/*
* 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 <stdint.h>
#include <sys/types.h>
#include <unistd.h>
#include <utils/RefBase.h>
#include <utils/Errors.h>
#include <binder/IInterface.h>
#include <media/IAudioTrack.h>
#include <media/IAudioRecord.h>
#include <media/IAudioFlingerClient.h>
#include <system/audio.h>
#include <system/audio_effect.h>
#include <system/audio_policy.h>
#include <media/IEffect.h>
#include <media/IEffectClient.h>
#include <utils/String8.h>
namespace android {
// ----------------------------------------------------------------------------
class IAudioFlinger : public IInterface
{
public:
DECLARE_META_INTERFACE(AudioFlinger);
// 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<IAudioTrack> createTrack(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t *pFrameCount,
audio_output_flags_t *flags,
const sp<IMemory>& 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,
audio_port_handle_t portId) = 0;
virtual sp<IAudioRecord> 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,
audio_input_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<IMemory>& cblk,
sp<IMemory>& buffers, // return value 0 means it follows cblk
status_t *status,
audio_port_handle_t portId) = 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<IAudioFlingerClient>& 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<IEffect> createEffect(
effect_descriptor_t *pDesc,
const sp<IEffectClient>& 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,
pid_t pid,
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<IAudioFlinger>
{
public:
virtual status_t onTransact( uint32_t code,
const Parcel& data,
Parcel* reply,
uint32_t flags = 0);
// Requests media.log to start merging log buffers
virtual void requestLogMerge() = 0;
};
// ----------------------------------------------------------------------------
}; // namespace android
#endif // ANDROID_IAUDIOFLINGER_H