/*
* 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_AUDIO_HARDWARE_INTERFACE_H
#define ANDROID_AUDIO_HARDWARE_INTERFACE_H
#include <stdint.h>
#include <sys/types.h>
#include <utils/Errors.h>
#include <utils/Vector.h>
#include <utils/String16.h>
#include <utils/String8.h>
#include <media/IAudioFlinger.h>
#include <hardware_legacy/AudioSystemLegacy.h>
#include <system/audio.h>
#include <hardware/audio.h>
#include <cutils/bitops.h>
namespace android_audio_legacy {
using android::Vector;
using android::String16;
using android::String8;
// ----------------------------------------------------------------------------
/**
* AudioStreamOut is the abstraction interface for the audio output hardware.
*
* It provides information about various properties of the audio output hardware driver.
*/
class AudioStreamOut {
public:
virtual ~AudioStreamOut() = 0;
/** return audio sampling rate in hz - eg. 44100 */
virtual uint32_t sampleRate() const = 0;
/** returns size of output buffer - eg. 4800 */
virtual size_t bufferSize() const = 0;
/**
* returns the output channel mask
*/
virtual uint32_t channels() const = 0;
/**
* return audio format in 8bit or 16bit PCM format -
* eg. AudioSystem:PCM_16_BIT
*/
virtual int format() const = 0;
/**
* return the frame size (number of bytes per sample).
*/
uint32_t frameSize() const { return audio_channel_count_from_out_mask(channels())*
((format()==AUDIO_FORMAT_PCM_16_BIT)?sizeof(int16_t):sizeof(int8_t)); }
/**
* return the audio hardware driver latency in milli seconds.
*/
virtual uint32_t latency() const = 0;
/**
* Use this method in situations where audio mixing is done in the
* hardware. This method serves as a direct interface with hardware,
* allowing you to directly set the volume as apposed to via the framework.
* This method might produce multiple PCM outputs or hardware accelerated
* codecs, such as MP3 or AAC.
*/
virtual status_t setVolume(float left, float right) = 0;
/** write audio buffer to driver. Returns number of bytes written */
virtual ssize_t write(const void* buffer, size_t bytes) = 0;
/**
* Put the audio hardware output into standby mode. Returns
* status based on include/utils/Errors.h
*/
virtual status_t standby() = 0;
/** dump the state of the audio output device */
virtual status_t dump(int fd, const Vector<String16>& args) = 0;
// set/get audio output 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).
// If the implementation does not accept a parameter change while the output is
// active but the parameter is acceptable otherwise, it must return INVALID_OPERATION.
// The audio flinger will put the output in standby and then change the parameter value.
virtual status_t setParameters(const String8& keyValuePairs) = 0;
virtual String8 getParameters(const String8& keys) = 0;
// return the number of audio frames written by the audio dsp to DAC since
// the output has exited standby
virtual status_t getRenderPosition(uint32_t *dspFrames) = 0;
/**
* get the local time at which the next write to the audio driver will be
* presented
*/
virtual status_t getNextWriteTimestamp(int64_t *timestamp);
/**
* Return a recent count of the number of audio frames presented to an external observer.
*/
virtual status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp);
};
/**
* AudioStreamIn is the abstraction interface for the audio input hardware.
*
* It defines the various properties of the audio hardware input driver.
*/
class AudioStreamIn {
public:
virtual ~AudioStreamIn() = 0;
/** return audio sampling rate in hz - eg. 44100 */
virtual uint32_t sampleRate() const = 0;
/** return the input buffer size allowed by audio driver */
virtual size_t bufferSize() const = 0;
/** return input channel mask */
virtual uint32_t channels() const = 0;
/**
* return audio format in 8bit or 16bit PCM format -
* eg. AudioSystem:PCM_16_BIT
*/
virtual int format() const = 0;
/**
* return the frame size (number of bytes per sample).
*/
uint32_t frameSize() const { return audio_channel_count_from_in_mask(channels())*
((format()==AudioSystem::PCM_16_BIT)?sizeof(int16_t):sizeof(int8_t)); }
/** set the input gain for the audio driver. This method is for
* for future use */
virtual status_t setGain(float gain) = 0;
/** read audio buffer in from audio driver */
virtual ssize_t read(void* buffer, ssize_t bytes) = 0;
/** dump the state of the audio input device */
virtual status_t dump(int fd, const Vector<String16>& args) = 0;
/**
* Put the audio hardware input into standby mode. Returns
* status based on include/utils/Errors.h
*/
virtual status_t standby() = 0;
// set/get audio input 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).
