Merge changes from topics "presubmit-am-87e43861d082492cad38dc75a9f31cdb",... (bd98611f) · Commits · e / os / android_frameworks_av

services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h

+1 −0

Original line number	Diff line number	Diff line
		@@ -305,6 +305,7 @@ public:
		{
		return !devices().isEmpty() ? devices().itemAt(0)->hasGainController() : false;
		}
		bool isRouted() const { return mPatchHandle != AUDIO_PATCH_HANDLE_NONE; }

		DeviceVector mDevices; /*< current devices this output is routed to /
		wp<AudioPolicyMix> mPolicyMix; // non NULL when used by a dynamic policy

services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h

+50 −2

Original line number	Diff line number	Diff line
		@@ -56,7 +56,13 @@ public:

		virtual void dump(String8 *dst, int spaces) const;
		virtual std::string toShortString() const;

		/**
		* @brief isInternal
		* @return true if the client corresponds to an audio patch created from createAudioPatch API or
		* for call audio routing, or false if the client corresponds to an AudioTrack, AudioRecord or
		* HW Audio Source.
		*/
		virtual bool isInternal() const { return false; }
		audio_port_handle_t portId() const { return mPortId; }
		uid_t uid() const { return mUid; }
		audio_session_t session() const { return mSessionId; };
		@@ -69,8 +75,16 @@ public:
		bool isPreferredDeviceForExclusiveUse() const { return mPreferredDeviceForExclusiveUse; }
		virtual void setActive(bool active) { mActive = active; }
		bool active() const { return mActive; }
		/**
		* @brief hasPreferredDevice Note that as internal clients use preferred device for convenience,
		* we do hide this internal behavior to prevent from regression (like invalidating track for
		* clients following same strategies...)
		* @param activeOnly
		* @return
		*/
		bool hasPreferredDevice(bool activeOnly = false) const {
		return mPreferredDeviceId != AUDIO_PORT_HANDLE_NONE && (!activeOnly \|\| mActive);
		return !isInternal() &&
		mPreferredDeviceId != AUDIO_PORT_HANDLE_NONE && (!activeOnly \|\| mActive);
		}

		private:
		@@ -211,6 +225,11 @@ public:
		mPatchHandle = AUDIO_PATCH_HANDLE_NONE;
		mSinkDevice = nullptr;
		}
		bool belongsToOutput(const sp<SwAudioOutputDescriptor> &swOutput) const {
		return swOutput != nullptr && mSwOutput.promote() == swOutput;
		}
		void setUseSwBridge() { mUseSwBridge = true; }
		bool useSwBridge() const { return mUseSwBridge; }
		bool isConnected() const { return mPatchHandle != AUDIO_PATCH_HANDLE_NONE; }
		audio_patch_handle_t getPatchHandle() const { return mPatchHandle; }
		sp<DeviceDescriptor> srcDevice() const { return mSrcDevice; }
		@@ -229,6 +248,35 @@ public:
		sp<DeviceDescriptor> mSinkDevice;
		wp<SwAudioOutputDescriptor> mSwOutput;
		wp<HwAudioOutputDescriptor> mHwOutput;
		bool mUseSwBridge = false;
		};

		/**
		* @brief The InternalSourceClientDescriptor class
		* Specialized Client Descriptor for either a raw patch created from @see createAudioPatch API
		* or for internal audio patches managed by APM (e.g. phone call patches).
		* Whatever the bridge created (software or hardware), we need a client to track the activity
		* and manage volumes.
		* The Audio Patch requested sink is expressed as a preferred device which allows to route
		* the SwOutput. Then APM will performs checks on the UID (against UID of Audioserver) of the
		* requester to prevent rerouting SwOutput involved in raw patches.
		*/
		class InternalSourceClientDescriptor: public SourceClientDescriptor
		{
		public:
		InternalSourceClientDescriptor(
		audio_port_handle_t portId, uid_t uid, audio_attributes_t attributes,
		const struct audio_port_config &config, const sp<DeviceDescriptor>& srcDevice,
		const sp<DeviceDescriptor>& sinkDevice,
		product_strategy_t strategy, VolumeSource volumeSource) :
		SourceClientDescriptor(
		portId, uid, attributes, config, srcDevice, AUDIO_STREAM_PATCH, strategy,
		volumeSource)
		{
		setPreferredDeviceId(sinkDevice->getId());
		}
		bool isInternal() const override { return true; }
		~InternalSourceClientDescriptor() override = default;
		};

		class SourceClientCollection :

services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp

+2 −1

Original line number	Diff line number	Diff line
		@@ -100,7 +100,8 @@ SourceClientDescriptor::SourceClientDescriptor(audio_port_handle_t portId, uid_t
		TrackClientDescriptor::TrackClientDescriptor(portId, uid, AUDIO_SESSION_NONE, attributes,
		{config.sample_rate, config.channel_mask, config.format}, AUDIO_PORT_HANDLE_NONE,
		stream, strategy, volumeSource, AUDIO_OUTPUT_FLAG_NONE, false,
		{} /* Sources do not support secondary outputs*/, nullptr), mSrcDevice(srcDevice)
		{} /* Sources do not support secondary outputs*/, nullptr),
		mSrcDevice(srcDevice)
		{
		}

services/audiopolicy/managerdefault/AudioPolicyManager.cpp

+244 −154

File changed.

