Merge "MediaCodec: more buffer state cleanup" into main
diff --git a/drm/libmediadrmrkp/include/DrmRemotelyProvisionedComponent.h b/drm/libmediadrmrkp/include/DrmRemotelyProvisionedComponent.h
index f046785..97a8cc4 100644
--- a/drm/libmediadrmrkp/include/DrmRemotelyProvisionedComponent.h
+++ b/drm/libmediadrmrkp/include/DrmRemotelyProvisionedComponent.h
@@ -35,7 +35,8 @@
 class DrmRemotelyProvisionedComponent : public BnRemotelyProvisionedComponent {
   public:
     DrmRemotelyProvisionedComponent(std::shared_ptr<IDrmPlugin> drm, std::string drmVendor,
-                                    std::string drmDesc, std::vector<uint8_t> bcc);
+                                    std::string drmDesc, std::vector<uint8_t> bcc,
+                                    std::vector<uint8_t> bcc_signature);
     ScopedAStatus getHardwareInfo(RpcHardwareInfo* info) override;
 
     ScopedAStatus generateEcdsaP256KeyPair(bool testMode, MacedPublicKey* macedPublicKey,
@@ -60,6 +61,7 @@
     std::string mDrmVendor;
     std::string mDrmDesc;
     std::vector<uint8_t> mBcc;
+    std::vector<uint8_t> mBccSignature;
 };
 }  // namespace android::mediadrm
 
diff --git a/drm/libmediadrmrkp/src/DrmRemotelyProvisionedComponent.cpp b/drm/libmediadrmrkp/src/DrmRemotelyProvisionedComponent.cpp
index 440be79..65054b0 100644
--- a/drm/libmediadrmrkp/src/DrmRemotelyProvisionedComponent.cpp
+++ b/drm/libmediadrmrkp/src/DrmRemotelyProvisionedComponent.cpp
@@ -28,11 +28,13 @@
 DrmRemotelyProvisionedComponent::DrmRemotelyProvisionedComponent(std::shared_ptr<IDrmPlugin> drm,
                                                                  std::string drmVendor,
                                                                  std::string drmDesc,
-                                                                 std::vector<uint8_t> bcc)
+                                                                 std::vector<uint8_t> bcc,
+                                                                 std::vector<uint8_t> bcc_signature)
     : mDrm(std::move(drm)),
       mDrmVendor(std::move(drmVendor)),
       mDrmDesc(std::move(drmDesc)),
-      mBcc(std::move(bcc)) {}
+      mBcc(std::move(bcc)),
+      mBccSignature(std::move(bcc_signature)) {}
 
 ScopedAStatus DrmRemotelyProvisionedComponent::getHardwareInfo(RpcHardwareInfo* info) {
     info->versionNumber = 3;
@@ -107,7 +109,7 @@
     for (auto i : keyToProp) {
         auto key = i.first;
         auto prop = i.second;
-        const auto& val= deviceInfoMap.get(key);
+        const auto& val = deviceInfoMap.get(key);
         if (val == nullptr || val->asTstr()->value().empty()) {
             std::string propValue = android::base::GetProperty(prop, "");
             if (propValue.empty()) {
@@ -161,12 +163,16 @@
     }
 
     // assemble AuthenticatedRequest (definition in IRemotelyProvisionedComponent.aidl)
-    *out = cppbor::Array()
-                   .add(1 /* version */)
-                   .add(cppbor::Map() /* UdsCerts */)
-                   .add(cppbor::EncodedItem(mBcc))
-                   .add(cppbor::EncodedItem(std::move(deviceSignedCsrPayload)))
-                   .encode();
+    cppbor::Array request_array = cppbor::Array().add(1 /* version */);
+    if (!mBccSignature.empty()) {
+        request_array.add(cppbor::EncodedItem(mBccSignature) /* UdsCerts */);
+    } else {
+        request_array.add(cppbor::Map() /* empty UdsCerts */);
+    }
+    request_array.add(cppbor::EncodedItem(mBcc))
+            .add(cppbor::EncodedItem(std::move(deviceSignedCsrPayload)));
+    *out = request_array.encode();
+
     return ScopedAStatus::ok();
 }
 }  // namespace android::mediadrm
\ No newline at end of file
diff --git a/drm/libmediadrmrkp/src/DrmRkpAdapter.cpp b/drm/libmediadrmrkp/src/DrmRkpAdapter.cpp
index 515d157..750b51e 100644
--- a/drm/libmediadrmrkp/src/DrmRkpAdapter.cpp
+++ b/drm/libmediadrmrkp/src/DrmRkpAdapter.cpp
@@ -87,13 +87,21 @@
                           status.getDescription().c_str());
                     return;
                 }
-
+                std::vector<uint8_t> bcc_signature;
+                status =
+                        mDrm->getPropertyByteArray("bootCertificateChainSignature", &bcc_signature);
+                if (!status.isOk()) {
+                    ALOGW("mDrm->getPropertyByteArray(\"bootCertificateChainSignature\") failed."
+                          "Detail: [%s].",
+                          status.getDescription().c_str());
+                    // bcc signature is optional, no need to return when it is unavailable.
+                }
                 std::string compName(instance);
                 auto comps = static_cast<
                         std::map<std::string, std::shared_ptr<IRemotelyProvisionedComponent>>*>(
                         context);
                 (*comps)[compName] = ::ndk::SharedRefBase::make<DrmRemotelyProvisionedComponent>(
-                        mDrm, drmVendor, drmDesc, bcc);
+                        mDrm, drmVendor, drmDesc, bcc, bcc_signature);
             });
     return comps;
 }
diff --git a/media/TEST_MAPPING b/media/TEST_MAPPING
index 1a637ac..695cad6 100644
--- a/media/TEST_MAPPING
+++ b/media/TEST_MAPPING
@@ -45,6 +45,32 @@
             "file_patterns": ["(?i)drm|crypto"]
         }
     ],
+    "postsubmit": [
+        {
+            "name": "MctsMediaCodecTestCases",
+            "options": [
+                {
+                    "include-filter": "android.media.codec.cts.EncodeDecodeTest"
+                }
+            ]
+        },
+        {
+            "name": "MctsMediaCodecTestCases",
+            "options": [
+                {
+                    "include-filter": "android.media.codec.cts.DecodeEditEncodeTest"
+                }
+            ]
+        },
+        {
+            "name": "MctsMediaCodecTestCases",
+            "options": [
+                {
+                    "include-filter": "android.media.codec.cts.ExtractDecodeEditEncodeMuxTest"
+                }
+            ]
+        }
+    ],
     // Postsubmit tests for TV devices
     "tv-postsubmit": [
         {
diff --git a/media/codec2/components/mp3/C2SoftMp3Dec.cpp b/media/codec2/components/mp3/C2SoftMp3Dec.cpp
index 149c6ee..aed5e68 100644
--- a/media/codec2/components/mp3/C2SoftMp3Dec.cpp
+++ b/media/codec2/components/mp3/C2SoftMp3Dec.cpp
@@ -114,7 +114,9 @@
 c2_status_t C2SoftMP3::onStop() {
     // Make sure that the next buffer output does not still
     // depend on fragments from the last one decoded.
-    pvmp3_InitDecoder(mConfig, mDecoderBuf);
+    if (mDecoderBuf) {
+        pvmp3_InitDecoder(mConfig, mDecoderBuf);
+    }
     mSignalledError = false;
     mIsFirst = true;
     mSignalledOutputEos = false;
diff --git a/media/codec2/hal/aidl/include/codec2/aidl/ComponentStore.h b/media/codec2/hal/aidl/include/codec2/aidl/ComponentStore.h
index de0f566..bb4c596 100644
--- a/media/codec2/hal/aidl/include/codec2/aidl/ComponentStore.h
+++ b/media/codec2/hal/aidl/include/codec2/aidl/ComponentStore.h
@@ -52,6 +52,13 @@
 using ::aidl::android::hardware::media::bufferpool2::IClientManager;
 
