AudioSystem.h revision caf7f48a0ef558689d39aafd187c1571ff4128b4 
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/IAudioFlingerClient.h>
22#include <media/IAudioPolicyServiceClient.h>
23#include <system/audio.h>
24#include <system/audio_policy.h>
25#include <utils/Errors.h>
26#include <utils/Mutex.h>
27
28namespace android {
29
30typedef void (*audio_error_callback)(status_t err);
31
32class IAudioFlinger;
33class IAudioPolicyService;
34class String8;
35
36class AudioSystem
37{
38public:
39
40    /* These are static methods to control the system-wide AudioFlinger
41     * only privileged processes can have access to them
42     */
43
44    // mute/unmute microphone
45    static status_t muteMicrophone(bool state);
46    static status_t isMicrophoneMuted(bool *state);
47
48    // set/get master volume
49    static status_t setMasterVolume(float value);
50    static status_t getMasterVolume(float* volume);
51
52    // mute/unmute audio outputs
53    static status_t setMasterMute(bool mute);
54    static status_t getMasterMute(bool* mute);
55
56    // set/get stream volume on specified output
57    static status_t setStreamVolume(audio_stream_type_t stream, float value,
58                                    audio_io_handle_t output);
59    static status_t getStreamVolume(audio_stream_type_t stream, float* volume,
60                                    audio_io_handle_t output);
61
62    // mute/unmute stream
63    static status_t setStreamMute(audio_stream_type_t stream, bool mute);
64    static status_t getStreamMute(audio_stream_type_t stream, bool* mute);
65
66    // set audio mode in audio hardware
67    static status_t setMode(audio_mode_t mode);
68
69    // returns true in *state if tracks are active on the specified stream or have been active
70    // in the past inPastMs milliseconds
71    static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs);
72    // returns true in *state if tracks are active for what qualifies as remote playback
73    // on the specified stream or have been active in the past inPastMs milliseconds. Remote
74    // playback isn't mutually exclusive with local playback.
75    static status_t isStreamActiveRemotely(audio_stream_type_t stream, bool *state,
76            uint32_t inPastMs);
77    // returns true in *state if a recorder is currently recording with the specified source
78    static status_t isSourceActive(audio_source_t source, bool *state);
79
80    // set/get audio hardware parameters. The function accepts a list of parameters
81    // key value pairs in the form: key1=value1;key2=value2;...
82    // Some keys are reserved for standard parameters (See AudioParameter class).
83    // The versions with audio_io_handle_t are intended for internal media framework use only.
84    static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
85    static String8  getParameters(audio_io_handle_t ioHandle, const String8& keys);
86    // The versions without audio_io_handle_t are intended for JNI.
87    static status_t setParameters(const String8& keyValuePairs);
88    static String8  getParameters(const String8& keys);
89
90    static void setErrorCallback(audio_error_callback cb);
91
92    // helper function to obtain AudioFlinger service handle
93    static const sp<IAudioFlinger> get_audio_flinger();
94
95    static float linearToLog(int volume);
96    static int logToLinear(float volume);
97
98    // Returned samplingRate and frameCount output values are guaranteed
99    // to be non-zero if status == NO_ERROR
100    static status_t getOutputSamplingRate(uint32_t* samplingRate,
101            audio_stream_type_t stream);
102    static status_t getOutputFrameCount(size_t* frameCount,
103            audio_stream_type_t stream);
104    static status_t getOutputLatency(uint32_t* latency,
105            audio_stream_type_t stream);
106    static status_t getSamplingRate(audio_io_handle_t output,
107                                          uint32_t* samplingRate);
108    // returns the number of frames per audio HAL write buffer. Corresponds to
109    // audio_stream->get_buffer_size()/audio_stream_out_frame_size()
110    static status_t getFrameCount(audio_io_handle_t output,
111                                  size_t* frameCount);
112    // returns the audio output stream latency in ms. Corresponds to
113    // audio_stream_out->get_latency()
114    static status_t getLatency(audio_io_handle_t output,
115                               uint32_t* latency);
116
117    // return status NO_ERROR implies *buffSize > 0
118    static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
119        audio_channel_mask_t channelMask, size_t* buffSize);
120
121    static status_t setVoiceVolume(float volume);
122
123    // return the number of audio frames written by AudioFlinger to audio HAL and
124    // audio dsp to DAC since the specified output I/O handle has exited standby.
125    // returned status (from utils/Errors.h) can be:
126    // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data
127    // - INVALID_OPERATION: Not supported on current hardware platform
128    // - BAD_VALUE: invalid parameter
129    // NOTE: this feature is not supported on all hardware platforms and it is
130    // necessary to check returned status before using the returned values.
131    static status_t getRenderPosition(audio_io_handle_t output,
132                                      uint32_t *halFrames,
133                                      uint32_t *dspFrames);
134
135    // return the number of input frames lost by HAL implementation, or 0 if the handle is invalid
136    static uint32_t getInputFramesLost(audio_io_handle_t ioHandle);
137
138    // Allocate a new unique ID for use as an audio session ID or I/O handle.
139    // If unable to contact AudioFlinger, returns AUDIO_UNIQUE_ID_ALLOCATE instead.
