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  /external/chromium_org/third_party/webrtc/modules/audio_processing/agc/
analog_agc.h 88 int16_t envSum; // Filtered scaled envelope in subframes
analog_agc.c 600 stt->envSum += tmpW16;
604 if (stt->envSum > 25000)
607 stt->envSum = 0;
610 /* stt->envSum *= 0.99; */
611 stt->envSum = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(stt->envSum,
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