 struct ComponentStore : public BnComponentStore {
+    /**
+     * Constructor for ComponentStore.
+     *
+     * IMPORTANT: SetPreferredCodec2ComponentStore() is called in the constructor.
+     * Be careful about the order of SetPreferredCodec2ComponentStore() and
+     * ComponentStore() in the code.
+     */
     ComponentStore(const std::shared_ptr<C2ComponentStore>& store);
     virtual ~ComponentStore();
 
diff --git a/media/codec2/hal/hidl/1.0/utils/include/codec2/hidl/1.0/ComponentStore.h b/media/codec2/hal/hidl/1.0/utils/include/codec2/hidl/1.0/ComponentStore.h
index 847c90c..028238b 100644
--- a/media/codec2/hal/hidl/1.0/utils/include/codec2/hidl/1.0/ComponentStore.h
+++ b/media/codec2/hal/hidl/1.0/utils/include/codec2/hidl/1.0/ComponentStore.h
@@ -55,6 +55,13 @@
 using ::android::sp;
 
 struct ComponentStore : public IComponentStore {
+    /**
+     * Constructor for ComponentStore.
+     *
+     * IMPORTANT: SetPreferredCodec2ComponentStore() is called in the constructor.
+     * Be careful about the order of SetPreferredCodec2ComponentStore() and
+     * ComponentStore() in the code.
+     */
     ComponentStore(const std::shared_ptr<C2ComponentStore>& store);
     virtual ~ComponentStore();
 
diff --git a/media/codec2/hal/hidl/1.1/utils/include/codec2/hidl/1.1/ComponentStore.h b/media/codec2/hal/hidl/1.1/utils/include/codec2/hidl/1.1/ComponentStore.h
index 9028149..b023115 100644
--- a/media/codec2/hal/hidl/1.1/utils/include/codec2/hidl/1.1/ComponentStore.h
+++ b/media/codec2/hal/hidl/1.1/utils/include/codec2/hidl/1.1/ComponentStore.h
@@ -56,6 +56,13 @@
 using ::android::sp;
 
 struct ComponentStore : public IComponentStore {
+    /**
+     * Constructor for ComponentStore.
+     *
+     * IMPORTANT: SetPreferredCodec2ComponentStore() is called in the constructor.
+     * Be careful about the order of SetPreferredCodec2ComponentStore() and
+     * ComponentStore() in the code.
+     */
     ComponentStore(const std::shared_ptr<C2ComponentStore>& store);
     virtual ~ComponentStore();
 
diff --git a/media/codec2/hal/hidl/1.2/utils/include/codec2/hidl/1.2/ComponentStore.h b/media/codec2/hal/hidl/1.2/utils/include/codec2/hidl/1.2/ComponentStore.h
index 4fd260b..a7e043b 100644
--- a/media/codec2/hal/hidl/1.2/utils/include/codec2/hidl/1.2/ComponentStore.h
+++ b/media/codec2/hal/hidl/1.2/utils/include/codec2/hidl/1.2/ComponentStore.h
@@ -56,6 +56,13 @@
 using ::android::sp;
 
 struct ComponentStore : public IComponentStore {
+    /**
+     * Constructor for ComponentStore.
+     *
+     * IMPORTANT: SetPreferredCodec2ComponentStore() is called in the constructor.
+     * Be careful about the order of SetPreferredCodec2ComponentStore() and
+     * ComponentStore() in the code.
+     */
     ComponentStore(const std::shared_ptr<C2ComponentStore>& store);
     virtual ~ComponentStore();
 
diff --git a/media/libaudioclient/TEST_MAPPING b/media/libaudioclient/TEST_MAPPING
index 68dba34..29b876c 100644
--- a/media/libaudioclient/TEST_MAPPING
+++ b/media/libaudioclient/TEST_MAPPING
@@ -47,12 +47,7 @@
       "name": "audioeffect_analysis"
     },
     {
-      "name": "CtsVirtualDevicesTestCases",
-      "options" : [
-        {
-          "include-filter": "android.virtualdevice.cts.VirtualAudioTest"
-        }
-      ]
+      "name": "CtsVirtualDevicesAudioTestCases"
     }
   ]
 }
diff --git a/media/libaudioclient/tests/audiorouting_tests.cpp b/media/libaudioclient/tests/audiorouting_tests.cpp
index 8151d39..a3ab9d2 100644
--- a/media/libaudioclient/tests/audiorouting_tests.cpp
+++ b/media/libaudioclient/tests/audiorouting_tests.cpp
@@ -86,7 +86,18 @@
     }
 }
 
-TEST(AudioTrackTest, DefaultRoutingTest) {
+class AudioTrackTest
+        : public ::testing::TestWithParam<int> {
+
+public:
+    AudioTrackTest()
+            : mSampleRate(GetParam()){};
+
+    const uint32_t mSampleRate;
+
+};
+
+TEST_P(AudioTrackTest, DefaultRoutingTest) {
     audio_port_v7 port;
     if (OK != getPortByAttributes(AUDIO_PORT_ROLE_SOURCE, AUDIO_PORT_TYPE_DEVICE,
                                   AUDIO_DEVICE_IN_REMOTE_SUBMIX, "0", port)) {
@@ -95,7 +106,8 @@
 
     // create record instance
     sp<AudioCapture> capture = sp<AudioCapture>::make(
-            AUDIO_SOURCE_REMOTE_SUBMIX, 48000, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_STEREO);
+            AUDIO_SOURCE_REMOTE_SUBMIX, mSampleRate, AUDIO_FORMAT_PCM_16_BIT,
+            AUDIO_CHANNEL_IN_STEREO);
     ASSERT_NE(nullptr, capture);
     ASSERT_EQ(OK, capture->create()) << "record creation failed";
     sp<OnAudioDeviceUpdateNotifier> cbCapture = sp<OnAudioDeviceUpdateNotifier>::make();
@@ -103,7 +115,7 @@
 
     // create playback instance
     sp<AudioPlayback> playback = sp<AudioPlayback>::make(
-            48000 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+            mSampleRate, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
             AUDIO_OUTPUT_FLAG_NONE, AUDIO_SESSION_NONE);
     ASSERT_NE(nullptr, playback);
     ASSERT_EQ(OK, playback->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"))
@@ -133,6 +145,12 @@
     playback->stop();
 }
 
