AudioSystem.h revision 4bcae82f9b07d1a39956c45a6f5bec0b696c4dd1 
1/*
2 * Copyright (C) 2008 The Android Open Source Project
3 *
4 * Licensed under the Apache License, Version 2.0 (the "License");
5 * you may not use this file except in compliance with the License.
6 * You may obtain a copy of the License at
7 *
8 *      http://www.apache.org/licenses/LICENSE-2.0
9 *
10 * Unless required by applicable law or agreed to in writing, software
11 * distributed under the License is distributed on an "AS IS" BASIS,
12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
13 * See the License for the specific language governing permissions and
14 * limitations under the License.
15 */
16
17#ifndef ANDROID_AUDIOSYSTEM_H_
18#define ANDROID_AUDIOSYSTEM_H_
19
20#include <utils/RefBase.h>
21#include <utils/threads.h>
22#include <media/IAudioFlinger.h>
23
24namespace android {
25
26typedef void (*audio_error_callback)(status_t err);
27typedef int audio_io_handle_t;
28
29class IAudioPolicyService;
30class String8;
31
32class AudioSystem
33{
34public:
35
36    // must match android/media/AudioSystem.java STREAM_* constants
37    enum stream_type {
38        DEFAULT          =-1,
39        VOICE_CALL       = 0,
40        SYSTEM           = 1,
41        RING             = 2,
42        MUSIC            = 3,
43        ALARM            = 4,
44        NOTIFICATION     = 5,
45        BLUETOOTH_SCO    = 6,
46        ENFORCED_AUDIBLE = 7, // Sounds that cannot be muted by user and must be routed to speaker
47        DTMF             = 8,
48        TTS              = 9,
49        NUM_STREAM_TYPES
50    };
51
52    // Audio sub formats (see AudioSystem::audio_format).
53    enum pcm_sub_format {
54        PCM_SUB_16_BIT          = 0x1, // must be 1 for backward compatibility
55        PCM_SUB_8_BIT           = 0x2, // must be 2 for backward compatibility
56    };
57
58    // FIXME These sub_format enums are currently unused
59
60    // MP3 sub format field definition : can use 11 LSBs in the same way as MP3 frame header to specify
61    // bit rate, stereo mode, version...
62    enum mp3_sub_format {
63        //TODO
64    };
65
66    // AMR NB/WB sub format field definition: specify frame block interleaving, bandwidth efficient or octet aligned,
67    // encoding mode for recording...
68    enum amr_sub_format {
69        //TODO
70    };
71
72    // AAC sub format field definition: specify profile or bitrate for recording...
73    enum aac_sub_format {
74        //TODO
75    };
76
77    // VORBIS sub format field definition: specify quality for recording...
78    enum vorbis_sub_format {
79        //TODO
80    };
81
82    // Audio format consists in a main format field (upper 8 bits) and a sub format field (lower 24 bits).
83    // The main format indicates the main codec type. The sub format field indicates options and parameters
84    // for each format. The sub format is mainly used for record to indicate for instance the requested bitrate
85    // or profile. It can also be used for certain formats to give informations not present in the encoded
86    // audio stream (e.g. octet alignement for AMR).
87    enum audio_format {
88        INVALID_FORMAT      = -1,
89        FORMAT_DEFAULT      = 0,
90        PCM                 = 0x00000000, // must be 0 for backward compatibility
91        MP3                 = 0x01000000,
92        AMR_NB              = 0x02000000,
93        AMR_WB              = 0x03000000,
94        AAC                 = 0x04000000,
95        HE_AAC_V1           = 0x05000000,
96        HE_AAC_V2           = 0x06000000,
97        VORBIS              = 0x07000000,
98        MAIN_FORMAT_MASK    = 0xFF000000,
99        SUB_FORMAT_MASK     = 0x00FFFFFF,
100        // Aliases
101        PCM_16_BIT          = (PCM|PCM_SUB_16_BIT),
102        PCM_8_BIT          = (PCM|PCM_SUB_8_BIT)
103    };
104
105
106    // Channel mask definitions must be kept in sync with values in /media/java/android/media/AudioFormat.java
107    enum audio_channels {
108        // output channels
109        CHANNEL_OUT_FRONT_LEFT = 0x4,
110        CHANNEL_OUT_FRONT_RIGHT = 0x8,
111        CHANNEL_OUT_FRONT_CENTER = 0x10,
112        CHANNEL_OUT_LOW_FREQUENCY = 0x20,
113        CHANNEL_OUT_BACK_LEFT = 0x40,
114        CHANNEL_OUT_BACK_RIGHT = 0x80,
115        CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x100,
116        CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x200,
117        CHANNEL_OUT_BACK_CENTER = 0x400,
118        CHANNEL_OUT_MONO = CHANNEL_OUT_FRONT_LEFT,
119        CHANNEL_OUT_STEREO = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT),
120        CHANNEL_OUT_QUAD = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
