| /external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
| arith_routines_logist.c | 286 newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(inSqrt, res) + res, 1);
290 newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(inSqrt, res) + res, 1);
|
| bandwidth_estimator.c | 300 bweStr->recBwInv = WEBRTC_SPL_DIV((1073741824 +
856 if (State->StillBuffered < WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL((512 - WEBRTC_SPL_DIV(512, BURST_LEN)), DelayBuildUp), 9)) {
858 inv_Q12 = WEBRTC_SPL_DIV(4096, WEBRTC_SPL_MUL(BURST_LEN, FrameSamples));
862 inv_Q12 = WEBRTC_SPL_DIV(4096, FrameSamples);
[all...] |
| decode.c | 79 frame_mode = (WebRtc_Word16)WEBRTC_SPL_DIV(*current_framesamples, MAX_FRAMESAMPLES); /* 0, or 1 */
80 processed_samples = (WebRtc_Word16)WEBRTC_SPL_DIV(*current_framesamples, frame_mode+1); /* either 320 (20ms) or 480 (30, 60 ms) */
|
| entropy_coding.c | 319 newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1);
323 newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1);
337 newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1);
341 newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1);
[all...] |
| /external/webrtc/src/modules/audio_processing/agc/ |
| analog_agc.c | 211 targetGainIdx = (WebRtc_UWord16)WEBRTC_SPL_DIV(tmp32, tmp16);
[all...] |
| /external/webrtc/src/common_audio/signal_processing/include/ |
| signal_processing_library.h | 79 #define WEBRTC_SPL_DIV(a, b) \
[all...] |
| /external/webrtc/src/common_audio/signal_processing/ |
| signal_processing_unittest.cc | 55 EXPECT_EQ(-5461, WEBRTC_SPL_DIV(a, b));
|
| /external/webrtc/src/modules/audio_processing/ns/ |
| nsx_core.c | [all...] |