+INSTANTIATE_TEST_SUITE_P(
+        AudioTrackParameterizedTest,
+        AudioTrackTest,
+        ::testing::Values(44100, 48000)
+);
+
 class AudioRoutingTest : public ::testing::Test {
   public:
     void SetUp() override {
diff --git a/media/libaudiohal/impl/DeviceHalHidl.cpp b/media/libaudiohal/impl/DeviceHalHidl.cpp
index 0a262e4..263ef96 100644
--- a/media/libaudiohal/impl/DeviceHalHidl.cpp
+++ b/media/libaudiohal/impl/DeviceHalHidl.cpp
@@ -619,7 +619,14 @@
             result != NO_ERROR) {
         return result;
     }
-    return processReturn("setConnectedState", mDevice->setConnectedState(hidlAddress, connected));
+    Return<Result> ret = mDevice->setConnectedState(hidlAddress, connected);
+    if (ret.isOk() || ret == Result::NOT_SUPPORTED) {
+        // The framework is only interested in errors occurring due to connection state handling,
+        // so it can decide whether retrying is needed. If the HAL does not support this operation,
+        // it's not an error.
+        return NO_ERROR;
+    }
+    return processReturn("setConnectedState", ret);
 }
 
 error::Result<audio_hw_sync_t> DeviceHalHidl::getHwAvSync() {
diff --git a/media/libaudiohal/impl/Hal2AidlMapper.cpp b/media/libaudiohal/impl/Hal2AidlMapper.cpp
index f352849..0cdf0f2 100644
--- a/media/libaudiohal/impl/Hal2AidlMapper.cpp
+++ b/media/libaudiohal/impl/Hal2AidlMapper.cpp
@@ -368,16 +368,21 @@
         const AudioConfig& config, const std::optional<AudioIoFlags>& flags, int32_t ioHandle,
         AudioSource source, const std::set<int32_t>& destinationPortIds,
         AudioPortConfig* portConfig, bool* created) {
-    // These flags get removed one by one in this order when retrying port finding.
-    static const std::vector<AudioInputFlags> kOptionalInputFlags{
-        AudioInputFlags::FAST, AudioInputFlags::RAW, AudioInputFlags::VOIP_TX };
     if (auto portConfigIt = findPortConfig(config, flags, ioHandle);
             portConfigIt == mPortConfigs.end() && flags.has_value()) {
-        auto optionalInputFlagsIt = kOptionalInputFlags.begin();
+        // These input flags get removed one by one in this order when retrying port finding.
+        std::vector<AudioInputFlags> optionalInputFlags {
+            AudioInputFlags::FAST, AudioInputFlags::RAW, AudioInputFlags::VOIP_TX };
+        // For remote submix input, retry with direct input flag removed as the remote submix
+        // input is not expected to manipulate the contents of the audio stream.
+        if (mRemoteSubmixIn.has_value()) {
+            optionalInputFlags.push_back(AudioInputFlags::DIRECT);
+        }
+        auto optionalInputFlagsIt = optionalInputFlags.begin();
         AudioIoFlags matchFlags = flags.value();
         auto portsIt = findPort(config, matchFlags, destinationPortIds);
         while (portsIt == mPorts.end() && matchFlags.getTag() == AudioIoFlags::Tag::input
-                && optionalInputFlagsIt != kOptionalInputFlags.end()) {
+                && optionalInputFlagsIt != optionalInputFlags.end()) {
             if (!isBitPositionFlagSet(
                             matchFlags.get<AudioIoFlags::Tag::input>(), *optionalInputFlagsIt)) {
                 ++optionalInputFlagsIt;
@@ -392,6 +397,36 @@
                         config.toString().c_str(), flags.value().toString().c_str(),
                         matchFlags.toString().c_str());
         }
+        // These output flags get removed one by one in this order when retrying port finding.
+        std::vector<AudioOutputFlags> optionalOutputFlags { };
+        // For remote submix output, retry with these output flags removed one by one:
+        // 1. DIRECT: remote submix outputs are expected not to manipulate the contents of the
+        //            audio stream.
+        // 2. IEC958_NONAUDIO: remote submix outputs are not connected to ALSA and do not require
+        //                     non audio signalling.
+        if (mRemoteSubmixOut.has_value()) {
+            optionalOutputFlags.push_back(AudioOutputFlags::DIRECT);
+            optionalOutputFlags.push_back(AudioOutputFlags::IEC958_NONAUDIO);
+        }
+        auto optionalOutputFlagsIt = optionalOutputFlags.begin();
+        matchFlags = flags.value();
+        while (portsIt == mPorts.end() && matchFlags.getTag() == AudioIoFlags::Tag::output
+                && optionalOutputFlagsIt != optionalOutputFlags.end()) {
+            if (!isBitPositionFlagSet(
+                            matchFlags.get<AudioIoFlags::Tag::output>(),*optionalOutputFlagsIt)) {
+                ++optionalOutputFlagsIt;
+                continue;
+            }
+            matchFlags.set<AudioIoFlags::Tag::output>(matchFlags.get<AudioIoFlags::Tag::output>() &
+                    ~makeBitPositionFlagMask(*optionalOutputFlagsIt++));
+            portsIt = findPort(config, matchFlags, destinationPortIds);
+            AUGMENT_LOG(I,
+                        "mix port for config %s, flags %s was not found"
+                        "retried with flags %s",
+                        config.toString().c_str(), flags.value().toString().c_str(),
+                        matchFlags.toString().c_str());
+        }
+
         if (portsIt == mPorts.end()) {
             AUGMENT_LOG(E, "mix port for config %s, flags %s is not found",
                         config.toString().c_str(), matchFlags.toString().c_str());
@@ -792,7 +827,8 @@
     status_t status = prepareToOpenStreamHelper(ioHandle, devicePortConfig.portId,
             devicePortConfig.id, flags, source, initialConfig, cleanups, config,
             mixPortConfig, patch);
-    if (status != OK) {
+    if (status != OK && !(mRemoteSubmixOut.has_value() &&
+                initialConfig.base.format.type != AudioFormatType::PCM)) {
         // If using the client-provided config did not work out for establishing a mix port config
         // or patching, try with the device port config. Note that in general device port config and
         // mix port config are not required to be the same, however they must match if the HAL
diff --git a/media/libstagefright/TEST_MAPPING b/media/libstagefright/TEST_MAPPING
index b7efbce..354fab0 100644
--- a/media/libstagefright/TEST_MAPPING
+++ b/media/libstagefright/TEST_MAPPING
@@ -85,13 +85,37 @@
     // writerTest fails about 5 out of 66
     // { "name": "writerTest" },
     {
-       "name": "BatteryChecker_test"
+        "name": "BatteryChecker_test"
     },
     {
         "name": "ExtractorFactoryTest"
     },
     {
         "name": "HEVCUtilsUnitTest"
+    },
+    {
+      "name": "MctsMediaDecoderTestCases",
+      "options": [
+        {
+          "include-annotation": "android.platform.test.annotations.Presubmit"
+        }
+      ]
+    },
+    {
+      "name": "MctsMediaEncoderTestCases",
+      "options": [
+        {
+          "include-annotation": "android.platform.test.annotations.Presubmit"
+        }
+      ]
+    },
+    {
+      "name": "MctsMediaCodecTestCases",
+      "options": [
+        {
+          "include-annotation": "android.platform.test.annotations.Presubmit"
+        }
+      ]
     }
   ]
 }
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 537a097..8215247 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -2923,7 +2923,7 @@
                                                         audio_config_base_t *mixerConfig,
                                                         audio_devices_t deviceType,
                                                         const String8& address,
-                                                        audio_output_flags_t flags,
+                                                        audio_output_flags_t *flags,
                                                         const audio_attributes_t attributes)
 {
     AudioHwDevice *outHwDev = findSuitableHwDev_l(module, deviceType);
@@ -2958,7 +2958,7 @@
     mHardwareStatus = AUDIO_HW_IDLE;
 
