【转】Android音频系统相关术语解释说明

【说明】

本文转载自Android官网文章《Audio Terminology》，以备不可翻墙的时候查阅。

【正文】

Audio Terminology

IN THIS DOCUMENT

Generic Terms

Digital Audio
Inter-device interconnect
Intra-device interconnect
Audio Signal Path
Sample Rate Conversion

Android-Specific
Terms

This glossary of audio-related terminology includes widely-used generic terms and Android-specific terms.

Generic Terms

Generic audio-related terms have conventional meanings.

Digital Audio

Digital audio terms relate to handling sound using audio signals encoded in digital form. For details, refer to Digital
Audio.
acousticsStudy of the mechanical properties of sound, such as how the physical placement of transducers (speakers, microphones, etc.) on a device affects perceived audio quality.attenuationMultiplicative factor less than or equal to 1.0, applied to an audio signal to decrease the signal level. Compare to gain.audiophilePerson concerned with a superior music reproduction experience, especially willing to make substantial tradeoffs (expense, component size, room design, etc.) for sound quality. For details, refer to audiophile.bits per sample or bit depthNumber of bits of information per sample.channelSingle stream of audio information, usually corresponding to one location of recording or playback.downmixingDecrease the number of channels, such as from stereo to mono or from 5.1 to stereo. Accomplished by dropping channels, mixing channels, or more advanced signal processing. Simple mixing without attenuation or
limiting has the potential for overflow and clipping. Compare to upmixing.DSDDirect Stream Digital. Proprietary audio encoding based on pulse-density
modulation. While Pulse Code Modulation (PCM) encodes a waveform as a sequence of individual audio samples of multiple bits, DSD encodes a waveform as a sequence of bits at a very high sample rate (without the concept of samples). Both PCM and DSD represent
multiple channels by independent sequences. DSD is better suited to content distribution than as an internal representation for processing as it can be difficult to apply traditional digital signal processing (DSP) algorithms to DSD. DSD is used in Super
Audio CD (SACD) and in DSD over PCM (DoP) for USB. For details, refer to Digital Stream Digital.duckTemporarily reduce the volume of a stream when another stream becomes active. For example, if music is playing when a notification arrives, the music ducks while the notification plays. Compare to mute.FIFOFirst In, First Out. Hardware module or software data structure that implements First
In, First Out queueing of data. In an audio context, the data stored in the queue are typically audio frames. FIFO can be implemented by a circular
buffer.frameSet of samples, one per channel, at a point in time.frames per bufferNumber of frames handed from one module to the next at one time. The audio HAL interface uses the concept of frames per buffer.gainMultiplicative factor greater than or equal to 1.0, applied to an audio signal to increase the signal level. Compare to attenuation.HD audioHigh-Definition audio. Synonym for high-resolution audio (but different than Intel High Definition Audio).HzUnits for sample rate or frame rate.high-resolution audioRepresentation with greater bit-depth and sample rate than CDs (stereo 16-bit PCM at 44.1 kHz) and without lossy data compression. Equivalent to HD audio. For details, refer to high-resolution
audio.latencyTime delay as a signal passes through a system.losslessA lossless data compression algorithm that preserves bit accuracy
across encoding and decoding, where the result of decoding previously encoded data is equivalent to the original data. Examples of lossless audio content distribution formats include CDs,
PCM within WAV, and FLAC.
The authoring process may reduce the bit depth or sample rate from that of the masters; distribution formats that
preserve the resolution and bit accuracy of masters are the subject of high-resolution audio.lossyA lossy data compression algorithm that attempts to preserve the
most important features of media across encoding and decoding where the result of decoding previously encoded data is perceptually similar to the original data but not identical. Examples of lossy audio compression algorithms include MP3 and AAC. As analog
values are from a continuous domain and digital values are discrete, ADC and DAC are lossy conversions with respect to amplitude. See also transparency.monoOne channel.multichannelSee surround sound. In strict terms, stereo is more than one channel and could be considered multichannel; however, such usage is confusing and thus avoided.muteTemporarily force volume to be zero, independent from the usual volume controls.overrunAudible glitch caused by failure to accept supplied data in sufficient time.
For details, refer to buffer underrun. Compare to underrun.panningDirect a signal to a desired position within a stereo or multichannel field.PCMPulse Code Modulation. Most common low-level encoding of digital audio. The audio signal is sampled at a regular interval, called the sample rate, then quantized to discrete values within a particular range
depending on the bit depth. For example, for 16-bit PCM the sample values are integers between -32768 and +32767.rampGradually increase or decrease the level of a particular audio parameter, such as the volume or the strength of an effect. A volume ramp is commonly applied when pausing and resuming music to avoid a hard audible
transition.sampleNumber representing the audio value for a single channel at a point in time.sample rate or frame rateNumber of frames per second. While frame rate is more accurate, sample rate is conventionally used to mean frame rate.sonificationUse of sound to express feedback or information, such as touch sounds and keyboard sounds.stereoTwo channels.stereo wideningEffect applied to a stereo signal to make another stereo signal that sounds fuller and richer. The effect can also be applied to a mono signal, where it is a type of upmixing.surround soundTechniques for increasing the ability of a listener to perceive sound position beyond stereo left and right.transparencyIdeal result of lossy data compression. Lossy data conversion is transparent if it is perceptually indistinguishable from the original by a human subject. For details, refer to Transparency.underrunAudible glitch caused by failure to supply needed data in sufficient time.
For details, refer to buffer underrun. Compare to overrun.upmixingIncrease the number of channels, such as from mono to stereo or from stereo to surround sound. Accomplished by duplication, panning, or more advanced signal processing. Compare to downmixing.virtualizerEffect that attempts to spatialize audio channels, such as trying to simulate more speakers or give the illusion that sound sources have position.volumeLoudness, the subjective strength of an audio signal.

