AudioSystem.h revision ffe9c25ce85e1af55d58ec025adc6367d70db7e8 
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        NUM_MODES  // not a valid entry, denotes end-of-list
160    };
161
162    enum audio_in_acoustics {
163        AGC_ENABLE    = 0x0001,
164        AGC_DISABLE   = 0,
165        NS_ENABLE     = 0x0002,
166        NS_DISABLE    = 0,
167        TX_IIR_ENABLE = 0x0004,
168        TX_DISABLE    = 0
169    };
170
171    /* These are static methods to control the system-wide AudioFlinger
172     * only privileged processes can have access to them
173     */
174
175    // mute/unmute microphone
176    static status_t muteMicrophone(bool state);
177    static status_t isMicrophoneMuted(bool *state);
178
179    // set/get master volume
180    static status_t setMasterVolume(float value);
181    static status_t getMasterVolume(float* volume);
182    // mute/unmute audio outputs
183    static status_t setMasterMute(bool mute);
184    static status_t getMasterMute(bool* mute);
185
186    // set/get stream volume on specified output
187    static status_t setStreamVolume(int stream, float value, int output);
188    static status_t getStreamVolume(int stream, float* volume, int output);
189
190    // mute/unmute stream
191    static status_t setStreamMute(int stream, bool mute);
192    static status_t getStreamMute(int stream, bool* mute);
193
194    // set audio mode in audio hardware (see AudioSystem::audio_mode)
195    static status_t setMode(int mode);
196
197    // returns true in *state if tracks are active on the specified stream
198    static status_t isStreamActive(int stream, bool *state);
199
200    // set/get audio hardware parameters. The function accepts a list of parameters
201    // key value pairs in the form: key1=value1;key2=value2;...
202    // Some keys are reserved for standard parameters (See AudioParameter class).
203    static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
204    static String8  getParameters(audio_io_handle_t ioHandle, const String8& keys);
205
206    static void setErrorCallback(audio_error_callback cb);
207
208    // helper function to obtain AudioFlinger service handle
209    static const sp<IAudioFlinger>& get_audio_flinger();
210
211    static float linearToLog(int volume);
212    static int logToLinear(float volume);
213
214    static status_t getOutputSamplingRate(int* samplingRate, int stream = DEFAULT);
215    static status_t getOutputFrameCount(int* frameCount, int stream = DEFAULT);
216    static status_t getOutputLatency(uint32_t* latency, int stream = DEFAULT);
217
218    static bool routedToA2dpOutput(int streamType);
219
220    static status_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount,
221        size_t* buffSize);
222
223    static status_t setVoiceVolume(float volume);
224
225    // return the number of audio frames written by AudioFlinger to audio HAL and
226    // audio dsp to DAC since the output on which the specificed stream is playing
227    // has exited standby.
228    // returned status (from utils/Errors.h) can be:
229    // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data
230    // - INVALID_OPERATION: Not supported on current hardware platform
231    // - BAD_VALUE: invalid parameter
232    // NOTE: this feature is not supported on all hardware platforms and it is
233    // necessary to check returned status before using the returned values.
234    static status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int stream = DEFAULT);
235
236    static unsigned int  getInputFramesLost(audio_io_handle_t ioHandle);
237
238    static int newAudioSessionId();
239    //
240    // AudioPolicyService interface
241    //
242
243    enum audio_devices {
244        // output devices
245        DEVICE_OUT_EARPIECE = 0x1,
246        DEVICE_OUT_SPEAKER = 0x2,
247        DEVICE_OUT_WIRED_HEADSET = 0x4,
248        DEVICE_OUT_WIRED_HEADPHONE = 0x8,
249        DEVICE_OUT_BLUETOOTH_SCO = 0x10,
250        DEVICE_OUT_BLUETOOTH_SCO_HEADSET = 0x20,
251        DEVICE_OUT_BLUETOOTH_SCO_CARKIT = 0x40,
252        DEVICE_OUT_BLUETOOTH_A2DP = 0x80,
253        DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100,
254        DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = 0x200,
255        DEVICE_OUT_AUX_DIGITAL = 0x400,
256        DEVICE_OUT_DEFAULT = 0x8000,
257        DEVICE_OUT_ALL = (DEVICE_OUT_EARPIECE | DEVICE_OUT_SPEAKER | DEVICE_OUT_WIRED_HEADSET |
258                DEVICE_OUT_WIRED_HEADPHONE | DEVICE_OUT_BLUETOOTH_SCO | DEVICE_OUT_BLUETOOTH_SCO_HEADSET |
259                DEVICE_OUT_BLUETOOTH_SCO_CARKIT | DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