140    // FIXME If AudioFlinger were to ever exhaust the unique ID namespace,
141    //       this method could fail by returning either AUDIO_UNIQUE_ID_ALLOCATE
142    //       or an unspecified existing unique ID.
143    static audio_unique_id_t newAudioUniqueId();
144
145    static void acquireAudioSessionId(int audioSession, pid_t pid);
146    static void releaseAudioSessionId(int audioSession, pid_t pid);
147
148    // Get the HW synchronization source used for an audio session.
149    // Return a valid source or AUDIO_HW_SYNC_INVALID if an error occurs
150    // or no HW sync source is used.
151    static audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId);
152
153    // types of io configuration change events received with ioConfigChanged()
154    enum io_config_event {
155        OUTPUT_OPENED,
156        OUTPUT_CLOSED,
157        OUTPUT_CONFIG_CHANGED,
158        INPUT_OPENED,
159        INPUT_CLOSED,
160        INPUT_CONFIG_CHANGED,
161        STREAM_CONFIG_CHANGED,
162        NUM_CONFIG_EVENTS
163    };
164
165    // audio output descriptor used to cache output configurations in client process to avoid
166    // frequent calls through IAudioFlinger
167    class OutputDescriptor {
168    public:
169        OutputDescriptor()
170        : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0)
171            {}
172
173        uint32_t samplingRate;
174        audio_format_t format;
175        audio_channel_mask_t channelMask;
176        size_t frameCount;
177        uint32_t latency;
178    };
179
180    // Events used to synchronize actions between audio sessions.
181    // For instance SYNC_EVENT_PRESENTATION_COMPLETE can be used to delay recording start until
182    // playback is complete on another audio session.
183    // See definitions in MediaSyncEvent.java
184    enum sync_event_t {
185        SYNC_EVENT_SAME = -1,             // used internally to indicate restart with same event
186        SYNC_EVENT_NONE = 0,
187        SYNC_EVENT_PRESENTATION_COMPLETE,
188
189        //
190        // Define new events here: SYNC_EVENT_START, SYNC_EVENT_STOP, SYNC_EVENT_TIME ...
191        //
192        SYNC_EVENT_CNT,
193    };
194
195    // Timeout for synchronous record start. Prevents from blocking the record thread forever
196    // if the trigger event is not fired.
197    static const uint32_t kSyncRecordStartTimeOutMs = 30000;
198
199    //
200    // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions)
201    //
202    static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state,
203                                                const char *device_address);
204    static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
205                                                                const char *device_address);
206    static status_t setPhoneState(audio_mode_t state);
207    static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config);
208    static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
209
210    // Client must successfully hand off the handle reference to AudioFlinger via createTrack(),
211    // or release it with releaseOutput().
212    static audio_io_handle_t getOutput(audio_stream_type_t stream,
213                                        uint32_t samplingRate = 0,
214                                        audio_format_t format = AUDIO_FORMAT_DEFAULT,
215                                        audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
216                                        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
217                                        const audio_offload_info_t *offloadInfo = NULL);
218    static status_t getOutputForAttr(const audio_attributes_t *attr,
219                                        audio_io_handle_t *output,
220                                        audio_session_t session,
221                                        audio_stream_type_t *stream,
222                                        uint32_t samplingRate = 0,
223                                        audio_format_t format = AUDIO_FORMAT_DEFAULT,
224                                        audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
225                                        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
226                                        const audio_offload_info_t *offloadInfo = NULL);
227    static status_t startOutput(audio_io_handle_t output,
228                                audio_stream_type_t stream,
229                                audio_session_t session);
230    static status_t stopOutput(audio_io_handle_t output,
231                               audio_stream_type_t stream,
232                               audio_session_t session);
233    static void releaseOutput(audio_io_handle_t output,
234                              audio_stream_type_t stream,
235                              audio_session_t session);
236
237    // Client must successfully hand off the handle reference to AudioFlinger via openRecord(),
238    // or release it with releaseInput().
239    static status_t getInputForAttr(const audio_attributes_t *attr,
240                                    audio_io_handle_t *input,
241                                    audio_session_t session,
242                                    uint32_t samplingRate,
243                                    audio_format_t format,
244                                    audio_channel_mask_t channelMask,
245                                    audio_input_flags_t flags);
246
247    static status_t startInput(audio_io_handle_t input,
248                               audio_session_t session);
249    static status_t stopInput(audio_io_handle_t input,
250                              audio_session_t session);
251    static void releaseInput(audio_io_handle_t input,
252                             audio_session_t session);
253    static status_t initStreamVolume(audio_stream_type_t stream,
254                                      int indexMin,
255                                      int indexMax);
256    static status_t setStreamVolumeIndex(audio_stream_type_t stream,
257                                         int index,
258                                         audio_devices_t device);
259    static status_t getStreamVolumeIndex(audio_stream_type_t stream,
260                                         int *index,
261                                         audio_devices_t device);
262
263    static uint32_t getStrategyForStream(audio_stream_type_t stream);
264    static audio_devices_t getDevicesForStream(audio_stream_type_t stream);
265
266    static audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc);
267    static status_t registerEffect(const effect_descriptor_t *desc,
268                                    audio_io_handle_t io,
269                                    uint32_t strategy,
270                                    int session,
271                                    int id);
272    static status_t unregisterEffect(int id);
273    static status_t setEffectEnabled(int id, bool enabled);
274
275    // clear stream to output mapping cache (gStreamOutputMap)
276    // and output configuration cache (gOutputs)
277    static void clearAudioConfigCache();
278
279    static const sp<IAudioPolicyService> get_audio_policy_service();
280
281    // helpers for android.media.AudioManager.getProperty(), see description there for meaning
282    static uint32_t getPrimaryOutputSamplingRate();
283    static size_t getPrimaryOutputFrameCount();
284
285    static status_t setLowRamDevice(bool isLowRamDevice);
286
287    // Check if hw offload is possible for given format, stream type, sample rate,
288    // bit rate, duration, video and streaming or offload property is enabled
289    static bool isOffloadSupported(const audio_offload_info_t& info);
290
291    // check presence of audio flinger service.