     if (status == NO_ERROR) {
-        if (flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) {
+        if (*flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) {
             const sp<IAfMmapPlaybackThread> thread = IAfMmapPlaybackThread::create(
                     this, *output, outHwDev, outputStream, mSystemReady);
             mMmapThreads.add(*output, thread);
@@ -2967,22 +2967,22 @@
             return thread;
         } else {
             sp<IAfPlaybackThread> thread;
-            if (flags & AUDIO_OUTPUT_FLAG_BIT_PERFECT) {
+            if (*flags & AUDIO_OUTPUT_FLAG_BIT_PERFECT) {
                 thread = IAfPlaybackThread::createBitPerfectThread(
                         this, outputStream, *output, mSystemReady);
                 ALOGV("%s() created bit-perfect output: ID %d thread %p",
                       __func__, *output, thread.get());
-            } else if (flags & AUDIO_OUTPUT_FLAG_SPATIALIZER) {
+            } else if (*flags & AUDIO_OUTPUT_FLAG_SPATIALIZER) {
                 thread = IAfPlaybackThread::createSpatializerThread(this, outputStream, *output,
                                                     mSystemReady, mixerConfig);
                 ALOGV("openOutput_l() created spatializer output: ID %d thread %p",
                       *output, thread.get());
-            } else if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+            } else if (*flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
                 thread = IAfPlaybackThread::createOffloadThread(this, outputStream, *output,
                         mSystemReady, halConfig->offload_info);
                 ALOGV("openOutput_l() created offload output: ID %d thread %p",
                       *output, thread.get());
-            } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
+            } else if ((*flags & AUDIO_OUTPUT_FLAG_DIRECT)
                     || !IAfThreadBase::isValidPcmSinkFormat(halConfig->format)
                     || !IAfThreadBase::isValidPcmSinkChannelMask(halConfig->channel_mask)) {
                 thread = IAfPlaybackThread::createDirectOutputThread(this, outputStream, *output,
@@ -3046,7 +3046,7 @@
     audio_utils::lock_guard _l(mutex());
 
     const sp<IAfThreadBase> thread = openOutput_l(module, &output, &halConfig,
-            &mixerConfig, deviceType, address, flags, attributes);
+            &mixerConfig, deviceType, address, &flags, attributes);
     if (thread != 0) {
         uint32_t latencyMs = 0;
         if ((flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) == 0) {
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 21c171d..6777075 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -337,7 +337,7 @@
             audio_config_base_t* mixerConfig,
             audio_devices_t deviceType,
             const String8& address,
-            audio_output_flags_t flags,
+            audio_output_flags_t* flags,
             audio_attributes_t attributes) final REQUIRES(mutex());
     const DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*>&
             getAudioHwDevs_l() const final REQUIRES(mutex(), hardwareMutex()) {
diff --git a/services/audioflinger/IAfPatchPanel.h b/services/audioflinger/IAfPatchPanel.h
index 37dce3a..15b6ddf 100644
--- a/services/audioflinger/IAfPatchPanel.h
+++ b/services/audioflinger/IAfPatchPanel.h
@@ -82,7 +82,7 @@
             audio_config_base_t* mixerConfig,
             audio_devices_t deviceType,
             const String8& address,
-            audio_output_flags_t flags,
+            audio_output_flags_t* flags,
             audio_attributes_t attributes) REQUIRES(mutex()) = 0;
     virtual audio_utils::mutex& mutex() const
             RETURN_CAPABILITY(audio_utils::AudioFlinger_Mutex) = 0;
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index 35f17c1..994dd47 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -268,7 +268,7 @@
                                                             &mixerConfig,
                                                             outputDevice,
                                                             outputDeviceAddress,
-                                                            flags,
+                                                            &flags,
                                                             attributes);
                     ALOGV("mAfPatchPanelCallback->openOutput_l() returned %p", thread.get());
                     if (thread == 0) {
diff --git a/services/audioflinger/datapath/AudioHwDevice.cpp b/services/audioflinger/datapath/AudioHwDevice.cpp
index 5314e9e..c2e538c 100644
--- a/services/audioflinger/datapath/AudioHwDevice.cpp
+++ b/services/audioflinger/datapath/AudioHwDevice.cpp
@@ -41,19 +41,20 @@
         AudioStreamOut **ppStreamOut,
         audio_io_handle_t handle,
         audio_devices_t deviceType,
-        audio_output_flags_t flags,
+        audio_output_flags_t *flags,
         struct audio_config *config,
         const char *address,
         const std::vector<playback_track_metadata_v7_t>& sourceMetadata)
 {
 
     struct audio_config originalConfig = *config;
-    auto outputStream = new AudioStreamOut(this, flags);
+    auto outputStream = new AudioStreamOut(this);
 
     // Try to open the HAL first using the current format.
     ALOGV("openOutputStream(), try sampleRate %d, format %#x, channelMask %#x", config->sample_rate,
             config->format, config->channel_mask);
-    status_t status = outputStream->open(handle, deviceType, config, address, sourceMetadata);
+    status_t status = outputStream->open(handle, deviceType, config, flags, address,
+                                        sourceMetadata);
 
     if (status != NO_ERROR) {
         delete outputStream;
@@ -67,19 +68,25 @@
 
         // If the data is encoded then try again using wrapped PCM.
         const bool wrapperNeeded = !audio_has_proportional_frames(originalConfig.format)
-                && ((flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0)
-                && ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0);
+                && ((*flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0)
+                && ((*flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0);
 
