AudioSystem.h revision 9ae8c597d9a0c8cedc4047fc4716d7361453f6ab 
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 <sys/types.h>
21
22#include <media/AudioPolicy.h>
23#include <media/AudioIoDescriptor.h>
24#include <media/IAudioFlingerClient.h>
25#include <media/IAudioPolicyServiceClient.h>
26#include <system/audio.h>
27#include <system/audio_effect.h>
28#include <system/audio_policy.h>
29#include <utils/Errors.h>
30#include <utils/Mutex.h>
31
32namespace android {
33
34typedef void (*audio_error_callback)(status_t err);
35typedef void (*dynamic_policy_callback)(int event, String8 regId, int val);
36typedef void (*record_config_callback)(int event, const record_client_info_t *clientInfo,
37                const audio_config_base_t *clientConfig, const audio_config_base_t *deviceConfig,
38                audio_patch_handle_t patchHandle);
39
40class IAudioFlinger;
41class IAudioPolicyService;
42class String8;
43
44class AudioSystem
45{
46public:
47
48    // FIXME Declare in binder opcode order, similarly to IAudioFlinger.h and IAudioFlinger.cpp
49
50    /* These are static methods to control the system-wide AudioFlinger
51     * only privileged processes can have access to them
52     */
53
54    // mute/unmute microphone
55    static status_t muteMicrophone(bool state);
56    static status_t isMicrophoneMuted(bool *state);
57
58    // set/get master volume
59    static status_t setMasterVolume(float value);
60    static status_t getMasterVolume(float* volume);
61
62    // mute/unmute audio outputs
63    static status_t setMasterMute(bool mute);
64    static status_t getMasterMute(bool* mute);
65
66    // set/get stream volume on specified output
67    static status_t setStreamVolume(audio_stream_type_t stream, float value,
68                                    audio_io_handle_t output);
69    static status_t getStreamVolume(audio_stream_type_t stream, float* volume,
70                                    audio_io_handle_t output);
71
72    // mute/unmute stream
73    static status_t setStreamMute(audio_stream_type_t stream, bool mute);
74    static status_t getStreamMute(audio_stream_type_t stream, bool* mute);
75
76    // set audio mode in audio hardware
77    static status_t setMode(audio_mode_t mode);
78
79    // returns true in *state if tracks are active on the specified stream or have been active
80    // in the past inPastMs milliseconds
81    static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs);
82    // returns true in *state if tracks are active for what qualifies as remote playback
83    // on the specified stream or have been active in the past inPastMs milliseconds. Remote
84    // playback isn't mutually exclusive with local playback.
85    static status_t isStreamActiveRemotely(audio_stream_type_t stream, bool *state,
86            uint32_t inPastMs);
87    // returns true in *state if a recorder is currently recording with the specified source
88    static status_t isSourceActive(audio_source_t source, bool *state);
89
90    // set/get audio hardware parameters. The function accepts a list of parameters
91    // key value pairs in the form: key1=value1;key2=value2;...
92    // Some keys are reserved for standard parameters (See AudioParameter class).
93    // The versions with audio_io_handle_t are intended for internal media framework use only.
94    static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
95    static String8  getParameters(audio_io_handle_t ioHandle, const String8& keys);
96    // The versions without audio_io_handle_t are intended for JNI.
97    static status_t setParameters(const String8& keyValuePairs);
98    static String8  getParameters(const String8& keys);
99
100    static void setErrorCallback(audio_error_callback cb);
101    static void setDynPolicyCallback(dynamic_policy_callback cb);
102    static void setRecordConfigCallback(record_config_callback);
103
104    // helper function to obtain AudioFlinger service handle
105    static const sp<IAudioFlinger> get_audio_flinger();
106
107    static float linearToLog(int volume);
108    static int logToLinear(float volume);
109
110    // Returned samplingRate and frameCount output values are guaranteed
111    // to be non-zero if status == NO_ERROR
112    // FIXME This API assumes a route, and so should be deprecated.