260                DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER | DEVICE_OUT_AUX_DIGITAL | DEVICE_OUT_DEFAULT),
261        DEVICE_OUT_ALL_A2DP = (DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
262                DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
263
264        // input devices
265        DEVICE_IN_COMMUNICATION = 0x10000,
266        DEVICE_IN_AMBIENT = 0x20000,
267        DEVICE_IN_BUILTIN_MIC = 0x40000,
268        DEVICE_IN_BLUETOOTH_SCO_HEADSET = 0x80000,
269        DEVICE_IN_WIRED_HEADSET = 0x100000,
270        DEVICE_IN_AUX_DIGITAL = 0x200000,
271        DEVICE_IN_VOICE_CALL = 0x400000,
272        DEVICE_IN_BACK_MIC = 0x800000,
273        DEVICE_IN_DEFAULT = 0x80000000,
274
275        DEVICE_IN_ALL = (DEVICE_IN_COMMUNICATION | DEVICE_IN_AMBIENT | DEVICE_IN_BUILTIN_MIC |
276                DEVICE_IN_BLUETOOTH_SCO_HEADSET | DEVICE_IN_WIRED_HEADSET | DEVICE_IN_AUX_DIGITAL |
277                DEVICE_IN_VOICE_CALL | DEVICE_IN_BACK_MIC | DEVICE_IN_DEFAULT)
278    };
279
280    // device connection states used for setDeviceConnectionState()
281    enum device_connection_state {
282        DEVICE_STATE_UNAVAILABLE,
283        DEVICE_STATE_AVAILABLE,
284        NUM_DEVICE_STATES
285    };
286
287    // request to open a direct output with getOutput() (by opposition to sharing an output with other AudioTracks)
288    enum output_flags {
289        OUTPUT_FLAG_INDIRECT = 0x0,
290        OUTPUT_FLAG_DIRECT = 0x1
291    };
292
293    // device categories used for setForceUse()
294    enum forced_config {
295        FORCE_NONE,
296        FORCE_SPEAKER,
297        FORCE_HEADPHONES,
298        FORCE_BT_SCO,
299        FORCE_BT_A2DP,
300        FORCE_WIRED_ACCESSORY,
301        FORCE_BT_CAR_DOCK,
302        FORCE_BT_DESK_DOCK,
303        NUM_FORCE_CONFIG,
304        FORCE_DEFAULT = FORCE_NONE
305    };
306
307    // usages used for setForceUse()
308    enum force_use {
309        FOR_COMMUNICATION,
310        FOR_MEDIA,
311        FOR_RECORD,
312        FOR_DOCK,
313        NUM_FORCE_USE
314    };
315
316    // types of io configuration change events received with ioConfigChanged()
317    enum io_config_event {
318        OUTPUT_OPENED,
319        OUTPUT_CLOSED,
320        OUTPUT_CONFIG_CHANGED,
321        INPUT_OPENED,
322        INPUT_CLOSED,
323        INPUT_CONFIG_CHANGED,
324        STREAM_CONFIG_CHANGED,
325        NUM_CONFIG_EVENTS
326    };
327
328    // audio output descritor used to cache output configurations in client process to avoid frequent calls
329    // through IAudioFlinger
330    class OutputDescriptor {
331    public:
332        OutputDescriptor()
333        : samplingRate(0), format(0), channels(0), frameCount(0), latency(0)  {}
334
335        uint32_t samplingRate;
336        int32_t format;
337        int32_t channels;
338        size_t frameCount;
339        uint32_t latency;
340    };
341
342    //
343    // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions)
344    //
345    static status_t setDeviceConnectionState(audio_devices device, device_connection_state state, const char *device_address);
346    static device_connection_state getDeviceConnectionState(audio_devices device, const char *device_address);
347    static status_t setPhoneState(int state);
348    static status_t setRingerMode(uint32_t mode, uint32_t mask);
349    static status_t setForceUse(force_use usage, forced_config config);
350    static forced_config getForceUse(force_use usage);
351    static audio_io_handle_t getOutput(stream_type stream,
352                                        uint32_t samplingRate = 0,
353                                        uint32_t format = FORMAT_DEFAULT,
354                                        uint32_t channels = CHANNEL_OUT_STEREO,
355                                        output_flags flags = OUTPUT_FLAG_INDIRECT);
356    static status_t startOutput(audio_io_handle_t output, AudioSystem::stream_type stream);
357    static status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream);
358    static void releaseOutput(audio_io_handle_t output);
359    static audio_io_handle_t getInput(int inputSource,
360                                    uint32_t samplingRate = 0,
361                                    uint32_t format = FORMAT_DEFAULT,
362                                    uint32_t channels = CHANNEL_IN_MONO,
363                                    audio_in_acoustics acoustics = (audio_in_acoustics)0);
364    static status_t startInput(audio_io_handle_t input);
365    static status_t stopInput(audio_io_handle_t input);
366    static void releaseInput(audio_io_handle_t input);
