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 <hardware/audio_effect.h>
21#include <media/AudioPolicy.h>
22#include <media/IAudioFlingerClient.h>
23#include <media/IAudioPolicyServiceClient.h>
24#include <system/audio.h>
25#include <system/audio_policy.h>
26#include <utils/Errors.h>
27#include <utils/Mutex.h>
28
29namespace android {
30
31typedef void (*audio_error_callback)(status_t err);
32
33class IAudioFlinger;
34class IAudioPolicyService;
35class String8;
36
37class AudioSystem
38{
39public:
40
41    /* These are static methods to control the system-wide AudioFlinger
42     * only privileged processes can have access to them
43     */
44
45    // mute/unmute microphone
46    static status_t muteMicrophone(bool state);
47    static status_t isMicrophoneMuted(bool *state);
48
49    // set/get master volume
50    static status_t setMasterVolume(float value);
51    static status_t getMasterVolume(float* volume);
52
53    // mute/unmute audio outputs
54    static status_t setMasterMute(bool mute);
55    static status_t getMasterMute(bool* mute);
56
57    // set/get stream volume on specified output
58    static status_t setStreamVolume(audio_stream_type_t stream, float value,
59                                    audio_io_handle_t output);
60    static status_t getStreamVolume(audio_stream_type_t stream, float* volume,
61                                    audio_io_handle_t output);
62
63    // mute/unmute stream
64    static status_t setStreamMute(audio_stream_type_t stream, bool mute);
65    static status_t getStreamMute(audio_stream_type_t stream, bool* mute);
66
67    // set audio mode in audio hardware
68    static status_t setMode(audio_mode_t mode);
69
70    // returns true in *state if tracks are active on the specified stream or have been active
71    // in the past inPastMs milliseconds
72    static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs);
73    // returns true in *state if tracks are active for what qualifies as remote playback
74    // on the specified stream or have been active in the past inPastMs milliseconds. Remote
75    // playback isn't mutually exclusive with local playback.
76    static status_t isStreamActiveRemotely(audio_stream_type_t stream, bool *state,
77            uint32_t inPastMs);
78    // returns true in *state if a recorder is currently recording with the specified source
79    static status_t isSourceActive(audio_source_t source, bool *state);
80
81    // set/get audio hardware parameters. The function accepts a list of parameters
82    // key value pairs in the form: key1=value1;key2=value2;...
83    // Some keys are reserved for standard parameters (See AudioParameter class).
84    // The versions with audio_io_handle_t are intended for internal media framework use only.
85    static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
86    static String8  getParameters(audio_io_handle_t ioHandle, const String8& keys);
87    // The versions without audio_io_handle_t are intended for JNI.
88    static status_t setParameters(const String8& keyValuePairs);
89    static String8  getParameters(const String8& keys);
90
91    static void setErrorCallback(audio_error_callback cb);
92
93    // helper function to obtain AudioFlinger service handle
94    static const sp<IAudioFlinger> get_audio_flinger();
95
96    static float linearToLog(int volume);
97    static int logToLinear(float volume);
98
99    // Returned samplingRate and frameCount output values are guaranteed
100    // to be non-zero if status == NO_ERROR
101    static status_t getOutputSamplingRate(uint32_t* samplingRate,
102            audio_stream_type_t stream);
103    static status_t getOutputFrameCount(size_t* frameCount,
104            audio_stream_type_t stream);
105    static status_t getOutputLatency(uint32_t* latency,
106            audio_stream_type_t stream);
107    static status_t getSamplingRate(audio_io_handle_t output,
108                                          uint32_t* samplingRate);
109    // returns the number of frames per audio HAL write buffer. Corresponds to
110    // audio_stream->get_buffer_size()/audio_stream_out_frame_size()
111    static status_t getFrameCount(audio_io_handle_t output,
112                                  size_t* frameCount);
113    // returns the audio output stream latency in ms. Corresponds to
114    // audio_stream_out->get_latency()
115    static status_t getLatency(audio_io_handle_t output,
116                               uint32_t* latency);
117
118    // return status NO_ERROR implies *buffSize > 0
119    static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
120        audio_channel_mask_t channelMask, size_t* buffSize);
121
122    static status_t setVoiceVolume(float volume);
123
124    // return the number of audio frames written by AudioFlinger to audio HAL and
125    // audio dsp to DAC since the specified output I/O handle has exited standby.