// If the implementation does not accept a parameter change while the output is
// active but the parameter is acceptable otherwise, it must return INVALID_OPERATION.
// The audio flinger will put the input in standby and then change the parameter value.
virtual status_t setParameters(const String8& keyValuePairs) = 0;
virtual String8 getParameters(const String8& keys) = 0;
// Return the number of input frames lost in the audio driver since the last call of this function.
// Audio driver is expected to reset the value to 0 and restart counting upon returning the current value by this function call.
// Such loss typically occurs when the user space process is blocked longer than the capacity of audio driver buffers.
// Unit: the number of input audio frames
virtual unsigned int getInputFramesLost() const = 0;
virtual status_t addAudioEffect(effect_handle_t effect) = 0;
virtual status_t removeAudioEffect(effect_handle_t effect) = 0;
};
/**
* AudioHardwareInterface.h defines the interface to the audio hardware abstraction layer.
*
* The interface supports setting and getting parameters, selecting audio routing
* paths, and defining input and output streams.
*
* AudioFlinger initializes the audio hardware and immediately opens an output stream.
* You can set Audio routing to output to handset, speaker, Bluetooth, or a headset.
*
* The audio input stream is initialized when AudioFlinger is called to carry out
* a record operation.
*/
class AudioHardwareInterface
{
public:
virtual ~AudioHardwareInterface() {}
/**
* check to see if the audio hardware interface has been initialized.
* return status based on values defined in include/utils/Errors.h
*/
virtual status_t initCheck() = 0;
/** set the audio volume of a voice call. Range is between 0.0 and 1.0 */
virtual status_t setVoiceVolume(float volume) = 0;
/**
* set the audio volume for all audio activities other than voice call.
* Range between 0.0 and 1.0. If any value other than NO_ERROR is returned,
* the software mixer will emulate this capability.
*/
virtual status_t setMasterVolume(float volume) = 0;
/**
* Get the current master volume value for the HAL, if the HAL supports
* master volume control. AudioFlinger will query this value from the
* primary audio HAL when the service starts and use the value for setting
* the initial master volume across all HALs.
*/
virtual status_t getMasterVolume(float *volume) = 0;
/**
* setMode is called when the audio mode changes. NORMAL mode is for
* standard audio playback, RINGTONE when a ringtone is playing, and IN_CALL
* when a call is in progress.
*/
virtual status_t setMode(int mode) = 0;
// mic mute
virtual status_t setMicMute(bool state) = 0;
virtual status_t getMicMute(bool* state) = 0;
// set/get global audio parameters
virtual status_t setParameters(const String8& keyValuePairs) = 0;
virtual String8 getParameters(const String8& keys) = 0;
// Returns audio input buffer size according to parameters passed or 0 if one of the
// parameters is not supported
virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount) = 0;
/** This method creates and opens the audio hardware output stream */
virtual AudioStreamOut* openOutputStream(
uint32_t devices,
int *format=0,
uint32_t *channels=0,
uint32_t *sampleRate=0,
status_t *status=0) = 0;
virtual AudioStreamOut* openOutputStreamWithFlags(
uint32_t devices,
audio_output_flags_t flags=(audio_output_flags_t)0,
int *format=0,
uint32_t *channels=0,
uint32_t *sampleRate=0,
status_t *status=0) = 0;
virtual void closeOutputStream(AudioStreamOut* out) = 0;
/** This method creates and opens the audio hardware input stream */
virtual AudioStreamIn* openInputStream(
uint32_t devices,
int *format,
uint32_t *channels,
uint32_t *sampleRate,
status_t *status,
AudioSystem::audio_in_acoustics acoustics) = 0;
virtual void closeInputStream(AudioStreamIn* in) = 0;
/**This method dumps the state of the audio hardware */
virtual status_t dumpState(int fd, const Vector<String16>& args) = 0;
virtual status_t setMasterMute(bool muted) = 0;
static AudioHardwareInterface* create();
virtual int createAudioPatch(unsigned int num_sources,
const struct audio_port_config *sources,
unsigned int num_sinks,
const struct audio_port_config *sinks,
audio_patch_handle_t *handle) = 0;
virtual int releaseAudioPatch(audio_patch_handle_t handle) = 0;
virtual int getAudioPort(struct audio_port *port) = 0;
virtual int setAudioPortConfig(const struct audio_port_config *config) = 0;
protected:
virtual status_t dump(int fd, const Vector<String16>& args) = 0;
};
// ----------------------------------------------------------------------------
extern "C" AudioHardwareInterface* createAudioHardware(void);
}; // namespace android
#endif // ANDROID_AUDIO_HARDWARE_INTERFACE_H