, (size_t)0 /* frameCount */,
                   nullptr /* buffer */, (size_t)0 /* bufferSize */,
@@ -3517,10 +3539,15 @@
                   TYPE_DEFAULT, portId,
                   std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_MMAP) + std::to_string(portId)),
         mPid(VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.pid))),
-            mSilenced(false), mSilencedNotified(false)
+            mSilenced(false), mSilencedNotified(false), mVolume(volume)
 {
     // Once this item is logged by the server, the client can add properties.
     mTrackMetrics.logConstructor(creatorPid, uid(), id());
+    if (isOut && (attr.usage == AUDIO_USAGE_CALL_ASSISTANT
+            || attr.usage == AUDIO_USAGE_VIRTUAL_SOURCE)) {
+        // Audio patch and call assistant volume are always max
+        mVolume = 1.0f;
+    }
 }
 
 MmapTrack::~MmapTrack()
@@ -3599,8 +3626,8 @@
 
 void MmapTrack::appendDumpHeader(String8& result) const
 {
-    result.appendFormat("Client Session Port Id  Format Chn mask  SRate Flags %s\n",
-                        isOut() ? "Usg CT": "Source");
+    result.appendFormat("Client Session Port Id  Format Chn mask  SRate Flags %s  %s\n",
+                        isOut() ? "Usg CT": "Source", isOut() ? "PortVol dB" : "");
 }
 
 void MmapTrack::appendDump(String8& result, bool active __unused) const
@@ -3615,6 +3642,7 @@
             mAttr.flags);
     if (isOut()) {
         result.appendFormat("%3x %2x", mAttr.usage, mAttr.content_type);
+        result.appendFormat("%11.2g", 20.0 * log10(mVolume));
     } else {
         result.appendFormat("%6x", mAttr.source);
     }
diff --git a/services/audioflinger/datapath/VolumePortInterface.h b/services/audioflinger/datapath/VolumePortInterface.h
new file mode 100644
index 0000000..fb1c463
--- /dev/null
+++ b/services/audioflinger/datapath/VolumePortInterface.h
@@ -0,0 +1,29 @@
+/*
+ * Copyright (C) 2024 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.
+ */
+
+#pragma once
+
+#include <system/audio.h>
+
+namespace android {
+
+class VolumePortInterface : public virtual RefBase {
+public:
+    virtual void setPortVolume(float volume) = 0;
+    virtual float getPortVolume() const = 0;
+};
+
+}  // namespace android
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index deb7345..8e8fac8 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -147,7 +147,8 @@
                                       std::vector<audio_io_handle_t> *secondaryOutputs,
                                       output_type_t *outputType,
                                       bool *isSpatialized,
-                                      bool *isBitPerfect) = 0;
+                                      bool *isBitPerfect,
+                                      float *volume) = 0;
     // indicates to the audio policy manager that the output starts being used by corresponding
     // stream.
     virtual status_t startOutput(audio_port_handle_t portId) = 0;
@@ -514,6 +515,18 @@
     // for each output (destination device) it is attached to.
     virtual status_t setStreamVolume(audio_stream_type_t stream, float volume,
                                      audio_io_handle_t output, int delayMs = 0) = 0;
+    /**
+     * Set volume for given AudioTrack port ids for a particular output.
+     * For the same user setting, a given volume group and associated output port id
+     * can have different volumes for each output (destination device) it is attached to.
+     * @param ports to consider
+     * @param volume to apply
+     * @param output to consider
+     * @param delayMs to use
+     * @return NO_ERROR if successful
+     */
+    virtual status_t setPortsVolume(const std::vector<audio_port_handle_t>& ports, float volume,
+            audio_io_handle_t output, int delayMs = 0) = 0;
 