367    static status_t initStreamVolume(stream_type stream,
368                                      int indexMin,
369                                      int indexMax);
370    static status_t setStreamVolumeIndex(stream_type stream, int index);
371    static status_t getStreamVolumeIndex(stream_type stream, int *index);
372
373    static const sp<IAudioPolicyService>& get_audio_policy_service();
374
375    // ----------------------------------------------------------------------------
376
377    static uint32_t popCount(uint32_t u);
378    static bool isOutputDevice(audio_devices device);
379    static bool isInputDevice(audio_devices device);
380    static bool isA2dpDevice(audio_devices device);
381    static bool isBluetoothScoDevice(audio_devices device);
382    static bool isLowVisibility(stream_type stream);
383    static bool isOutputChannel(uint32_t channel);
384    static bool isInputChannel(uint32_t channel);
385    static bool isValidFormat(uint32_t format);
386    static bool isLinearPCM(uint32_t format);
387
388private:
389
390    class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient
391    {
392    public:
393        AudioFlingerClient() {
394        }
395
396        // DeathRecipient
397        virtual void binderDied(const wp<IBinder>& who);
398
399        // IAudioFlingerClient
400
401        // indicate a change in the configuration of an output or input: keeps the cached
402        // values for output/input parameters upto date in client process
403        virtual void ioConfigChanged(int event, int ioHandle, void *param2);
404    };
405
406    class AudioPolicyServiceClient: public IBinder::DeathRecipient
407    {
408    public:
409        AudioPolicyServiceClient() {
410        }
411
412        // DeathRecipient
413        virtual void binderDied(const wp<IBinder>& who);
414    };
415
416    static sp<AudioFlingerClient> gAudioFlingerClient;
417    static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient;
418    friend class AudioFlingerClient;
419    friend class AudioPolicyServiceClient;
420
421    static Mutex gLock;
422    static sp<IAudioFlinger> gAudioFlinger;
423    static audio_error_callback gAudioErrorCallback;
424
425    static size_t gInBuffSize;
426    // previous parameters for recording buffer size queries
427    static uint32_t gPrevInSamplingRate;
428    static int gPrevInFormat;
429    static int gPrevInChannelCount;
430
431    static sp<IAudioPolicyService> gAudioPolicyService;
432
433    // mapping between stream types and outputs
434    static DefaultKeyedVector<int, audio_io_handle_t> gStreamOutputMap;
435    // list of output descritor containing cached parameters (sampling rate, framecount, channel count...)
436    static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs;
437};
438
439class AudioParameter {
440
441public:
442    AudioParameter() {}
443    AudioParameter(const String8& keyValuePairs);
444    virtual ~AudioParameter();
445
446    // reserved parameter keys for changeing standard parameters with setParameters() function.
447    // Using these keys is mandatory for AudioFlinger to properly monitor audio output/input
448    // configuration changes and act accordingly.
449    //  keyRouting: to change audio routing, value is an int in AudioSystem::audio_devices
450    //  keySamplingRate: to change sampling rate routing, value is an int
451    //  keyFormat: to change audio format, value is an int in AudioSystem::audio_format
452    //  keyChannels: to change audio channel configuration, value is an int in AudioSystem::audio_channels
453    //  keyFrameCount: to change audio output frame count, value is an int
454    static const char *keyRouting;
455    static const char *keySamplingRate;
456    static const char *keyFormat;
457    static const char *keyChannels;
458    static const char *keyFrameCount;
459
460    String8 toString();
461
462    status_t add(const String8& key, const String8& value);
463    status_t addInt(const String8& key, const int value);
464    status_t addFloat(const String8& key, const float value);
465
466    status_t remove(const String8& key);
467
468    status_t get(const String8& key, String8& value);
469    status_t getInt(const String8& key, int& value);
470    status_t getFloat(const String8& key, float& value);
471    status_t getAt(size_t index, String8& key, String8& value);
472
473    size_t size() { return mParameters.size(); }
474
475private:
476    String8 mKeyValuePairs;
477    KeyedVector <String8, String8> mParameters;
478};
479
480};  // namespace android
481
482#endif  /*ANDROID_AUDIOSYSTEM_H_*/
483