Create a snapshot of the hardware_legacy (7a7fd84c) · Commits · e / os / android_hardware_interfaces

wifi/1.6/default/hal_legacy/AudioHardwareBase.h

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		/*
		* Copyright (C) 2007 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.
		*/

		#ifndef ANDROID_AUDIO_HARDWARE_BASE_H
		#define ANDROID_AUDIO_HARDWARE_BASE_H

		#include <hardware_legacy/AudioHardwareInterface.h>

		#include <system/audio.h>

		namespace android_audio_legacy {

		// ----------------------------------------------------------------------------

		/**
		* AudioHardwareBase is a convenient base class used for implementing the
		* AudioHardwareInterface interface.
		*/
		class AudioHardwareBase : public AudioHardwareInterface {
		public:
		AudioHardwareBase();
		virtual ~AudioHardwareBase() {}

		/**
		* setMode is called when the audio mode changes. NORMAL mode is for
		* standard audio playback, RINGTONE when a ringtone is playing, IN_CALL
		* when a telephony call is in progress, IN_COMMUNICATION when a VoIP call is in progress.
		*/
		virtual status_t setMode(int mode);

		virtual status_t setParameters(const String8& keyValuePairs);
		virtual String8 getParameters(const String8& keys);

		virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount);
		virtual status_t getMasterVolume(float* volume);

		/*This method dumps the state of the audio hardware /
		virtual status_t dumpState(int fd, const Vector<String16>& args);

		protected:
		/** returns true if the given mode maps to a telephony or VoIP call is in progress */
		virtual bool isModeInCall(int mode) {
		return ((mode == AudioSystem::MODE_IN_CALL) \|\|
		(mode == AudioSystem::MODE_IN_COMMUNICATION));
		};
		/** returns true if a telephony or VoIP call is in progress */
		virtual bool isInCall() { return isModeInCall(mMode); };
		int mMode;
		};

		}; // namespace android_audio_legacy

		#endif // ANDROID_AUDIO_HARDWARE_BASE_H

wifi/1.6/default/hal_legacy/AudioHardwareInterface.h

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		/*
		* Copyright (C) 2007 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.
		*/

		#ifndef ANDROID_AUDIO_HARDWARE_INTERFACE_H
		#define ANDROID_AUDIO_HARDWARE_INTERFACE_H

		#include <stdint.h>
		#include <sys/types.h>

		#include <utils/Errors.h>
		#include <utils/String16.h>
		#include <utils/String8.h>
		#include <utils/Vector.h>

		#include <hardware_legacy/AudioSystemLegacy.h>

		#include <hardware/audio.h>
		#include <system/audio.h>

		#include <cutils/bitops.h>

		namespace android_audio_legacy {
		using android::String16;
		using android::String8;
		using android::Vector;

		// ----------------------------------------------------------------------------

		/**
		* AudioStreamOut is the abstraction interface for the audio output hardware.
		*
		* It provides information about various properties of the audio output hardware driver.
		*/
		class AudioStreamOut {
		public:
		virtual ~AudioStreamOut() = 0;

		/** return audio sampling rate in hz - eg. 44100 */
		virtual uint32_t sampleRate() const = 0;

		/** returns size of output buffer - eg. 4800 */
		virtual size_t bufferSize() const = 0;

		/**
		* returns the output channel mask
		*/
		virtual uint32_t channels() const = 0;

		/**
		* return audio format in 8bit or 16bit PCM format -
		* eg. AudioSystem:PCM_16_BIT
		*/
		virtual int format() const = 0;

		/**
		* return the frame size (number of bytes per sample).
		*/
		uint32_t frameSize() const {
		return audio_channel_count_from_out_mask(channels()) *
		((format() == AUDIO_FORMAT_PCM_16_BIT) ? sizeof(int16_t) : sizeof(int8_t));
		}

		/**
		* return the audio hardware driver latency in milli seconds.
		*/
		virtual uint32_t latency() const = 0;

		/**
		* Use this method in situations where audio mixing is done in the
		* hardware. This method serves as a direct interface with hardware,
		* allowing you to directly set the volume as apposed to via the framework.
		* This method might produce multiple PCM outputs or hardware accelerated
		* codecs, such as MP3 or AAC.
		*/
		virtual status_t setVolume(float left, float right) = 0;