     // function enabling to send proprietary informations directly from audio policy manager to
     // audio hardware interface.
diff --git a/services/audiopolicy/common/managerdefinitions/Android.bp b/services/audiopolicy/common/managerdefinitions/Android.bp
index 051e975..4dedcd6 100644
--- a/services/audiopolicy/common/managerdefinitions/Android.bp
+++ b/services/audiopolicy/common/managerdefinitions/Android.bp
@@ -39,6 +39,8 @@
         "android.media.audiopolicy-aconfig-cc",
         "audioclient-types-aidl-cpp",
         "audiopolicy-types-aidl-cpp",
+        "com.android.media.audioserver-aconfig-cc",
+        "libaconfig_storage_read_api_cc",
         "libaudioclient_aidl_conversion",
         "libaudiofoundation",
         "libaudiopolicy",
@@ -51,6 +53,7 @@
         "libmedia_helper",
         "libutils",
         "libxml2",
+        "server_configurable_flags",
     ],
     export_shared_lib_headers: [
         "libaudiofoundation",
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
index 914f3fe..203fa80 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
@@ -490,6 +490,13 @@
 
     virtual std::string info() const override;
 
+    /**
+     * Finds all ports matching the given volume source.
+     * @param vs to be considered
+     * @return vector of ports following the given volume source.
+     */
+    std::vector<audio_port_handle_t> getPortsForVolumeSource(const VolumeSource& vs);
+
     const sp<IOProfile> mProfile;          // I/O profile this output derives from
     audio_io_handle_t mIoHandle;           // output handle
     uint32_t mLatency;                  //
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 0131ba0..a0f1006 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -27,6 +27,7 @@
 #include "HwModule.h"
 #include "TypeConverter.h"
 #include "policy.h"
+#include <com_android_media_audioserver.h>
 #include <media/AudioGain.h>
 #include <media/AudioParameter.h>
 #include <media/AudioPolicy.h>
@@ -34,6 +35,8 @@
 // A device mask for all audio output devices that are considered "remote" when evaluating
 // active output devices in isStreamActiveRemotely()
 
+namespace audioserver_flags = com::android::media::audioserver;
+
 namespace android {
 
