584c285492
Add an implementation of get_next_write_timestamp. Change-Id: Ie8e19307bafdb057d06bf3c536380c5740f830a6
281 lines
10 KiB
C++
281 lines
10 KiB
C++
/*
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* Copyright (C) 2007 The Android Open Source Project
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*/
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#ifndef ANDROID_AUDIO_HARDWARE_INTERFACE_H
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#define ANDROID_AUDIO_HARDWARE_INTERFACE_H
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#include <stdint.h>
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#include <sys/types.h>
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#include <utils/Errors.h>
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#include <utils/Vector.h>
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#include <utils/String16.h>
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#include <utils/String8.h>
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#include <media/IAudioFlinger.h>
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#include <hardware_legacy/AudioSystemLegacy.h>
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#include <system/audio.h>
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#include <hardware/audio.h>
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#include <cutils/bitops.h>
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namespace android_audio_legacy {
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using android::Vector;
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using android::String16;
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using android::String8;
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// ----------------------------------------------------------------------------
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/**
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* AudioStreamOut is the abstraction interface for the audio output hardware.
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*
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* It provides information about various properties of the audio output hardware driver.
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*/
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class AudioStreamOut {
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public:
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virtual ~AudioStreamOut() = 0;
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/** return audio sampling rate in hz - eg. 44100 */
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virtual uint32_t sampleRate() const = 0;
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/** returns size of output buffer - eg. 4800 */
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virtual size_t bufferSize() const = 0;
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/**
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* returns the output channel mask
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*/
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virtual uint32_t channels() const = 0;
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/**
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* return audio format in 8bit or 16bit PCM format -
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* eg. AudioSystem:PCM_16_BIT
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*/
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virtual int format() const = 0;
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/**
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* return the frame size (number of bytes per sample).
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*/
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uint32_t frameSize() const { return popcount(channels())*((format()==AUDIO_FORMAT_PCM_16_BIT)?sizeof(int16_t):sizeof(int8_t)); }
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/**
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* return the audio hardware driver latency in milli seconds.
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*/
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virtual uint32_t latency() const = 0;
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/**
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* Use this method in situations where audio mixing is done in the
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* hardware. This method serves as a direct interface with hardware,
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* allowing you to directly set the volume as apposed to via the framework.
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* This method might produce multiple PCM outputs or hardware accelerated
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* codecs, such as MP3 or AAC.
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*/
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virtual status_t setVolume(float left, float right) = 0;
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/** write audio buffer to driver. Returns number of bytes written */
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virtual ssize_t write(const void* buffer, size_t bytes) = 0;
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/**
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* Put the audio hardware output into standby mode. Returns
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* status based on include/utils/Errors.h
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*/
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virtual status_t standby() = 0;
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/** dump the state of the audio output device */
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virtual status_t dump(int fd, const Vector<String16>& args) = 0;
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// set/get audio output parameters. The function accepts a list of parameters
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// key value pairs in the form: key1=value1;key2=value2;...
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// Some keys are reserved for standard parameters (See AudioParameter class).
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// If the implementation does not accept a parameter change while the output is
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// active but the parameter is acceptable otherwise, it must return INVALID_OPERATION.
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// The audio flinger will put the output in standby and then change the parameter value.
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virtual status_t setParameters(const String8& keyValuePairs) = 0;
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virtual String8 getParameters(const String8& keys) = 0;
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// return the number of audio frames written by the audio dsp to DAC since
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// the output has exited standby
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virtual status_t getRenderPosition(uint32_t *dspFrames) = 0;
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/**
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* get the local time at which the next write to the audio driver will be
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* presented
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*/
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virtual status_t getNextWriteTimestamp(int64_t *timestamp);
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};
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/**
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* AudioStreamIn is the abstraction interface for the audio input hardware.
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*
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* It defines the various properties of the audio hardware input driver.
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*/
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class AudioStreamIn {
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public:
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virtual ~AudioStreamIn() = 0;
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/** return audio sampling rate in hz - eg. 44100 */
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virtual uint32_t sampleRate() const = 0;
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/** return the input buffer size allowed by audio driver */
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virtual size_t bufferSize() const = 0;
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/** return input channel mask */
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virtual uint32_t channels() const = 0;
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/**
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* return audio format in 8bit or 16bit PCM format -
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* eg. AudioSystem:PCM_16_BIT
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*/
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virtual int format() const = 0;
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/**
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* return the frame size (number of bytes per sample).