         if (wrapperNeeded) {
             if (SPDIFEncoder::isFormatSupported(originalConfig.format)) {
-                outputStream = new SpdifStreamOut(this, flags, originalConfig.format);
-                status = outputStream->open(handle, deviceType, &originalConfig, address,
+                outputStream = new SpdifStreamOut(this, originalConfig.format);
+                status = outputStream->open(handle, deviceType, &originalConfig, flags, address,
                                             sourceMetadata);
                 if (status != NO_ERROR) {
                     ALOGE("ERROR - openOutputStream(), SPDIF open returned %d",
                         status);
                     delete outputStream;
                     outputStream = nullptr;
+                } else {
+                    // on success, we need to assign the actual HAL stream config so that clients
+                    // know and can later patch correctly.
+                    config->format = originalConfig.format;
+                    config->channel_mask = originalConfig.channel_mask;
+                    config->sample_rate = originalConfig.sample_rate;
                 }
             } else {
                 ALOGE("ERROR - openOutputStream(), SPDIFEncoder does not support format 0x%08x",
@@ -153,6 +160,12 @@
                         status);
                     delete inputStream;
                     inputStream = nullptr;
+                } else {
+                    // on success, we need to assign the actual HAL stream config so that clients
+                    // know and can later patch correctly.
+                    config->format = originalConfig.format;
+                    config->channel_mask = originalConfig.channel_mask;
+                    config->sample_rate = originalConfig.sample_rate;
                 }
             } else {
                 ALOGE("ERROR - openInputStream(), SPDIFDecoder does not support format 0x%08x",
diff --git a/services/audioflinger/datapath/AudioHwDevice.h b/services/audioflinger/datapath/AudioHwDevice.h
index e1a9018..6a35b91 100644
--- a/services/audioflinger/datapath/AudioHwDevice.h
+++ b/services/audioflinger/datapath/AudioHwDevice.h
@@ -85,7 +85,7 @@
             AudioStreamOut **ppStreamOut,
             audio_io_handle_t handle,
             audio_devices_t deviceType,
-            audio_output_flags_t flags,
+            audio_output_flags_t *flags,
             struct audio_config *config,
             const char *address,
             const std::vector<playback_track_metadata_v7_t>& sourceMetadata);
diff --git a/services/audioflinger/datapath/AudioStreamOut.cpp b/services/audioflinger/datapath/AudioStreamOut.cpp
index c65373e..7aadda3 100644
--- a/services/audioflinger/datapath/AudioStreamOut.cpp
+++ b/services/audioflinger/datapath/AudioStreamOut.cpp
@@ -30,9 +30,8 @@
 namespace android {
 
 // ----------------------------------------------------------------------------
-AudioStreamOut::AudioStreamOut(AudioHwDevice *dev, audio_output_flags_t flags)
+AudioStreamOut::AudioStreamOut(AudioHwDevice *dev)
         : audioHwDev(dev)
-        , flags(flags)
 {
 }
 
@@ -93,14 +92,16 @@
         audio_io_handle_t handle,
         audio_devices_t deviceType,
         struct audio_config *config,
+        audio_output_flags_t *flagsPtr,
         const char *address,
         const std::vector<playback_track_metadata_v7_t>& sourceMetadata)
 {
     sp<StreamOutHalInterface> outStream;
 
-    const audio_output_flags_t customFlags = (config->format == AUDIO_FORMAT_IEC61937)
-                ? (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO)
-                : flags;
+    audio_output_flags_t customFlags = (config->format == AUDIO_FORMAT_IEC61937)
+                ? (audio_output_flags_t)(*flagsPtr | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO)
+                : *flagsPtr;
+    *flagsPtr = flags = customFlags;
 
     int status = hwDev()->openOutputStream(
             handle,
diff --git a/services/audioflinger/datapath/AudioStreamOut.h b/services/audioflinger/datapath/AudioStreamOut.h
index 2bf94a1..1857099 100644
--- a/services/audioflinger/datapath/AudioStreamOut.h
+++ b/services/audioflinger/datapath/AudioStreamOut.h
@@ -37,16 +37,17 @@
 public:
     AudioHwDevice * const audioHwDev;
     sp<StreamOutHalInterface> stream;
-    const audio_output_flags_t flags;
+    audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE;
 
     [[nodiscard]] sp<DeviceHalInterface> hwDev() const;
 
-    AudioStreamOut(AudioHwDevice *dev, audio_output_flags_t flags);
+    explicit AudioStreamOut(AudioHwDevice *dev);
 
     virtual status_t open(
             audio_io_handle_t handle,
             audio_devices_t deviceType,
             struct audio_config *config,
+            audio_output_flags_t *flagsPtr,
             const char *address,
             const std::vector<playback_track_metadata_v7_t>& sourceMetadata);
 
diff --git a/services/audioflinger/datapath/SpdifStreamIn.cpp b/services/audioflinger/datapath/SpdifStreamIn.cpp
index 98ce712..0090bc5 100644
--- a/services/audioflinger/datapath/SpdifStreamIn.cpp
+++ b/services/audioflinger/datapath/SpdifStreamIn.cpp
@@ -81,6 +81,11 @@
             outputDevice,
             outputDeviceAddress);
 
+    // reset config back to whatever is returned by HAL
+    config->sample_rate = customConfig.sample_rate;
+    config->format = customConfig.format;
+    config->channel_mask = customConfig.channel_mask;
+
     ALOGI("SpdifStreamIn::open() status = %d", status);
 
 #ifdef TEE_SINK
diff --git a/services/audioflinger/datapath/SpdifStreamOut.cpp b/services/audioflinger/datapath/SpdifStreamOut.cpp
index d3983b0..a565955 100644
--- a/services/audioflinger/datapath/SpdifStreamOut.cpp
+++ b/services/audioflinger/datapath/SpdifStreamOut.cpp
@@ -33,10 +33,8 @@
  * PCM then we need to wrap the data in an SPDIF wrapper.
  */
 SpdifStreamOut::SpdifStreamOut(AudioHwDevice *dev,
-            audio_output_flags_t flags,
             audio_format_t format)
-        // Tell the HAL that the data will be compressed audio wrapped in a data burst.
-        : AudioStreamOut(dev, (audio_output_flags_t) (flags | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO))
+        : AudioStreamOut(dev)
         , mSpdifEncoder(this, format)
 {
 }
@@ -45,6 +43,7 @@
         audio_io_handle_t handle,
         audio_devices_t devices,
         struct audio_config *config,
+        audio_output_flags_t *flags,
         const char *address,
         const std::vector<playback_track_metadata_v7_t>& sourceMetadata)
 {
@@ -63,6 +62,8 @@
 
     customConfig.format = AUDIO_FORMAT_PCM_16_BIT;
     customConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+    // Tell the HAL that the data will be compressed audio wrapped in a data burst.
+    *flags = (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO);
 
     // Always print this because otherwise it could be very confusing if the
     // HAL and AudioFlinger are using different formats.
@@ -76,9 +77,15 @@
             handle,
             devices,
             &customConfig,
+            flags,
             address,
             sourceMetadata);
 
+    // reset config back to whatever is returned by HAL
+    config->sample_rate = customConfig.sample_rate;
+    config->format = customConfig.format;
+    config->channel_mask = customConfig.channel_mask;
+
     ALOGI("SpdifStreamOut::open() status = %d", status);
 
 #ifdef TEE_SINK
diff --git a/services/audioflinger/datapath/SpdifStreamOut.h b/services/audioflinger/datapath/SpdifStreamOut.h
index 1cd8f65..3241d6f 100644
--- a/services/audioflinger/datapath/SpdifStreamOut.h
+++ b/services/audioflinger/datapath/SpdifStreamOut.h
@@ -36,13 +36,13 @@
 class SpdifStreamOut : public AudioStreamOut {
 public:
 
-    SpdifStreamOut(AudioHwDevice *dev, audio_output_flags_t flags,
-            audio_format_t format);
+    SpdifStreamOut(AudioHwDevice *dev, audio_format_t format);
 
     status_t open(
             audio_io_handle_t handle,
             audio_devices_t devices,
             struct audio_config *config,
+            audio_output_flags_t *flags,
             const char *address,
             const std::vector<playback_track_metadata_v7_t>& sourceMetadata) override;
 