 static const DeviceTypeSet& getAllOutRemoteDevices() {
@@ -498,17 +501,33 @@
         const DeviceTypeSet& deviceTypes, uint32_t delayMs) {
     // volume source active and more than one volume source is active, otherwise, no-op or let
     // setVolume controlling SW and/or HW Gains
-    if (!streamTypes.empty() && isActive(vs) && (getActiveVolumeSources().size() > 1)) {
-        for (const auto& devicePort : devices()) {
-            if (isSingleDeviceType(deviceTypes, devicePort->type()) &&
+    if (!audioserver_flags::portid_volume_management()) {
+        if (!streamTypes.empty() && isActive(vs) && (getActiveVolumeSources().size() > 1)) {
+            for (const auto& devicePort : devices()) {
+                if (isSingleDeviceType(deviceTypes, devicePort->type()) &&
                     devicePort->hasGainController(true /*canUseForVolume*/)) {
-                float volumeAmpl = muted ? 0.0f : Volume::DbToAmpl(0);
-                ALOGV("%s: output: %d, vs: %d, muted: %d, active vs count: %zu", __func__,
-                      mIoHandle, vs, muted, getActiveVolumeSources().size());
-                for (const auto &stream : streamTypes) {
-                    mClientInterface->setStreamVolume(stream, volumeAmpl, mIoHandle, delayMs);
+                    float volumeAmpl = muted ? 0.0f : Volume::DbToAmpl(0);
+                    ALOGV("%s: output: %d, vs: %d, muted: %d, active vs count: %zu", __func__,
+                          mIoHandle, vs, muted, getActiveVolumeSources().size());
+                    for (const auto &stream : streamTypes) {
+                        mClientInterface->setStreamVolume(stream, volumeAmpl, mIoHandle, delayMs);
+                    }
+                    return;
                 }
-                return;
+            }
+        }
+    } else {
+        if (isActive(vs) && (getActiveVolumeSources().size() > 1)) {
+            for (const auto &devicePort: devices()) {
+                if (isSingleDeviceType(deviceTypes, devicePort->type()) &&
+                    devicePort->hasGainController(true /*canUseForVolume*/)) {
+                    float volumeAmpl = muted ? 0.0f : Volume::DbToAmpl(0);
+                    ALOGV("%s: output: %d, vs: %d, muted: %d, active vs count: %zu", __func__,
+                          mIoHandle, vs, muted, getActiveVolumeSources().size());
+                    mClientInterface->setPortsVolume(
+                            getPortsForVolumeSource(vs), volumeAmpl, mIoHandle, delayMs);
+                    return;
+                }
             }
         }
     }
@@ -528,8 +547,14 @@
             VolumeSource callVolSrc = getVoiceSource();
             if (callVolSrc != VOLUME_SOURCE_NONE && volumeDb != getCurVolume(callVolSrc)) {
                 setCurVolume(callVolSrc, volumeDb, true);
-                mClientInterface->setStreamVolume(
-                        AUDIO_STREAM_VOICE_CALL, Volume::DbToAmpl(volumeDb), mIoHandle, delayMs);
+                float volumeAmpl = Volume::DbToAmpl(volumeDb);
+                if (audioserver_flags::portid_volume_management()) {
+                    mClientInterface->setPortsVolume(getPortsForVolumeSource(callVolSrc),
+                            volumeAmpl, mIoHandle, delayMs);
+                } else {
+                    mClientInterface->setStreamVolume(AUDIO_STREAM_VOICE_CALL,
+                            volumeAmpl, mIoHandle, delayMs);
+                }
             }
         }
         return false;
@@ -539,25 +564,34 @@
     }
     for (const auto& devicePort : devices()) {
         // APM loops on all group, so filter on active group to set the port gain,
-        // let the other groups set the stream volume as per legacy
+        // let the other groups set the sw volume as per legacy
         // TODO: Pass in the device address and check against it.
         if (isSingleDeviceType(deviceTypes, devicePort->type()) &&
                 devicePort->hasGainController(true) && isActive(vs)) {
             ALOGV("%s: device %s has gain controller", __func__, devicePort->toString().c_str());
             // @todo: here we might be in trouble if the SwOutput has several active clients with
             // different Volume Source (or if we allow several curves within same volume group)
-            //
-            // @todo: default stream volume to max (0) when using HW Port gain?
-            // Allows to set SW Gain on AudioFlinger if:
-            //    -volume group has explicit stream(s) associated
-            //    -volume group with no explicit stream(s) is the only active source on this output
-            // Allows to mute SW Gain on AudioFlinger only for volume group with explicit stream(s)
-            if (!streamTypes.empty() || (getActiveVolumeSources().size() == 1)) {
-                const bool canMute = muted && (volumeDb != 0.0f) && !streamTypes.empty();
-                float volumeAmpl = canMute ? 0.0f : Volume::DbToAmpl(0);
-                for (const auto &stream : streams) {
-                    mClientInterface->setStreamVolume(stream, volumeAmpl, mIoHandle, delayMs);
+            if (!audioserver_flags::portid_volume_management()) {
+                // @todo: default stream volume to max (0) when using HW Port gain?
+                // Allows to set SW Gain on AudioFlinger if:
+                //    -volume group has explicit stream(s) associated
+                //    -volume group with no explicit stream(s) is the only active source on this
+                //    output
+                // Allows to mute SW Gain on AudioFlinger only for volume group with explicit
+                // stream(s)
+                if (!streamTypes.empty() || (getActiveVolumeSources().size() == 1)) {
+                    const bool canMute = muted && (volumeDb != 0.0f) && !streamTypes.empty();
+                    float volumeAmpl = canMute ? 0.0f : Volume::DbToAmpl(0);
+                    for (const auto &stream: streams) {
+                        mClientInterface->setStreamVolume(stream, volumeAmpl, mIoHandle, delayMs);
+                    }
                 }
+            } else {
+                float volumeAmpl = (muted && volumeDb != 0.0f) ? 0.0f : Volume::DbToAmpl(0);
+                ALOGV("%s: output: %d, vs: %d, active vs count: %zu", __func__,
+                      mIoHandle, vs, getActiveVolumeSources().size());
+                mClientInterface->setPortsVolume(
+                        getPortsForVolumeSource(vs), volumeAmpl, mIoHandle, delayMs);
             }
             AudioGains gains = devicePort->getGains();
             int gainMinValueInMb = gains[0]->getMinValueInMb();
@@ -577,20 +611,47 @@
     // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is enabled
     float volumeAmpl = Volume::DbToAmpl(getCurVolume(vs));
     if (hasStream(streams, AUDIO_STREAM_BLUETOOTH_SCO)) {
-        mClientInterface->setStreamVolume(AUDIO_STREAM_VOICE_CALL, volumeAmpl, mIoHandle, delayMs);
         VolumeSource callVolSrc = getVoiceSource();
+        if (audioserver_flags::portid_volume_management()) {
+            if (callVolSrc != VOLUME_SOURCE_NONE) {
+                mClientInterface->setPortsVolume(getPortsForVolumeSource(callVolSrc), volumeAmpl,
+                        mIoHandle, delayMs);
+            }
+        } else {
+            mClientInterface->setStreamVolume(AUDIO_STREAM_VOICE_CALL, volumeAmpl, mIoHandle,
+                    delayMs);
+        }
         if (callVolSrc != VOLUME_SOURCE_NONE) {
             setCurVolume(callVolSrc, getCurVolume(vs), true);
         }
     }
-    for (const auto &stream : streams) {
-        ALOGV("%s output %d for volumeSource %d, volume %f, delay %d stream=%s", __func__,
-              mIoHandle, vs, volumeDb, delayMs, toString(stream).c_str());
-        mClientInterface->setStreamVolume(stream, volumeAmpl, mIoHandle, delayMs);
+    if (audioserver_flags::portid_volume_management()) {
+        ALOGV("%s output %d for volumeSource %d, volume %f, delay %d active=%d", __func__,
+              mIoHandle, vs, volumeDb, delayMs, isActive(vs));
+        mClientInterface->setPortsVolume(getPortsForVolumeSource(vs), volumeAmpl, mIoHandle,
+                                         delayMs);
+    } else {
+        for (const auto &stream : streams) {
+            ALOGV("%s output %d for volumeSource %d, volume %f, delay %d stream=%s", __func__,
+                  mIoHandle, vs, volumeDb, delayMs, toString(stream).c_str());
+            mClientInterface->setStreamVolume(stream, volumeAmpl, mIoHandle, delayMs);
+        }
     }
     return true;
 }
 
