/*
* Copyright (C) 2008 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_AUDIOSYSTEM_H_
#define ANDROID_AUDIOSYSTEM_H_
#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>
#include <system/audio_policy.h>
#include <utils/Errors.h>
#include <utils/Mutex.h>
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;
class String8;
class AudioSystem
{
public:
/* These are static methods to control the system-wide AudioFlinger
* only privileged processes can have access to them
*/
// mute/unmute microphone
static status_t muteMicrophone(bool state);
static status_t isMicrophoneMuted(bool *state);
// set/get master volume
static status_t setMasterVolume(float value);
static status_t getMasterVolume(float* volume);
// mute/unmute audio outputs
static status_t setMasterMute(bool mute);
static status_t getMasterMute(bool* mute);
// set/get stream volume on specified output
static status_t setStreamVolume(audio_stream_type_t stream, float value,
audio_io_handle_t output);
static status_t getStreamVolume(audio_stream_type_t stream, float* volume,
audio_io_handle_t output);
// mute/unmute stream
static status_t setStreamMute(audio_stream_type_t stream, bool mute);
static status_t getStreamMute(audio_stream_type_t stream, bool* mute);
// set audio mode in audio hardware
static status_t setMode(audio_mode_t mode);
// 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);
// 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);
// 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);
static void setDynPolicyCallback(dynamic_policy_callback cb);
// helper function to obtain AudioFlinger service handle
static const sp<IAudioFlinger> get_audio_flinger();
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
// 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,
uint32_t* samplingRate);
// returns the number of frames per audio HAL write buffer. Corresponds to
// 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 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 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
// - BAD_VALUE: invalid parameter
// NOTE: this feature is not supported on all hardware platforms and it is
// 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);
// return the number of input frames lost by HAL implementation, or 0 if the handle is invalid
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 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);
// 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
// playback is complete on another audio session.
// See definitions in MediaSyncEvent.java
enum sync_event_t {
SYNC_EVENT_SAME = -1, // used internally to indicate restart with same event
SYNC_EVENT_NONE = 0,
SYNC_EVENT_PRESENTATION_COMPLETE,
//
// Define new events here: SYNC_EVENT_START, SYNC_EVENT_STOP, SYNC_EVENT_TIME ...
//
SYNC_EVENT_CNT,
};
// Timeout for synchronous record start. Prevents from blocking the record thread forever
// if the trigger event is not fired.
static const uint32_t kSyncRecordStartTimeOutMs = 30000;
//
// 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_name);
static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
const char *device_address);
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 status_t getOutputForAttr(const audio_attributes_t *attr,
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);
static status_t stopOutput(audio_io_handle_t output,
audio_stream_type_t stream,
audio_session_t session);
static void releaseOutput(audio_io_handle_t output,
audio_stream_type_t stream,
audio_session_t session);
// Client must successfully hand off the handle reference to AudioFlinger via openRecord(),
// or release it with releaseInput().
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_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE);
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);
static status_t setStreamVolumeIndex(audio_stream_type_t stream,
int index,
audio_devices_t device);
static status_t getStreamVolumeIndex(audio_stream_type_t stream,
int *index,
audio_devices_t device);
static uint32_t getStrategyForStream(audio_stream_type_t stream);
static audio_devices_t getDevicesForStream(audio_stream_type_t stream);
static audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc);
static status_t registerEffect(const effect_descriptor_t *desc,
audio_io_handle_t io,
uint32_t strategy,
int session,
int id);
static status_t unregisterEffect(int id);
static status_t setEffectEnabled(int id, bool enabled);
// clear stream to output mapping cache (gStreamOutputMap)
// and output configuration cache (gOutputs)
static void clearAudioConfigCache();
static const sp<IAudioPolicyService> get_audio_policy_service();
// helpers for android.media.AudioManager.getProperty(), see description there for meaning
static uint32_t getPrimaryOutputSamplingRate();
static size_t getPrimaryOutputFrameCount();
static status_t setLowRamDevice(bool isLowRamDevice);
// Check if hw offload is possible for given format, stream type, sample rate,
// bit rate, duration, video and streaming or offload property is enabled
static bool isOffloadSupported(const audio_offload_info_t& info);
// 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);
static audio_mode_t getPhoneState();
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
{
public:
AudioPortCallback() {}
virtual ~AudioPortCallback() {}
virtual void onAudioPortListUpdate() = 0;
virtual void onAudioPatchListUpdate() = 0;
virtual void onServiceDied() = 0;
};
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() :
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);
// IAudioFlingerClient
// 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(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;
sp<AudioIoDescriptor> getIoDescriptor_l(audio_io_handle_t ioHandle);
};
class AudioPolicyServiceClient: public IBinder::DeathRecipient,
public BnAudioPolicyServiceClient
{
public:
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 gLockAPS; // protects gAudioPolicyService and gAudioPolicyServiceClient
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
static uint32_t gPrevInSamplingRate;
static audio_format_t gPrevInFormat;
static audio_channel_mask_t gPrevInChannelMask;
static sp<IAudioPolicyService> gAudioPolicyService;
};
}; // namespace android
#endif /*ANDROID_AUDIOSYSTEM_H_*/