/*

 * Copied from frameworks/base/media/AudioSystems.h in the Android sources.

 * Modified by commenting out parts that we don't need

 * (especially dependencies on other header files).

 */

/*

 * Copyright (C) 2008 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_AUDIOSYSTEM_H_

#define ANDROID_AUDIOSYSTEM_H_



#if 0

#include <utils/RefBase.h>

#include <utils/threads.h>

#include <media/IAudioFlinger.h>

#endif



namespace android {



class AudioSystem

{

public:



    enum stream_type {

        DEFAULT          =-1,

        VOICE_CALL       = 0,

        SYSTEM           = 1,

        RING             = 2,

        MUSIC            = 3,

        ALARM            = 4,

        NOTIFICATION     = 5,

        BLUETOOTH_SCO    = 6,

        ENFORCED_AUDIBLE = 7, // Sounds that cannot be muted by user and must be routed to speaker

        DTMF             = 8,

        TTS              = 9,

        NUM_STREAM_TYPES

    };



    // Audio sub formats (see AudioSystem::audio_format).

    enum pcm_sub_format {

        PCM_SUB_16_BIT          = 0x1, // must be 1 for backward compatibility

        PCM_SUB_8_BIT           = 0x2, // must be 2 for backward compatibility

    };



    // MP3 sub format field definition : can use 11 LSBs in the same way as MP3 frame header to specify

    // bit rate, stereo mode, version...

    enum mp3_sub_format {

        //TODO

    };



    // AMR NB/WB sub format field definition: specify frame block interleaving, bandwidth efficient or octet aligned,

    // encoding mode for recording...

    enum amr_sub_format {

        //TODO

    };



    // AAC sub format field definition: specify profile or bitrate for recording...

    enum aac_sub_format {

        //TODO

    };



    // VORBIS sub format field definition: specify quality for recording...

    enum vorbis_sub_format {

        //TODO

    };



    // Audio format consists in a main format field (upper 8 bits) and a sub format field (lower 24 bits).

    // The main format indicates the main codec type. The sub format field indicates options and parameters

    // for each format. The sub format is mainly used for record to indicate for instance the requested bitrate

    // or profile. It can also be used for certain formats to give informations not present in the encoded

    // audio stream (e.g. octet alignement for AMR).

    enum audio_format {

        INVALID_FORMAT      = -1,

        FORMAT_DEFAULT      = 0,

        PCM                 = 0x00000000, // must be 0 for backward compatibility

        MP3                 = 0x01000000,

        AMR_NB              = 0x02000000,

        AMR_WB              = 0x03000000,

        AAC                 = 0x04000000,

        HE_AAC_V1           = 0x05000000,

        HE_AAC_V2           = 0x06000000,

        VORBIS              = 0x07000000,

        MAIN_FORMAT_MASK    = 0xFF000000,

        SUB_FORMAT_MASK     = 0x00FFFFFF,

        // Aliases

        PCM_16_BIT          = (PCM|PCM_SUB_16_BIT),

        PCM_8_BIT          = (PCM|PCM_SUB_8_BIT)

    };





    // Channel mask definitions must be kept in sync with JAVA values in /media/java/android/media/AudioFormat.java

    enum audio_channels {

        // output channels

        CHANNEL_OUT_FRONT_LEFT = 0x4,

        CHANNEL_OUT_FRONT_RIGHT = 0x8,

        CHANNEL_OUT_FRONT_CENTER = 0x10,

        CHANNEL_OUT_LOW_FREQUENCY = 0x20,

        CHANNEL_OUT_BACK_LEFT = 0x40,

        CHANNEL_OUT_BACK_RIGHT = 0x80,

        CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x100,

        CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x200,

        CHANNEL_OUT_BACK_CENTER = 0x400,

        CHANNEL_OUT_MONO = CHANNEL_OUT_FRONT_LEFT,

        CHANNEL_OUT_STEREO = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT),

        CHANNEL_OUT_QUAD = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |

                CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT),

        CHANNEL_OUT_SURROUND = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |

                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_BACK_CENTER),

        CHANNEL_OUT_5POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |

                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT),

        CHANNEL_OUT_7POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |

                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT |

                CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER),

        CHANNEL_OUT_ALL = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |

                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT |

                CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER | CHANNEL_OUT_BACK_CENTER),