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index 1bac259..35973c1 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -477,7 +477,7 @@
                                 audio_config_base_t *mixerConfig,
                                 const sp<DeviceDescriptorBase>& device,
                                 uint32_t *latencyMs,
-                                audio_output_flags_t flags,
+                                audio_output_flags_t *flags,
                                 audio_attributes_t audioAttributes) = 0;
     // creates a special output that is duplicated to the two outputs passed as arguments.
     // The duplication is performed by a special mixer thread in the AudioFlinger.
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
index bfb28a5..a18cf1f 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
@@ -412,7 +412,7 @@
                       const audio_config_base_t *mixerConfig,
                       const DeviceVector &devices,
                       audio_stream_type_t stream,
-                      audio_output_flags_t flags,
+                      audio_output_flags_t *flags,
                       audio_io_handle_t *output,
                       audio_attributes_t attributes);
 
diff --git a/services/audiopolicy/common/managerdefinitions/include/HwModule.h b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
index d206637..26bb94f 100644
--- a/services/audiopolicy/common/managerdefinitions/include/HwModule.h
+++ b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
@@ -137,6 +137,7 @@
 class HwModuleCollection : public Vector<sp<HwModule> >
 {
 public:
+    sp<HwModule> getModuleFromHandle(audio_module_handle_t handle) const;
     sp<HwModule> getModuleFromName(const char *name) const;
 
     /**
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index a8663fa..3c2f46a 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -587,7 +587,7 @@
                                        const audio_config_base_t *mixerConfig,
                                        const DeviceVector &devices,
                                        audio_stream_type_t stream,
-                                       audio_output_flags_t flags,
+                                       audio_output_flags_t *flags,
                                        audio_io_handle_t *output,
                                        audio_attributes_t attributes)
 {
@@ -617,7 +617,7 @@
     // create a default one
     if ((mProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
             lHalConfig.offload_info.format == AUDIO_FORMAT_DEFAULT) {
-        flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
+        *flags = (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
         lHalConfig.offload_info = AUDIO_INFO_INITIALIZER;
         lHalConfig.offload_info.sample_rate = lHalConfig.sample_rate;
         lHalConfig.offload_info.channel_mask = lHalConfig.channel_mask;
@@ -635,7 +635,7 @@
         lMixerConfig = *mixerConfig;
     }
 
-    mFlags = (audio_output_flags_t)(mFlags | flags);
+    mFlags = (audio_output_flags_t)(mFlags | *flags);
 
     // If no mixer config is specified for a spatializer output, default to 5.1 for proper
     // configuration of the final downmixer or spatializer
@@ -653,8 +653,9 @@
                                                    &lMixerConfig,
                                                    device,
                                                    &mLatency,
-                                                   mFlags,
+                                                   &mFlags,
                                                    attributes);
+    *flags = mFlags;
 
     if (status == NO_ERROR) {
         LOG_ALWAYS_FATAL_IF(*output == AUDIO_IO_HANDLE_NONE,
diff --git a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
index 6696b45..2d8231a 100644
--- a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
@@ -283,6 +283,16 @@
     dumpAudioRouteVector(mRoutes, dst, spaces);
 }
 
+sp<HwModule> HwModuleCollection::getModuleFromHandle(audio_module_handle_t handle) const
+{
+    for (const auto& module : *this) {
+        if (module->getHandle() == handle) {
+            return module;
+        }
+    }
+    return nullptr;
+}
+
 sp <HwModule> HwModuleCollection::getModuleFromName(const char *name) const
 {
     for (const auto& module : *this) {
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 2ff2907..91d0430 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -122,17 +122,16 @@
     }
 }
 
-void AudioPolicyManager::broadcastDeviceConnectionState(const sp<DeviceDescriptor> &device,
+status_t AudioPolicyManager::broadcastDeviceConnectionState(const sp<DeviceDescriptor> &device,
                                                         media::DeviceConnectedState state)
 {
     audio_port_v7 devicePort;
     device->toAudioPort(&devicePort);
-    if (status_t status = mpClientInterface->setDeviceConnectedState(&devicePort, state);
-            status != OK) {
-        ALOGE("Error %d while setting connected state %d for device %s",
-                status, static_cast<int>(state),
-                device->getDeviceTypeAddr().toString(false).c_str());
-    }
+    status_t status = mpClientInterface->setDeviceConnectedState(&devicePort, state);
+    ALOGE_IF(status != OK, "Error %d while setting connected state %d for device %s", status,
+             static_cast<int>(state), device->getDeviceTypeAddr().toString(false).c_str());
+
+    return status;
 }
 
 status_t AudioPolicyManager::setDeviceConnectionStateInt(
@@ -213,7 +212,14 @@
 
             // Before checking outputs, broadcast connect event to allow HAL to retrieve dynamic
             // parameters on newly connected devices (instead of opening the outputs...)
-            broadcastDeviceConnectionState(device, media::DeviceConnectedState::CONNECTED);
+            if (broadcastDeviceConnectionState(
+                        device, media::DeviceConnectedState::CONNECTED) != NO_ERROR) {
+                mAvailableOutputDevices.remove(device);
+                mHwModules.cleanUpForDevice(device);
+                ALOGE("%s() device %s format %x connection failed", __func__,
+                      device->toString().c_str(), device->getEncodedFormat());
+                return INVALID_OPERATION;
+            }
 
             if (checkOutputsForDevice(device, state, outputs) != NO_ERROR) {
                 mAvailableOutputDevices.remove(device);
@@ -398,7 +404,14 @@
 
             // Before checking intputs, broadcast connect event to allow HAL to retrieve dynamic
             // parameters on newly connected devices (instead of opening the inputs...)
-            broadcastDeviceConnectionState(device, media::DeviceConnectedState::CONNECTED);
+            if (broadcastDeviceConnectionState(
+                        device, media::DeviceConnectedState::CONNECTED) != NO_ERROR) {
+                mAvailableInputDevices.remove(device);
+                mHwModules.cleanUpForDevice(device);
+                ALOGE("%s() device %s format %x connection failed", __func__,
+                      device->toString().c_str(), device->getEncodedFormat());
+                return INVALID_OPERATION;
+            }
             // Propagate device availability to Engine
             setEngineDeviceConnectionState(device, state);
 
@@ -1610,14 +1623,19 @@
     releaseMsdOutputPatches(devices);
 
     status_t status =
-            outputDesc->open(config, nullptr /* mixerConfig */, devices, stream, flags, output,
+            outputDesc->open(config, nullptr /* mixerConfig */, devices, stream, &flags, output,
                              attributes);
 