126    // returned status (from utils/Errors.h) can be:
127    // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data
128    // - INVALID_OPERATION: Not supported on current hardware platform
129    // - BAD_VALUE: invalid parameter
130    // NOTE: this feature is not supported on all hardware platforms and it is
131    // necessary to check returned status before using the returned values.
132    static status_t getRenderPosition(audio_io_handle_t output,
133                                      uint32_t *halFrames,
134                                      uint32_t *dspFrames);
135
136    // return the number of input frames lost by HAL implementation, or 0 if the handle is invalid
137    static uint32_t getInputFramesLost(audio_io_handle_t ioHandle);
138
139    // Allocate a new unique ID for use as an audio session ID or I/O handle.
140    // If unable to contact AudioFlinger, returns AUDIO_UNIQUE_ID_ALLOCATE instead.
141    // FIXME If AudioFlinger were to ever exhaust the unique ID namespace,
142    //       this method could fail by returning either AUDIO_UNIQUE_ID_ALLOCATE
143    //       or an unspecified existing unique ID.
144    static audio_unique_id_t newAudioUniqueId();
145
146    static void acquireAudioSessionId(int audioSession, pid_t pid);
147    static void releaseAudioSessionId(int audioSession, pid_t pid);
148
149    // Get the HW synchronization source used for an audio session.
150    // Return a valid source or AUDIO_HW_SYNC_INVALID if an error occurs
151    // or no HW sync source is used.
152    static audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId);
153
154    // types of io configuration change events received with ioConfigChanged()
155    enum io_config_event {
156        OUTPUT_OPENED,
157        OUTPUT_CLOSED,
158        OUTPUT_CONFIG_CHANGED,
159        INPUT_OPENED,
160        INPUT_CLOSED,
161        INPUT_CONFIG_CHANGED,
162        STREAM_CONFIG_CHANGED,
163        NUM_CONFIG_EVENTS
164    };
165
166    // audio output descriptor used to cache output configurations in client process to avoid
167    // frequent calls through IAudioFlinger
168    class OutputDescriptor {
169    public:
170        OutputDescriptor()
171        : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0)
172            {}
173
174        uint32_t samplingRate;
175        audio_format_t format;
176        audio_channel_mask_t channelMask;
177        size_t frameCount;
178        uint32_t latency;
179    };
180
181    // Events used to synchronize actions between audio sessions.
182    // For instance SYNC_EVENT_PRESENTATION_COMPLETE can be used to delay recording start until
183    // playback is complete on another audio session.
184    // See definitions in MediaSyncEvent.java
185    enum sync_event_t {
186        SYNC_EVENT_SAME = -1,             // used internally to indicate restart with same event
187        SYNC_EVENT_NONE = 0,
188        SYNC_EVENT_PRESENTATION_COMPLETE,
189
190        //
191        // Define new events here: SYNC_EVENT_START, SYNC_EVENT_STOP, SYNC_EVENT_TIME ...
192        //
193        SYNC_EVENT_CNT,
194    };
195
196    // Timeout for synchronous record start. Prevents from blocking the record thread forever
197    // if the trigger event is not fired.
198    static const uint32_t kSyncRecordStartTimeOutMs = 30000;
199
200    //
201    // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions)
202    //
203    static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state,
204                                                const char *device_address);
205    static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
206                                                                const char *device_address);
207    static status_t setPhoneState(audio_mode_t state);
208    static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config);
209    static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
210
211    // Client must successfully hand off the handle reference to AudioFlinger via createTrack(),
212    // or release it with releaseOutput().
