AudioFlinger: Control volume using Port ID

This CL migrates the volume management within AudioFlinger
from stream type to port ID.

It gives full power to AudioPolicy to compute the list of port
(so MmapThreads/Tracks) on which volume control is required.

It prevents from overwritting MUSIC stream type which is
the default for volume groups without associated stream type.

Bug: 317212590
Test: build & play audio
Test: atest audiopolicy_tests
Flag: com.android.media.audioserver.portid_volume_management

Change-Id: I4c9e8bb45660c9ceffcc0f4029b0617f9795ab3c
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..f2b59b7 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 +1648,37 @@
     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());
+    for (const auto& port : ports) {
+        sp<VolumePortInterface> volumePortInterface = getVolumePortInterface_l(output, port);
+        if (volumePortInterface == nullptr) {
+            return BAD_VALUE;
+        }
+        volumePortInterface->setPortVolume(volume);
+    }
+    const sp<IAfMmapThread> mmapThread = checkMmapThread_l(output);
+    if (mmapThread) {
+        // send broadcast event only when all tracks volume is updated
+        audio_utils::lock_guard _l(mmapThread->mutex());
+        mmapThread->broadcast_l();
+    }
+    return NO_ERROR;
+}
+
 status_t AudioFlinger::setRequestedLatencyMode(
         audio_io_handle_t output, audio_latency_mode_t mode) {
     if (output == AUDIO_IO_HANDLE_NONE) {
@@ -3824,8 +3859,7 @@
 
 
 // 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();
@@ -3840,6 +3874,21 @@
     return volumeInterface;
 }
 
+sp<VolumePortInterface> AudioFlinger::getVolumePortInterface_l(audio_io_handle_t output,
+        audio_port_handle_t port) const
+{
+    IAfPlaybackThread *thread = checkPlaybackThread_l(output);
+    if (thread != nullptr) {
+        return thread->getVolumePortInterface(port);
+    }
+    const sp<IAfMmapThread> mmapThread = checkMmapThread_l(output);
+    if (mmapThread != nullptr && mmapThread->isOutput()) {
+        IAfMmapPlaybackThread *mmapPlaybackThread = mmapThread->asIAfMmapPlaybackThread().get();
+        return mmapPlaybackThread->getVolumePortInterface(port);
+    }
+    return nullptr;
+}
+
 std::vector<sp<VolumeInterface>> AudioFlinger::getAllVolumeInterfaces_l() const
 {
     std::vector<sp<VolumeInterface>> volumeInterfaces;
@@ -5119,6 +5168,7 @@
         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 aee4d7a..8b1f7ad 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;
@@ -549,6 +552,9 @@
     IAfPlaybackThread* checkMixerThread_l(audio_io_handle_t output) const REQUIRES(mutex());
 
     sp<VolumeInterface> getVolumeInterface_l(audio_io_handle_t output) const REQUIRES(mutex());
+
+    sp<VolumePortInterface> getVolumePortInterface_l(
+            audio_io_handle_t output, audio_port_handle_t port) const REQUIRES(mutex());
     std::vector<sp<VolumeInterface>> getAllVolumeInterfaces_l() const REQUIRES(mutex());
 
 
@@ -753,6 +759,7 @@
     bool mIsDeviceTypeKnown GUARDED_BY(mutex()) = false;
     int64_t mTotalMemory GUARDED_BY(mutex()) = 0;
     std::atomic<size_t> mClientSharedHeapSize = kMinimumClientSharedHeapSizeBytes;
+
     static constexpr size_t kMinimumClientSharedHeapSizeBytes = 1024 * 1024; // 1MB
 
     // when a global effect was last enabled
diff --git a/services/audioflinger/IAfThread.h b/services/audioflinger/IAfThread.h
index 4d26aa0..b6259a8 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 +557,8 @@
 
     virtual void setTracksInternalMute(std::map<audio_port_handle_t, bool>* tracksInternalMute)
             EXCLUDES_ThreadBase_Mutex = 0;
+
+    virtual sp<VolumePortInterface> getVolumePortInterface(audio_port_handle_t port) const = 0;
 };
 
 class IAfDirectOutputThread : public virtual IAfPlaybackThread {
@@ -694,6 +698,8 @@
             AudioHwDevice* hwDev, AudioStreamOut* output, bool systemReady);
 
     virtual AudioStreamOut* clearOutput() EXCLUDES_ThreadBase_Mutex = 0;
+
+    virtual sp<VolumePortInterface> getVolumePortInterface(audio_port_handle_t port) const = 0;
 };
 
 class IAfMmapCaptureThread : public virtual IAfMmapThread {
diff --git a/services/audioflinger/IAfTrack.h b/services/audioflinger/IAfTrack.h
index a9c87ad..d123052 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;
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/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 2cc6236..11d82b4 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 +223,14 @@
 
     bool getInternalMute() const final { return mInternalMute; }
     void setInternalMute(bool muted) final { mInternalMute = muted; }
+
+    // VolumePortInterface implementation
+    void setPortVolume(float volume) override {
+        mVolume = volume;
+        signal();
+    }
+    float getPortVolume() const override { return mVolume; }
+
 protected:
 
     DISALLOW_COPY_AND_ASSIGN(Track);
@@ -403,8 +412,8 @@
     // 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
 