113    static status_t getOutputSamplingRate(uint32_t* samplingRate,
114            audio_stream_type_t stream);
115    // FIXME This API assumes a route, and so should be deprecated.
116    static status_t getOutputFrameCount(size_t* frameCount,
117            audio_stream_type_t stream);
118    // FIXME This API assumes a route, and so should be deprecated.
119    static status_t getOutputLatency(uint32_t* latency,
120            audio_stream_type_t stream);
121    // returns the audio HAL sample rate
122    static status_t getSamplingRate(audio_io_handle_t ioHandle,
123                                          uint32_t* samplingRate);
124    // For output threads with a fast mixer, returns the number of frames per normal mixer buffer.
125    // For output threads without a fast mixer, or for input, this is same as getFrameCountHAL().
126    static status_t getFrameCount(audio_io_handle_t ioHandle,
127                                  size_t* frameCount);
128    // returns the audio output latency in ms. Corresponds to
129    // audio_stream_out->get_latency()
130    static status_t getLatency(audio_io_handle_t output,
131                               uint32_t* latency);
132
133    // return status NO_ERROR implies *buffSize > 0
134    // FIXME This API assumes a route, and so should deprecated.
135    static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
136        audio_channel_mask_t channelMask, size_t* buffSize);
137
138    static status_t setVoiceVolume(float volume);
139
140    // return the number of audio frames written by AudioFlinger to audio HAL and
141    // audio dsp to DAC since the specified output has exited standby.
142    // returned status (from utils/Errors.h) can be:
143    // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data
144    // - INVALID_OPERATION: Not supported on current hardware platform
145    // - BAD_VALUE: invalid parameter
146    // NOTE: this feature is not supported on all hardware platforms and it is
147    // necessary to check returned status before using the returned values.
148    static status_t getRenderPosition(audio_io_handle_t output,
149                                      uint32_t *halFrames,
150                                      uint32_t *dspFrames);
151
152    // return the number of input frames lost by HAL implementation, or 0 if the handle is invalid
153    static uint32_t getInputFramesLost(audio_io_handle_t ioHandle);
154
155    // Allocate a new unique ID for use as an audio session ID or I/O handle.
156    // If unable to contact AudioFlinger, returns AUDIO_UNIQUE_ID_ALLOCATE instead.
157    // FIXME If AudioFlinger were to ever exhaust the unique ID namespace,
158    //       this method could fail by returning either a reserved ID like AUDIO_UNIQUE_ID_ALLOCATE
159    //       or an unspecified existing unique ID.
160    static audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use);
161
162    static void acquireAudioSessionId(audio_session_t audioSession, pid_t pid);
163    static void releaseAudioSessionId(audio_session_t audioSession, pid_t pid);
164
165    // Get the HW synchronization source used for an audio session.
166    // Return a valid source or AUDIO_HW_SYNC_INVALID if an error occurs
167    // or no HW sync source is used.
168    static audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId);
169
170    // Indicate JAVA services are ready (scheduling, power management ...)
171    static status_t systemReady();
172
173    // Returns the number of frames per audio HAL buffer.
174    // Corresponds to audio_stream->get_buffer_size()/audio_stream_in_frame_size() for input.
175    // See also getFrameCount().
176    static status_t getFrameCountHAL(audio_io_handle_t ioHandle,
177                                     size_t* frameCount);
178
179    // Events used to synchronize actions between audio sessions.
180    // For instance SYNC_EVENT_PRESENTATION_COMPLETE can be used to delay recording start until
181    // playback is complete on another audio session.
182    // See definitions in MediaSyncEvent.java
183    enum sync_event_t {
184        SYNC_EVENT_SAME = -1,             // used internally to indicate restart with same event
185        SYNC_EVENT_NONE = 0,
186        SYNC_EVENT_PRESENTATION_COMPLETE,
187
188        //
189        // Define new events here: SYNC_EVENT_START, SYNC_EVENT_STOP, SYNC_EVENT_TIME ...