Inter-device interconnect

Inter-device interconnection technologies connect audio and video components between devices and are readily visible at the external connectors. The HAL implementer and end user should be aware of these terms.
BluetoothShort range wireless technology. For details on the audio-related Bluetooth
profiles and Bluetooth protocols, refer to A2DP for
music, SCO for telephony, and Audio/Video
Remote Control Profile (AVRCP).DisplayPortDigital display interface by the Video Electronics Standards Association (VESA).dongleA dongle is a small gadget, especially one that hangs off another device.HDMIHigh-Definition Multimedia Interface. Interface for transferring audio and video data. For mobile devices, a micro-HDMI (type D) or MHL connector is used.Intel HDAIntel High Definition Audio (do not confuse with generic high-definition audio or high-resolution audio). Specification for a front-panel connector. For details, refer to Intel
High Definition Audio.interfaceAn interface converts a signal from one representation to
another. Common interfaces include a USB audio interface and MIDI interface.line levelLine level is the strength of an analog audio signal that passes between
audio components, not transducers.MHLMobile High-Definition Link. Mobile audio/video interface, often over micro-USB connector.phone connectorMini or sub-mini component that connects a device to wired headphones, headset, or line-level amplifier.SlimPortAdapter from micro-USB to HDMI.S/PDIFSony/Philips Digital Interface Format. Interconnect for uncompressed PCM. For details, refer to S/PDIF.
S/PDIF is the consumer grade variant of AES3.ThunderboltMultimedia interface that competes with USB and HDMI for connecting to high-end peripherals. For details, refer to Thunderbolt.TOSLINKTOSLINK is an optical audio cable used with S/PDIF.USBUniversal Serial Bus. For details, refer to USB.

