diff options
Diffstat (limited to 'include/media/AudioSystem.h')
| -rw-r--r-- | include/media/AudioSystem.h | 142 |
1 files changed, 92 insertions, 50 deletions
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h index 843a354..06116a5 100644 --- a/include/media/AudioSystem.h +++ b/include/media/AudioSystem.h @@ -19,6 +19,7 @@ #include <hardware/audio_effect.h> #include <media/AudioPolicy.h> +#include <media/AudioIoDescriptor.h> #include <media/IAudioFlingerClient.h> #include <media/IAudioPolicyServiceClient.h> #include <system/audio.h> @@ -29,6 +30,7 @@ namespace android { typedef void (*audio_error_callback)(status_t err); +typedef void (*dynamic_policy_callback)(int event, String8 regId, int val); class IAudioFlinger; class IAudioPolicyService; @@ -89,6 +91,7 @@ public: static String8 getParameters(const String8& keys); static void setErrorCallback(audio_error_callback cb); + static void setDynPolicyCallback(dynamic_policy_callback cb); // helper function to obtain AudioFlinger service handle static const sp<IAudioFlinger> get_audio_flinger(); @@ -98,10 +101,13 @@ public: // Returned samplingRate and frameCount output values are guaranteed // to be non-zero if status == NO_ERROR + // FIXME This API assumes a route, and so should be deprecated. static status_t getOutputSamplingRate(uint32_t* samplingRate, audio_stream_type_t stream); + // FIXME This API assumes a route, and so should be deprecated. static status_t getOutputFrameCount(size_t* frameCount, audio_stream_type_t stream); + // FIXME This API assumes a route, and so should be deprecated. static status_t getOutputLatency(uint32_t* latency, audio_stream_type_t stream); static status_t getSamplingRate(audio_io_handle_t output, @@ -110,19 +116,20 @@ public: // audio_stream->get_buffer_size()/audio_stream_out_frame_size() static status_t getFrameCount(audio_io_handle_t output, size_t* frameCount); - // returns the audio output stream latency in ms. Corresponds to + // returns the audio output latency in ms. Corresponds to // audio_stream_out->get_latency() static status_t getLatency(audio_io_handle_t output, uint32_t* latency); // return status NO_ERROR implies *buffSize > 0 + // FIXME This API assumes a route, and so should deprecated. static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t* buffSize); static status_t setVoiceVolume(float volume); // return the number of audio frames written by AudioFlinger to audio HAL and - // audio dsp to DAC since the specified output I/O handle has exited standby. + // audio dsp to DAC since the specified output has exited standby. // returned status (from utils/Errors.h) can be: // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data // - INVALID_OPERATION: Not supported on current hardware platform @@ -151,32 +158,8 @@ public: // or no HW sync source is used. static audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); - // types of io configuration change events received with ioConfigChanged() - enum io_config_event { - OUTPUT_OPENED, - OUTPUT_CLOSED, - OUTPUT_CONFIG_CHANGED, - INPUT_OPENED, - INPUT_CLOSED, - INPUT_CONFIG_CHANGED, - STREAM_CONFIG_CHANGED, - NUM_CONFIG_EVENTS - }; - - // audio output descriptor used to cache output configurations in client process to avoid - // frequent calls through IAudioFlinger - class OutputDescriptor { - public: - OutputDescriptor() - : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0) - {} - - uint32_t samplingRate; - audio_format_t format; - audio_channel_mask_t channelMask; - size_t frameCount; - uint32_t latency; - }; + // Indicate JAVA services are ready (scheduling, power management ...) + static status_t systemReady(); // Events used to synchronize actions between audio sessions. // For instance SYNC_EVENT_PRESENTATION_COMPLETE can be used to delay recording start until @@ -201,7 +184,7 @@ public: // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions) // static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, - const char *device_address); + const char *device_address, const char *device_name); static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, const char *device_address); static status_t setPhoneState(audio_mode_t state); @@ -217,14 +200,16 @@ public: audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, const audio_offload_info_t *offloadInfo = NULL); static status_t getOutputForAttr(const audio_attributes_t *attr, - audio_io_handle_t *output, - audio_session_t session, - audio_stream_type_t *stream, - uint32_t samplingRate = 0, - audio_format_t format = AUDIO_FORMAT_DEFAULT, - audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO, - audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, - const audio_offload_info_t *offloadInfo = NULL); + audio_io_handle_t *output, + audio_session_t session, + audio_stream_type_t *stream, + uid_t uid, + uint32_t samplingRate = 0, + audio_format_t format = AUDIO_FORMAT_DEFAULT, + audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO, + audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, + audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE, + const audio_offload_info_t *offloadInfo = NULL); static status_t startOutput(audio_io_handle_t output, audio_stream_type_t