190        //
191        SYNC_EVENT_CNT,
192    };
193
194    // Timeout for synchronous record start. Prevents from blocking the record thread forever
195    // if the trigger event is not fired.
196    static const uint32_t kSyncRecordStartTimeOutMs = 30000;
197
198    //
199    // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions)
200    //
201    static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state,
202                                             const char *device_address, const char *device_name);
203    static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
204                                                                const char *device_address);
205    static status_t handleDeviceConfigChange(audio_devices_t device,
206                                             const char *device_address,
207                                             const char *device_name);
208    static status_t setPhoneState(audio_mode_t state);
209    static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config);
210    static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
211
212    // Client must successfully hand off the handle reference to AudioFlinger via createTrack(),
213    // or release it with releaseOutput().
214    static audio_io_handle_t getOutput(audio_stream_type_t stream,
215                                        uint32_t samplingRate = 0,
216                                        audio_format_t format = AUDIO_FORMAT_DEFAULT,
217                                        audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
218                                        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
219                                        const audio_offload_info_t *offloadInfo = NULL);
220    static status_t getOutputForAttr(const audio_attributes_t *attr,
221                                     audio_io_handle_t *output,
222                                     audio_session_t session,
223                                     audio_stream_type_t *stream,
224                                     uid_t uid,
225                                     const audio_config_t *config,
226                                     audio_output_flags_t flags,
227                                     audio_port_handle_t *selectedDeviceId,
228                                     audio_port_handle_t *portId);
229    static status_t startOutput(audio_io_handle_t output,
230                                audio_stream_type_t stream,
231                                audio_session_t session);
232    static status_t stopOutput(audio_io_handle_t output,
233                               audio_stream_type_t stream,
234                               audio_session_t session);
235    static void releaseOutput(audio_io_handle_t output,
236                              audio_stream_type_t stream,
237                              audio_session_t session);
238
239    // Client must successfully hand off the handle reference to AudioFlinger via openRecord(),
240    // or release it with releaseInput().
241    static status_t getInputForAttr(const audio_attributes_t *attr,
242                                    audio_io_handle_t *input,
243                                    audio_session_t session,
244                                    pid_t pid,
245                                    uid_t uid,
246                                    const audio_config_base_t *config,
247                                    audio_input_flags_t flags,
248                                    audio_port_handle_t *selectedDeviceId,
249                                    audio_port_handle_t *portId);
250
251    static status_t startInput(audio_io_handle_t input,
252                               audio_session_t session);
253    static status_t stopInput(audio_io_handle_t input,
254                              audio_session_t session);
255    static void releaseInput(audio_io_handle_t input,
256                             audio_session_t session);
257    static status_t initStreamVolume(audio_stream_type_t stream,
258                                      int indexMin,
259                                      int indexMax);
260    static status_t setStreamVolumeIndex(audio_stream_type_t stream,
261                                         int index,
262                                         audio_devices_t device);
263    static status_t getStreamVolumeIndex(audio_stream_type_t stream,
264                                         int *index,
265                                         audio_devices_t device);
266
267    static uint32_t getStrategyForStream(audio_stream_type_t stream);
268    static audio_devices_t getDevicesForStream(audio_stream_type_t stream);
269
270    static audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc);
271    static status_t registerEffect(const effect_descriptor_t *desc,
272                                    audio_io_handle_t io,
273                                    uint32_t strategy,
274                                    audio_session_t session,
275                                    int id);
276    static status_t unregisterEffect(int id);
277    static status_t setEffectEnabled(int id, bool enabled);
278
279    // clear stream to output mapping cache (gStreamOutputMap)
280    // and output configuration cache (gOutputs)
281    static void clearAudioConfigCache();
282
283    static const sp<IAudioPolicyService> get_audio_policy_service();
284
285    // helpers for android.media.AudioManager.getProperty(), see description there for meaning
286    static uint32_t getPrimaryOutputSamplingRate();
287    static size_t getPrimaryOutputFrameCount();
288
289    static status_t setLowRamDevice(bool isLowRamDevice);
290
291    // Check if hw offload is possible for given format, stream type, sample rate,
292    // bit rate, duration, video and streaming or offload property is enabled
293    static bool isOffloadSupported(const audio_offload_info_t& info);
294
295    // check presence of audio flinger service.
