Index: webrtc/modules/audio_processing/aec/aec_resampler.c |
diff --git a/webrtc/modules/audio_processing/aec/aec_resampler.c b/webrtc/modules/audio_processing/aec/aec_resampler.c |
index 99c39efa8818e16be8e8d95f2dbc77b8a1c717ed..ae64ddf0f22decc0181ed650a80941e418cc10f3 100644 |
--- a/webrtc/modules/audio_processing/aec/aec_resampler.c |
+++ b/webrtc/modules/audio_processing/aec/aec_resampler.c |
@@ -21,9 +21,7 @@ |
#include "webrtc/modules/audio_processing/aec/aec_core.h" |
-enum { |
- kEstimateLengthFrames = 400 |
-}; |
+enum { kEstimateLengthFrames = 400 }; |
typedef struct { |
float buffer[kResamplerBufferSize]; |
@@ -81,8 +79,7 @@ void WebRtcAec_ResampleLinear(void* resampInst, |
assert(size_out != NULL); |
// Add new frame data in lookahead |
- memcpy(&obj->buffer[FRAME_LEN + kResamplingDelay], |
- inspeech, |
+ memcpy(&obj->buffer[FRAME_LEN + kResamplingDelay], inspeech, |
size * sizeof(inspeech[0])); |
// Sample rate ratio |
@@ -96,7 +93,6 @@ void WebRtcAec_ResampleLinear(void* resampInst, |
tn = (size_t)tnew; |
while (tn < size) { |
- |
// Interpolation |
outspeech[mm] = y[tn] + (tnew - tn) * (y[tn + 1] - y[tn]); |
mm++; |
@@ -109,8 +105,7 @@ void WebRtcAec_ResampleLinear(void* resampInst, |
obj->position += (*size_out) * be - size; |
// Shift buffer |
- memmove(obj->buffer, |
- &obj->buffer[size], |
+ memmove(obj->buffer, &obj->buffer[size], |
(kResamplerBufferSize - size) * sizeof(obj->buffer[0])); |
} |
@@ -122,8 +117,8 @@ int WebRtcAec_GetSkew(void* resampInst, int rawSkew, float* skewEst) { |
obj->skewData[obj->skewDataIndex] = rawSkew; |
obj->skewDataIndex++; |
} else if (obj->skewDataIndex == kEstimateLengthFrames) { |
- err = EstimateSkew( |
- obj->skewData, kEstimateLengthFrames, obj->deviceSampleRateHz, skewEst); |
+ err = EstimateSkew(obj->skewData, kEstimateLengthFrames, |
+ obj->deviceSampleRateHz, skewEst); |
obj->skewEstimate = *skewEst; |
obj->skewDataIndex++; |
} else { |