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Diffstat (limited to 'audio/AudioSystem.h')
| -rwxr-xr-x | audio/AudioSystem.h | 562 |
1 files changed, 562 insertions, 0 deletions
diff --git a/audio/AudioSystem.h b/audio/AudioSystem.h new file mode 100755 index 0000000..bd75a73 --- /dev/null +++ b/audio/AudioSystem.h @@ -0,0 +1,562 @@ +/* + * 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_ +#define ANDROID_AUDIOPARAMETER_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, +#ifdef HAVE_FM_RADIO + FM = 10, +#endif + 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, +#ifdef OMAP_ENHANCEMENT + CHANNEL_IN_VOICE_UPLINK_DNLINK = 0x10000, +#endif + 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 | +#ifdef OMAP_ENHANCEMENT + CHANNEL_IN_VOICE_UPLINK | CHANNEL_IN_VOICE_DNLINK | CHANNEL_IN_VOICE_UPLINK_DNLINK) +#else + CHANNEL_IN_VOICE_UPLINK | CHANNEL_IN_VOICE_DNLINK ) +#endif + }; + + enum audio_mode { + MODE_INVALID = -2, + MODE_CURRENT = -1, + MODE_NORMAL = 0, + MODE_RINGTONE, + MODE_IN_CALL, + MODE_IN_COMMUNICATION, + 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 + }; + + // special audio session values + enum audio_sessions { + SESSION_OUTPUT_STAGE = -1, // session for effects attached to a particular output stream + // (value must be less than 0) + SESSION_OUTPUT_MIX = 0, // session for effects applied to output mix. These effects can + // be moved by audio policy manager to another output stream + // (value must be 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); +#ifdef HAVE_FM_RADIO + static status_t setFmVolume(float volume); +#endif + + // 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 = 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, +#ifdef HAVE_FM_RADIO + DEVICE_OUT_FM = 0x800, + DEVICE_OUT_FM_SPEAKER = 0x1000, + DEVICE_OUT_FM_ALL = (DEVICE_OUT_FM | DEVICE_OUT_FM_SPEAKER), +#elif defined(OMAP_ENHANCEMENT) + DEVICE_OUT_FM_TRANSMIT = 0x800, + DEVICE_OUT_LOW_POWER = 0x1000, +#endif + DEVICE_OUT_HDMI = 0x2000, + DEVICE_OUT_DEFAULT = 0x8000, + DEVICE_OUT_ALL = (DEVICE_OUT_EARPIECE | DEVICE_OUT_SPEAKER | DEVICE_OUT_WIRED_HEADSET | +#ifdef HAVE_FM_RADIO + DEVICE_OUT_WIRED_HEADPHONE | DEVICE_OUT_FM | DEVICE_OUT_FM_SPEAKER | DEVICE_OUT_BLUETOOTH_SCO | DEVICE_OUT_BLUETOOTH_SCO_HEADSET | +#else + DEVICE_OUT_WIRED_HEADPHONE | DEVICE_OUT_BLUETOOTH_SCO | DEVICE_OUT_BLUETOOTH_SCO_HEADSET | +#endif + DEVICE_OUT_BLUETOOTH_SCO_CARKIT | DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | +#if defined(OMAP_ENHANCEMENT) && !defined(HAVE_FM_RADIO) + DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER | DEVICE_OUT_AUX_DIGITAL | DEVICE_OUT_LOW_POWER | + DEVICE_OUT_FM_TRANSMIT | DEVICE_OUT_DEFAULT), +#else + DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER | DEVICE_OUT_AUX_DIGITAL | DEVICE_OUT_HDMI | DEVICE_OUT_DEFAULT), +#endif + 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, +#ifdef HAVE_FM_RADIO + DEVICE_IN_FM_RX = 0x1000000, + DEVICE_IN_FM_RX_A2DP = 0x2000000, +#endif +#ifdef OMAP_ENHANCEMENT + DEVICE_IN_FM_ANALOG = 0x1000000, +#endif + 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 | +#ifdef HAVE_FM_RADIO + DEVICE_IN_VOICE_CALL | DEVICE_IN_BACK_MIC | DEVICE_IN_FM_RX | DEVICE_IN_FM_RX_A2DP | DEVICE_IN_DEFAULT) +#elif OMAP_ENHANCEMENT + DEVICE_IN_VOICE_CALL | DEVICE_IN_BACK_MIC | DEVICE_IN_FM_ANALOG | DEVICE_IN_DEFAULT) +#else + DEVICE_IN_VOICE_CALL | DEVICE_IN_BACK_MIC | DEVICE_IN_DEFAULT) +#endif + + }; + + // 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, + int session = 0); + static status_t stopOutput(audio_io_handle_t output, + AudioSystem::stream_type stream, + int session = 0); + 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 uint32_t getStrategyForStream(stream_type stream); + + static audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc); + static status_t registerEffect(effect_descriptor_t *desc, + audio_io_handle_t output, + uint32_t strategy, + int session, + int id); + static status_t unregisterEffect(int id); + + 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); +#ifdef HAVE_FM_RADIO + static bool isFmDevice(audio_devices device); +#endif + 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 changing 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 + // keyInputSource: to change audio input source, value is an int in audio_source + // (defined in media/mediarecorder.h) + static const char *keyRouting; + static const char *keySamplingRate; + static const char *keyFormat; + static const char *keyChannels; + static const char *keyFrameCount; +#ifdef HAVE_FM_RADIO + static const char *keyFmOn; + static const char *keyFmOff; +#endif + static const char *keyInputSource; + + 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_*/ |