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*/
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uint32_t frameSize() const { return AudioSystem::popCount(channels())*((format()==AudioSystem::PCM_16_BIT)?sizeof(int16_t):sizeof(int8_t)); }
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/** set the input gain for the audio driver. This method is for
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* for future use */
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virtual status_t setGain(float gain) = 0;
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/** read audio buffer in from audio driver */
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virtual ssize_t read(void* buffer, ssize_t bytes) = 0;
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/** dump the state of the audio input device */
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virtual status_t dump(int fd, const Vector<String16>& args) = 0;
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/**
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* Put the audio hardware input into standby mode. Returns
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* status based on include/utils/Errors.h
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*/
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virtual status_t standby() = 0;
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// set/get audio input parameters. The function accepts a list of parameters
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// key value pairs in the form: key1=value1;key2=value2;...
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// Some keys are reserved for standard parameters (See AudioParameter class).
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// If the implementation does not accept a parameter change while the output is
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// active but the parameter is acceptable otherwise, it must return INVALID_OPERATION.
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// The audio flinger will put the input in standby and then change the parameter value.
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virtual status_t setParameters(const String8& keyValuePairs) = 0;
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virtual String8 getParameters(const String8& keys) = 0;
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// Return the number of input frames lost in the audio driver since the last call of this function.
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// Audio driver is expected to reset the value to 0 and restart counting upon returning the current value by this function call.
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// Such loss typically occurs when the user space process is blocked longer than the capacity of audio driver buffers.
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// Unit: the number of input audio frames
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virtual unsigned int getInputFramesLost() const = 0;
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virtual status_t addAudioEffect(effect_handle_t effect) = 0;
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virtual status_t removeAudioEffect(effect_handle_t effect) = 0;
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};
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/**
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* AudioHardwareInterface.h defines the interface to the audio hardware abstraction layer.
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*
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* The interface supports setting and getting parameters, selecting audio routing
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* paths, and defining input and output streams.
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*
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* AudioFlinger initializes the audio hardware and immediately opens an output stream.
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* You can set Audio routing to output to handset, speaker, Bluetooth, or a headset.
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*
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* The audio input stream is initialized when AudioFlinger is called to carry out
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* a record operation.
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*/
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class AudioHardwareInterface
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{
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public:
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virtual ~AudioHardwareInterface() {}
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/**
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* check to see if the audio hardware interface has been initialized.
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* return status based on values defined in include/utils/Errors.h
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*/
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virtual status_t initCheck() = 0;
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/** set the audio volume of a voice call. Range is between 0.0 and 1.0 */
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virtual status_t setVoiceVolume(float volume) = 0;
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/**
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* set the audio volume for all audio activities other than voice call.
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* Range between 0.0 and 1.0. If any value other than NO_ERROR is returned,
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* the software mixer will emulate this capability.
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*/
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virtual status_t setMasterVolume(float volume) = 0;
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/**
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* Get the current master volume value for the HAL, if the HAL supports
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* master volume control. AudioFlinger will query this value from the
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* primary audio HAL when the service starts and use the value for setting
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* the initial master volume across all HALs.
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*/
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virtual status_t getMasterVolume(float *volume) = 0;
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/**
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* setMode is called when the audio mode changes. NORMAL mode is for
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* standard audio playback, RINGTONE when a ringtone is playing, and IN_CALL
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* when a call is in progress.
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*/
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virtual status_t setMode(int mode) = 0;
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// mic mute
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virtual status_t setMicMute(bool state) = 0;
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virtual status_t getMicMute(bool* state) = 0;
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// set/get global audio parameters
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virtual status_t setParameters(const String8& keyValuePairs) = 0;
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virtual String8 getParameters(const String8& keys) = 0;
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// Returns audio input buffer size according to parameters passed or 0 if one of the
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// parameters is not supported
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virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount) = 0;
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/** This method creates and opens the audio hardware output stream */
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virtual AudioStreamOut* openOutputStream(
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uint32_t devices,
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int *format=0,
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uint32_t *channels=0,
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uint32_t *sampleRate=0,
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status_t *status=0) = 0;
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virtual void closeOutputStream(AudioStreamOut* out) = 0;
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/** This method creates and opens the audio hardware input stream */
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virtual AudioStreamIn* openInputStream(
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uint32_t devices,
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int *format,
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uint32_t *channels,
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uint32_t *sampleRate,
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status_t *status,
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AudioSystem::audio_in_acoustics acoustics) = 0;
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virtual void closeInputStream(AudioStreamIn* in) = 0;
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/**This method dumps the state of the audio hardware */
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virtual status_t dumpState(int fd, const Vector<String16>& args) = 0;
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static AudioHardwareInterface* create();
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protected:
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virtual status_t dump(int fd, const Vector<String16>& args) = 0;
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};
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// ----------------------------------------------------------------------------
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extern "C" AudioHardwareInterface* createAudioHardware(void);
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}; // namespace android
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#endif // ANDROID_AUDIO_HARDWARE_INTERFACE_H
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