121                CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT),
122        CHANNEL_OUT_SURROUND = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
123                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_BACK_CENTER),
124        CHANNEL_OUT_5POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
125                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT),
126        CHANNEL_OUT_7POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
127                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT |
128                CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER),
129        CHANNEL_OUT_ALL = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
130                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT |
131                CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER | CHANNEL_OUT_BACK_CENTER),
132
133        // input channels
134        CHANNEL_IN_LEFT = 0x4,
135        CHANNEL_IN_RIGHT = 0x8,
136        CHANNEL_IN_FRONT = 0x10,
137        CHANNEL_IN_BACK = 0x20,
138        CHANNEL_IN_LEFT_PROCESSED = 0x40,
139        CHANNEL_IN_RIGHT_PROCESSED = 0x80,
140        CHANNEL_IN_FRONT_PROCESSED = 0x100,
141        CHANNEL_IN_BACK_PROCESSED = 0x200,
142        CHANNEL_IN_PRESSURE = 0x400,
143        CHANNEL_IN_X_AXIS = 0x800,
144        CHANNEL_IN_Y_AXIS = 0x1000,
145        CHANNEL_IN_Z_AXIS = 0x2000,
146        CHANNEL_IN_VOICE_UPLINK = 0x4000,
147        CHANNEL_IN_VOICE_DNLINK = 0x8000,
148        CHANNEL_IN_MONO = CHANNEL_IN_FRONT,
149        CHANNEL_IN_STEREO = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT),
150        CHANNEL_IN_ALL = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT | CHANNEL_IN_FRONT | CHANNEL_IN_BACK|
151                CHANNEL_IN_LEFT_PROCESSED | CHANNEL_IN_RIGHT_PROCESSED | CHANNEL_IN_FRONT_PROCESSED | CHANNEL_IN_BACK_PROCESSED|
152                CHANNEL_IN_PRESSURE | CHANNEL_IN_X_AXIS | CHANNEL_IN_Y_AXIS | CHANNEL_IN_Z_AXIS |
153                CHANNEL_IN_VOICE_UPLINK | CHANNEL_IN_VOICE_DNLINK)
154    };
155
156    // must match android/media/AudioSystem.java MODE_* values
157    enum audio_mode {
158        MODE_INVALID = -2,
159        MODE_CURRENT = -1,
160        MODE_NORMAL = 0,
161        MODE_RINGTONE,
162        MODE_IN_CALL,
163        MODE_IN_COMMUNICATION,
164        NUM_MODES  // not a valid entry, denotes end-of-list
165    };
166
167    enum audio_in_acoustics {
168        AGC_ENABLE    = 0x0001,
169        AGC_DISABLE   = 0,
170        NS_ENABLE     = 0x0002,
171        NS_DISABLE    = 0,
172        TX_IIR_ENABLE = 0x0004,
173        TX_DISABLE    = 0
174    };
175
176    // special audio session values
177    enum audio_sessions {
178        SESSION_OUTPUT_STAGE = -1, // session for effects attached to a particular output stream
179                                   // (value must be less than 0)
180        SESSION_OUTPUT_MIX = 0,    // session for effects applied to output mix. These effects can
181                                   // be moved by audio policy manager to another output stream
182                                   // (value must be 0)
183    };
184
185    /* These are static methods to control the system-wide AudioFlinger
186     * only privileged processes can have access to them
187     */
188
189    // mute/unmute microphone
190    static status_t muteMicrophone(bool state);
191    static status_t isMicrophoneMuted(bool *state);
192
193    // set/get master volume
194    static status_t setMasterVolume(float value);
195    static status_t getMasterVolume(float* volume);
196
197    // mute/unmute audio outputs
198    static status_t setMasterMute(bool mute);
199    static status_t getMasterMute(bool* mute);
200
201    // set/get stream volume on specified output
202    static status_t setStreamVolume(int stream, float value, int output);
203    static status_t getStreamVolume(int stream, float* volume, int output);
204
205    // mute/unmute stream
206    static status_t setStreamMute(int stream, bool mute);
207    static status_t getStreamMute(int stream, bool* mute);
208
209    // set audio mode in audio hardware (see AudioSystem::audio_mode)
210    static status_t setMode(int mode);
211
212    // returns true in *state if tracks are active on the specified stream or has been active
213    // in the past inPastMs milliseconds
214    static status_t isStreamActive(int stream, bool *state, uint32_t inPastMs = 0);
215
216    // set/get audio hardware parameters. The function accepts a list of parameters
217    // key value pairs in the form: key1=value1;key2=value2;...