+std::vector<audio_port_handle_t> SwAudioOutputDescriptor::getPortsForVolumeSource(
+        const VolumeSource& vs)
+{
+    std::vector<audio_port_handle_t> portsForVolumeSource;
+    for (const auto& client : getClientIterable()) {
+        if (client->volumeSource() == vs) {
+            portsForVolumeSource.push_back(client->portId());
+        }
+    }
+    return portsForVolumeSource;
+}
+
 status_t SwAudioOutputDescriptor::open(const audio_config_t *halConfig,
                                        const audio_config_base_t *mixerConfig,
                                        const DeviceVector &devices,
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/plugin/Android.bp b/services/audiopolicy/engineconfigurable/parameter-framework/plugin/Android.bp
index 3dc2229..c9a77a4 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/plugin/Android.bp
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/plugin/Android.bp
@@ -38,6 +38,8 @@
     shared_libs: [
         "libaudiopolicycomponents",
         "libaudiopolicyengineconfigurable",
+        "libbase",
+        "libcutils",
         "liblog",
         "libmedia_helper",
         "libparameter",
diff --git a/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp b/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp
index 6416a47..fd40c04 100644
--- a/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp
+++ b/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp
@@ -265,6 +265,7 @@
     AudioPolicyInterface::output_type_t outputType;
     bool isSpatialized;
     bool isBitPerfect;
+    float volume;
 
     // TODO b/182392769: use attribution source util
     AttributionSourceState attributionSource;
@@ -272,7 +273,7 @@
     attributionSource.token = sp<BBinder>::make();
     if (mManager->getOutputForAttr(&attr, output, AUDIO_SESSION_NONE, &stream, attributionSource,
             &config, &flags, selectedDeviceId, portId, {}, &outputType, &isSpatialized,
-            &isBitPerfect) != OK) {
+            &isBitPerfect, &volume) != OK) {
         return false;
     }
     if (*output == AUDIO_IO_HANDLE_NONE || *portId == AUDIO_PORT_HANDLE_NONE) {
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 739e201..7cc6791 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -1488,7 +1488,8 @@
                                               std::vector<audio_io_handle_t> *secondaryOutputs,
                                               output_type_t *outputType,
                                               bool *isSpatialized,
-                                              bool *isBitPerfect)
+                                              bool *isBitPerfect,
+                                              float *volume)
 {
     // The supplied portId must be AUDIO_PORT_HANDLE_NONE
     if (*portId != AUDIO_PORT_HANDLE_NONE) {
@@ -1544,6 +1545,8 @@
                                   outputDesc->mPolicyMix);
     outputDesc->addClient(clientDesc);
 
