AudioSystem.h revision 4dc680607181e6a76f4e91a39366c4f5dfb7b03e
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 getOutputSamplingRateForAttr(uint32_t* samplingRate,
103                const audio_attributes_t *attr);
104    static status_t getOutputFrameCount(size_t* frameCount,
105            audio_stream_type_t stream);
106    static status_t getOutputLatency(uint32_t* latency,
107            audio_stream_type_t stream);
108    static status_t getSamplingRate(audio_io_handle_t output,
109                                          uint32_t* samplingRate);
110    // returns the number of frames per audio HAL write buffer. Corresponds to
111    // audio_stream->get_buffer_size()/audio_stream_out_frame_size()
112    static status_t getFrameCount(audio_io_handle_t output,
113                                  size_t* frameCount);
114    // returns the audio output stream latency in ms. Corresponds to
115    // audio_stream_out->get_latency()
116    static status_t getLatency(audio_io_handle_t output,
117                               uint32_t* latency);
118
119    static bool routedToA2dpOutput(audio_stream_type_t streamType);
120
121    // return status NO_ERROR implies *buffSize > 0
122    static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
123        audio_channel_mask_t channelMask, size_t* buffSize);
124
125    static status_t setVoiceVolume(float volume);
126
127    // return the number of audio frames written by AudioFlinger to audio HAL and
128    // audio dsp to DAC since the specified output I/O handle has exited standby.
129    // returned status (from utils/Errors.h) can be:
130    // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data
131    // - INVALID_OPERATION: Not supported on current hardware platform
132    // - BAD_VALUE: invalid parameter
133    // NOTE: this feature is not supported on all hardware platforms and it is
134    // necessary to check returned status before using the returned values.
135    static status_t getRenderPosition(audio_io_handle_t output,
136                                      uint32_t *halFrames,
137                                      uint32_t *dspFrames);
138
139    // return the number of input frames lost by HAL implementation, or 0 if the handle is invalid
140    static uint32_t getInputFramesLost(audio_io_handle_t ioHandle);
141
142    // Allocate a new unique ID for use as an audio session ID or I/O handle.
143    // If unable to contact AudioFlinger, returns AUDIO_UNIQUE_ID_ALLOCATE instead.
144    // FIXME If AudioFlinger were to ever exhaust the unique ID namespace,
145    //       this method could fail by returning either AUDIO_UNIQUE_ID_ALLOCATE
146    //       or an unspecified existing unique ID.
147    static audio_unique_id_t newAudioUniqueId();
148
149    static void acquireAudioSessionId(int audioSession, pid_t pid);
150    static void releaseAudioSessionId(int audioSession, pid_t pid);
151
152    // types of io configuration change events received with ioConfigChanged()
153    enum io_config_event {
154        OUTPUT_OPENED,
155        OUTPUT_CLOSED,
156        OUTPUT_CONFIG_CHANGED,
157        INPUT_OPENED,
158        INPUT_CLOSED,
159        INPUT_CONFIG_CHANGED,
160        STREAM_CONFIG_CHANGED,
161        NUM_CONFIG_EVENTS
162    };
163
164    // audio output descriptor used to cache output configurations in client process to avoid
165    // frequent calls through IAudioFlinger
166    class OutputDescriptor {
167    public:
168        OutputDescriptor()
169        : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0)
170            {}
171
172        uint32_t samplingRate;
173        audio_format_t format;
174        audio_channel_mask_t channelMask;
175        size_t frameCount;
176        uint32_t latency;
177    };
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);
203    static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
204                                                                const char *device_address);
205    static status_t setPhoneState(audio_mode_t state);
206    static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config);
207    static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
208
209    // Client must successfully hand off the handle reference to AudioFlinger via createTrack(),
210    // or release it with releaseOutput().
211    static audio_io_handle_t getOutput(audio_stream_type_t stream,
212                                        uint32_t samplingRate = 0,
213                                        audio_format_t format = AUDIO_FORMAT_DEFAULT,
214                                        audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
215                                        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
216                                        const audio_offload_info_t *offloadInfo = NULL);
217    static audio_io_handle_t getOutputForAttr(const audio_attributes_t *attr,
218                                        uint32_t samplingRate = 0,
219                                        audio_format_t format = AUDIO_FORMAT_DEFAULT,
220                                        audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
221                                        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
222                                        const audio_offload_info_t *offloadInfo = NULL);
223    static status_t startOutput(audio_io_handle_t output,
224                                audio_stream_type_t stream,
225                                int session);
226    static status_t stopOutput(audio_io_handle_t output,
227                               audio_stream_type_t stream,
228                               int session);
229    static void releaseOutput(audio_io_handle_t output);
230
231    // Client must successfully hand off the handle reference to AudioFlinger via openRecord(),
232    // or release it with releaseInput().