-    // only accept an output with the requested parameters
+    // only accept an output with the requested parameters, unless the format can be IEC61937
+    // encapsulated and opened by AudioFlinger as wrapped IEC61937.
+    const bool ignoreRequestedParametersCheck = audio_is_iec61937_compatible(config->format)
+            && (flags & AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO)
+            && audio_has_proportional_frames(outputDesc->getFormat());
     if (status != NO_ERROR ||
-        (config->sample_rate != 0 && config->sample_rate != outputDesc->getSamplingRate()) ||
-        (config->format != AUDIO_FORMAT_DEFAULT && config->format != outputDesc->getFormat()) ||
-        (config->channel_mask != 0 && config->channel_mask != outputDesc->getChannelMask())) {
+        (!ignoreRequestedParametersCheck &&
+        ((config->sample_rate != 0 && config->sample_rate != outputDesc->getSamplingRate()) ||
+         (config->format != AUDIO_FORMAT_DEFAULT && config->format != outputDesc->getFormat()) ||
+         (config->channel_mask != 0 && config->channel_mask != outputDesc->getChannelMask())))) {
         ALOGV("%s failed opening direct output: output %d sample rate %d %d,"
                 "format %d %d, channel mask %04x %04x", __func__, *output, config->sample_rate,
                 outputDesc->getSamplingRate(), config->format, outputDesc->getFormat(),
@@ -1637,11 +1655,11 @@
     outputDesc->mDirectClientSession = session;
 
     addOutput(*output, outputDesc);
-    setOutputDevices(__func__, outputDesc,
-                     devices,
-                     true,
-                     0,
-                     NULL);
+    // The version check is essentially to avoid making this call in the case of the HIDL HAL.
+    if (auto hwModule = mHwModules.getModuleFromHandle(mPrimaryModuleHandle); hwModule &&
+            hwModule->getHalVersionMajor() >= 3) {
+        setOutputDevices(__func__, outputDesc, devices, true, 0, NULL);
+    }
     mPreviousOutputs = mOutputs;
     ALOGV("%s returns new direct output %d", __func__, *output);
     mpClientInterface->onAudioPortListUpdate();
@@ -6608,11 +6626,12 @@
             sp<SwAudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(outProfile,
                                                                                  mpClientInterface);
             audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
+            audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE;
             audio_attributes_t attributes = AUDIO_ATTRIBUTES_INITIALIZER;
             status_t status = outputDesc->open(nullptr /* halConfig */, nullptr /* mixerConfig */,
                                                DeviceVector(supportedDevice),
                                                AUDIO_STREAM_DEFAULT,
-                                               AUDIO_OUTPUT_FLAG_NONE, &output, attributes);
+                                               &flags, &output, attributes);
             if (status != NO_ERROR) {
                 ALOGW("Cannot open output stream for devices %s on hw module %s",
                       supportedDevice->toString().c_str(), hwModule->getName());
@@ -7982,9 +8001,21 @@
                         if (result.source == AUDIO_SOURCE_HOTWORD && !inputDesc->isSoundTrigger()) {
                             result.source = AUDIO_SOURCE_VOICE_RECOGNITION;
                         }
-                        return result; }).
+                        return result; });
             //only one input device for now
-                    addSource(device);
+            if (audio_is_remote_submix_device(device->type())) {
+                // remote submix HAL does not support audio conversion, need source device
+                // audio config to match the sink input descriptor audio config, otherwise AIDL
+                // HAL patching will fail
+                audio_port_config srcDevicePortConfig = {};
+                device->toAudioPortConfig(&srcDevicePortConfig, nullptr);
+                srcDevicePortConfig.sample_rate = inputDesc->getSamplingRate();
+                srcDevicePortConfig.channel_mask = inputDesc->getChannelMask();
+                srcDevicePortConfig.format = inputDesc->getFormat();
+                patchBuilder.addSource(srcDevicePortConfig);
+            } else {
+                patchBuilder.addSource(device);
+            }
             status = installPatch(__func__, patchHandle, inputDesc.get(), patchBuilder.patch(), 0);
         }
     }
@@ -8827,7 +8858,7 @@
     audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
     audio_attributes_t attributes = AUDIO_ATTRIBUTES_INITIALIZER;
     status_t status = desc->open(halConfig, mixerConfig, devices,
-            AUDIO_STREAM_DEFAULT, flags, &output, attributes);
+            AUDIO_STREAM_DEFAULT, &flags, &output, attributes);
     if (status != NO_ERROR) {
         ALOGE("%s failed to open output %d", __func__, status);
         return nullptr;
@@ -8865,7 +8896,7 @@
         config.offload_info.channel_mask = config.channel_mask;
         config.offload_info.format = config.format;
 
-        status = desc->open(&config, mixerConfig, devices, AUDIO_STREAM_DEFAULT, flags, &output,
+        status = desc->open(&config, mixerConfig, devices, AUDIO_STREAM_DEFAULT, &flags, &output,
                             attributes);
         if (status != NO_ERROR) {
             return nullptr;
@@ -8873,11 +8904,11 @@
     }
 
     addOutput(output, desc);
-    setOutputDevices(__func__, desc,
-                     devices,
-                     true,
-                     0,
-                     NULL);
+    // The version check is essentially to avoid making this call in the case of the HIDL HAL.
+    if (auto hwModule = mHwModules.getModuleFromHandle(mPrimaryModuleHandle); hwModule &&
+            hwModule->getHalVersionMajor() >= 3) {
+        setOutputDevices(__func__, desc, devices, true, 0, NULL);
+    }
     sp<DeviceDescriptor> speaker = mAvailableOutputDevices.getDevice(
             AUDIO_DEVICE_OUT_SPEAKER, String8(""), AUDIO_FORMAT_DEFAULT);
 
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 953fd2a..9d2166a 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -1105,8 +1105,8 @@
         // It can give a chance to HAL implementer to retrieve dynamic capabilities associated
         // to this device for example.
         // TODO avoid opening stream to retrieve capabilities of a profile.
-        void broadcastDeviceConnectionState(const sp<DeviceDescriptor> &device,
-                                            media::DeviceConnectedState state);
+        status_t broadcastDeviceConnectionState(const sp<DeviceDescriptor> &device,
+                                                media::DeviceConnectedState state);
 
         // updates device caching and output for streams that can influence the
         //    routing of notifications
diff --git a/services/audiopolicy/service/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
index 22fc151..6d2c772 100644
--- a/services/audiopolicy/service/AudioPolicyClientImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
@@ -56,7 +56,7 @@
                                                            audio_config_base_t *mixerConfig,
                                                            const sp<DeviceDescriptorBase>& device,
                                                            uint32_t *latencyMs,
-                                                           audio_output_flags_t flags,
+                                                           audio_output_flags_t *flags,
                                                            audio_attributes_t attributes)
 {
     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
@@ -74,7 +74,7 @@
     request.mixerConfig = VALUE_OR_RETURN_STATUS(
             legacy2aidl_audio_config_base_t_AudioConfigBase(*mixerConfig, false /*isInput*/));
     request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_DeviceDescriptorBase(device));
-    request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
+    request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_output_flags_t_int32_t_mask(*flags));
     request.attributes = VALUE_OR_RETURN_STATUS(
             legacy2aidl_audio_attributes_t_AudioAttributes(attributes));
 
