AudioSystem.h revision f1fb01a7f00b8da90a36268aba8584a872e99175 
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    enum stream_type {
37        DEFAULT          =-1,
38        VOICE_CALL       = 0,
39        SYSTEM           = 1,
40        RING             = 2,
41        MUSIC            = 3,
42        ALARM            = 4,
43        NOTIFICATION     = 5,
44        BLUETOOTH_SCO    = 6,
45        ENFORCED_AUDIBLE = 7, // Sounds that cannot be muted by user and must be routed to speaker
46        DTMF             = 8,
47        TTS              = 9,
48        NUM_STREAM_TYPES
49    };
50
51    // Audio sub formats (see AudioSystem::audio_format).
52    enum pcm_sub_format {
53        PCM_SUB_16_BIT          = 0x1, // must be 1 for backward compatibility
54        PCM_SUB_8_BIT           = 0x2, // must be 2 for backward compatibility
55    };
56
57    // MP3 sub format field definition : can use 11 LSBs in the same way as MP3 frame header to specify
58    // bit rate, stereo mode, version...
59    enum mp3_sub_format {
60        //TODO
61    };
62
63    // AMR NB/WB sub format field definition: specify frame block interleaving, bandwidth efficient or octet aligned,
64    // encoding mode for recording...
65    enum amr_sub_format {
66        //TODO
67    };
68
69    // AAC sub format field definition: specify profile or bitrate for recording...
70    enum aac_sub_format {
71        //TODO
72    };
73
74    // VORBIS sub format field definition: specify quality for recording...
75    enum vorbis_sub_format {
76        //TODO
77    };
78
79    // Audio format consists in a main format field (upper 8 bits) and a sub format field (lower 24 bits).
80    // The main format indicates the main codec type. The sub format field indicates options and parameters
81    // for each format. The sub format is mainly used for record to indicate for instance the requested bitrate
82    // or profile. It can also be used for certain formats to give informations not present in the encoded
83    // audio stream (e.g. octet alignement for AMR).
84    enum audio_format {
85        INVALID_FORMAT      = -1,
86        FORMAT_DEFAULT      = 0,
87        PCM                 = 0x00000000, // must be 0 for backward compatibility
88        MP3                 = 0x01000000,
89        AMR_NB              = 0x02000000,
90        AMR_WB              = 0x03000000,
91        AAC                 = 0x04000000,
92        HE_AAC_V1           = 0x05000000,
93        HE_AAC_V2           = 0x06000000,
94        VORBIS              = 0x07000000,
95        MAIN_FORMAT_MASK    = 0xFF000000,
96        SUB_FORMAT_MASK     = 0x00FFFFFF,
97        // Aliases
98        PCM_16_BIT          = (PCM|PCM_SUB_16_BIT),
99        PCM_8_BIT          = (PCM|PCM_SUB_8_BIT)
100    };
101
102
103    // Channel mask definitions must be kept in sync with JAVA values in /media/java/android/media/AudioFormat.java
104    enum audio_channels {
105        // output channels
106        CHANNEL_OUT_FRONT_LEFT = 0x4,
107        CHANNEL_OUT_FRONT_RIGHT = 0x8,
108        CHANNEL_OUT_FRONT_CENTER = 0x10,
109        CHANNEL_OUT_LOW_FREQUENCY = 0x20,
110        CHANNEL_OUT_BACK_LEFT = 0x40,
111        CHANNEL_OUT_BACK_RIGHT = 0x80,
112        CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x100,
113        CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x200,
114        CHANNEL_OUT_BACK_CENTER = 0x400,
115        CHANNEL_OUT_MONO = CHANNEL_OUT_FRONT_LEFT,
116        CHANNEL_OUT_STEREO = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT),
117        CHANNEL_OUT_QUAD = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
118                CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT),
119        CHANNEL_OUT_SURROUND = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
120                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_BACK_CENTER),
121        CHANNEL_OUT_5POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
122                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT),