292    // returns NO_ERROR if binding to service succeeds, DEAD_OBJECT otherwise
293    static status_t checkAudioFlinger();
294
295    /* List available audio ports and their attributes */
296    static status_t listAudioPorts(audio_port_role_t role,
297                                   audio_port_type_t type,
298                                   unsigned int *num_ports,
299                                   struct audio_port *ports,
300                                   unsigned int *generation);
301
302    /* Get attributes for a given audio port */
303    static status_t getAudioPort(struct audio_port *port);
304
305    /* Create an audio patch between several source and sink ports */
306    static status_t createAudioPatch(const struct audio_patch *patch,
307                                       audio_patch_handle_t *handle);
308
309    /* Release an audio patch */
310    static status_t releaseAudioPatch(audio_patch_handle_t handle);
311
312    /* List existing audio patches */
313    static status_t listAudioPatches(unsigned int *num_patches,
314                                      struct audio_patch *patches,
315                                      unsigned int *generation);
316    /* Set audio port configuration */
317    static status_t setAudioPortConfig(const struct audio_port_config *config);
318
319
320    static status_t acquireSoundTriggerSession(audio_session_t *session,
321                                           audio_io_handle_t *ioHandle,
322                                           audio_devices_t *device);
323    static status_t releaseSoundTriggerSession(audio_session_t session);
324
325    static audio_mode_t getPhoneState();
326
327    // ----------------------------------------------------------------------------
328
329    class AudioPortCallback : public RefBase
330    {
331    public:
332
333                AudioPortCallback() {}
334        virtual ~AudioPortCallback() {}
335
336        virtual void onAudioPortListUpdate() = 0;
337        virtual void onAudioPatchListUpdate() = 0;
338        virtual void onServiceDied() = 0;
339
340    };
341
342    static void setAudioPortCallback(sp<AudioPortCallback> callBack);
343
344private:
345
346    class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient
347    {
348    public:
349        AudioFlingerClient() {
350        }
351
352        // DeathRecipient
353        virtual void binderDied(const wp<IBinder>& who);
354
355        // IAudioFlingerClient
356
357        // indicate a change in the configuration of an output or input: keeps the cached
358        // values for output/input parameters up-to-date in client process
359        virtual void ioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2);
360    };
361
362    class AudioPolicyServiceClient: public IBinder::DeathRecipient,
363                                    public BnAudioPolicyServiceClient
364    {
365    public:
366        AudioPolicyServiceClient() {
367        }
368
369        // DeathRecipient
370        virtual void binderDied(const wp<IBinder>& who);
371
372        // IAudioPolicyServiceClient
373        virtual void onAudioPortListUpdate();
374        virtual void onAudioPatchListUpdate();
375    };
376
377    static sp<AudioFlingerClient> gAudioFlingerClient;
378    static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient;
379    friend class AudioFlingerClient;
380    friend class AudioPolicyServiceClient;
381
382    static Mutex gLock;      // protects gAudioFlinger and gAudioErrorCallback,
383    static Mutex gLockCache; // protects gOutputs, gPrevInSamplingRate, gPrevInFormat,
384                             // gPrevInChannelMask and gInBuffSize
385    static Mutex gLockAPS;   // protects gAudioPolicyService and gAudioPolicyServiceClient
386    static Mutex gLockAPC;   // protects gAudioPortCallback
387    static sp<IAudioFlinger> gAudioFlinger;
388    static audio_error_callback gAudioErrorCallback;
389
390    static size_t gInBuffSize;
391    // previous parameters for recording buffer size queries
392    static uint32_t gPrevInSamplingRate;
393    static audio_format_t gPrevInFormat;
394    static audio_channel_mask_t gPrevInChannelMask;
395
396    static sp<IAudioPolicyService> gAudioPolicyService;
397
398    // list of output descriptors containing cached parameters
399    // (sampling rate, framecount, channel count...)
400    static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs;
401
402    static sp<AudioPortCallback> gAudioPortCallback;
403};
404
405};  // namespace android
406
407#endif  /*ANDROID_AUDIOSYSTEM_H_*/
408