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  /external/webrtc/webrtc/modules/audio_coding/neteq/
normal.cc 48 const int fs_mult = fs_hz_ / 8000; local
49 assert(fs_mult > 0);
50 // fs_shift = log2(fs_mult), rounded down.
53 const int fs_shift = 30 - WebRtcSpl_NormW32(fs_mult);
80 std::min(static_cast<size_t>(fs_mult * 64), length_per_channel);
114 int increment = 64 / fs_mult;
134 for (size_t i = 0; i < static_cast<size_t>(8 * fs_mult); i++) {
163 memcpy(cng_output, signal, fs_mult * 8 * sizeof(int16_t));
170 for (size_t i = 0; i < static_cast<size_t>(8 * fs_mult); i++) {
181 int increment = 64 / fs_mult;
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expand.cc 80 int fs_mult = fs_hz_ / 8000; local
182 // max_lag_ = fs_mult * 120;
183 // expand_lags_[0] = fs_mult * 120;
184 // expand_lags_[1] = fs_mult * 120;
185 // expand_lags_[2] = fs_mult * 120;
211 // <= 31 * fs_mult => go from 1 to 0 in about 8 ms;
212 // (>= 31 .. <= 63) * fs_mult => go from 1 to 0 in about 16 ms;
213 // >= 64 * fs_mult => go from 1 to 0 in about 32 ms.
248 // mute_slope = 0.0010 / fs_mult in Q20.
249 parameters.mute_slope = std::max(parameters.mute_slope, 1049 / fs_mult);
365 int fs_mult = fs_hz_ \/ 8000; local
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