/*
* 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>
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 = 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
};
/* 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);
//
// 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;
};
}; // namespace android
#endif /*ANDROID_AUDIOSYSTEM_H_*/