        // input channels

        CHANNEL_IN_LEFT = 0x4,

        CHANNEL_IN_RIGHT = 0x8,

        CHANNEL_IN_FRONT = 0x10,

        CHANNEL_IN_BACK = 0x20,

        CHANNEL_IN_LEFT_PROCESSED = 0x40,

        CHANNEL_IN_RIGHT_PROCESSED = 0x80,

        CHANNEL_IN_FRONT_PROCESSED = 0x100,

        CHANNEL_IN_BACK_PROCESSED = 0x200,

        CHANNEL_IN_PRESSURE = 0x400,

        CHANNEL_IN_X_AXIS = 0x800,

        CHANNEL_IN_Y_AXIS = 0x1000,

        CHANNEL_IN_Z_AXIS = 0x2000,

        CHANNEL_IN_VOICE_UPLINK = 0x4000,

        CHANNEL_IN_VOICE_DNLINK = 0x8000,

        CHANNEL_IN_MONO = CHANNEL_IN_FRONT,

        CHANNEL_IN_STEREO = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT),

        CHANNEL_IN_ALL = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT | CHANNEL_IN_FRONT | CHANNEL_IN_BACK|

                CHANNEL_IN_LEFT_PROCESSED | CHANNEL_IN_RIGHT_PROCESSED | CHANNEL_IN_FRONT_PROCESSED | CHANNEL_IN_BACK_PROCESSED|

                CHANNEL_IN_PRESSURE | CHANNEL_IN_X_AXIS | CHANNEL_IN_Y_AXIS | CHANNEL_IN_Z_AXIS |

                CHANNEL_IN_VOICE_UPLINK | CHANNEL_IN_VOICE_DNLINK)

    };



    enum audio_mode {

        MODE_INVALID = -2,

        MODE_CURRENT = -1,

        MODE_NORMAL = 0,

        MODE_RINGTONE,

        MODE_IN_CALL,

        NUM_MODES  // not a valid entry, denotes end-of-list

    };



    enum audio_in_acoustics {

        AGC_ENABLE    = 0x0001,

        AGC_DISABLE   = 0,

        NS_ENABLE     = 0x0002,

        NS_DISABLE    = 0,

        TX_IIR_ENABLE = 0x0004,

        TX_DISABLE    = 0

    };



#if 0

    /* These are static methods to control the system-wide AudioFlinger

     * only privileged processes can have access to them

     */



    // mute/unmute microphone

    static status_t muteMicrophone(bool state);

    static status_t isMicrophoneMuted(bool *state);



    // set/get master volume

    static status_t setMasterVolume(float value);

    static status_t getMasterVolume(float* volume);

    // mute/unmute audio outputs

    static status_t setMasterMute(bool mute);

    static status_t getMasterMute(bool* mute);



    // set/get stream volume on specified output

    static status_t setStreamVolume(int stream, float value, int output);

    static status_t getStreamVolume(int stream, float* volume, int output);



    // mute/unmute stream

    static status_t setStreamMute(int stream, bool mute);

    static status_t getStreamMute(int stream, bool* mute);



    // set audio mode in audio hardware (see AudioSystem::audio_mode)

    static status_t setMode(int mode);



    // returns true in *state if tracks are active on the specified stream

    static status_t isStreamActive(int stream, bool *state);



    // set/get audio hardware 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).

    static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);

    static String8  getParameters(audio_io_handle_t ioHandle, const String8& keys);



    static void setErrorCallback(audio_error_callback cb);



    // helper function to obtain AudioFlinger service handle

    static const sp<IAudioFlinger>& get_audio_flinger();



    static float linearToLog(int volume);

    static int logToLinear(float volume);



    static status_t getOutputSamplingRate(int* samplingRate, int stream = DEFAULT);

    static status_t getOutputFrameCount(int* frameCount, int stream = DEFAULT);

    static status_t getOutputLatency(uint32_t* latency, int stream = DEFAULT);



    static bool routedToA2dpOutput(int streamType);



    static status_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount,

        size_t* buffSize);



    static status_t setVoiceVolume(float volume);



    // return the number of audio frames written by AudioFlinger to audio HAL and

    // audio dsp to DAC since the output on which the specificed stream is playing

    // has exited standby.

    // returned status (from utils/Errors.h) can be:

    // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data

    // - INVALID_OPERATION: Not supported on current hardware platform

    // - BAD_VALUE: invalid parameter

    // NOTE: this feature is not supported on all hardware platforms and it is

    // necessary to check returned status before using the returned values.

    static status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int stream = DEFAULT);