stream, audio_session_t session); @@ -240,10 +225,12 @@ public: static status_t getInputForAttr(const audio_attributes_t *attr, audio_io_handle_t *input, audio_session_t session, + uid_t uid, uint32_t samplingRate, audio_format_t format, audio_channel_mask_t channelMask, - audio_input_flags_t flags); + audio_input_flags_t flags, + audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE); static status_t startInput(audio_io_handle_t input, audio_session_t session); @@ -327,6 +314,12 @@ public: static status_t registerPolicyMixes(Vector<AudioMix> mixes, bool registration); + static status_t startAudioSource(const struct audio_port_config *source, + const audio_attributes_t *attributes, + audio_io_handle_t *handle); + static status_t stopAudioSource(audio_io_handle_t handle); + + // ---------------------------------------------------------------------------- class AudioPortCallback : public RefBase @@ -342,16 +335,42 @@ public: }; - static void setAudioPortCallback(sp<AudioPortCallback> callBack); + static status_t addAudioPortCallback(const sp<AudioPortCallback>& callback); + static status_t removeAudioPortCallback(const sp<AudioPortCallback>& callback); + + class AudioDeviceCallback : public RefBase + { + public: + + AudioDeviceCallback() {} + virtual ~AudioDeviceCallback() {} + + virtual void onAudioDeviceUpdate(audio_io_handle_t audioIo, + audio_port_handle_t deviceId) = 0; + }; + + static status_t addAudioDeviceCallback(const sp<AudioDeviceCallback>& callback, + audio_io_handle_t audioIo); + static status_t removeAudioDeviceCallback(const sp<AudioDeviceCallback>& callback, + audio_io_handle_t audioIo); + + static audio_port_handle_t getDeviceIdForIo(audio_io_handle_t audioIo); private: class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient { public: - AudioFlingerClient() { + AudioFlingerClient() : + mInBuffSize(0), mInSamplingRate(0), + mInFormat(AUDIO_FORMAT_DEFAULT), mInChannelMask(AUDIO_CHANNEL_NONE) { } + void clearIoCache(); + status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, + audio_channel_mask_t channelMask, size_t* buffSize); + sp<AudioIoDescriptor> getIoDescriptor(audio_io_handle_t ioHandle); + // DeathRecipient virtual void binderDied(const wp<IBinder>& who); @@ -359,7 +378,27 @@ private: // indicate a change in the configuration of an output or input: keeps the cached // values for output/input parameters up-to-date in client process - virtual void ioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2); + virtual void ioConfigChanged(audio_io_config_event event, + const sp<AudioIoDescriptor>& ioDesc); + + + status_t addAudioDeviceCallback(const sp<AudioDeviceCallback>& callback, + audio_io_handle_t audioIo); + status_t removeAudioDeviceCallback(const sp<AudioDeviceCallback>& callback, + audio_io_handle_t audioIo); + + audio_port_handle_t getDeviceIdForIo(audio_io_handle_t audioIo); + + private: + Mutex mLock; + DefaultKeyedVector<audio_io_handle_t, sp<AudioIoDescriptor> > mIoDescriptors; + DefaultKeyedVector<audio_io_handle_t, Vector < sp<AudioDeviceCallback> > > + mAudioDeviceCallbacks; + // cached values for recording getInputBufferSize() queries + size_t mInBuffSize; // zero indicates cache is invalid + uint32_t mInSamplingRate; + audio_format_t mInFormat; + audio_channel_mask_t mInChannelMask; }; class AudioPolicyServiceClient: public IBinder::DeathRecipient, @@ -369,26 +408,35 @@ private: AudioPolicyServiceClient() { } + int addAudioPortCallback(const sp<AudioPortCallback>& callback); + int removeAudioPortCallback(const sp<AudioPortCallback>& callback); + // DeathRecipient virtual void binderDied(const wp<IBinder>& who); // IAudioPolicyServiceClient virtual void onAudioPortListUpdate(); virtual void onAudioPatchListUpdate(); + virtual void onDynamicPolicyMixStateUpdate(String8 regId, int32_t state); + + private: + Mutex mLock; + Vector <sp <AudioPortCallback> > mAudioPortCallbacks; }; + static const sp<AudioFlingerClient> getAudioFlingerClient(); + static sp<AudioIoDescriptor> getIoDescriptor(audio_io_handle_t ioHandle); + static sp<AudioFlingerClient> gAudioFlingerClient; static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient; friend class AudioFlingerClient; friend class AudioPolicyServiceClient; static Mutex gLock; // protects gAudioFlinger and gAudioErrorCallback, - static Mutex gLockCache; // protects gOutputs, gPrevInSamplingRate, gPrevInFormat, - // gPrevInChannelMask and gInBuffSize static Mutex gLockAPS; // protects gAudioPolicyService and gAudioPolicyServiceClient - static Mutex gLockAPC; // protects gAudioPortCallback static sp<IAudioFlinger> gAudioFlinger; static audio_error_callback gAudioErrorCallback; + static dynamic_policy_callback gDynPolicyCallback; static size_t gInBuffSize; // previous parameters for recording buffer size queries @@ -397,12 +445,6 @@ private: static audio_channel_mask_t gPrevInChannelMask; static sp<IAudioPolicyService> gAudioPolicyService; - - // list of output descriptors containing cached parameters - // (sampling rate, framecount, channel count...) - static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs; - - static sp<AudioPortCallback> gAudioPortCallback; }; }; // namespace android |