123        CHANNEL_OUT_7POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
124                CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT |
125                CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER),
126        CHANNEL_OUT_ALL = (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 | CHANNEL_OUT_BACK_CENTER),
129
130        // input channels
131        CHANNEL_IN_LEFT = 0x4,
132        CHANNEL_IN_RIGHT = 0x8,
133        CHANNEL_IN_FRONT = 0x10,
134        CHANNEL_IN_BACK = 0x20,
135        CHANNEL_IN_LEFT_PROCESSED = 0x40,
136        CHANNEL_IN_RIGHT_PROCESSED = 0x80,
137        CHANNEL_IN_FRONT_PROCESSED = 0x100,
138        CHANNEL_IN_BACK_PROCESSED = 0x200,
139        CHANNEL_IN_PRESSURE = 0x400,
140        CHANNEL_IN_X_AXIS = 0x800,
141        CHANNEL_IN_Y_AXIS = 0x1000,
142        CHANNEL_IN_Z_AXIS = 0x2000,
143        CHANNEL_IN_VOICE_UPLINK = 0x4000,
144        CHANNEL_IN_VOICE_DNLINK = 0x8000,
145        CHANNEL_IN_MONO = CHANNEL_IN_FRONT,
146        CHANNEL_IN_STEREO = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT),
147        CHANNEL_IN_ALL = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT | CHANNEL_IN_FRONT | CHANNEL_IN_BACK|
148                CHANNEL_IN_LEFT_PROCESSED | CHANNEL_IN_RIGHT_PROCESSED | CHANNEL_IN_FRONT_PROCESSED | CHANNEL_IN_BACK_PROCESSED|
149                CHANNEL_IN_PRESSURE | CHANNEL_IN_X_AXIS | CHANNEL_IN_Y_AXIS | CHANNEL_IN_Z_AXIS |
150                CHANNEL_IN_VOICE_UPLINK | CHANNEL_IN_VOICE_DNLINK)
151    };
152
153    enum audio_mode {
154        MODE_INVALID = -2,
155        MODE_CURRENT = -1,
156        MODE_NORMAL = 0,
157        MODE_RINGTONE,
158        MODE_IN_CALL,
159        MODE_IN_COMMUNICATION,
160        NUM_MODES  // not a valid entry, denotes end-of-list
161    };
162
163    enum audio_in_acoustics {
164        AGC_ENABLE    = 0x0001,
165        AGC_DISABLE   = 0,
166        NS_ENABLE     = 0x0002,
167        NS_DISABLE    = 0,
168        TX_IIR_ENABLE = 0x0004,
169        TX_DISABLE    = 0
170    };
171
172    // special audio session values
173    enum audio_sessions {
174        SESSION_OUTPUT_STAGE = -1, // session for effects attached to a particular output stream
175                                   // (value must be less than 0)
176        SESSION_OUTPUT_MIX = 0,    // session for effects applied to output mix. These effects can
177                                   // be moved by audio policy manager to another output stream
178                                   // (value must be 0)
179    };
180
181    /* These are static methods to control the system-wide AudioFlinger
182     * only privileged processes can have access to them
183     */
184
185    // mute/unmute microphone
186    static status_t muteMicrophone(bool state);
187    static status_t isMicrophoneMuted(bool *state);
188
189    // set/get master volume
190    static status_t setMasterVolume(float value);
191    static status_t getMasterVolume(float* volume);
192    // mute/unmute audio outputs
193    static status_t setMasterMute(bool mute);
194    static status_t getMasterMute(bool* mute);
195
196    // set/get stream volume on specified output
197    static status_t setStreamVolume(int stream, float value, int output);
198    static status_t getStreamVolume(int stream, float* volume, int output);
199
200    // mute/unmute stream
201    static status_t setStreamMute(int stream, bool mute);
202    static status_t getStreamMute(int stream, bool* mute);
203
204    // set audio mode in audio hardware (see AudioSystem::audio_mode)
205    static status_t setMode(int mode);
206
207    // returns true in *state if tracks are active on the specified stream
208    static status_t isStreamActive(int stream, bool *state);
209
210    // set/get audio hardware parameters. The function accepts a list of parameters
211    // key value pairs in the form: key1=value1;key2=value2;...
212    // Some keys are reserved for standard parameters (See AudioParameter class).