213    static audio_io_handle_t getOutput(audio_stream_type_t stream,
214                                        uint32_t samplingRate = 0,
215                                        audio_format_t format = AUDIO_FORMAT_DEFAULT,
216                                        audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
217                                        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
218                                        const audio_offload_info_t *offloadInfo = NULL);
219    static status_t getOutputForAttr(const audio_attributes_t *attr,
220                                        audio_io_handle_t *output,
221                                        audio_session_t session,
222                                        audio_stream_type_t *stream,
223                                        uint32_t samplingRate = 0,
224                                        audio_format_t format = AUDIO_FORMAT_DEFAULT,
225                                        audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
226                                        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
227                                        const audio_offload_info_t *offloadInfo = NULL);
228    static status_t startOutput(audio_io_handle_t output,
229                                audio_stream_type_t stream,
230                                audio_session_t session);
231    static status_t stopOutput(audio_io_handle_t output,
232                               audio_stream_type_t stream,
233                               audio_session_t session);
234    static void releaseOutput(audio_io_handle_t output,
235                              audio_stream_type_t stream,
236                              audio_session_t session);
237
238    // Client must successfully hand off the handle reference to AudioFlinger via openRecord(),
239    // or release it with releaseInput().
240    static status_t getInputForAttr(const audio_attributes_t *attr,
241                                    audio_io_handle_t *input,
242                                    audio_session_t session,
243                                    uint32_t samplingRate,
244                                    audio_format_t format,
245                                    audio_channel_mask_t channelMask,
246                                    audio_input_flags_t flags);
247
248    static status_t startInput(audio_io_handle_t input,
249                               audio_session_t session);
250    static status_t stopInput(audio_io_handle_t input,
251                              audio_session_t session);
252    static void releaseInput(audio_io_handle_t input,
253                             audio_session_t session);
254    static status_t initStreamVolume(audio_stream_type_t stream,
255                                      int indexMin,
256                                      int indexMax);
257    static status_t setStreamVolumeIndex(audio_stream_type_t stream,
258                                         int index,
259                                         audio_devices_t device);
260    static status_t getStreamVolumeIndex(audio_stream_type_t stream,
261                                         int *index,
262                                         audio_devices_t device);
263
264    static uint32_t getStrategyForStream(audio_stream_type_t stream);
265    static audio_devices_t getDevicesForStream(audio_stream_type_t stream);
266
267    static audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc);
268    static status_t registerEffect(const effect_descriptor_t *desc,
269                                    audio_io_handle_t io,
270                                    uint32_t strategy,
271                                    int session,
272                                    int id);
273    static status_t unregisterEffect(int id);
274    static status_t setEffectEnabled(int id, bool enabled);
275
276    // clear stream to output mapping cache (gStreamOutputMap)
277    // and output configuration cache (gOutputs)
278    static void clearAudioConfigCache();
279
280    static const sp<IAudioPolicyService> get_audio_policy_service();
281
282    // helpers for android.media.AudioManager.getProperty(), see description there for meaning
283    static uint32_t getPrimaryOutputSamplingRate();
284    static size_t getPrimaryOutputFrameCount();
285
286    static status_t setLowRamDevice(bool isLowRamDevice);
287
288    // Check if hw offload is possible for given format, stream type, sample rate,
289    // bit rate, duration, video and streaming or offload property is enabled
290    static bool isOffloadSupported(const audio_offload_info_t& info);
291
292    // check presence of audio flinger service.