218    // Some keys are reserved for standard parameters (See AudioParameter class).
219    static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
220    static String8  getParameters(audio_io_handle_t ioHandle, const String8& keys);
221
222    static void setErrorCallback(audio_error_callback cb);
223
224    // helper function to obtain AudioFlinger service handle
225    static const sp<IAudioFlinger>& get_audio_flinger();
226
227    static float linearToLog(int volume);
228    static int logToLinear(float volume);
229
230    static status_t getOutputSamplingRate(int* samplingRate, int stream = DEFAULT);
231    static status_t getOutputFrameCount(int* frameCount, int stream = DEFAULT);
232    static status_t getOutputLatency(uint32_t* latency, int stream = DEFAULT);
233
234    static bool routedToA2dpOutput(int streamType);
235
236    static status_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount,
237        size_t* buffSize);
238
239    static status_t setVoiceVolume(float volume);
240
241    // return the number of audio frames written by AudioFlinger to audio HAL and
242    // audio dsp to DAC since the output on which the specified stream is playing
243    // has exited standby.
244    // returned status (from utils/Errors.h) can be:
245    // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data
246    // - INVALID_OPERATION: Not supported on current hardware platform
247    // - BAD_VALUE: invalid parameter
248    // NOTE: this feature is not supported on all hardware platforms and it is
249    // necessary to check returned status before using the returned values.
250    static status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int stream = DEFAULT);
251
252    static unsigned int  getInputFramesLost(audio_io_handle_t ioHandle);
253
254    static int newAudioSessionId();
255    //
256    // AudioPolicyService interface
257    //
258
259    enum audio_devices {
260        // output devices
261        DEVICE_OUT_EARPIECE = 0x1,
262        DEVICE_OUT_SPEAKER = 0x2,
263        DEVICE_OUT_WIRED_HEADSET = 0x4,
264        DEVICE_OUT_WIRED_HEADPHONE = 0x8,
265        DEVICE_OUT_BLUETOOTH_SCO = 0x10,
266        DEVICE_OUT_BLUETOOTH_SCO_HEADSET = 0x20,
267        DEVICE_OUT_BLUETOOTH_SCO_CARKIT = 0x40,
268        DEVICE_OUT_BLUETOOTH_A2DP = 0x80,
269        DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100,
270        DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = 0x200,
271        DEVICE_OUT_AUX_DIGITAL = 0x400,
272        DEVICE_OUT_ANLG_DOCK_HEADSET = 0x800,
273        DEVICE_OUT_DGTL_DOCK_HEADSET = 0x1000,
274        DEVICE_OUT_DEFAULT = 0x8000,
275        DEVICE_OUT_ALL = (DEVICE_OUT_EARPIECE | DEVICE_OUT_SPEAKER | DEVICE_OUT_WIRED_HEADSET |
276                DEVICE_OUT_WIRED_HEADPHONE | DEVICE_OUT_BLUETOOTH_SCO | DEVICE_OUT_BLUETOOTH_SCO_HEADSET |
277                DEVICE_OUT_BLUETOOTH_SCO_CARKIT | DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
278                DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER | DEVICE_OUT_AUX_DIGITAL |
279                DEVICE_OUT_ANLG_DOCK_HEADSET | DEVICE_OUT_DGTL_DOCK_HEADSET |
280                DEVICE_OUT_DEFAULT),
281        DEVICE_OUT_ALL_A2DP = (DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
282                DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
283
284        // input devices
285        DEVICE_IN_COMMUNICATION = 0x10000,
286        DEVICE_IN_AMBIENT = 0x20000,
287        DEVICE_IN_BUILTIN_MIC = 0x40000,
288        DEVICE_IN_BLUETOOTH_SCO_HEADSET = 0x80000,
289        DEVICE_IN_WIRED_HEADSET = 0x100000,
290        DEVICE_IN_AUX_DIGITAL = 0x200000,
291        DEVICE_IN_VOICE_CALL = 0x400000,
292        DEVICE_IN_BACK_MIC = 0x800000,
293        DEVICE_IN_DEFAULT = 0x80000000,
294
295        DEVICE_IN_ALL = (DEVICE_IN_COMMUNICATION | DEVICE_IN_AMBIENT | DEVICE_IN_BUILTIN_MIC |
296                DEVICE_IN_BLUETOOTH_SCO_HEADSET | DEVICE_IN_WIRED_HEADSET | DEVICE_IN_AUX_DIGITAL |
297                DEVICE_IN_VOICE_CALL | DEVICE_IN_BACK_MIC | DEVICE_IN_DEFAULT)
298    };
299
300    // device connection states used for setDeviceConnectionState()
301    enum device_connection_state {
302        DEVICE_STATE_UNAVAILABLE,
303        DEVICE_STATE_AVAILABLE,
304        NUM_DEVICE_STATES
305    };
306
307    // request to open a direct output with getOutput() (by opposition to sharing an output with other AudioTracks)