213    static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
214    static String8  getParameters(audio_io_handle_t ioHandle, const String8& keys);
215
216    static void setErrorCallback(audio_error_callback cb);
217
218    // helper function to obtain AudioFlinger service handle
219    static const sp<IAudioFlinger>& get_audio_flinger();
220
221    static float linearToLog(int volume);
222    static int logToLinear(float volume);
223
224    static status_t getOutputSamplingRate(int* samplingRate, int stream = DEFAULT);
225    static status_t getOutputFrameCount(int* frameCount, int stream = DEFAULT);
226    static status_t getOutputLatency(uint32_t* latency, int stream = DEFAULT);
227
228    static bool routedToA2dpOutput(int streamType);
229
230    static status_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount,
231        size_t* buffSize);
232
233    static status_t setVoiceVolume(float volume);
234
235    // return the number of audio frames written by AudioFlinger to audio HAL and
236    // audio dsp to DAC since the output on which the specificed stream is playing
237    // has exited standby.
238    // returned status (from utils/Errors.h) can be:
239    // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data
240    // - INVALID_OPERATION: Not supported on current hardware platform
241    // - BAD_VALUE: invalid parameter
242    // NOTE: this feature is not supported on all hardware platforms and it is
243    // necessary to check returned status before using the returned values.
244    static status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int stream = DEFAULT);
245
246    static unsigned int  getInputFramesLost(audio_io_handle_t ioHandle);
247
248    static int newAudioSessionId();
249    //
250    // AudioPolicyService interface
251    //
252
253    enum audio_devices {
254        // output devices
255        DEVICE_OUT_EARPIECE = 0x1,
256        DEVICE_OUT_SPEAKER = 0x2,
257        DEVICE_OUT_WIRED_HEADSET = 0x4,
258        DEVICE_OUT_WIRED_HEADPHONE = 0x8,
259        DEVICE_OUT_BLUETOOTH_SCO = 0x10,
260        DEVICE_OUT_BLUETOOTH_SCO_HEADSET = 0x20,
261        DEVICE_OUT_BLUETOOTH_SCO_CARKIT = 0x40,
262        DEVICE_OUT_BLUETOOTH_A2DP = 0x80,
263        DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100,
264        DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = 0x200,
265        DEVICE_OUT_AUX_DIGITAL = 0x400,
266        DEVICE_OUT_ANLG_DOCK_HEADSET = 0x800,
267        DEVICE_OUT_DGTL_DOCK_HEADSET = 0x1000,
268        DEVICE_OUT_DEFAULT = 0x8000,
269        DEVICE_OUT_ALL = (DEVICE_OUT_EARPIECE | DEVICE_OUT_SPEAKER | DEVICE_OUT_WIRED_HEADSET |
270                DEVICE_OUT_WIRED_HEADPHONE | DEVICE_OUT_BLUETOOTH_SCO | DEVICE_OUT_BLUETOOTH_SCO_HEADSET |
271                DEVICE_OUT_BLUETOOTH_SCO_CARKIT | DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
272                DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER | DEVICE_OUT_AUX_DIGITAL |
273                DEVICE_OUT_ANLG_DOCK_HEADSET | DEVICE_OUT_DGTL_DOCK_HEADSET |
274                DEVICE_OUT_DEFAULT),
275        DEVICE_OUT_ALL_A2DP = (DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
276                DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
277
278        // input devices
279        DEVICE_IN_COMMUNICATION = 0x10000,
280        DEVICE_IN_AMBIENT = 0x20000,
281        DEVICE_IN_BUILTIN_MIC = 0x40000,
282        DEVICE_IN_BLUETOOTH_SCO_HEADSET = 0x80000,
283        DEVICE_IN_WIRED_HEADSET = 0x100000,
284        DEVICE_IN_AUX_DIGITAL = 0x200000,
285        DEVICE_IN_VOICE_CALL = 0x400000,
286        DEVICE_IN_BACK_MIC = 0x800000,
287        DEVICE_IN_DEFAULT = 0x80000000,
288
289        DEVICE_IN_ALL = (DEVICE_IN_COMMUNICATION | DEVICE_IN_AMBIENT | DEVICE_IN_BUILTIN_MIC |
290                DEVICE_IN_BLUETOOTH_SCO_HEADSET | DEVICE_IN_WIRED_HEADSET | DEVICE_IN_AUX_DIGITAL |
291                DEVICE_IN_VOICE_CALL | DEVICE_IN_BACK_MIC | DEVICE_IN_DEFAULT)
292    };
293
294    // device connection states used for setDeviceConnectionState()
295    enum device_connection_state {
296        DEVICE_STATE_UNAVAILABLE,
297        DEVICE_STATE_AVAILABLE,
298        NUM_DEVICE_STATES
299    };
300
301    // request to open a direct output with getOutput() (by opposition to sharing an output with other AudioTracks)
302    enum output_flags {