+    *volume = Volume::DbToAmpl(outputDesc->getCurVolume(toVolumeSource(resultAttr)));
+
     ALOGV("%s() returns output %d requestedPortId %d selectedDeviceId %d for port ID %d", __func__,
           *output, requestedPortId, *selectedDeviceId, *portId);
 
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 98853ce..a67ba78 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -128,7 +128,8 @@
                                   std::vector<audio_io_handle_t> *secondaryOutputs,
                                   output_type_t *outputType,
                                   bool *isSpatialized,
-                                  bool *isBitPerfect) override;
+                                  bool *isBitPerfect,
+                                  float *volume) override;
         virtual status_t startOutput(audio_port_handle_t portId);
         virtual status_t stopOutput(audio_port_handle_t portId);
         virtual bool releaseOutput(audio_port_handle_t portId);
diff --git a/services/audiopolicy/service/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
index f70dc52..5008d68 100644
--- a/services/audiopolicy/service/AudioPolicyClientImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
@@ -188,6 +188,16 @@
                                                delay_ms);
 }
 
+status_t AudioPolicyService::AudioPolicyClient::setPortsVolume(
+        const std::vector<audio_port_handle_t> &ports, float volume, audio_io_handle_t output,
+        int delayMs)
+{
+    if (ports.empty()) {
+        return NO_ERROR;
+    }
+    return mAudioPolicyService->setPortsVolume(ports, volume, output, delayMs);
+}
+
 void AudioPolicyService::AudioPolicyClient::setParameters(audio_io_handle_t io_handle,
                    const String8& keyValuePairs,
                    int delay_ms)
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index f414862..6194002 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -423,6 +423,7 @@
     AudioPolicyInterface::output_type_t outputType;
     bool isSpatialized = false;
     bool isBitPerfect = false;
+    float volume;
     status_t result = mAudioPolicyManager->getOutputForAttr(&attr, &output, session,
                                                             &stream,
                                                             attributionSource,
@@ -431,7 +432,8 @@
                                                             &secondaryOutputs,
                                                             &outputType,
                                                             &isSpatialized,
-                                                            &isBitPerfect);
+                                                            &isBitPerfect,
+                                                            &volume);
 
     // FIXME: Introduce a way to check for the the telephony device before opening the output
     if (result == NO_ERROR) {
@@ -495,6 +497,7 @@
         _aidl_return->isBitPerfect = isBitPerfect;
         _aidl_return->attr = VALUE_OR_RETURN_BINDER_STATUS(
                 legacy2aidl_audio_attributes_t_AudioAttributes(attr));
+        _aidl_return->volume = volume;
     } else {
         _aidl_return->configBase.format = VALUE_OR_RETURN_BINDER_STATUS(
                 legacy2aidl_audio_format_t_AudioFormatDescription(config.format));
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index cc67481..a8b7954 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -1815,6 +1815,16 @@
                                                                     data->mIO);
                     ul.lock();
                     }break;
+                case SET_PORTS_VOLUME: {
+                    VolumePortsData *data = (VolumePortsData *)command->mParam.get();
+                    ALOGV("AudioCommandThread() processing set volume Ports %s volume %f, \
+                            output %d", data->dumpPorts().c_str(), data->mVolume, data->mIO);
+                    ul.unlock();
+                    command->mStatus = AudioSystem::setPortsVolume(data->mPorts,
+                                                                   data->mVolume,
+                                                                   data->mIO);
+                    ul.lock();
+                    } break;
                 case SET_PARAMETERS: {
                     ParametersData *data = (ParametersData *)command->mParam.get();
                     ALOGV("AudioCommandThread() processing set parameters string %s, io %d",
@@ -2127,6 +2137,23 @@
     return sendCommand(command, delayMs);
 }
 