308    enum output_flags {
309        OUTPUT_FLAG_INDIRECT = 0x0,
310        OUTPUT_FLAG_DIRECT = 0x1
311    };
312
313    // device categories used for setForceUse()
314    enum forced_config {
315        FORCE_NONE,
316        FORCE_SPEAKER,
317        FORCE_HEADPHONES,
318        FORCE_BT_SCO,
319        FORCE_BT_A2DP,
320        FORCE_WIRED_ACCESSORY,
321        FORCE_BT_CAR_DOCK,
322        FORCE_BT_DESK_DOCK,
323        FORCE_ANALOG_DOCK,
324        FORCE_DIGITAL_DOCK,
325        NUM_FORCE_CONFIG,
326        FORCE_DEFAULT = FORCE_NONE
327    };
328
329    // usages used for setForceUse(), must match AudioSystem.java
330    enum force_use {
331        FOR_COMMUNICATION,
332        FOR_MEDIA,
333        FOR_RECORD,
334        FOR_DOCK,
335        NUM_FORCE_USE
336    };
337
338    // types of io configuration change events received with ioConfigChanged()
339    enum io_config_event {
340        OUTPUT_OPENED,
341        OUTPUT_CLOSED,
342        OUTPUT_CONFIG_CHANGED,
343        INPUT_OPENED,
344        INPUT_CLOSED,
345        INPUT_CONFIG_CHANGED,
346        STREAM_CONFIG_CHANGED,
347        NUM_CONFIG_EVENTS
348    };
349
350    // audio output descritor used to cache output configurations in client process to avoid frequent calls
351    // through IAudioFlinger
352    class OutputDescriptor {
353    public:
354        OutputDescriptor()
355        : samplingRate(0), format(0), channels(0), frameCount(0), latency(0)  {}
356
357        uint32_t samplingRate;
358        int32_t format;
359        int32_t channels;
360        size_t frameCount;
361        uint32_t latency;
362    };
363
364    //
365    // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions)
366    //
367    static status_t setDeviceConnectionState(audio_devices device, device_connection_state state, const char *device_address);
368    static device_connection_state getDeviceConnectionState(audio_devices device, const char *device_address);
369    static status_t setPhoneState(int state);
370    static status_t setRingerMode(uint32_t mode, uint32_t mask);
371    static status_t setForceUse(force_use usage, forced_config config);
372    static forced_config getForceUse(force_use usage);
373    static audio_io_handle_t getOutput(stream_type stream,
374                                        uint32_t samplingRate = 0,
375                                        uint32_t format = FORMAT_DEFAULT,
376                                        uint32_t channels = CHANNEL_OUT_STEREO,
377                                        output_flags flags = OUTPUT_FLAG_INDIRECT);
378    static status_t startOutput(audio_io_handle_t output,
379                                AudioSystem::stream_type stream,
380                                int session = 0);
381    static status_t stopOutput(audio_io_handle_t output,
382                               AudioSystem::stream_type stream,
383                               int session = 0);
384    static void releaseOutput(audio_io_handle_t output);
385    static audio_io_handle_t getInput(int inputSource,
386                                    uint32_t samplingRate = 0,
387                                    uint32_t format = FORMAT_DEFAULT,
388                                    uint32_t channels = CHANNEL_IN_MONO,
389                                    audio_in_acoustics acoustics = (audio_in_acoustics)0);
390    static status_t startInput(audio_io_handle_t input);
391    static status_t stopInput(audio_io_handle_t input);
392    static void releaseInput(audio_io_handle_t input);
393    static status_t initStreamVolume(stream_type stream,
394                                      int indexMin,
395                                      int indexMax);
396    static status_t setStreamVolumeIndex(stream_type stream, int index);
397    static status_t getStreamVolumeIndex(stream_type stream, int *index);
398
399    static uint32_t getStrategyForStream(stream_type stream);
400    static uint32_t getDevicesForStream(stream_type stream);
401
402    static audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc);
403    static status_t registerEffect(effect_descriptor_t *desc,
404                                    audio_io_handle_t output,
405                                    uint32_t strategy,
406                                    int session,
407                                    int id);
408    static status_t unregisterEffect(int id);
409