303        OUTPUT_FLAG_INDIRECT = 0x0,
304        OUTPUT_FLAG_DIRECT = 0x1
305    };
306
307    // device categories used for setForceUse()
308    enum forced_config {
309        FORCE_NONE,
310        FORCE_SPEAKER,
311        FORCE_HEADPHONES,
312        FORCE_BT_SCO,
313        FORCE_BT_A2DP,
314        FORCE_WIRED_ACCESSORY,
315        FORCE_BT_CAR_DOCK,
316        FORCE_BT_DESK_DOCK,
317        FORCE_ANALOG_DOCK,
318        FORCE_DIGITAL_DOCK,
319        NUM_FORCE_CONFIG,
320        FORCE_DEFAULT = FORCE_NONE
321    };
322
323    // usages used for setForceUse()
324    enum force_use {
325        FOR_COMMUNICATION,
326        FOR_MEDIA,
327        FOR_RECORD,
328        FOR_DOCK,
329        NUM_FORCE_USE
330    };
331
332    // types of io configuration change events received with ioConfigChanged()
333    enum io_config_event {
334        OUTPUT_OPENED,
335        OUTPUT_CLOSED,
336        OUTPUT_CONFIG_CHANGED,
337        INPUT_OPENED,
338        INPUT_CLOSED,
339        INPUT_CONFIG_CHANGED,
340        STREAM_CONFIG_CHANGED,
341        NUM_CONFIG_EVENTS
342    };
343
344    // audio output descritor used to cache output configurations in client process to avoid frequent calls
345    // through IAudioFlinger
346    class OutputDescriptor {
347    public:
348        OutputDescriptor()
349        : samplingRate(0), format(0), channels(0), frameCount(0), latency(0)  {}
350
351        uint32_t samplingRate;
352        int32_t format;
353        int32_t channels;
354        size_t frameCount;
355        uint32_t latency;
356    };
357
358    //
359    // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions)
360    //
361    static status_t setDeviceConnectionState(audio_devices device, device_connection_state state, const char *device_address);
362    static device_connection_state getDeviceConnectionState(audio_devices device, const char *device_address);
363    static status_t setPhoneState(int state);
364    static status_t setRingerMode(uint32_t mode, uint32_t mask);
365    static status_t setForceUse(force_use usage, forced_config config);
366    static forced_config getForceUse(force_use usage);
367    static audio_io_handle_t getOutput(stream_type stream,
368                                        uint32_t samplingRate = 0,
369                                        uint32_t format = FORMAT_DEFAULT,
370                                        uint32_t channels = CHANNEL_OUT_STEREO,
371                                        output_flags flags = OUTPUT_FLAG_INDIRECT);
372    static status_t startOutput(audio_io_handle_t output,
373                                AudioSystem::stream_type stream,
374                                int session = 0);
375    static status_t stopOutput(audio_io_handle_t output,
376                               AudioSystem::stream_type stream,
377                               int session = 0);
378    static void releaseOutput(audio_io_handle_t output);
379    static audio_io_handle_t getInput(int inputSource,
380                                    uint32_t samplingRate = 0,
381                                    uint32_t format = FORMAT_DEFAULT,
382                                    uint32_t channels = CHANNEL_IN_MONO,
383                                    audio_in_acoustics acoustics = (audio_in_acoustics)0);
384    static status_t startInput(audio_io_handle_t input);
385    static status_t stopInput(audio_io_handle_t input);
386    static void releaseInput(audio_io_handle_t input);
387    static status_t initStreamVolume(stream_type stream,
388                                      int indexMin,
389                                      int indexMax);
390    static status_t setStreamVolumeIndex(stream_type stream, int index);
391    static status_t getStreamVolumeIndex(stream_type stream, int *index);
392
393    static uint32_t getStrategyForStream(stream_type stream);
394
395    static audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc);
396    static status_t registerEffect(effect_descriptor_t *desc,
397                                    audio_io_handle_t output,
398                                    uint32_t strategy,
399                                    int session,
400                                    int id);
401    static status_t unregisterEffect(int id);
402
403    static const sp<IAudioPolicyService>& get_audio_policy_service();
404
