diff options
Diffstat (limited to 'include/media/AudioSystem.h')
| -rw-r--r-- | include/media/AudioSystem.h | 145 |
1 files changed, 115 insertions, 30 deletions
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h index 4c22412..dd63a23 100644 --- a/include/media/AudioSystem.h +++ b/include/media/AudioSystem.h @@ -19,6 +19,7 @@ #include <hardware/audio_effect.h> #include <media/IAudioFlingerClient.h> +#include <media/IAudioPolicyServiceClient.h> #include <system/audio.h> #include <system/audio_policy.h> #include <utils/Errors.h> @@ -67,20 +68,24 @@ public: // returns true in *state if tracks are active on the specified stream or have been active // in the past inPastMs milliseconds - static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs = 0); + static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs); // returns true in *state if tracks are active for what qualifies as remote playback // on the specified stream or have been active in the past inPastMs milliseconds. Remote // playback isn't mutually exclusive with local playback. static status_t isStreamActiveRemotely(audio_stream_type_t stream, bool *state, - uint32_t inPastMs = 0); + uint32_t inPastMs); // returns true in *state if a recorder is currently recording with the specified source static status_t isSourceActive(audio_source_t source, bool *state); // set/get audio hardware parameters. The function accepts a list of parameters // key value pairs in the form: key1=value1;key2=value2;... // Some keys are reserved for standard parameters (See AudioParameter class). + // The versions with audio_io_handle_t are intended for internal media framework use only. static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); static String8 getParameters(audio_io_handle_t ioHandle, const String8& keys); + // The versions without audio_io_handle_t are intended for JNI. + static status_t setParameters(const String8& keyValuePairs); + static String8 getParameters(const String8& keys); static void setErrorCallback(audio_error_callback cb); @@ -90,36 +95,37 @@ public: static float linearToLog(int volume); static int logToLinear(float volume); + // Returned samplingRate and frameCount output values are guaranteed + // to be non-zero if status == NO_ERROR static status_t getOutputSamplingRate(uint32_t* samplingRate, - audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); + audio_stream_type_t stream); + static status_t getOutputSamplingRateForAttr(uint32_t* samplingRate, + const audio_attributes_t *attr); static status_t getOutputFrameCount(size_t* frameCount, - audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); + audio_stream_type_t stream); static status_t getOutputLatency(uint32_t* latency, - audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); + audio_stream_type_t stream); static status_t getSamplingRate(audio_io_handle_t output, - audio_stream_type_t streamType, uint32_t* samplingRate); // returns the number of frames per audio HAL write buffer. Corresponds to - // audio_stream->get_buffer_size()/audio_stream_frame_size() + // audio_stream->get_buffer_size()/audio_stream_out_frame_size() static status_t getFrameCount(audio_io_handle_t output, - audio_stream_type_t stream, size_t* frameCount); // returns the audio output stream latency in ms. Corresponds to // audio_stream_out->get_latency() static status_t getLatency(audio_io_handle_t output, - audio_stream_type_t stream, uint32_t* latency); static bool routedToA2dpOutput(audio_stream_type_t streamType); + // return status NO_ERROR implies *buffSize > 0 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 output on which the specified stream is playing - // has exited standby. + // audio dsp to DAC since the specified output I/O handle 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 @@ -128,15 +134,25 @@ public: // necessary to check returned status before using the returned values. static status_t getRenderPosition(audio_io_handle_t output, uint32_t *halFrames, - uint32_t *dspFrames, - audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); + uint32_t *dspFrames); // return the number of input frames lost by HAL implementation, or 0 if the handle is invalid - static size_t getInputFramesLost(audio_io_handle_t ioHandle); + static uint32_t getInputFramesLost(audio_io_handle_t ioHandle); + + // Allocate a new unique ID for use as an audio session ID or I/O handle. + // If unable to contact AudioFlinger, returns AUDIO_UNIQUE_ID_ALLOCATE instead. + // FIXME If AudioFlinger were to ever exhaust the unique ID namespace, + // this method could fail by returning either AUDIO_UNIQUE_ID_ALLOCATE + // or an unspecified existing unique ID. + static audio_unique_id_t newAudioUniqueId(); - static int newAudioSessionId(); - static void acquireAudioSessionId(int audioSession); - static void releaseAudioSessionId(int audioSession); + static void acquireAudioSessionId(int audioSession, pid_t pid); + static