410    static const sp<IAudioPolicyService>& get_audio_policy_service();
411
412    // ----------------------------------------------------------------------------
413
414    static uint32_t popCount(uint32_t u);
415    static bool isOutputDevice(audio_devices device);
416    static bool isInputDevice(audio_devices device);
417    static bool isA2dpDevice(audio_devices device);
418    static bool isBluetoothScoDevice(audio_devices device);
419    static bool isLowVisibility(stream_type stream);
420    static bool isOutputChannel(uint32_t channel);
421    static bool isInputChannel(uint32_t channel);
422    static bool isValidFormat(uint32_t format);
423    static bool isLinearPCM(uint32_t format);
424
425private:
426
427    class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient
428    {
429    public:
430        AudioFlingerClient() {
431        }
432
433        // DeathRecipient
434        virtual void binderDied(const wp<IBinder>& who);
435
436        // IAudioFlingerClient
437
438        // indicate a change in the configuration of an output or input: keeps the cached
439        // values for output/input parameters upto date in client process
440        virtual void ioConfigChanged(int event, int ioHandle, void *param2);
441    };
442
443    class AudioPolicyServiceClient: public IBinder::DeathRecipient
444    {
445    public:
446        AudioPolicyServiceClient() {
447        }
448
449        // DeathRecipient
450        virtual void binderDied(const wp<IBinder>& who);
451    };
452
453    static sp<AudioFlingerClient> gAudioFlingerClient;
454    static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient;
455    friend class AudioFlingerClient;
456    friend class AudioPolicyServiceClient;
457
458    static Mutex gLock;
459    static sp<IAudioFlinger> gAudioFlinger;
460    static audio_error_callback gAudioErrorCallback;
461
462    static size_t gInBuffSize;
463    // previous parameters for recording buffer size queries
464    static uint32_t gPrevInSamplingRate;
465    static int gPrevInFormat;
466    static int gPrevInChannelCount;
467
468    static sp<IAudioPolicyService> gAudioPolicyService;
469
470    // mapping between stream types and outputs
471    static DefaultKeyedVector<int, audio_io_handle_t> gStreamOutputMap;
472    // list of output descritor containing cached parameters (sampling rate, framecount, channel count...)
473    static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs;
474};
475
476class AudioParameter {
477
478public:
479    AudioParameter() {}
480    AudioParameter(const String8& keyValuePairs);
481    virtual ~AudioParameter();
482
483    // reserved parameter keys for changing standard parameters with setParameters() function.
484    // Using these keys is mandatory for AudioFlinger to properly monitor audio output/input
485    // configuration changes and act accordingly.
486    //  keyRouting: to change audio routing, value is an int in AudioSystem::audio_devices
487    //  keySamplingRate: to change sampling rate routing, value is an int
488    //  keyFormat: to change audio format, value is an int in AudioSystem::audio_format
489    //  keyChannels: to change audio channel configuration, value is an int in AudioSystem::audio_channels
490    //  keyFrameCount: to change audio output frame count, value is an int
491    //  keyInputSource: to change audio input source, value is an int in audio_source
492    //     (defined in media/mediarecorder.h)
493    static const char *keyRouting;
494    static const char *keySamplingRate;
495    static const char *keyFormat;
496    static const char *keyChannels;
497    static const char *keyFrameCount;
498    static const char *keyInputSource;
499
500    String8 toString();
501
502    status_t add(const String8& key, const String8& value);
503    status_t addInt(const String8& key, const int value);
504    status_t addFloat(const String8& key, const float value);
505
506    status_t remove(const String8& key);
507
508    status_t get(const String8& key, String8& value);
509    status_t getInt(const String8& key, int& value);
510    status_t getFloat(const String8& key, float& value);
511    status_t getAt(size_t index, String8& key, String8& value);
512
513    size_t size() { return mParameters.size(); }
514
515private:
516    String8 mKeyValuePairs;
517    KeyedVector <String8, String8> mParameters;
518};
519
520};  // namespace android
521
522#endif  /*ANDROID_AUDIOSYSTEM_H_*/
523