WebRTC AGC (Automatic Gain Control)

Question

I am testing the WebRTC AGC but I must be doing something wrong because the signal just passes through unmodified.

Here's how I create and initialize the AGC:

agcConfig.compressionGaindB = 9; agcConfig.limiterEnable = 1; agcConfig.targetLevelDbfs = 9;   /* 9dB below full scale */  WebRtcAgc_Create(&agc); WebRtcAgc_Init(agc, minLevel, maxLevel, kAgcModeFixedDigital, 8000); WebRtcAgc_set_config(agc, agcConfig); 

And then for each 10ms sample block I do the following:

WebRtcAgc_Process(agc, micData, NULL, 80, micData, NULL, micLevelIn, &micLevelOut, 0, &saturationWarning); 

Where micLevelIn is set to 0.

Can somebody tell me what I'm doing wrong?

I expected that a full scale sine tone would be attenuated to the target DBFS level; and a low level sine tone (i.e. -30dBFS) would be amplified to match the target DBFS level. But that's not what I'm seeing.

Answer 1

Here is the sequence of operations to be used for Webrtc_AGC:

1. Create AGC: WebRtcAgc_Create
2. Initialize AGC: WebRtcAgc_Init
3. Set Config: WebRtcAgc_set_config
4. Initialize capture_level = 0
5. For kAgcModeAdaptiveDigital, invoke VirtualMic: WebRtcAgc_VirtualMic
6. Process Buffer with capture_level: WebRtcAgc_Process
7. Get the out capture level returned from WebRtcAgc_Process and set it to capture_level
8. Repeat 5 to 7 for the audio buffers
9. Destroy the AGC: WebRtcAgc_Free

Check webrtc/modules/audio_processing/gain_control_impl.cc for reference.