+status_t AudioPolicyService::AudioCommandThread::volumePortsCommand(
+        const std::vector<audio_port_handle_t> &ports, float volume, audio_io_handle_t output,
+        int delayMs)
+{
+    sp<AudioCommand> command = new AudioCommand();
+    command->mCommand = SET_PORTS_VOLUME;
+    sp<VolumePortsData> data = new VolumePortsData();
+    data->mPorts = ports;
+    data->mVolume = volume;
+    data->mIO = output;
+    command->mParam = data;
+    command->mWaitStatus = true;
+    ALOGV("AudioCommandThread() adding set volume ports %s, volume %f, output %d",
+            data->dumpPorts().c_str(), volume, output);
+    return sendCommand(command, delayMs);
+}
+
 status_t AudioPolicyService::AudioCommandThread::parametersCommand(audio_io_handle_t ioHandle,
                                                                    const char *keyValuePairs,
                                                                    int delayMs)
@@ -2457,6 +2484,31 @@
             delayMs = 1;
         } break;
 
+        case SET_PORTS_VOLUME: {
+            VolumePortsData *data = (VolumePortsData *)command->mParam.get();
+            VolumePortsData *data2 = (VolumePortsData *)command2->mParam.get();
+            if (data->mIO != data2->mIO) break;
+            // Can remove command only if port ids list is the same, otherwise, remove from
+            // command 2 all port whose volume will be replaced with command 1 volume.
+            std::vector<audio_port_handle_t> portsOnlyInCommand2{};
+            std::copy_if(data2->mPorts.begin(), data2->mPorts.end(),
+                    std::back_inserter(portsOnlyInCommand2), [&](const auto &portId) {
+                return std::find(data->mPorts.begin(), data->mPorts.end(), portId) ==
+                        data->mPorts.end();
+            });
+            if (!portsOnlyInCommand2.empty()) {
+                data2->mPorts = portsOnlyInCommand2;
+                break;
+            }
+            ALOGV("Filtering out volume command on output %d for ports %s",
+                    data->mIO, data->dumpPorts().c_str());
+            removedCommands.add(command2);
+            command->mTime = command2->mTime;
+            // force delayMs to non 0 so that code below does not request to wait for
+            // command status as the command is now delayed
+            delayMs = 1;
+        } break;
+
         case SET_VOICE_VOLUME: {
             VoiceVolumeData *data = (VoiceVolumeData *)command->mParam.get();
             VoiceVolumeData *data2 = (VoiceVolumeData *)command2->mParam.get();
@@ -2603,6 +2655,12 @@
                                                    output, delayMs);
 }
 
+int AudioPolicyService::setPortsVolume(const std::vector<audio_port_handle_t> &ports, float volume,
+                                       audio_io_handle_t output, int delayMs)
+{
+    return (int)mAudioCommandThread->volumePortsCommand(ports, volume, output, delayMs);
+}
+
 int AudioPolicyService::setVoiceVolume(float volume, int delayMs)
 {
     return (int)mAudioCommandThread->voiceVolumeCommand(volume, delayMs);
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index 720ba84..0492cd3 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -47,6 +47,7 @@
 #include <android/hardware/BnSensorPrivacyListener.h>
 #include <android/content/AttributionSourceState.h>
 
+#include <numeric>
 #include <unordered_map>
 
 namespace android {
@@ -354,6 +355,21 @@
                                      float volume,
                                      audio_io_handle_t output,
                                      int delayMs = 0);
+
+    /**
+     * Set a volume on AudioTrack port id(s) for a particular output.
+     * For the same user setting, a volume group (and associated given port of the
+     * client's track) can have different volumes for each output destination device
+     * it is attached to.
+     *
+     * @param ports to consider
+     * @param volume to set
+     * @param output to consider
+     * @param delayMs to use
+     * @return NO_ERROR if successful
+     */
+    virtual status_t setPortsVolume(const std::vector<audio_port_handle_t> &ports, float volume,
+            audio_io_handle_t output, int delayMs = 0);
     virtual status_t setVoiceVolume(float volume, int delayMs = 0);
 
