/* * Copyright (C) 2007 The Android Open Source Project * * Licensed under the Apache License, Version 2.0 (the "License"); * you may not use this file except in compliance with the License. * You may obtain a copy of the License at * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. * See the License for the specific language governing permissions and * limitations under the License. */ #ifndef ANDROID_IAUDIOFLINGER_H #define ANDROID_IAUDIOFLINGER_H #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include namespace android { // ---------------------------------------------------------------------------- class IAudioFlinger : public IInterface { public: DECLARE_META_INTERFACE(AudioFlinger); // or-able bits shared by createTrack and openRecord, but not all combinations make sense enum { TRACK_DEFAULT = 0, // client requests a default AudioTrack // FIXME: obsolete // TRACK_TIMED= 1, // client requests a TimedAudioTrack TRACK_FAST = 2, // client requests a fast AudioTrack or AudioRecord TRACK_OFFLOAD = 4, // client requests offload to hw codec TRACK_DIRECT = 8, // client requests a direct output }; typedef uint32_t track_flags_t; // invariant on exit for all APIs that return an sp<>: // (return value != 0) == (*status == NO_ERROR) /* create an audio track and registers it with AudioFlinger. * return null if the track cannot be created. */ virtual sp createTrack( audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t *pFrameCount, track_flags_t *flags, const sp& sharedBuffer, // On successful return, AudioFlinger takes over the handle // reference and will release it when the track is destroyed. // However on failure, the client is responsible for release. audio_io_handle_t output, pid_t pid, pid_t tid, // -1 means unused, otherwise must be valid non-0 audio_session_t *sessionId, int clientUid, status_t *status) = 0; virtual sp openRecord( // On successful return, AudioFlinger takes over the handle // reference and will release it when the track is destroyed. // However on failure, the client is responsible for release. audio_io_handle_t input, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, const String16& callingPackage, size_t *pFrameCount, track_flags_t *flags, pid_t pid, pid_t tid, // -1 means unused, otherwise must be valid non-0 int clientUid, audio_session_t *sessionId, size_t *notificationFrames, sp& cblk, sp& buffers, // return value 0 means it follows cblk status_t *status) = 0; // FIXME Surprisingly, format/latency don't work for input handles /* query the audio hardware state. This state never changes, * and therefore can be cached. */ virtual uint32_t sampleRate(audio_io_handle_t ioHandle) const = 0; // reserved; formerly channelCount() virtual audio_format_t format(audio_io_handle_t output) const = 0; virtual size_t frameCount(audio_io_handle_t ioHandle) const = 0; // return estimated latency in milliseconds virtual uint32_t latency(audio_io_handle_t output) const = 0; /* set/get the audio hardware state. This will probably be used by * the preference panel, mostly. */ virtual status_t setMasterVolume(float value) = 0; virtual status_t setMasterMute(bool muted) = 0; virtual float masterVolume() const = 0; virtual bool masterMute() const = 0; /* set/get stream type state. This will probably be used by * the preference panel, mostly. */ virtual status_t setStreamVolume(audio_stream_type_t stream, float value, audio_io_handle_t output) = 0; virtual status_t setStreamMute(audio_stream_type_t stream, bool muted) = 0; virtual float streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const = 0; virtual bool streamMute(audio_stream_type_t stream) const = 0; // set audio mode virtual status_t setMode(audio_mode_t mode) = 0; // mic mute/state virtual status_t setMicMute(bool state) = 0; virtual bool getMicMute() const = 0; virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) = 0; virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const = 0; // Register an object to receive audio input/output change and track notifications. // For a given calling pid, AudioFlinger disregards any registrations after the first. // Thus the IAudioFlingerClient must be a singleton per process. virtual void registerClient(const sp& client) = 0; // retrieve the audio recording buffer size // FIXME This API assumes a route, and so should be deprecated. virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask) const = 0; virtual status_t openOutput(audio_module_handle_t module, audio_io_handle_t *output, audio_config_t *config, audio_devices_t *devices, const String8& address, uint32_t *latencyMs, audio_output_flags_t flags) = 0; virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, audio_io_handle_t output2) = 0; virtual status_t closeOutput(audio_io_handle_t output) = 0; virtual status_t suspendOutput(audio_io_handle_t output) = 0; virtual status_t restoreOutput(audio_io_handle_t output) = 0; virtual status_t openInput(audio_module_handle_t module, audio_io_handle_t *input, audio_config_t *config, audio_devices_t *device, const String8& address, audio_source_t source, audio_input_flags_t flags) = 0; virtual status_t closeInput(audio_io_handle_t input) = 0; virtual status_t invalidateStream(audio_stream_type_t stream) = 0; virtual status_t setVoiceVolume(float volume) = 0; virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, audio_io_handle_t output) const = 0; virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const = 0; virtual audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use) = 0; virtual void acquireAudioSessionId(audio_session_t audioSession, pid_t pid) = 0; virtual void releaseAudioSessionId(audio_session_t audioSession, pid_t pid) = 0; virtual status_t queryNumberEffects(uint32_t *numEffects) const = 0; virtual status_t queryEffect(uint32_t index, effect_descriptor_t *pDescriptor) const = 0; virtual status_t getEffectDescriptor(const effect_uuid_t *pEffectUUID, effect_descriptor_t *pDescriptor) const = 0; virtual sp createEffect( effect_descriptor_t *pDesc, const sp& client, int32_t priority, // AudioFlinger doesn't take over handle reference from client audio_io_handle_t output, audio_session_t sessionId, const String16& callingPackage, status_t *status, int *id, int *enabled) = 0; virtual status_t moveEffects(audio_session_t session, audio_io_handle_t srcOutput, audio_io_handle_t dstOutput) = 0; virtual audio_module_handle_t loadHwModule(const char *name) = 0; // helpers for android.media.AudioManager.getProperty(), see description there for meaning // FIXME move these APIs to AudioPolicy to permit a more accurate implementation // that looks on primary device for a stream with fast flag, primary flag, or first one. virtual uint32_t getPrimaryOutputSamplingRate() = 0; virtual size_t getPrimaryOutputFrameCount() = 0; // Intended for AudioService to inform AudioFlinger of device's low RAM attribute, // and should be called at most once. For a definition of what "low RAM" means, see // android.app.ActivityManager.isLowRamDevice(). virtual status_t setLowRamDevice(bool isLowRamDevice) = 0; /* List available audio ports and their attributes */ virtual status_t listAudioPorts(unsigned int *num_ports, struct audio_port *ports) = 0; /* Get attributes for a given audio port */ virtual status_t getAudioPort(struct audio_port *port) = 0; /* Create an audio patch between several source and sink ports */ virtual status_t createAudioPatch(const struct audio_patch *patch, audio_patch_handle_t *handle) = 0; /* Release an audio patch */ virtual status_t releaseAudioPatch(audio_patch_handle_t handle) = 0; /* List existing audio patches */ virtual status_t listAudioPatches(unsigned int *num_patches, struct audio_patch *patches) = 0; /* Set audio port configuration */ virtual status_t setAudioPortConfig(const struct audio_port_config *config) = 0; /* Get the HW synchronization source used for an audio session */ virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId) = 0; /* Indicate JAVA services are ready (scheduling, power management ...) */ virtual status_t systemReady() = 0; // Returns the number of frames per audio HAL buffer. virtual size_t frameCountHAL(audio_io_handle_t ioHandle) const = 0; }; // ---------------------------------------------------------------------------- class BnAudioFlinger : public BnInterface { public: virtual status_t onTransact( uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags = 0); }; // ---------------------------------------------------------------------------- }; // namespace android #endif // ANDROID_IAUDIOFLINGER_H