293    // returns NO_ERROR if binding to service succeeds, DEAD_OBJECT otherwise
294    static status_t checkAudioFlinger();
295
296    /* List available audio ports and their attributes */
297    static status_t listAudioPorts(audio_port_role_t role,
298                                   audio_port_type_t type,
299                                   unsigned int *num_ports,
300                                   struct audio_port *ports,
301                                   unsigned int *generation);
302
303    /* Get attributes for a given audio port */
304    static status_t getAudioPort(struct audio_port *port);
305
306    /* Create an audio patch between several source and sink ports */
307    static status_t createAudioPatch(const struct audio_patch *patch,
308                                       audio_patch_handle_t *handle);
309
310    /* Release an audio patch */
311    static status_t releaseAudioPatch(audio_patch_handle_t handle);
312
313    /* List existing audio patches */
314    static status_t listAudioPatches(unsigned int *num_patches,
315                                      struct audio_patch *patches,
316                                      unsigned int *generation);
317    /* Set audio port configuration */
318    static status_t setAudioPortConfig(const struct audio_port_config *config);
319
320
321    static status_t acquireSoundTriggerSession(audio_session_t *session,
322                                           audio_io_handle_t *ioHandle,
323                                           audio_devices_t *device);
324    static status_t releaseSoundTriggerSession(audio_session_t session);
325
326    static audio_mode_t getPhoneState();
327
328    static status_t registerPolicyMixes(Vector<AudioMix> mixes, bool registration);
329
330    // ----------------------------------------------------------------------------
331
332    class AudioPortCallback : public RefBase
333    {
334    public:
335
336                AudioPortCallback() {}
337        virtual ~AudioPortCallback() {}
338
339        virtual void onAudioPortListUpdate() = 0;
340        virtual void onAudioPatchListUpdate() = 0;
341        virtual void onServiceDied() = 0;
342
343    };
344
345    static void setAudioPortCallback(sp<AudioPortCallback> callBack);
346
347private:
348
349    class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient
350    {
351    public:
352        AudioFlingerClient() {
353        }
354
355        // DeathRecipient
356        virtual void binderDied(const wp<IBinder>& who);
357
358        // IAudioFlingerClient
359
360        // indicate a change in the configuration of an output or input: keeps the cached
361        // values for output/input parameters up-to-date in client process
362        virtual void ioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2);
363    };
364
365    class AudioPolicyServiceClient: public IBinder::DeathRecipient,
366                                    public BnAudioPolicyServiceClient
367    {
368    public:
369        AudioPolicyServiceClient() {
370        }
371
372        // DeathRecipient
373        virtual void binderDied(const wp<IBinder>& who);
374
375        // IAudioPolicyServiceClient
376        virtual void onAudioPortListUpdate();
377        virtual void onAudioPatchListUpdate();
378    };
379
380    static sp<AudioFlingerClient> gAudioFlingerClient;
381    static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient;
382    friend class AudioFlingerClient;
383    friend class AudioPolicyServiceClient;
384
385    static Mutex gLock;      // protects gAudioFlinger and gAudioErrorCallback,
386    static Mutex gLockCache; // protects gOutputs, gPrevInSamplingRate, gPrevInFormat,
387                             // gPrevInChannelMask and gInBuffSize
388    static Mutex gLockAPS;   // protects gAudioPolicyService and gAudioPolicyServiceClient
389    static Mutex gLockAPC;   // protects gAudioPortCallback
390    static sp<IAudioFlinger> gAudioFlinger;
391    static audio_error_callback gAudioErrorCallback;
392
393    static size_t gInBuffSize;
394    // previous parameters for recording buffer size queries
395    static uint32_t gPrevInSamplingRate;
396    static audio_format_t gPrevInFormat;
397    static audio_channel_mask_t gPrevInChannelMask;
398
399    static sp<IAudioPolicyService> gAudioPolicyService;
400
401    // list of output descriptors containing cached parameters
402    // (sampling rate, framecount, channel count...)
403    static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs;
404
405    static sp<AudioPortCallback> gAudioPortCallback;
406};
407
408};  // namespace android
409
410#endif  /*ANDROID_AUDIOSYSTEM_H_*/
411