Intra-device interconnect

Intra-device interconnection technologies connect internal audio components within a given device and are not visible without disassembling the device. The HAL implementer may need to be aware of these, but not
the end user. For details on intra-device interconnections, refer to the following articles:

GPIO
I²C, for control channel
I²S, for audio data, simpler than SLIMbus
McASP
SLIMbus
SPI
AC'97
Intel HDA
SoundWire

In ALSA System on Chip (ASoC), these are collectively called Digital
Audio Interfaces (DAI).

Audio Signal Path

Audio signal path terms relate to the signal path that audio data follows from an application to the transducer or vice-versa.
ADCAnalog-to-digital converter. Module that converts an analog signal (continuous in time and amplitude) to a digital signal (discrete in time and amplitude). Conceptually, an ADC consists of a periodic sample-and-hold
followed by a quantizer, although it does not have to be implemented that way. An ADC is usually preceded by a low-pass filter to remove any high frequency components that are not representable using the desired sample rate. For details, refer to Analog-to-digital
converter.APApplication processor. Main general-purpose computer on a mobile device.codecCoder-decoder. Module that encodes and/or decodes an audio signal from one representation to another (typically analog to PCM or PCM to analog). In strict terms, codec is reserved for modules that both
encode and decode but can be used loosely to refer to only one of these. For details, refer to Audio codec.DACDigital-to-analog converter. Module that converts a digital signal (discrete in time and amplitude) to an analog signal (continuous in time and amplitude). Often followed by a low-pass filter to remove high-frequency
components introduced by digital quantization. For details, refer to Digital-to-analog converter.DSPDigital Signal Processor. Optional component typically located after the application processor (for output) or before the application processor (for input). Primary purpose is to off-load the application processor
and provide signal processing features at a lower power cost.PDMPulse-density modulation. Form of modulation used to represent an analog signal by a digital signal, where the relative density of 1s versus 0s indicates the signal level. Commonly used by digital to analog
converters. For details, refer to Pulse-density modulation.PWMPulse-width modulation. Form of modulation used to represent an analog signal by a digital signal, where the relative width of a digital pulse indicates the signal level. Commonly used by analog-to-digital converters.
For details, refer to Pulse-width modulation.transducerConverts variations in physical real-world quantities to electrical signals. In audio, the physical quantity is sound pressure, and the transducers are the loudspeaker and microphone. For details, refer to Transducer.

Sample Rate Conversion

Sample rate conversion terms relate to the process of converting from one sampling rate to another.
downsampleResample, where sink sample rate < source sample rate.Nyquist frequencyMaximum frequency component that can be represented by a discretized signal at 1/2 of a given sample rate. For example, the human hearing range extends to approximately 20 kHz, so a digital audio signal must
have a sample rate of at least 40 kHz to represent that range. In practice, sample rates of 44.1 kHz and 48 kHz are commonly used, with Nyquist frequencies of 22.05 kHz and 24 kHz respectively. For details, refer to Nyquist
frequency and Hearing range.resamplerSynonym for sample rate converter.resamplingProcess of converting sample rate.sample rate converterModule that resamples.sinkOutput of a resampler.sourceInput to a resampler.upsampleResample, where sink sample rate > source sample rate.

