webrtc/modules/audio_coding/codecs/isac/main/source/codec.h - Issue 1230503003: Update a ton of audio code to use size_t more correctly and in general reduce

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Unified Diff: webrtc/modules/audio_coding/codecs/isac/main/source/codec.h

Issue 1230503003: Update a ton of audio code to use size_t more correctly and in general reduce (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master

Patch Set: Resync Created 5 years, 4 months ago

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Index: webrtc/modules/audio_coding/codecs/isac/main/source/codec.h

diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/codec.h b/webrtc/modules/audio_coding/codecs/isac/main/source/codec.h

index 4b36fffed686b3154ebdd1c53ee7e9354b31150e..7ef64b55fe09c25a9762946cbb43cdbe702e060f 100644

--- a/webrtc/modules/audio_coding/codecs/isac/main/source/codec.h

+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/codec.h

@@ -25,7 +25,7 @@

void WebRtcIsac_ResetBitstream(Bitstr* bit_stream);

int WebRtcIsac_EstimateBandwidth(BwEstimatorstr* bwest_str, Bitstr* streamdata,

- int32_t packet_size,

+ size_t packet_size,

uint16_t rtp_seq_number,

uint32_t send_ts, uint32_t arr_ts,

enum IsacSamplingRate encoderSampRate,

@@ -195,14 +195,14 @@ void WebRtcIsac_Spec2time(const TransformTables* tables,

/******************************* filter functions ****************************/

-void WebRtcIsac_AllPoleFilter(double* InOut, double* Coef, int lengthInOut,

+void WebRtcIsac_AllPoleFilter(double* InOut, double* Coef, size_t lengthInOut,

int orderCoef);

-void WebRtcIsac_AllZeroFilter(double* In, double* Coef, int lengthInOut,

+void WebRtcIsac_AllZeroFilter(double* In, double* Coef, size_t lengthInOut,

int orderCoef, double* Out);

void WebRtcIsac_ZeroPoleFilter(double* In, double* ZeroCoef, double* PoleCoef,

- int lengthInOut, int orderCoef, double* Out);

+ size_t lengthInOut, int orderCoef, double* Out);

/***************************** filterbank functions **************************/

@@ -228,6 +228,6 @@ void WebRtcIsac_NormLatticeFilterAr(int orderCoef, float* stateF, float* stateG,

void WebRtcIsac_Dir2Lat(double* a, int orderCoef, float* sth, float* cth);

-void WebRtcIsac_AutoCorr(double* r, const double* x, int N, int order);

+void WebRtcIsac_AutoCorr(double* r, const double* x, size_t N, size_t order);

#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_ */