/*
* 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 <utils/RefBase.h>
#include <utils/threads.h>
#include <media/IAudioFlinger.h>
#include <system/audio.h>
#include <system/audio_policy.h>
/* XXX: Should be include by all the users instead */
#include <media/AudioParameter.h>
namespace android {
typedef void (*audio_error_callback)(status_t err);
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 = 0);
// 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);
// 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).
static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
static String8 getParameters(audio_io_handle_t ioHandle, const String8& keys);
static void setErrorCallback(audio_error_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);
static status_t getOutputSamplingRate(uint32_t* samplingRate,
audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
static status_t getOutputFrameCount(size_t* frameCount,
audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
static status_t getOutputLatency(uint32_t* latency,
audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
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()
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);
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.
// 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(size_t *halFrames, size_t *dspFrames,
audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
// 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 int newAudioSessionId();
static void acquireAudioSessionId(int audioSession);
static void releaseAudioSessionId(int audioSession);
// 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), channels(0), frameCount(0), latency(0) {}
uint32_t samplingRate;
int32_t format;
int32_t channels;
size_t frameCount;
uint32_t latency;
};
// 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);
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);
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);
static status_t startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
int session = 0);
static status_t stopOutput(audio_io_handle_t output,
audio_stream_type_t stream,
int session = 0);
static void releaseOutput(audio_io_handle_t output);
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);
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();
// ----------------------------------------------------------------------------
private:
class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient
{
public:
AudioFlingerClient() {
}
// 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(int event, audio_io_handle_t ioHandle, const void *param2);
};
class AudioPolicyServiceClient: public IBinder::DeathRecipient
{
public:
AudioPolicyServiceClient() {
}
// DeathRecipient
virtual void binderDied(const wp<IBinder>& who);
};
static sp<AudioFlingerClient> gAudioFlingerClient;
static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient;
friend class AudioFlingerClient;
friend class AudioPolicyServiceClient;
static Mutex gLock;
static sp<IAudioFlinger> gAudioFlinger;
static audio_error_callback gAudioErrorCallback;
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;
// 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;
};
}; // namespace android
#endif /*ANDROID_AUDIOSYSTEM_H_*/