296    // returns NO_ERROR if binding to service succeeds, DEAD_OBJECT otherwise
297    static status_t checkAudioFlinger();
298
299    /* List available audio ports and their attributes */
300    static status_t listAudioPorts(audio_port_role_t role,
301                                   audio_port_type_t type,
302                                   unsigned int *num_ports,
303                                   struct audio_port *ports,
304                                   unsigned int *generation);
305
306    /* Get attributes for a given audio port */
307    static status_t getAudioPort(struct audio_port *port);
308
309    /* Create an audio patch between several source and sink ports */
310    static status_t createAudioPatch(const struct audio_patch *patch,
311                                       audio_patch_handle_t *handle);
312
313    /* Release an audio patch */
314    static status_t releaseAudioPatch(audio_patch_handle_t handle);
315
316    /* List existing audio patches */
317    static status_t listAudioPatches(unsigned int *num_patches,
318                                      struct audio_patch *patches,
319                                      unsigned int *generation);
320    /* Set audio port configuration */
321    static status_t setAudioPortConfig(const struct audio_port_config *config);
322
323
324    static status_t acquireSoundTriggerSession(audio_session_t *session,
325                                           audio_io_handle_t *ioHandle,
326                                           audio_devices_t *device);
327    static status_t releaseSoundTriggerSession(audio_session_t session);
328
329    static audio_mode_t getPhoneState();
330
331    static status_t registerPolicyMixes(const Vector<AudioMix>& mixes, bool registration);
332
333    static status_t startAudioSource(const struct audio_port_config *source,
334                                      const audio_attributes_t *attributes,
335                                      audio_patch_handle_t *handle);
336    static status_t stopAudioSource(audio_patch_handle_t handle);
337
338    static status_t setMasterMono(bool mono);
339    static status_t getMasterMono(bool *mono);
340
341    static float    getStreamVolumeDB(
342            audio_stream_type_t stream, int index, audio_devices_t device);
343
344    // ----------------------------------------------------------------------------
345
346    class AudioPortCallback : public RefBase
347    {
348    public:
349
350                AudioPortCallback() {}
351        virtual ~AudioPortCallback() {}
352
353        virtual void onAudioPortListUpdate() = 0;
354        virtual void onAudioPatchListUpdate() = 0;
355        virtual void onServiceDied() = 0;
356
357    };
358
359    static status_t addAudioPortCallback(const sp<AudioPortCallback>& callback);
360    static status_t removeAudioPortCallback(const sp<AudioPortCallback>& callback);
361
362    class AudioDeviceCallback : public RefBase
363    {
364    public:
365
366                AudioDeviceCallback() {}
367        virtual ~AudioDeviceCallback() {}
368
369        virtual void onAudioDeviceUpdate(audio_io_handle_t audioIo,
370                                         audio_port_handle_t deviceId) = 0;
371    };
372
373    static status_t addAudioDeviceCallback(const sp<AudioDeviceCallback>& callback,
374                                           audio_io_handle_t audioIo);
375    static status_t removeAudioDeviceCallback(const sp<AudioDeviceCallback>& callback,
376                                              audio_io_handle_t audioIo);
377
378    static audio_port_handle_t getDeviceIdForIo(audio_io_handle_t audioIo);
379
380private:
381
382    class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient
383    {
384    public:
385        AudioFlingerClient() :
386            mInBuffSize(0), mInSamplingRate(0),
387            mInFormat(AUDIO_FORMAT_DEFAULT), mInChannelMask(AUDIO_CHANNEL_NONE) {
388        }
389
390        void clearIoCache();
391        status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