     void doOnNewAudioModulesAvailable();
@@ -577,6 +593,7 @@
         // commands for tone AudioCommand
         enum {
             SET_VOLUME,
+            SET_PORTS_VOLUME,
             SET_PARAMETERS,
             SET_VOICE_VOLUME,
             STOP_OUTPUT,
@@ -610,6 +627,8 @@
                     void        exit();
                     status_t    volumeCommand(audio_stream_type_t stream, float volume,
                                             audio_io_handle_t output, int delayMs = 0);
+                    status_t    volumePortsCommand(const std::vector<audio_port_handle_t> &ports,
+                            float volume, audio_io_handle_t output, int delayMs = 0);
                     status_t    parametersCommand(audio_io_handle_t ioHandle,
                                             const char *keyValuePairs, int delayMs = 0);
                     status_t    voiceVolumeCommand(float volume, int delayMs = 0);
@@ -684,6 +703,20 @@
             audio_io_handle_t mIO;
         };
 
+        class VolumePortsData : public AudioCommandData {
+        public:
+            std::vector<audio_port_handle_t> mPorts;
+            float mVolume;
+            audio_io_handle_t mIO;
+            std::string dumpPorts() {
+                return std::string("volume ") + std::to_string(mVolume) + " on IO " +
+                        std::to_string(mIO) + " and ports " +
+                        std::accumulate(std::begin(mPorts), std::end(mPorts), std::string{},
+                                       [] (const std::string& ls, int rs) {
+                                return ls + std::to_string(rs) + " "; });
+            }
+        };
+
         class ParametersData : public AudioCommandData {
         public:
             audio_io_handle_t mIO;
@@ -823,6 +856,19 @@
         // set a stream volume for a particular output. For the same user setting, a given stream type can have different volumes
         // for each output (destination device) it is attached to.
         virtual status_t setStreamVolume(audio_stream_type_t stream, float volume, audio_io_handle_t output, int delayMs = 0);
+        /**
+         * Set a volume on port(s) for a particular output. For the same user setting, a volume
+         * group (and associated given port of the client's track) can have different volumes for
+         * each output (destination device) it is attached to.
+         *
+         * @param ports to consider
+         * @param volume to set
+         * @param output to consider
+         * @param delayMs to use
+         * @return NO_ERROR if successful
+         */
+        status_t setPortsVolume(const std::vector<audio_port_handle_t> &ports, float volume,
+                audio_io_handle_t output, int delayMs = 0) override;
 
         // function enabling to send proprietary informations directly from audio policy manager to audio hardware interface.
         virtual void setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs, int delayMs = 0);
diff --git a/services/audiopolicy/tests/AudioPolicyTestClient.h b/services/audiopolicy/tests/AudioPolicyTestClient.h
index c15adcb..ea76685 100644
--- a/services/audiopolicy/tests/AudioPolicyTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyTestClient.h
@@ -57,6 +57,10 @@
                              float /*volume*/,
                              audio_io_handle_t /*output*/,
                              int /*delayMs*/) override { return NO_INIT; }
+
+    status_t setPortsVolume(const std::vector<audio_port_handle_t>& /*ports*/, float /*volume*/,
+            audio_io_handle_t /*output*/, int /*delayMs*/) override { return NO_INIT; }
+
     void setParameters(audio_io_handle_t /*ioHandle*/,
                        const String8& /*keyValuePairs*/,
                        int /*delayMs*/) override { }
diff --git a/services/audiopolicy/tests/audiopolicymanager_tests.cpp b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
index 07aad0c..eb4240a 100644
--- a/services/audiopolicy/tests/audiopolicymanager_tests.cpp
+++ b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
@@ -299,11 +299,12 @@
     AudioPolicyInterface::output_type_t outputType;
     bool isSpatialized;
     bool isBitPerfectInternal;
+    float volume;
     AttributionSourceState attributionSource = createAttributionSourceState(uid);
     ASSERT_EQ(OK, mManager->getOutputForAttr(
                     &attr, output, session, &stream, attributionSource, &config, &flags,
                     selectedDeviceId, portId, {}, &outputType, &isSpatialized,
-                    isBitPerfect == nullptr ? &isBitPerfectInternal : isBitPerfect));
+                    isBitPerfect == nullptr ? &isBitPerfectInternal : isBitPerfect, &volume));
     ASSERT_NE(AUDIO_PORT_HANDLE_NONE, *portId);
     ASSERT_NE(AUDIO_IO_HANDLE_NONE, *output);
 }
@@ -2065,6 +2066,7 @@
     audio_attributes_t attr = AUDIO_ATTRIBUTES_INITIALIZER;
     bool mIsSpatialized;
     bool mIsBitPerfect;
+    float mVolume;
 };
 