void releaseAudioSessionId(int audioSession, pid_t pid); + + // Get the HW synchronization source used for an audio session. + // Return a valid source or AUDIO_HW_SYNC_INVALID if an error occurs + // 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 { @@ -155,7 +171,8 @@ public: class OutputDescriptor { public: OutputDescriptor() - : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0) {} + : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0) + {} uint32_t samplingRate; audio_format_t format; @@ -193,27 +210,44 @@ public: static status_t setPhoneState(audio_mode_t state); static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config); static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage); + + // Client must successfully hand off the handle reference to AudioFlinger via createTrack(), + // or release it with releaseOutput(). static audio_io_handle_t getOutput(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); + static audio_io_handle_t getOutputForAttr(const audio_attributes_t *attr, + 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); static status_t startOutput(audio_io_handle_t output, audio_stream_type_t stream, - int session = 0); + int session); static status_t stopOutput(audio_io_handle_t output, audio_stream_type_t stream, - int session = 0); + int session); static void releaseOutput(audio_io_handle_t output); + + // Client must successfully hand off the handle reference to AudioFlinger via openRecord(), + // or release it with releaseInput(). static audio_io_handle_t getInput(audio_source_t inputSource, - uint32_t samplingRate = 0, - audio_format_t format = AUDIO_FORMAT_DEFAULT, - audio_channel_mask_t channelMask = AUDIO_CHANNEL_IN_MONO, - int sessionId = 0); - static status_t startInput(audio_io_handle_t input); - static status_t stopInput(audio_io_handle_t input); - static void releaseInput(audio_io_handle_t input); + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + int sessionId, + audio_input_flags_t); + + static status_t startInput(audio_io_handle_t input, + audio_session_t session); + static status_t stopInput(audio_io_handle_t input, + audio_session_t session); + static void releaseInput(audio_io_handle_t input, + audio_session_t session); static status_t initStreamVolume(audio_stream_type_t stream, int indexMin, int indexMax); @@ -255,8 +289,54 @@ public: // check presence of audio flinger service. // returns NO_ERROR if binding to service succeeds, DEAD_OBJECT otherwise static status_t checkAudioFlinger(); + + /* List available audio ports and their attributes */ + static status_t listAudioPorts(audio_port_role_t role, + audio_port_type_t type, + unsigned int *num_ports, + struct audio_port *ports, + unsigned int *generation); + + /* Get attributes for a given audio port */ + static status_t getAudioPort(struct audio_port *port); + + /* Create an audio patch between several source and sink ports */ + static status_t createAudioPatch(const struct audio_patch *patch, + audio_patch_handle_t *handle); + + /* Release an audio patch */ + static status_t releaseAudioPatch(audio_patch_handle_t handle); + + /* List existing audio patches */ + static status_t listAudioPatches(unsigned int *num_patches, + struct audio_patch *patches, + unsigned int *generation); + /* Set audio port configuration */ + static status_t setAudioPortConfig(const struct audio_port_config *config); + + + static status_t acquireSoundTriggerSession(audio_session_t *session, + audio_io_handle_t *ioHandle, + audio_devices_t *device); + static status_t releaseSoundTriggerSession(audio_session_t session); + // ---------------------------------------------------------------------------- + class AudioPortCallback : public RefBase + { + public: + + AudioPortCallback() {} + virtual ~AudioPortCallback() {} + + virtual void onAudioPortListUpdate() = 0; + virtual void onAudioPatchListUpdate() = 0; + virtual void onServiceDied() = 0; + + }; + + static void setAudioPortCallback(sp<AudioPortCallback> callBack); + private: class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient @@ -275,7 +355,8 @@ private: virtual void ioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2); }; - class AudioPolicyServiceClient: public IBinder::DeathRecipient + class AudioPolicyServiceClient: public IBinder::DeathRecipient, + public BnAudioPolicyServiceClient { public: AudioPolicyServiceClient() { @@ -283,6 +364,10 @@ private: // DeathRecipient virtual void binderDied(const wp<IBinder>& who); + + // IAudioPolicyServiceClient + virtual void onAudioPortListUpdate(); + virtual void onAudioPatchListUpdate(); }; static sp<AudioFlingerClient> gAudioFlingerClient; @@ -302,11 +387,11 @@ private: static sp<IAudioPolicyService> gAudioPolicyService; - // mapping between stream types and outputs - static DefaultKeyedVector<audio_stream_type_t, audio_io_handle_t> gStreamOutputMap; // 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 |