    //

    // AudioPolicyService interface

    //



    enum audio_devices {

        // output devices

        DEVICE_OUT_EARPIECE = 0x1,

        DEVICE_OUT_SPEAKER = 0x2,

        DEVICE_OUT_WIRED_HEADSET = 0x4,

        DEVICE_OUT_WIRED_HEADPHONE = 0x8,

        DEVICE_OUT_BLUETOOTH_SCO = 0x10,

        DEVICE_OUT_BLUETOOTH_SCO_HEADSET = 0x20,

        DEVICE_OUT_BLUETOOTH_SCO_CARKIT = 0x40,

        DEVICE_OUT_BLUETOOTH_A2DP = 0x80,

        DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100,

        DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = 0x200,

        DEVICE_OUT_AUX_DIGITAL = 0x400,

        DEVICE_OUT_DEFAULT = 0x8000,

        DEVICE_OUT_ALL = (DEVICE_OUT_EARPIECE | DEVICE_OUT_SPEAKER | DEVICE_OUT_WIRED_HEADSET |

                DEVICE_OUT_WIRED_HEADPHONE | DEVICE_OUT_BLUETOOTH_SCO | DEVICE_OUT_BLUETOOTH_SCO_HEADSET |

                DEVICE_OUT_BLUETOOTH_SCO_CARKIT | DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |

                DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER | DEVICE_OUT_AUX_DIGITAL | DEVICE_OUT_DEFAULT),

        DEVICE_OUT_ALL_A2DP = (DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |

                DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),



        // input devices

        DEVICE_IN_COMMUNICATION = 0x10000,

        DEVICE_IN_AMBIENT = 0x20000,

        DEVICE_IN_BUILTIN_MIC = 0x40000,

        DEVICE_IN_BLUETOOTH_SCO_HEADSET = 0x80000,

        DEVICE_IN_WIRED_HEADSET = 0x100000,

        DEVICE_IN_AUX_DIGITAL = 0x200000,

        DEVICE_IN_VOICE_CALL = 0x400000,

        DEVICE_IN_BACK_MIC = 0x800000,

        DEVICE_IN_DEFAULT = 0x80000000,



        DEVICE_IN_ALL = (DEVICE_IN_COMMUNICATION | DEVICE_IN_AMBIENT | DEVICE_IN_BUILTIN_MIC |

                DEVICE_IN_BLUETOOTH_SCO_HEADSET | DEVICE_IN_WIRED_HEADSET | DEVICE_IN_AUX_DIGITAL |

                DEVICE_IN_VOICE_CALL | DEVICE_IN_BACK_MIC | DEVICE_IN_DEFAULT)

    };



    // device connection states used for setDeviceConnectionState()

    enum device_connection_state {

        DEVICE_STATE_UNAVAILABLE,

        DEVICE_STATE_AVAILABLE,

        NUM_DEVICE_STATES

    };



    // request to open a direct output with getOutput() (by opposition to sharing an output with other AudioTracks)

    enum output_flags {

        OUTPUT_FLAG_INDIRECT = 0x0,

        OUTPUT_FLAG_DIRECT = 0x1

    };



    // device categories used for setForceUse()

    enum forced_config {

        FORCE_NONE,

        FORCE_SPEAKER,

        FORCE_HEADPHONES,

        FORCE_BT_SCO,

        FORCE_BT_A2DP,

        FORCE_WIRED_ACCESSORY,

        FORCE_BT_CAR_DOCK,

        FORCE_BT_DESK_DOCK,

        NUM_FORCE_CONFIG,

        FORCE_DEFAULT = FORCE_NONE

    };



    // usages used for setForceUse()

    enum force_use {

        FOR_COMMUNICATION,

        FOR_MEDIA,

        FOR_RECORD,

        FOR_DOCK,

        NUM_FORCE_USE

    };



    // types of io configuration change events received with ioConfigChanged()

    enum io_config_event {

        OUTPUT_OPENED,

        OUTPUT_CLOSED,

        OUTPUT_CONFIG_CHANGED,

        INPUT_OPENED,

        INPUT_CLOSED,

        INPUT_CONFIG_CHANGED,

        STREAM_CONFIG_CHANGED,

        NUM_CONFIG_EVENTS

    };



    // audio output descritor used to cache output configurations in client process to avoid frequent calls

    // through IAudioFlinger

    class OutputDescriptor {

    public:

        OutputDescriptor()

        : samplingRate(0), format(0), channels(0), frameCount(0), latency(0)  {}



        uint32_t samplingRate;

        int32_t format;

        int32_t channels;

        size_t frameCount;

        uint32_t latency;

    };



    //

    // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions)