233    static audio_io_handle_t getInput(audio_source_t inputSource,
234                                    uint32_t samplingRate,
235                                    audio_format_t format,
236                                    audio_channel_mask_t channelMask,
237                                    int sessionId,
238                                    audio_input_flags_t);
239
240    static status_t startInput(audio_io_handle_t input,
241                               audio_session_t session);
242    static status_t stopInput(audio_io_handle_t input,
243                              audio_session_t session);
244    static void releaseInput(audio_io_handle_t input,
245                             audio_session_t session);
246    static status_t initStreamVolume(audio_stream_type_t stream,
247                                      int indexMin,
248                                      int indexMax);
249    static status_t setStreamVolumeIndex(audio_stream_type_t stream,
250                                         int index,
251                                         audio_devices_t device);
252    static status_t getStreamVolumeIndex(audio_stream_type_t stream,
253                                         int *index,
254                                         audio_devices_t device);
255
256    static uint32_t getStrategyForStream(audio_stream_type_t stream);
257    static audio_devices_t getDevicesForStream(audio_stream_type_t stream);
258
259    static audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc);
260    static status_t registerEffect(const effect_descriptor_t *desc,
261                                    audio_io_handle_t io,
262                                    uint32_t strategy,
263                                    int session,
264                                    int id);
265    static status_t unregisterEffect(int id);
266    static status_t setEffectEnabled(int id, bool enabled);
267
268    // clear stream to output mapping cache (gStreamOutputMap)
269    // and output configuration cache (gOutputs)
270    static void clearAudioConfigCache();
271
272    static const sp<IAudioPolicyService>& get_audio_policy_service();
273
274    // helpers for android.media.AudioManager.getProperty(), see description there for meaning
275    static uint32_t getPrimaryOutputSamplingRate();
276    static size_t getPrimaryOutputFrameCount();
277
278    static status_t setLowRamDevice(bool isLowRamDevice);
279
280    // Check if hw offload is possible for given format, stream type, sample rate,
281    // bit rate, duration, video and streaming or offload property is enabled
282    static bool isOffloadSupported(const audio_offload_info_t& info);
283
284    // check presence of audio flinger service.
285    // returns NO_ERROR if binding to service succeeds, DEAD_OBJECT otherwise
286    static status_t checkAudioFlinger();
287
288    /* List available audio ports and their attributes */
289    static status_t listAudioPorts(audio_port_role_t role,
290                                   audio_port_type_t type,
291                                   unsigned int *num_ports,
292                                   struct audio_port *ports,
293                                   unsigned int *generation);
294
295    /* Get attributes for a given audio port */
296    static status_t getAudioPort(struct audio_port *port);
297
298    /* Create an audio patch between several source and sink ports */
299    static status_t createAudioPatch(const struct audio_patch *patch,
300                                       audio_patch_handle_t *handle);
301
302    /* Release an audio patch */
303    static status_t releaseAudioPatch(audio_patch_handle_t handle);
304
305    /* List existing audio patches */
306    static status_t listAudioPatches(unsigned int *num_patches,
307                                      struct audio_patch *patches,
308                                      unsigned int *generation);
309    /* Set audio port configuration */
310    static status_t setAudioPortConfig(const struct audio_port_config *config);
311
312    // ----------------------------------------------------------------------------
313
314    class AudioPortCallback : public RefBase
315    {
316    public:
317
318                AudioPortCallback() {}
319        virtual ~AudioPortCallback() {}
320
321        virtual void onAudioPortListUpdate() = 0;
322        virtual void onAudioPatchListUpdate() = 0;
323        virtual void onServiceDied() = 0;
324
325    };
326
327    static void setAudioPortCallback(sp<AudioPortCallback> callBack);
328
329private:
330
331    class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient
332    {
333    public:
334        AudioFlingerClient() {
335        }
336
337        // DeathRecipient
338        virtual void binderDied(const wp<IBinder>& who);
339
340        // IAudioFlingerClient
341
342        // indicate a change in the configuration of an output or input: keeps the cached
343        // values for output/input parameters up-to-date in client process
344        virtual void ioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2);
345    };
346
347    class AudioPolicyServiceClient: public IBinder::DeathRecipient,
348                                    public BnAudioPolicyServiceClient
349    {
350    public:
351        AudioPolicyServiceClient() {
352        }
353
354        // DeathRecipient
355        virtual void binderDied(const wp<IBinder>& who);
356
357        // IAudioPolicyServiceClient
358        virtual void onAudioPortListUpdate();
359        virtual void onAudioPatchListUpdate();
360    };
361
362    static sp<AudioFlingerClient> gAudioFlingerClient;
363    static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient;
364    friend class AudioFlingerClient;
365    friend class AudioPolicyServiceClient;
366
367    static Mutex gLock;
368    static sp<IAudioFlinger> gAudioFlinger;
369    static audio_error_callback gAudioErrorCallback;
370
371    static size_t gInBuffSize;
372    // previous parameters for recording buffer size queries
373    static uint32_t gPrevInSamplingRate;
374    static audio_format_t gPrevInFormat;
375    static audio_channel_mask_t gPrevInChannelMask;
376
377    static sp<IAudioPolicyService> gAudioPolicyService;
378
379    // list of output descriptors containing cached parameters
380    // (sampling rate, framecount, channel count...)
381    static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs;
382
383    static sp<AudioPortCallback> gAudioPortCallback;
384};
385
386};  // namespace android
387
388#endif  /*ANDROID_AUDIOSYSTEM_H_*/
389