Android-Specific Terms

Android-specific terms include terms used only in the Android audio framework and generic terms that have special meaning within Android.
ALSAAdvanced Linux Sound Architecture. An audio framework for Linux that has also influenced other systems. For a generic definition, refer to ALSA.
In Android, ALSA refers to the kernel audio framework and drivers and not to the user-mode API. See also tinyalsa.audio deviceAudio I/O endpoint backed by a HAL implementation.AudioEffectAPI and implementation framework for output (post-processing) effects and input (pre-processing) effects. The API is defined at android.media.audiofx.AudioEffect.AudioFlingerAndroid sound server implementation. AudioFlinger runs within the mediaserver process. For a generic definition, refer to Sound
server.audio focusSet of APIs for managing audio interactions across multiple independent apps. For details, see Managing
Audio Focus and the focus-related methods and constants of android.media.AudioManager.AudioMixerModule in AudioFlinger responsible for combining multiple tracks and applying attenuation (volume) and effects. For a generic definition, refer to Audio
mixing (recorded music) (discusses a mixer as a hardware device or software application, rather than a software module within a system).audio policyService responsible for all actions that require a policy decision to be made first, such as opening a new I/O stream, re-routing after a change, and stream volume management.AudioRecordPrimary low-level client API for receiving data from an audio input device such as a microphone. The data is usually PCM format. The API is defined at android.media.AudioRecord.AudioResamplerModule in AudioFlinger responsible for sample rate conversion.audio sourceAn enumeration of constants that indicates the desired use case for capturing audio input. For details, see audio
source. As of API level 21 and above, audio attributes are preferred.AudioTrackPrimary low-level client API for sending data to an audio output device such as a speaker. The data is usually in PCM format. The API is defined at android.media.AudioTrack.audio_utilsAudio utility library for features such as PCM format conversion, WAV file I/O, and non-blocking
FIFO, which is largely independent of the Android platform.clientUsually an application or app client. However, an AudioFlinger client can be a thread running within the mediaserver system process, such as when playing media decoded by a MediaPlayer object.HALHardware Abstraction Layer. HAL is a generic term in Android; in audio, it is a layer between AudioFlinger and the kernel device driver with a C API (which replaces the C++ libaudio).FastCaptureThread within AudioFlinger that sends audio data to lower latency fast tracks and drives the input device when configured for reduced latency.FastMixerThread within AudioFlinger that receives and mixes audio data from lower latency fast tracks and drives the primary output device when configured for reduced latency.fast trackAudioTrack or AudioRecord client with lower latency but fewer features on some devices and routes.MediaPlayerHigher-level client API than AudioTrack. Plays encoded content or content that includes multimedia audio and video tracks.media.logAudioFlinger debugging feature available in custom builds only. Used for logging audio events to a circular buffer where they can then be retroactively dumped when needed.mediaserverAndroid system process that contains media-related services, including AudioFlinger.NBAIONon-blocking audio input/output. Abstraction for AudioFlinger ports. The term can be misleading as some implementations of the NBAIO API support blocking. The key implementations of NBAIO are for different types
of pipes.normal mixerThread within AudioFlinger that services most full-featured AudioTrack clients. Directly drives an output device or feeds its sub-mix into FastMixer via a pipe.OpenSL ESAudio API standard by The Khronos Group. Android versions since API level 9 support a
native audio API that is based on a subset of OpenSL ES 1.0.1.silent modeUser-settable feature to mute the phone ringer and notifications without affecting media playback (music, videos, games) or alarms.SoundPoolHigher-level client API than AudioTrack. Plays sampled audio clips. Useful for triggering UI feedback, game sounds, etc. The API is defined at android.media.SoundPool.StagefrightSee Media.StateQueueModule within AudioFlinger responsible for synchronizing state among threads. Whereas NBAIO is used to pass data, StateQueue is used to pass control information.strategyGroup of stream types with similar behavior. Used by the audio policy service.stream typeEnumeration that expresses a use case for audio output. The audio policy implementation uses the stream type, along with other parameters, to determine volume and routing decisions. For a list of stream types,
see android.media.AudioManager.tee sinkSee Audio Debugging.tinyalsaSmall user-mode API above ALSA kernel with BSD license. Recommended for HAL implementations.ToneGeneratorHigher-level client API than AudioTrack. Plays dual-tone multi-frequency (DTMF) signals. For details, refer to Dual-tone
multi-frequency signaling and the API definition at android.media.ToneGenerator.trackAudio stream. Controlled by the AudioTrack or AudioRecord API.volume attenuation curveDevice-specific mapping from a generic volume index to a specific attenuation factor for a given output.volume indexUnitless integer that expresses the desired relative volume of a stream. The volume-related APIs ofandroid.media.AudioManager operate
in volume indices rather than absolute attenuation factors.