 TEST_P(AudioPolicyManagerTestMMapPlaybackRerouting, MmapPlaybackStreamMatchingLoopbackDapMixFails) {
@@ -2083,7 +2085,7 @@
               mManager->getOutputForAttr(&attr, &mOutput, AUDIO_SESSION_NONE, &mStream,
                                          createAttributionSourceState(testUid), &audioConfig,
                                          &outputFlags, &mSelectedDeviceId, &mPortId, {},
-                                         &mOutputType, &mIsSpatialized, &mIsBitPerfect));
+                                         &mOutputType, &mIsSpatialized, &mIsBitPerfect, &mVolume));
 }
 
 TEST_P(AudioPolicyManagerTestMMapPlaybackRerouting,
@@ -2102,7 +2104,7 @@
               mManager->getOutputForAttr(&attr, &mOutput, AUDIO_SESSION_NONE, &mStream,
                                          createAttributionSourceState(testUid), &audioConfig,
                                          &outputFlags, &mSelectedDeviceId, &mPortId, {},
-                                         &mOutputType, &mIsSpatialized, &mIsBitPerfect));
+                                         &mOutputType, &mIsSpatialized, &mIsBitPerfect, &mVolume));
 }
 
 TEST_F(AudioPolicyManagerTestMMapPlaybackRerouting,
@@ -2133,7 +2135,7 @@
               mManager->getOutputForAttr(&attr, &mOutput, AUDIO_SESSION_NONE, &mStream,
                                          createAttributionSourceState(testUid), &audioConfig,
                                          &outputFlags, &mSelectedDeviceId, &mPortId, {},
-                                         &mOutputType, &mIsSpatialized, &mIsBitPerfect));
+                                         &mOutputType, &mIsSpatialized, &mIsBitPerfect, &mVolume));
     ASSERT_EQ(usbDevicePort.id, mSelectedDeviceId);
     auto outputDesc = mManager->getOutputs().valueFor(mOutput);
     ASSERT_NE(nullptr, outputDesc);
@@ -2149,7 +2151,7 @@
               mManager->getOutputForAttr(&attr, &mOutput, AUDIO_SESSION_NONE, &mStream,
                                          createAttributionSourceState(testUid), &audioConfig,
                                          &outputFlags, &mSelectedDeviceId, &mPortId, {},
-                                         &mOutputType, &mIsSpatialized, &mIsBitPerfect));
+                                         &mOutputType, &mIsSpatialized, &mIsBitPerfect, &mVolume));
     ASSERT_EQ(usbDevicePort.id, mSelectedDeviceId);
     outputDesc = mManager->getOutputs().valueFor(mOutput);
     ASSERT_NE(nullptr, outputDesc);
@@ -2178,7 +2180,7 @@
               mManager->getOutputForAttr(&attr, &mOutput, AUDIO_SESSION_NONE, &mStream,
                                          createAttributionSourceState(testUid), &audioConfig,
                                          &outputFlags, &mSelectedDeviceId, &mPortId, {},
-                                         &mOutputType, &mIsSpatialized, &mIsBitPerfect));
+                                         &mOutputType, &mIsSpatialized, &mIsBitPerfect, &mVolume));
 }
 
 INSTANTIATE_TEST_SUITE_P(
@@ -3632,11 +3634,12 @@
     AudioPolicyInterface::output_type_t outputType;
     bool isSpatialized;
     bool isBitPerfect;
+    float volume;
     EXPECT_EQ(expected,
               mManager->getOutputForAttr(&sMediaAttr, &mBitPerfectOutput, AUDIO_SESSION_NONE,
                                          &stream, attributionSource, &config, &flags,
                                          &mSelectedDeviceId, &mBitPerfectPortId, {}, &outputType,
-                                         &isSpatialized, &isBitPerfect));
+                                         &isSpatialized, &isBitPerfect, &volume));
 }
 
 class AudioPolicyManagerTestBitPerfect : public AudioPolicyManagerTestBitPerfectBase {