		/** write audio buffer to driver. Returns number of bytes written */
		virtual ssize_t write(const void* buffer, size_t bytes) = 0;

		/**
		* Put the audio hardware output into standby mode. Returns
		* status based on include/utils/Errors.h
		*/
		virtual status_t standby() = 0;

		/** dump the state of the audio output device */
		virtual status_t dump(int fd, const Vector<String16>& args) = 0;

		// set/get audio output parameters. The function accepts a list of parameters
		// key value pairs in the form: key1=value1;key2=value2;...
		// Some keys are reserved for standard parameters (See AudioParameter class).
		// If the implementation does not accept a parameter change while the output is
		// active but the parameter is acceptable otherwise, it must return INVALID_OPERATION.
		// The audio flinger will put the output in standby and then change the parameter value.
		virtual status_t setParameters(const String8& keyValuePairs) = 0;
		virtual String8 getParameters(const String8& keys) = 0;

		// return the number of audio frames written by the audio dsp to DAC since
		// the output has exited standby
		virtual status_t getRenderPosition(uint32_t* dspFrames) = 0;

		/**
		* get the local time at which the next write to the audio driver will be
		* presented
		*/
		virtual status_t getNextWriteTimestamp(int64_t* timestamp);

		/**
		* Return a recent count of the number of audio frames presented to an external observer.
		*/
		virtual status_t getPresentationPosition(uint64_t* frames, struct timespec* timestamp);
		};

		/**
		* AudioStreamIn is the abstraction interface for the audio input hardware.
		*
		* It defines the various properties of the audio hardware input driver.
		*/
		class AudioStreamIn {
		public:
		virtual ~AudioStreamIn() = 0;

		/** return audio sampling rate in hz - eg. 44100 */
		virtual uint32_t sampleRate() const = 0;

		/** return the input buffer size allowed by audio driver */
		virtual size_t bufferSize() const = 0;

		/** return input channel mask */
		virtual uint32_t channels() const = 0;

		/**
		* return audio format in 8bit or 16bit PCM format -
		* eg. AudioSystem:PCM_16_BIT
		*/
		virtual int format() const = 0;

		/**
		* return the frame size (number of bytes per sample).
		*/
		uint32_t frameSize() const {
		return audio_channel_count_from_in_mask(channels()) *
		((format() == AudioSystem::PCM_16_BIT) ? sizeof(int16_t) : sizeof(int8_t));
		}

		/** set the input gain for the audio driver. This method is for
		* for future use */
		virtual status_t setGain(float gain) = 0;

		/** read audio buffer in from audio driver */
		virtual ssize_t read(void* buffer, ssize_t bytes) = 0;

		/** dump the state of the audio input device */
		virtual status_t dump(int fd, const Vector<String16>& args) = 0;

		/**
		* Put the audio hardware input into standby mode. Returns
		* status based on include/utils/Errors.h
		*/
		virtual status_t standby() = 0;

		// set/get audio input parameters. The function accepts a list of parameters
		// key value pairs in the form: key1=value1;key2=value2;...
		// Some keys are reserved for standard parameters (See AudioParameter class).
		// If the implementation does not accept a parameter change while the output is
		// active but the parameter is acceptable otherwise, it must return INVALID_OPERATION.
		// The audio flinger will put the input in standby and then change the parameter value.
		virtual status_t setParameters(const String8& keyValuePairs) = 0;
		virtual String8 getParameters(const String8& keys) = 0;

		// Return the number of input frames lost in the audio driver since the last call of this
		// function. Audio driver is expected to reset the value to 0 and restart counting upon
		// returning the current value by this function call. Such loss typically occurs when the user
		// space process is blocked longer than the capacity of audio driver buffers. Unit: the number
		// of input audio frames
		virtual unsigned int getInputFramesLost() const = 0;

		virtual status_t addAudioEffect(effect_handle_t effect) = 0;
		virtual status_t removeAudioEffect(effect_handle_t effect) = 0;
		};