405    // ----------------------------------------------------------------------------
406
407    static uint32_t popCount(uint32_t u);
408    static bool isOutputDevice(audio_devices device);
409    static bool isInputDevice(audio_devices device);
410    static bool isA2dpDevice(audio_devices device);
411    static bool isBluetoothScoDevice(audio_devices device);
412    static bool isLowVisibility(stream_type stream);
413    static bool isOutputChannel(uint32_t channel);
414    static bool isInputChannel(uint32_t channel);
415    static bool isValidFormat(uint32_t format);
416    static bool isLinearPCM(uint32_t format);
417
418private:
419
420    class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient
421    {
422    public:
423        AudioFlingerClient() {
424        }
425
426        // DeathRecipient
427        virtual void binderDied(const wp<IBinder>& who);
428
429        // IAudioFlingerClient
430
431        // indicate a change in the configuration of an output or input: keeps the cached
432        // values for output/input parameters upto date in client process
433        virtual void ioConfigChanged(int event, int ioHandle, void *param2);
434    };
435
436    class AudioPolicyServiceClient: public IBinder::DeathRecipient
437    {
438    public:
439        AudioPolicyServiceClient() {
440        }
441
442        // DeathRecipient
443        virtual void binderDied(const wp<IBinder>& who);
444    };
445
446    static sp<AudioFlingerClient> gAudioFlingerClient;
447    static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient;
448    friend class AudioFlingerClient;
449    friend class AudioPolicyServiceClient;
450
451    static Mutex gLock;
452    static sp<IAudioFlinger> gAudioFlinger;
453    static audio_error_callback gAudioErrorCallback;
454
455    static size_t gInBuffSize;
456    // previous parameters for recording buffer size queries
457    static uint32_t gPrevInSamplingRate;
458    static int gPrevInFormat;
459    static int gPrevInChannelCount;
460
461    static sp<IAudioPolicyService> gAudioPolicyService;
462
463    // mapping between stream types and outputs
464    static DefaultKeyedVector<int, audio_io_handle_t> gStreamOutputMap;
465    // list of output descritor containing cached parameters (sampling rate, framecount, channel count...)
466    static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs;
467};
468
469class AudioParameter {
470
471public:
472    AudioParameter() {}
473    AudioParameter(const String8& keyValuePairs);
474    virtual ~AudioParameter();
475
476    // reserved parameter keys for changing standard parameters with setParameters() function.
477    // Using these keys is mandatory for AudioFlinger to properly monitor audio output/input
478    // configuration changes and act accordingly.
479    //  keyRouting: to change audio routing, value is an int in AudioSystem::audio_devices
480    //  keySamplingRate: to change sampling rate routing, value is an int
481    //  keyFormat: to change audio format, value is an int in AudioSystem::audio_format
482    //  keyChannels: to change audio channel configuration, value is an int in AudioSystem::audio_channels
483    //  keyFrameCount: to change audio output frame count, value is an int
484    //  keyInputSource: to change audio input source, value is an int in audio_source
485    //     (defined in media/mediarecorder.h)
486    static const char *keyRouting;
487    static const char *keySamplingRate;
488    static const char *keyFormat;
489    static const char *keyChannels;
490    static const char *keyFrameCount;
491    static const char *keyInputSource;
492
493    String8 toString();
494
495    status_t add(const String8& key, const String8& value);
496    status_t addInt(const String8& key, const int value);
497    status_t addFloat(const String8& key, const float value);
498
499    status_t remove(const String8& key);
500
501    status_t get(const String8& key, String8& value);
502    status_t getInt(const String8& key, int& value);
503    status_t getFloat(const String8& key, float& value);
504    status_t getAt(size_t index, String8& key, String8& value);
505
506    size_t size() { return mParameters.size(); }
507
508private:
509    String8 mKeyValuePairs;
510    KeyedVector <String8, String8> mParameters;
511};
512
513};  // namespace android
514
515#endif  /*ANDROID_AUDIOSYSTEM_H_*/
516