392                                    audio_channel_mask_t channelMask, size_t* buffSize);
393        sp<AudioIoDescriptor> getIoDescriptor(audio_io_handle_t ioHandle);
394
395        // DeathRecipient
396        virtual void binderDied(const wp<IBinder>& who);
397
398        // IAudioFlingerClient
399
400        // indicate a change in the configuration of an output or input: keeps the cached
401        // values for output/input parameters up-to-date in client process
402        virtual void ioConfigChanged(audio_io_config_event event,
403                                     const sp<AudioIoDescriptor>& ioDesc);
404
405
406        status_t addAudioDeviceCallback(const sp<AudioDeviceCallback>& callback,
407                                               audio_io_handle_t audioIo);
408        status_t removeAudioDeviceCallback(const sp<AudioDeviceCallback>& callback,
409                                           audio_io_handle_t audioIo);
410
411        audio_port_handle_t getDeviceIdForIo(audio_io_handle_t audioIo);
412
413    private:
414        Mutex                               mLock;
415        DefaultKeyedVector<audio_io_handle_t, sp<AudioIoDescriptor> >   mIoDescriptors;
416        DefaultKeyedVector<audio_io_handle_t, Vector < sp<AudioDeviceCallback> > >
417                                                                        mAudioDeviceCallbacks;
418        // cached values for recording getInputBufferSize() queries
419        size_t                              mInBuffSize;    // zero indicates cache is invalid
420        uint32_t                            mInSamplingRate;
421        audio_format_t                      mInFormat;
422        audio_channel_mask_t                mInChannelMask;
423        sp<AudioIoDescriptor> getIoDescriptor_l(audio_io_handle_t ioHandle);
424    };
425
426    class AudioPolicyServiceClient: public IBinder::DeathRecipient,
427                                    public BnAudioPolicyServiceClient
428    {
429    public:
430        AudioPolicyServiceClient() {
431        }
432
433        int addAudioPortCallback(const sp<AudioPortCallback>& callback);
434        int removeAudioPortCallback(const sp<AudioPortCallback>& callback);
435
436        // DeathRecipient
437        virtual void binderDied(const wp<IBinder>& who);
438
439        // IAudioPolicyServiceClient
440        virtual void onAudioPortListUpdate();
441        virtual void onAudioPatchListUpdate();
442        virtual void onDynamicPolicyMixStateUpdate(String8 regId, int32_t state);
443        virtual void onRecordingConfigurationUpdate(int event,
444                        const record_client_info_t *clientInfo,
445                        const audio_config_base_t *clientConfig,
446                        const audio_config_base_t *deviceConfig, audio_patch_handle_t patchHandle);
447
448    private:
449        Mutex                               mLock;
450        Vector <sp <AudioPortCallback> >    mAudioPortCallbacks;
451    };
452
453    static const sp<AudioFlingerClient> getAudioFlingerClient();
454    static sp<AudioIoDescriptor> getIoDescriptor(audio_io_handle_t ioHandle);
455
456    static sp<AudioFlingerClient> gAudioFlingerClient;
457    static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient;
458    friend class AudioFlingerClient;
459    friend class AudioPolicyServiceClient;
460
461    static Mutex gLock;      // protects gAudioFlinger and gAudioErrorCallback,
462    static Mutex gLockAPS;   // protects gAudioPolicyService and gAudioPolicyServiceClient
463    static sp<IAudioFlinger> gAudioFlinger;
464    static audio_error_callback gAudioErrorCallback;
465    static dynamic_policy_callback gDynPolicyCallback;
466    static record_config_callback gRecordConfigCallback;
467
468    static size_t gInBuffSize;
469    // previous parameters for recording buffer size queries
470    static uint32_t gPrevInSamplingRate;
471    static audio_format_t gPrevInFormat;
472    static audio_channel_mask_t gPrevInChannelMask;
473
474    static sp<IAudioPolicyService> gAudioPolicyService;
475};
476
477};  // namespace android
478
479#endif  /*ANDROID_AUDIOSYSTEM_H_*/
480