问题描述
我试图使用WebRTC技术库中创建一个独立的AGC。 （输入 - wav文件，输出 - 与调节增益wav文件）。但就在这个时候，我有一些问题，这个问题。
I'm trying to create a standalone AGC using WebRtc library. (Input - wav file, output - wav file with adjusted gain). But at this time I have some problems with this issue.
我想使用的在 gain_control.h 文件中声明功能。当我使用WebRtcAgc_Process（....）我获得恒定的增益，它适用于整个信号，而不是非线性增益取决于从输入信号幅度。
I'm trying to use functions which are declared in gain_control.h file. When I'm using WebRtcAgc_Process(....) I obtain constant gain, which applies to whole signal, but not nonlinear gain which depends from input signal magnitude.
也许我应该使用其他功能为我的目的是什么？如何通过WebRTC技术库实现AGC？
May be I should use another functions for my purpose? How can I implement AGC via WebRTC library?
推荐答案
的AGC的主要目的是提供一种用户预期通过OS来设置一个建议的系统的麦克风的音量。如果你想申请一个纯粹的数字增益，可以在两种模式（一个配置从模块/ audio_processing /有/ audio_processing.h ，但 gain_control.h 有类似的模式）：
The AGC's main purpose is to provide a recommended system mic volume which the user is expected to set through the OS. If you would like to apply a purely digital gain, you can configure it in one of two modes (from modules/audio_processing/include/audio_processing.h, but gain_control.h has analogous modes):
// Adaptive mode intended for situations in which an analog volume control
// is unavailable. It operates in a similar fashion to the adaptive analog
// mode, but with scaling instead applied in the digital domain. As with
// the analog mode, it additionally uses a digital compression stage.
kAdaptiveDigital,
// Fixed mode which enables only the digital compression stage also used by
// the two adaptive modes.
//
// It is distinguished from the adaptive modes by considering only a
// short time-window of the input signal. It applies a fixed gain through
// most of the input level range, and compresses (gradually reduces gain
// with increasing level) the input signal at higher levels. This mode is
// preferred on embedded devices where the capture signal level is
// predictable, so that a known gain can be applied.
kFixedDigital
您可以通过 WebRtcAgc_Init设置这些（），但除非你需要避免的开销，我建议只使用AudioProcessing类。
You can set these through WebRtcAgc_Init(), though unless you need to avoid the overhead, I'd recommend just using the AudioProcessing class.
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