include/media/AudioSystem.h

/*
 * 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_

#include <utils/RefBase.h>
#include <utils/threads.h>
#include <media/IAudioFlinger.h>
#include <media/AudioCommon.h>

namespace android {

typedef void (*audio_error_callback)(status_t err);
typedef int audio_io_handle_t;

class IAudioPolicyService;
class String8;

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          = PCM_FORMAT_S15, // must be 1 for backward compatibility
        PCM_SUB_8_BIT           = PCM_FORMAT_U8, // 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 = CHANNEL_FRONT_LEFT,
        CHANNEL_OUT_FRONT_RIGHT = CHANNEL_FRONT_RIGHT,
        CHANNEL_OUT_FRONT_CENTER = CHANNEL_FRONT_CENTER,
        CHANNEL_OUT_LOW_FREQUENCY = CHANNEL_LOW_FREQUENCY,
        CHANNEL_OUT_BACK_LEFT = CHANNEL_BACK_LEFT,
        CHANNEL_OUT_BACK_RIGHT = CHANNEL_BACK_RIGHT,
        CHANNEL_OUT_FRONT_LEFT_OF_CENTER = CHANNEL_FRONT_LEFT_OF_CENTER,
        CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = CHANNEL_FRONT_RIGHT_OF_CENTER,
        CHANNEL_OUT_BACK_CENTER = CHANNEL_BACK_CENTER,
        CHANNEL_OUT_MONO = CHANNEL_MONO,
        CHANNEL_OUT_STEREO = CHANNEL_STEREO,
        CHANNEL_OUT_QUAD = CHANNEL_QUAD,
        CHANNEL_OUT_SURROUND = CHANNEL_SURROUND,
        CHANNEL_OUT_5POINT1 = CHANNEL_5POINT1,
        CHANNEL_OUT_7POINT1 = CHANNEL_7POINT1,
        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
    };

    /* 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);

    static unsigned int  getInputFramesLost(audio_io_handle_t ioHandle);

    static int newAudioSessionId();
    //
    // AudioPolicyService interface
    //

    enum audio_devices {
        // output devices
        DEVICE_OUT_EARPIECE = DEVICE_EARPIECE,
        DEVICE_OUT_SPEAKER = DEVICE_SPEAKER,
        DEVICE_OUT_WIRED_HEADSET = DEVICE_WIRED_HEADSET,
        DEVICE_OUT_WIRED_HEADPHONE = DEVICE_WIRED_HEADPHONE,
        DEVICE_OUT_BLUETOOTH_SCO = DEVICE_BLUETOOTH_SCO,
        DEVICE_OUT_BLUETOOTH_SCO_HEADSET = DEVICE_BLUETOOTH_SCO_HEADSET,
        DEVICE_OUT_BLUETOOTH_SCO_CARKIT = DEVICE_BLUETOOTH_SCO_CARKIT,
        DEVICE_OUT_BLUETOOTH_A2DP = DEVICE_BLUETOOTH_A2DP,
        DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = DEVICE_BLUETOOTH_A2DP_HEADPHONES,
        DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = DEVICE_BLUETOOTH_A2DP_SPEAKER,
        DEVICE_OUT_AUX_DIGITAL = DEVICE_AUX_DIGITAL,
        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;
};

};  // namespace android

#endif  /*ANDROID_AUDIOSYSTEM_H_*/