 
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 2dcbbce..8c9e7c8 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 {
 
@@ -2212,17 +2214,18 @@
                 (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()
@@ -2273,16 +2276,17 @@
 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");
@@ -2390,7 +2394,8 @@
         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;
@@ -2719,7 +2724,7 @@
                           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) {
@@ -2847,6 +2852,21 @@
     return mStreamTypes[stream].volume;
 }
 
+sp<VolumePortInterface> PlaybackThread::getVolumePortInterface(audio_port_handle_t port) const
+{
+    audio_utils::lock_guard _l(mutex());
+    if (port == AUDIO_PORT_HANDLE_NONE) {
+        return nullptr;
+    }
+    for (size_t i = 0; i < mTracks.size(); i++) {
+        sp<IAfTrack> track = mTracks[i].get();
+        if (port == track->portId()) {
+            return track;
+        }
+    }
+    return nullptr;
+}
+
 void PlaybackThread::setVolumeForOutput_l(float left, float right) const
 {
     mOutput->stream->setVolume(left, right);
@@ -5778,12 +5798,19 @@
                 }
                 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
@@ -5795,15 +5822,23 @@
                 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;
 
@@ -5954,16 +5989,22 @@
             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()) {
@@ -5983,15 +6024,23 @@
                     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;
@@ -6717,34 +6766,64 @@
 
     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);
@@ -7838,7 +7917,9 @@
         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();
@@ -10325,6 +10406,7 @@
 
     const auto localSessionId = mSessionId;
     auto localAttr = mAttr;
+    float volume = 0.0f;
     if (isOutput()) {
         audio_config_t config = AUDIO_CONFIG_INITIALIZER;
         config.sample_rate = mSampleRate;
@@ -10348,7 +10430,8 @@
                                             &portId,
                                             &secondaryOutputs,
                                             &isSpatialized,
-                                            &isBitPerfect);
+                                            &isBitPerfect,
+                                            &volume);
         mutex().lock();
         mAttr = localAttr;
         ALOGD_IF(!secondaryOutputs.empty(),
@@ -10417,7 +10500,8 @@
             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));
     }
@@ -11002,18 +11086,18 @@
     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;
@@ -11092,6 +11176,20 @@
     }
 }
 
+sp<VolumePortInterface> MmapPlaybackThread::getVolumePortInterface(audio_port_handle_t port) const
+{
+    audio_utils::lock_guard _l(mutex());
+    if (port == AUDIO_PORT_HANDLE_NONE) {
+        return nullptr;
+    }
+    for (const sp<IAfMmapTrack>& track : mActiveTracks) {
+        if (port == track->portId()) {
+            return track;
+        }
+    }
+    return nullptr;
+}
+
 void MmapPlaybackThread::invalidateTracks(audio_stream_type_t streamType)
 {
     audio_utils::lock_guard _l(mutex());
@@ -11125,14 +11223,26 @@
 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));
@@ -11165,14 +11275,25 @@
         }
         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*/});
+                }
         }
     }
 }
@@ -11279,9 +11400,13 @@
 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..ba5c09c 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 +1017,7 @@
     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;
+    sp<VolumePortInterface> getVolumePortInterface(audio_port_handle_t port) const;
     void setVolumeForOutput_l(float left, float right) const final;
 
     sp<IAfTrack> createTrack_l(
@@ -1035,7 +1042,8 @@
                                 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 {
@@ -2386,6 +2394,8 @@
     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;
 
+    sp<VolumePortInterface> getVolumePortInterface(audio_port_handle_t port) const;
+
     void setMasterMute_l(bool muted) REQUIRES(mutex()) { mMasterMute = muted; }
 
     void invalidateTracks(audio_stream_type_t streamType) final EXCLUDES_ThreadBase_Mutex;
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index f5f11cc..83116dc 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));
@@ -842,6 +846,10 @@
         mFastIndex = i;
         thread->fastTrackAvailMask_l() &= ~(1 << i);
     }
+    if (attr.usage == AUDIO_USAGE_CALL_ASSISTANT || attr.usage == AUDIO_USAGE_VIRTUAL_SOURCE) {
+        // Audio patch and call assistant volume are always max
+        mVolume = 1.0f;
+    }
 
     mServerLatencySupported = checkServerLatencySupported(format, flags);
 #ifdef TEE_SINK
@@ -923,7 +931,7 @@
     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 +1017,7 @@
     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 +1039,7 @@
             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,
@@ -2191,14 +2200,13 @@
             size_t frameCount,
             const AttributionSourceState& attributionSource)
     :   Track(playbackThread, NULL, AUDIO_STREAM_PATCH,
-              audio_attributes_t{} /* currently unused for output track */,
+              audio_attributes_t{ .usage = AUDIO_USAGE_VIRTUAL_SOURCE } /* for volume init only */,
               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);
@@ -3482,7 +3490,8 @@
           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 +3503,8 @@
             isOut,
             attributionSource,
             creatorPid,
-            portId);
+            portId,
+            volume);
 }
 
 MmapTrack::MmapTrack(IAfThreadBase* thread,
@@ -3506,7 +3516,8 @@
         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 +3528,15 @@
                   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 +3615,8 @@
 
 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 +3631,7 @@
             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