AnotherFoxGuy/webrtc-audio-processing
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Update to current webrtc library

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This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.
This commit is contained in:
Arun Raghavan
2020-10-12 18:08:02 -04:00
parent b1b02581d3
commit bcec8b0b21
859 changed files with 76187 additions and 49580 deletions
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8
webrtc/common_audio/signal_processing/resample_by_2.c
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@ -15,7 +15,7 @@
*
*/
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
#ifdef WEBRTC_ARCH_ARM_V7
@ -83,7 +83,7 @@ void WebRtcSpl_DownsampleBy2(const int16_t* in, size_t len,
for (i = (len >> 1); i > 0; i--) {
// lower allpass filter
in32 = (int32_t)(*in++) << 10;
in32 = (int32_t)(*in++) * (1 << 10);
diff = in32 - state1;
tmp1 = MUL_ACCUM_1(kResampleAllpass2[0], diff, state0);
state0 = in32;
@ -95,7 +95,7 @@ void WebRtcSpl_DownsampleBy2(const int16_t* in, size_t len,
state2 = tmp2;
// upper allpass filter
in32 = (int32_t)(*in++) << 10;
in32 = (int32_t)(*in++) * (1 << 10);
diff = in32 - state5;
tmp1 = MUL_ACCUM_1(kResampleAllpass1[0], diff, state4);
state4 = in32;
@ -141,7 +141,7 @@ void WebRtcSpl_UpsampleBy2(const int16_t* in, size_t len,
for (i = len; i > 0; i--) {
// lower allpass filter
in32 = (int32_t)(*in++) << 10;
in32 = (int32_t)(*in++) * (1 << 10);
diff = in32 - state1;
tmp1 = MUL_ACCUM_1(kResampleAllpass1[0], diff, state0);
state0 = in32;