@@ -89,7 +89,9 @@
             .channel_mask = halConfig->channel_mask,
             .format = halConfig->format,
         };
-        mAudioPolicyService->registerOutput(*output, config, flags);
+        *flags = VALUE_OR_RETURN_STATUS(
+                aidl2legacy_int32_t_audio_output_flags_t_mask(response.flags));
+        mAudioPolicyService->registerOutput(*output, config, *flags);
     }
     return status;
 }
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index 5d9813f..eccefa7 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -790,7 +790,7 @@
                                     audio_config_base_t *mixerConfig,
                                     const sp<DeviceDescriptorBase>& device,
                                     uint32_t *latencyMs,
-                                    audio_output_flags_t flags,
+                                    audio_output_flags_t *flags,
                                     audio_attributes_t attributes);
         // creates a special output that is duplicated to the two outputs passed as arguments. The duplication is performed by
         // a special mixer thread in the AudioFlinger.
diff --git a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
index ca7ad40..483f827 100644
--- a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
@@ -42,7 +42,7 @@
                         audio_config_base_t *mixerConfig,
                         const sp<DeviceDescriptorBase>& /*device*/,
                         uint32_t * /*latencyMs*/,
-                        audio_output_flags_t flags,
+                        audio_output_flags_t *flags,
                         audio_attributes_t /*attributes*/) override {
         if (module >= mNextModuleHandle) {
             ALOGE("%s: Module handle %d has not been allocated yet (next is %d)",
@@ -50,13 +50,13 @@
             return BAD_VALUE;
         }
         *output = mNextIoHandle++;
-        mOpenedOutputs[*output] = flags;
+        mOpenedOutputs[*output] = *flags;
         ALOGD("%s: opened output %d: HAL(%s %s %d) Mixer(%s %s %d) %s", __func__, *output,
               audio_channel_out_mask_to_string(halConfig->channel_mask),
               audio_format_to_string(halConfig->format), halConfig->sample_rate,
               audio_channel_out_mask_to_string(mixerConfig->channel_mask),
               audio_format_to_string(mixerConfig->format), mixerConfig->sample_rate,
-              android::toString(flags).c_str());
+              android::toString(*flags).c_str());
         return NO_ERROR;
     }
 
diff --git a/services/audiopolicy/tests/AudioPolicyTestClient.h b/services/audiopolicy/tests/AudioPolicyTestClient.h
index 0299160..6116eab 100644
--- a/services/audiopolicy/tests/AudioPolicyTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyTestClient.h
@@ -37,7 +37,7 @@
                         audio_config_base_t* /*mixerConfig*/,
                         const sp<DeviceDescriptorBase>& /*device*/,
                         uint32_t* /*latencyMs*/,
-                        audio_output_flags_t /*flags*/,
+                        audio_output_flags_t* /*flags*/,
                         audio_attributes_t /*attributes*/) override { return NO_INIT; }
     audio_io_handle_t openDuplicateOutput(audio_io_handle_t /*output1*/,
                                           audio_io_handle_t /*output2*/) override {
diff --git a/services/audiopolicy/tests/audiopolicymanager_tests.cpp b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
index 08855c9..5278b73 100644
--- a/services/audiopolicy/tests/audiopolicymanager_tests.cpp
+++ b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
@@ -498,6 +498,9 @@
 void AudioPolicyManagerTestMsd::SetUpManagerConfig() {
     // TODO: Consider using Serializer to load part of the config from a string.
     ASSERT_NO_FATAL_FAILURE(AudioPolicyManagerTest::SetUpManagerConfig());
+    mConfig->getHwModules().getModuleFromName(
+            AUDIO_HARDWARE_MODULE_ID_PRIMARY)->setHalVersion(3, 0);
+
     mMsdOutputDevice = new DeviceDescriptor(AUDIO_DEVICE_OUT_BUS);
     sp<AudioProfile> pcmOutputProfile = new AudioProfile(
             AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO, k48000SamplingRate);
@@ -529,7 +532,7 @@
                 addOutputProfile(spdifOutputProfile);
     }
 
-    sp<HwModule> msdModule = new HwModule(AUDIO_HARDWARE_MODULE_ID_MSD, 2 /*halVersionMajor*/);
+    sp<HwModule> msdModule = new HwModule(AUDIO_HARDWARE_MODULE_ID_MSD, 3 /*halVersionMajor*/);
     HwModuleCollection modules = mConfig->getHwModules();
     modules.add(msdModule);
     mConfig->setHwModules(modules);
@@ -2517,7 +2520,7 @@
                         audio_config_base_t * mixerConfig,
                         const sp<DeviceDescriptorBase>& device,
                         uint32_t * latencyMs,
-                        audio_output_flags_t flags,
+                        audio_output_flags_t *flags,
                         audio_attributes_t attributes) override {
         return mSimulateFailure ? BAD_VALUE :
                 AudioPolicyManagerTestClient::openOutput(
@@ -2539,8 +2542,29 @@
 
     void setSimulateFailure(bool simulateFailure) { mSimulateFailure = simulateFailure; }
 
+    void setSimulateBroadcastDeviceStatus(audio_devices_t device, status_t status) {
+        if (status != NO_ERROR) {
+            // simulate device connect status
+            mSimulateBroadcastDeviceStatus[device] = status;
+        } else {
+            // remove device connection fixed status
+            mSimulateBroadcastDeviceStatus.erase(device);
+        }
+    }
+
+    status_t setDeviceConnectedState(const struct audio_port_v7* port,
+                                     media::DeviceConnectedState state) override {
+        if (mSimulateBroadcastDeviceStatus.find(port->ext.device.type) !=
+            mSimulateBroadcastDeviceStatus.end()) {
+            // If a simulated status exists, return a status value
+            return mSimulateBroadcastDeviceStatus[port->ext.device.type];
+        }
+        return AudioPolicyManagerTestClient::setDeviceConnectedState(port, state);
+    }
+
   private:
     bool mSimulateFailure = false;
+    std::map<audio_devices_t, status_t> mSimulateBroadcastDeviceStatus;
 };
 
 }  // namespace
@@ -2561,6 +2585,9 @@
     void setSimulateOpenFailure(bool simulateFailure) {
         mFullClient->setSimulateFailure(simulateFailure); }
 
+    void setSimulateBroadcastDeviceStatus(audio_devices_t device, status_t status) {
+        mFullClient->setSimulateBroadcastDeviceStatus(device, status); }
+
     static const std::string sBluetoothConfig;
 
   private:
@@ -2604,6 +2631,30 @@
     }
 }
 
+TEST_P(AudioPolicyManagerTestDeviceConnectionFailed, BroadcastDeviceFailure) {
+    const audio_devices_t type = std::get<0>(GetParam());
+    const std::string name = std::get<1>(GetParam());
+    const std::string address = std::get<2>(GetParam());
+    const audio_format_t format = std::get<3>(GetParam());
+
+    // simulate broadcastDeviceConnectionState return failure
+    setSimulateBroadcastDeviceStatus(type, INVALID_OPERATION);
+    ASSERT_EQ(INVALID_OPERATION, mManager->setDeviceConnectionState(
+            type, AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+            address.c_str(), name.c_str(), format));
+
+    // if broadcast is fail, device should not be added to available devices list
+    if (audio_is_output_device(type)) {
+        auto availableDevices = mManager->getAvailableOutputDevices();
+        EXPECT_FALSE(availableDevices.containsDeviceWithType(type));
+    } else if (audio_is_input_device(type)) {
+        auto availableDevices = mManager->getAvailableInputDevices();
+        EXPECT_FALSE(availableDevices.containsDeviceWithType(type));
+    }
+
+    setSimulateBroadcastDeviceStatus(type, NO_ERROR);
+}
+
 INSTANTIATE_TEST_CASE_P(
         DeviceConnectionFailure,
         AudioPolicyManagerTestDeviceConnectionFailed,