    //

    static status_t setDeviceConnectionState(audio_devices device, device_connection_state state, const char *device_address);

    static device_connection_state getDeviceConnectionState(audio_devices device, const char *device_address);

    static status_t setPhoneState(int state);

    static status_t setRingerMode(uint32_t mode, uint32_t mask);

    static status_t setForceUse(force_use usage, forced_config config);

    static forced_config getForceUse(force_use usage);

    static audio_io_handle_t getOutput(stream_type stream,

                                        uint32_t samplingRate = 0,

                                        uint32_t format = FORMAT_DEFAULT,

                                        uint32_t channels = CHANNEL_OUT_STEREO,

                                        output_flags flags = OUTPUT_FLAG_INDIRECT);

    static status_t startOutput(audio_io_handle_t output, AudioSystem::stream_type stream);

    static status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream);

    static void releaseOutput(audio_io_handle_t output);

    static audio_io_handle_t getInput(int inputSource,

                                    uint32_t samplingRate = 0,

                                    uint32_t format = FORMAT_DEFAULT,

                                    uint32_t channels = CHANNEL_IN_MONO,

                                    audio_in_acoustics acoustics = (audio_in_acoustics)0);

    static status_t startInput(audio_io_handle_t input);

    static status_t stopInput(audio_io_handle_t input);

    static void releaseInput(audio_io_handle_t input);

    static status_t initStreamVolume(stream_type stream,

                                      int indexMin,

                                      int indexMax);

    static status_t setStreamVolumeIndex(stream_type stream, int index);

    static status_t getStreamVolumeIndex(stream_type stream, int *index);



    static const sp<IAudioPolicyService>& get_audio_policy_service();



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



    static uint32_t popCount(uint32_t u);

    static bool isOutputDevice(audio_devices device);

    static bool isInputDevice(audio_devices device);

    static bool isA2dpDevice(audio_devices device);

    static bool isBluetoothScoDevice(audio_devices device);

    static bool isLowVisibility(stream_type stream);

    static bool isOutputChannel(uint32_t channel);

    static bool isInputChannel(uint32_t channel);

    static bool isValidFormat(uint32_t format);

    static bool isLinearPCM(uint32_t format);



private:



    class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient

    {

    public:

        AudioFlingerClient() {

        }



        // DeathRecipient

        virtual void binderDied(const wp<IBinder>& who);



        // IAudioFlingerClient



        // indicate a change in the configuration of an output or input: keeps the cached

        // values for output/input parameters upto date in client process

        virtual void ioConfigChanged(int event, int ioHandle, void *param2);

    };



    class AudioPolicyServiceClient: public IBinder::DeathRecipient

    {

    public:

        AudioPolicyServiceClient() {

        }



        // DeathRecipient

        virtual void binderDied(const wp<IBinder>& who);

    };



    static sp<AudioFlingerClient> gAudioFlingerClient;

    static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient;

    friend class AudioFlingerClient;

    friend class AudioPolicyServiceClient;



    static Mutex gLock;

    static sp<IAudioFlinger> gAudioFlinger;

    static audio_error_callback gAudioErrorCallback;



    static size_t gInBuffSize;

    // previous parameters for recording buffer size queries

    static uint32_t gPrevInSamplingRate;

    static int gPrevInFormat;

    static int gPrevInChannelCount;



    static sp<IAudioPolicyService> gAudioPolicyService;



    // mapping between stream types and outputs

    static DefaultKeyedVector<int, audio_io_handle_t> gStreamOutputMap;

    // list of output descritor containing cached parameters (sampling rate, framecount, channel count...)

    static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs;

};



class AudioParameter {



public:

    AudioParameter() {}

    AudioParameter(const String8& keyValuePairs);

    virtual ~AudioParameter();



    // reserved parameter keys for changeing standard parameters with setParameters() function.

    // Using these keys is mandatory for AudioFlinger to properly monitor audio output/input

    // configuration changes and act accordingly.

    //  keyRouting: to change audio routing, value is an int in AudioSystem::audio_devices

    //  keySamplingRate: to change sampling rate routing, value is an int

    //  keyFormat: to change audio format, value is an int in AudioSystem::audio_format

    //  keyChannels: to change audio channel configuration, value is an int in AudioSystem::audio_channels

    //  keyFrameCount: to change audio output frame count, value is an int

    static const char *keyRouting;

    static const char *keySamplingRate;

    static const char *keyFormat;

    static const char *keyChannels;

    static const char *keyFrameCount;



    String8 toString();



    status_t add(const String8& key, const String8& value);

    status_t addInt(const String8& key, const int value);

    status_t addFloat(const String8& key, const float value);



    status_t remove(const String8& key);



    status_t get(const String8& key, String8& value);

    status_t getInt(const String8& key, int& value);

    status_t getFloat(const String8& key, float& value);

    status_t getAt(size_t index, String8& key, String8& value);



    size_t size() { return mParameters.size(); }



private:

    String8 mKeyValuePairs;

    KeyedVector <String8, String8> mParameters;

#endif

};



};  // namespace android



#endif  /*ANDROID_AUDIOSYSTEM_H_*/