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      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 
     32 namespace android {
     33 
     34 typedef void (*audio_error_callback)(status_t err);
     35 typedef void (*dynamic_policy_callback)(int event, String8 regId, int val);
     36 typedef void (*record_config_callback)(int event, audio_session_t session, int source,
     37                 const audio_config_base_t *clientConfig, const audio_config_base_t *deviceConfig,
     38                 audio_patch_handle_t patchHandle);
     39 
     40 class IAudioFlinger;
     41 class IAudioPolicyService;
     42 class String8;
     43 
     44 class AudioSystem
     45 {
     46 public:
     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     // ----------------------------------------------------------------------------
    342 
    343     class AudioPortCallback : public RefBase
    344     {
    345     public:
    346 
    347                 AudioPortCallback() {}
    348         virtual ~AudioPortCallback() {}
    349 
    350         virtual void onAudioPortListUpdate() = 0;
    351         virtual void onAudioPatchListUpdate() = 0;
    352         virtual void onServiceDied() = 0;
    353 
    354     };
    355 
    356     static status_t addAudioPortCallback(const sp<AudioPortCallback>& callback);
    357     static status_t removeAudioPortCallback(const sp<AudioPortCallback>& callback);
    358 
    359     class AudioDeviceCallback : public RefBase
    360     {
    361     public:
    362 
    363                 AudioDeviceCallback() {}
    364         virtual ~AudioDeviceCallback() {}
    365 
    366         virtual void onAudioDeviceUpdate(audio_io_handle_t audioIo,
    367                                          audio_port_handle_t deviceId) = 0;
    368     };
    369 
    370     static status_t addAudioDeviceCallback(const sp<AudioDeviceCallback>& callback,
    371                                            audio_io_handle_t audioIo);
    372     static status_t removeAudioDeviceCallback(const sp<AudioDeviceCallback>& callback,
    373                                               audio_io_handle_t audioIo);
    374 
    375     static audio_port_handle_t getDeviceIdForIo(audio_io_handle_t audioIo);
    376 
    377 private:
    378 
    379     class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient
    380     {
    381     public:
    382         AudioFlingerClient() :
    383             mInBuffSize(0), mInSamplingRate(0),
    384             mInFormat(AUDIO_FORMAT_DEFAULT), mInChannelMask(AUDIO_CHANNEL_NONE) {
    385         }
    386 
    387         void clearIoCache();
    388         status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
    389                                     audio_channel_mask_t channelMask, size_t* buffSize);
    390         sp<AudioIoDescriptor> getIoDescriptor(audio_io_handle_t ioHandle);
    391 
    392         // DeathRecipient
    393         virtual void binderDied(const wp<IBinder>& who);
    394 
    395         // IAudioFlingerClient
    396 
    397         // indicate a change in the configuration of an output or input: keeps the cached
    398         // values for output/input parameters up-to-date in client process
    399         virtual void ioConfigChanged(audio_io_config_event event,
    400                                      const sp<AudioIoDescriptor>& ioDesc);
    401 
    402 
    403         status_t addAudioDeviceCallback(const sp<AudioDeviceCallback>& callback,
    404                                                audio_io_handle_t audioIo);
    405         status_t removeAudioDeviceCallback(const sp<AudioDeviceCallback>& callback,
    406                                            audio_io_handle_t audioIo);
    407 
    408         audio_port_handle_t getDeviceIdForIo(audio_io_handle_t audioIo);
    409 
    410     private:
    411         Mutex                               mLock;
    412         DefaultKeyedVector<audio_io_handle_t, sp<AudioIoDescriptor> >   mIoDescriptors;
    413         DefaultKeyedVector<audio_io_handle_t, Vector < sp<AudioDeviceCallback> > >
    414                                                                         mAudioDeviceCallbacks;
    415         // cached values for recording getInputBufferSize() queries
    416         size_t                              mInBuffSize;    // zero indicates cache is invalid
    417         uint32_t                            mInSamplingRate;
    418         audio_format_t                      mInFormat;
    419         audio_channel_mask_t                mInChannelMask;
    420         sp<AudioIoDescriptor> getIoDescriptor_l(audio_io_handle_t ioHandle);
    421     };
    422 
    423     class AudioPolicyServiceClient: public IBinder::DeathRecipient,
    424                                     public BnAudioPolicyServiceClient
    425     {
    426     public:
    427         AudioPolicyServiceClient() {
    428         }
    429 
    430         int addAudioPortCallback(const sp<AudioPortCallback>& callback);
    431         int removeAudioPortCallback(const sp<AudioPortCallback>& callback);
    432 
    433         // DeathRecipient
    434         virtual void binderDied(const wp<IBinder>& who);
    435 
    436         // IAudioPolicyServiceClient
    437         virtual void onAudioPortListUpdate();
    438         virtual void onAudioPatchListUpdate();
    439         virtual void onDynamicPolicyMixStateUpdate(String8 regId, int32_t state);
    440         virtual void onRecordingConfigurationUpdate(int event, audio_session_t session,
    441                         audio_source_t source, const audio_config_base_t *clientConfig,
    442                         const audio_config_base_t *deviceConfig, audio_patch_handle_t patchHandle);
    443 
    444     private:
    445         Mutex                               mLock;
    446         Vector <sp <AudioPortCallback> >    mAudioPortCallbacks;
    447     };
    448 
    449     static const sp<AudioFlingerClient> getAudioFlingerClient();
    450     static sp<AudioIoDescriptor> getIoDescriptor(audio_io_handle_t ioHandle);
    451 
    452     static sp<AudioFlingerClient> gAudioFlingerClient;
    453     static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient;
    454     friend class AudioFlingerClient;
    455     friend class AudioPolicyServiceClient;
    456 
    457     static Mutex gLock;      // protects gAudioFlinger and gAudioErrorCallback,
    458     static Mutex gLockAPS;   // protects gAudioPolicyService and gAudioPolicyServiceClient
    459     static sp<IAudioFlinger> gAudioFlinger;
    460     static audio_error_callback gAudioErrorCallback;
    461     static dynamic_policy_callback gDynPolicyCallback;
    462     static record_config_callback gRecordConfigCallback;
    463 
    464     static size_t gInBuffSize;
    465     // previous parameters for recording buffer size queries
    466     static uint32_t gPrevInSamplingRate;
    467     static audio_format_t gPrevInFormat;
    468     static audio_channel_mask_t gPrevInChannelMask;
    469 
    470     static sp<IAudioPolicyService> gAudioPolicyService;
    471 };
    472 
    473 };  // namespace android
    474 
    475 #endif  /*ANDROID_AUDIOSYSTEM_H_*/
    476