		/**
		* AudioHardwareInterface.h defines the interface to the audio hardware abstraction layer.
		*
		* The interface supports setting and getting parameters, selecting audio routing
		* paths, and defining input and output streams.
		*
		* AudioFlinger initializes the audio hardware and immediately opens an output stream.
		* You can set Audio routing to output to handset, speaker, Bluetooth, or a headset.
		*
		* The audio input stream is initialized when AudioFlinger is called to carry out
		* a record operation.
		*/
		class AudioHardwareInterface {
		public:
		virtual ~AudioHardwareInterface() {}

		/**
		* check to see if the audio hardware interface has been initialized.
		* return status based on values defined in include/utils/Errors.h
		*/
		virtual status_t initCheck() = 0;

		/** set the audio volume of a voice call. Range is between 0.0 and 1.0 */
		virtual status_t setVoiceVolume(float volume) = 0;

		/**
		* set the audio volume for all audio activities other than voice call.
		* Range between 0.0 and 1.0. If any value other than NO_ERROR is returned,
		* the software mixer will emulate this capability.
		*/
		virtual status_t setMasterVolume(float volume) = 0;

		/**
		* Get the current master volume value for the HAL, if the HAL supports
		* master volume control. AudioFlinger will query this value from the
		* primary audio HAL when the service starts and use the value for setting
		* the initial master volume across all HALs.
		*/
		virtual status_t getMasterVolume(float* volume) = 0;

		/**
		* setMode is called when the audio mode changes. NORMAL mode is for
		* standard audio playback, RINGTONE when a ringtone is playing, and IN_CALL
		* when a call is in progress.
		*/
		virtual status_t setMode(int mode) = 0;

		// mic mute
		virtual status_t setMicMute(bool state) = 0;
		virtual status_t getMicMute(bool* state) = 0;

		// set/get global audio parameters
		virtual status_t setParameters(const String8& keyValuePairs) = 0;
		virtual String8 getParameters(const String8& keys) = 0;

		// Returns audio input buffer size according to parameters passed or 0 if one of the
		// parameters is not supported
		virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount) = 0;

		/** This method creates and opens the audio hardware output stream */
		virtual AudioStreamOut* openOutputStream(uint32_t devices, int* format = 0,
		uint32_t* channels = 0, uint32_t* sampleRate = 0,
		status_t* status = 0) = 0;
		virtual AudioStreamOut* openOutputStreamWithFlags(
		uint32_t devices, audio_output_flags_t flags = (audio_output_flags_t)0, int* format = 0,
		uint32_t* channels = 0, uint32_t* sampleRate = 0, status_t* status = 0) = 0;
		virtual void closeOutputStream(AudioStreamOut* out) = 0;

		/** This method creates and opens the audio hardware input stream */
		virtual AudioStreamIn* openInputStream(uint32_t devices, int* format, uint32_t* channels,
		uint32_t* sampleRate, status_t* status,
		AudioSystem::audio_in_acoustics acoustics) = 0;
		virtual void closeInputStream(AudioStreamIn* in) = 0;

		/*This method dumps the state of the audio hardware /
		virtual status_t dumpState(int fd, const Vector<String16>& args) = 0;

		virtual status_t setMasterMute(bool muted) = 0;

		static AudioHardwareInterface* create();

		virtual int createAudioPatch(unsigned int num_sources, const struct audio_port_config* sources,
		unsigned int num_sinks, const struct audio_port_config* sinks,
		audio_patch_handle_t* handle) = 0;

		virtual int releaseAudioPatch(audio_patch_handle_t handle) = 0;

		virtual int getAudioPort(struct audio_port* port) = 0;

		virtual int setAudioPortConfig(const struct audio_port_config* config) = 0;

		protected:
		virtual status_t dump(int fd, const Vector<String16>& args) = 0;
		};

		// ----------------------------------------------------------------------------

		extern "C" AudioHardwareInterface* createAudioHardware(void);

		}; // namespace android_audio_legacy

		#endif // ANDROID_AUDIO_HARDWARE_INTERFACE_H

wifi/1.6/default/hal_legacy/AudioPolicyInterface.h

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wifi/1.6/default/hal_legacy/AudioPolicyManagerBase.h

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wifi/1.6/default/hal_legacy/AudioSystemLegacy.h

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