Preview size limit exceeded, changes collapsed.

services/audiopolicy/managerdefault/AudioPolicyManager.h

+36 −15

Original line number	Diff line number	Diff line
		@@ -263,10 +263,7 @@ public:
		virtual status_t getAudioPort(struct audio_port_v7 *port);
		virtual status_t createAudioPatch(const struct audio_patch *patch,
		audio_patch_handle_t *handle,
		uid_t uid) {
		return createAudioPatchInternal(patch, handle, uid);
		}

		uid_t uid);
		virtual status_t releaseAudioPatch(audio_patch_handle_t handle,
		uid_t uid);
		virtual status_t listAudioPatches(unsigned int *num_patches,
		@@ -638,13 +635,22 @@ protected:
		void updateCallAndOutputRouting(bool forceVolumeReeval = true, uint32_t delayMs = 0);

		bool isCallRxAudioSource(const sp<SourceClientDescriptor> &source) {
		return mCallRxSourceClientPort != AUDIO_PORT_HANDLE_NONE
		&& source == mAudioSources.valueFor(mCallRxSourceClientPort);
		return mCallRxSourceClient != nullptr && source == mCallRxSourceClient;
		}

		void connectTelephonyRxAudioSource();

		void disconnectTelephonyRxAudioSource();
		void disconnectTelephonyAudioSource(sp<SourceClientDescriptor> &clientDesc);

		void connectTelephonyTxAudioSource(const sp<DeviceDescriptor> &srcdevice,
		const sp<DeviceDescriptor> &sinkDevice,
		uint32_t delayMs);

		bool isTelephonyRxOrTx(const sp<SwAudioOutputDescriptor>& desc) const {
		return (mCallRxSourceClient != nullptr && mCallRxSourceClient->belongsToOutput(desc))
		\|\| (mCallTxSourceClient != nullptr
		&& mCallTxSourceClient->belongsToOutput(desc));
		}

		/**
		* @brief updates routing for all inputs.
		@@ -851,6 +857,12 @@ protected:
		status_t connectAudioSource(const sp<SourceClientDescriptor>& sourceDesc);
		status_t disconnectAudioSource(const sp<SourceClientDescriptor>& sourceDesc);

		status_t connectAudioSourceToSink(const sp<SourceClientDescriptor>& sourceDesc,
		const sp<DeviceDescriptor> &sinkDevice,
		const struct audio_patch *patch,
		audio_patch_handle_t &handle,
		uid_t uid, uint32_t delayMs);

		sp<SourceClientDescriptor> getSourceForAttributesOnOutput(audio_io_handle_t output,
		const audio_attributes_t &attr);
		void clearAudioSourcesForOutput(audio_io_handle_t output);
		@@ -901,8 +913,6 @@ protected:

		SoundTriggerSessionCollection mSoundTriggerSessions;

		sp<AudioPatch> mCallTxPatch;

		HwAudioOutputCollection mHwOutputs;
		SourceClientCollection mAudioSources;

		@@ -943,7 +953,8 @@ protected:

		// The port handle of the hardware audio source created internally for the Call RX audio
		// end point.
		audio_port_handle_t mCallRxSourceClientPort = AUDIO_PORT_HANDLE_NONE;
		sp<SourceClientDescriptor> mCallRxSourceClient;
		sp<SourceClientDescriptor> mCallTxSourceClient;

		// Support for Multi-Stream Decoder (MSD) module
		sp<DeviceDescriptor> getMsdAudioInDevice() const;
		@@ -975,7 +986,13 @@ protected:
		// Called by setDeviceConnectionState()
		status_t deviceToAudioPort(audio_devices_t deviceType, const char* device_address,
		const char* device_name, media::AudioPort* aidPort);
		bool isMsdPatch(const audio_patch_handle_t &handle) const;

		private:
		sp<SourceClientDescriptor> startAudioSourceInternal(
		const struct audio_port_config source, const audio_attributes_t attributes,
		uid_t uid);

		void onNewAudioModulesAvailableInt(DeviceVector *newDevices);

		// Add or remove AC3 DTS encodings based on user preferences.
		@@ -1120,21 +1137,25 @@ private:
		* @param[out] handle patch handle to be provided if patch installed correctly
		* @param[in] uid of the client
		* @param[in] delayMs if required
		* @param[in] sourceDesc [optional] in case of external source, source client to be
		* configured by the patch, i.e. assigning an Output (HW or SW)
		* @param[in] sourceDesc source client to be configured when creating the patch, i.e.
		* assigning an Output (HW or SW) used for volume control.
		* @return NO_ERROR if patch installed correctly, error code otherwise.
		*/
		status_t createAudioPatchInternal(const struct audio_patch *patch,
		audio_patch_handle_t *handle,
		uid_t uid, uint32_t delayMs = 0,
		const sp<SourceClientDescriptor>& sourceDesc = nullptr);
		uid_t uid, uint32_t delayMs,
		const sp<SourceClientDescriptor>& sourceDesc);
		/**
		* @brief releaseAudioPatchInternal internal function to remove an audio patch
		* @param[in] handle of the patch to be removed
		* @param[in] delayMs if required
		* @param[in] sourceDesc [optional] in case of external source, source client to be
		* unrouted from the patch, i.e. assigning an Output (HW or SW)
		* @return NO_ERROR if patch removed correctly, error code otherwise.
		*/
		status_t releaseAudioPatchInternal(audio_patch_handle_t handle, uint32_t delayMs = 0);
		status_t releaseAudioPatchInternal(audio_patch_handle_t handle,
		uint32_t delayMs = 0,
		const sp<SourceClientDescriptor>& sourceDesc = nullptr);

		status_t installPatch(const char *caller,
		audio_patch_handle_t *patchHandle,