Index: webrtc/modules/audio_processing/agc/legacy/digital_agc.h |
diff --git a/webrtc/modules/audio_processing/agc/legacy/digital_agc.h b/webrtc/modules/audio_processing/agc/legacy/digital_agc.h |
index 819844d774c7af8ab0c808d2bd43583e0ad740bb..4664b59dc8eaa85359c3ca5d77ff0400779d1468 100644 |
--- a/webrtc/modules/audio_processing/agc/legacy/digital_agc.h |
+++ b/webrtc/modules/audio_processing/agc/legacy/digital_agc.h |
@@ -18,37 +18,36 @@ |
#include "webrtc/typedefs.h" |
// the 32 most significant bits of A(19) * B(26) >> 13 |
-#define AGC_MUL32(A, B) (((B)>>13)*(A) + ( ((0x00001FFF & (B))*(A)) >> 13 )) |
+#define AGC_MUL32(A, B) (((B) >> 13) * (A) + (((0x00001FFF & (B)) * (A)) >> 13)) |
// C + the 32 most significant bits of A * B |
-#define AGC_SCALEDIFF32(A, B, C) ((C) + ((B)>>16)*(A) + ( ((0x0000FFFF & (B))*(A)) >> 16 )) |
+#define AGC_SCALEDIFF32(A, B, C) \ |
+ ((C) + ((B) >> 16) * (A) + (((0x0000FFFF & (B)) * (A)) >> 16)) |
-typedef struct |
-{ |
- int32_t downState[8]; |
- int16_t HPstate; |
- int16_t counter; |
- int16_t logRatio; // log( P(active) / P(inactive) ) (Q10) |
- int16_t meanLongTerm; // Q10 |
- int32_t varianceLongTerm; // Q8 |
- int16_t stdLongTerm; // Q10 |
- int16_t meanShortTerm; // Q10 |
- int32_t varianceShortTerm; // Q8 |
- int16_t stdShortTerm; // Q10 |
-} AgcVad; // total = 54 bytes |
+typedef struct { |
+ int32_t downState[8]; |
+ int16_t HPstate; |
+ int16_t counter; |
+ int16_t logRatio; // log( P(active) / P(inactive) ) (Q10) |
+ int16_t meanLongTerm; // Q10 |
+ int32_t varianceLongTerm; // Q8 |
+ int16_t stdLongTerm; // Q10 |
+ int16_t meanShortTerm; // Q10 |
+ int32_t varianceShortTerm; // Q8 |
+ int16_t stdShortTerm; // Q10 |
+} AgcVad; // total = 54 bytes |
-typedef struct |
-{ |
- int32_t capacitorSlow; |
- int32_t capacitorFast; |
- int32_t gain; |
- int32_t gainTable[32]; |
- int16_t gatePrevious; |
- int16_t agcMode; |
- AgcVad vadNearend; |
- AgcVad vadFarend; |
+typedef struct { |
+ int32_t capacitorSlow; |
+ int32_t capacitorFast; |
+ int32_t gain; |
+ int32_t gainTable[32]; |
+ int16_t gatePrevious; |
+ int16_t agcMode; |
+ AgcVad vadNearend; |
+ AgcVad vadFarend; |
#ifdef WEBRTC_AGC_DEBUG_DUMP |
- FILE* logFile; |
- int frameCounter; |
+ FILE* logFile; |
+ int frameCounter; |
#endif |
} DigitalAgc; |
@@ -67,14 +66,14 @@ int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc* digitalAgcInst, |
void WebRtcAgc_InitVad(AgcVad* vadInst); |
-int16_t WebRtcAgc_ProcessVad(AgcVad* vadInst, // (i) VAD state |
- const int16_t* in, // (i) Speech signal |
+int16_t WebRtcAgc_ProcessVad(AgcVad* vadInst, // (i) VAD state |
+ const int16_t* in, // (i) Speech signal |
size_t nrSamples); // (i) number of samples |
-int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16 |
- int16_t compressionGaindB, // Q0 (in dB) |
- int16_t targetLevelDbfs,// Q0 (in dB) |
+int32_t WebRtcAgc_CalculateGainTable(int32_t* gainTable, // Q16 |
+ int16_t compressionGaindB, // Q0 (in dB) |
+ int16_t targetLevelDbfs, // Q0 (in dB) |
uint8_t limiterEnable, |
int16_t analogTarget); |
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_ |
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_ |