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Reformat the WebRTC code base

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Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This commit is contained in:
Yves Gerey
2018-06-19 15:03:05 +02:00
parent b602123a5a
commit 665174fdbb
1569 changed files with 30495 additions and 30309 deletions
Download Patch File Download Diff File Expand all files Collapse all files

51
modules/audio_coding/codecs/isac/main/source/arith_routines.h
View File

@ -21,42 +21,47 @@
#include "modules/audio_coding/codecs/isac/main/source/structs.h"
int WebRtcIsac_EncLogisticMulti2(
Bitstr *streamdata, /* in-/output struct containing bitstream */
int16_t *dataQ7, /* input: data vector */
const uint16_t *env, /* input: side info vector defining the width of the pdf */
const int N, /* input: data vector length */
Bitstr* streamdata, /* in-/output struct containing bitstream */
int16_t* dataQ7, /* input: data vector */
const uint16_t*
env, /* input: side info vector defining the width of the pdf */
const int N, /* input: data vector length */
const int16_t isSWB12kHz); /* if the codec is working in 12kHz bandwidth */
/* returns the number of bytes in the stream */
int WebRtcIsac_EncTerminate(Bitstr *streamdata); /* in-/output struct containing bitstream */
int WebRtcIsac_EncTerminate(
Bitstr* streamdata); /* in-/output struct containing bitstream */
/* returns the number of bytes in the stream so far */
int WebRtcIsac_DecLogisticMulti2(
int16_t *data, /* output: data vector */
Bitstr *streamdata, /* in-/output struct containing bitstream */
const uint16_t *env, /* input: side info vector defining the width of the pdf */
const int16_t *dither, /* input: dither vector */
const int N, /* input: data vector length */
int16_t* data, /* output: data vector */
Bitstr* streamdata, /* in-/output struct containing bitstream */
const uint16_t*
env, /* input: side info vector defining the width of the pdf */
const int16_t* dither, /* input: dither vector */
const int N, /* input: data vector length */
const int16_t isSWB12kHz); /* if the codec is working in 12kHz bandwidth */
void WebRtcIsac_EncHistMulti(
Bitstr *streamdata, /* in-/output struct containing bitstream */
const int *data, /* input: data vector */
const uint16_t *const *cdf, /* input: array of cdf arrays */
Bitstr* streamdata, /* in-/output struct containing bitstream */
const int* data, /* input: data vector */
const uint16_t* const* cdf, /* input: array of cdf arrays */
const int N); /* input: data vector length */
int WebRtcIsac_DecHistBisectMulti(
int *data, /* output: data vector */
Bitstr *streamdata, /* in-/output struct containing bitstream */
const uint16_t *const *cdf, /* input: array of cdf arrays */
const uint16_t *cdf_size, /* input: array of cdf table sizes+1 (power of two: 2^k) */
const int N); /* input: data vector length */
int* data, /* output: data vector */
Bitstr* streamdata, /* in-/output struct containing bitstream */
const uint16_t* const* cdf, /* input: array of cdf arrays */
const uint16_t*
cdf_size, /* input: array of cdf table sizes+1 (power of two: 2^k) */
const int N); /* input: data vector length */
int WebRtcIsac_DecHistOneStepMulti(
int *data, /* output: data vector */
Bitstr *streamdata, /* in-/output struct containing bitstream */
const uint16_t *const *cdf, /* input: array of cdf arrays */
const uint16_t *init_index,/* input: vector of initial cdf table search entries */
const int N); /* input: data vector length */
int* data, /* output: data vector */
Bitstr* streamdata, /* in-/output struct containing bitstream */
const uint16_t* const* cdf, /* input: array of cdf arrays */
const uint16_t*
init_index, /* input: vector of initial cdf table search entries */
const int N); /* input: data vector length */
#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ARITH_ROUTINES_H_ */

6
modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc
View File

@ -29,7 +29,11 @@ void TestGoodConfig(const AudioEncoderIsacFloatImpl::Config& config) {
// Wrap subroutine calls that test things in this, so that the error messages
// will be accompanied by stack traces that make it possible to tell which
// subroutine invocation caused the failure.
#define S(x) do { SCOPED_TRACE(#x); x; } while (0)
#define S(x) \
do { \
SCOPED_TRACE(#x); \
x; \
} while (0)
} // namespace

209
modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h
View File

@ -24,162 +24,151 @@
#include "modules/audio_coding/codecs/isac/main/source/settings.h"
#include "modules/audio_coding/codecs/isac/main/source/structs.h"
#define MIN_ISAC_BW 10000
#define MIN_ISAC_BW_LB 10000
#define MIN_ISAC_BW_UB 25000
#define MIN_ISAC_BW 10000
#define MIN_ISAC_BW_LB 10000
#define MIN_ISAC_BW_UB 25000
#define MAX_ISAC_BW 56000
#define MAX_ISAC_BW_UB 32000
#define MAX_ISAC_BW_LB 32000
#define MAX_ISAC_BW 56000
#define MAX_ISAC_BW_UB 32000
#define MAX_ISAC_BW_LB 32000
#define MIN_ISAC_MD 5
#define MAX_ISAC_MD 25
#define MIN_ISAC_MD 5
#define MAX_ISAC_MD 25
// assumed header size, in bytes; we don't know the exact number
// (header compression may be used)
#define HEADER_SIZE 35
#define HEADER_SIZE 35
// Initial Frame-Size, in ms, for Wideband & Super-Wideband Mode
#define INIT_FRAME_LEN_WB 60
#define INIT_FRAME_LEN_WB 60
#define INIT_FRAME_LEN_SWB 30
// Initial Bottleneck Estimate, in bits/sec, for
// Wideband & Super-wideband mode
#define INIT_BN_EST_WB 20e3f
#define INIT_BN_EST_SWB 56e3f
#define INIT_BN_EST_WB 20e3f
#define INIT_BN_EST_SWB 56e3f
// Initial Header rate (header rate depends on frame-size),
// in bits/sec, for Wideband & Super-Wideband mode.
#define INIT_HDR_RATE_WB \
#define INIT_HDR_RATE_WB \
((float)HEADER_SIZE * 8.0f * 1000.0f / (float)INIT_FRAME_LEN_WB)
#define INIT_HDR_RATE_SWB \
#define INIT_HDR_RATE_SWB \
((float)HEADER_SIZE * 8.0f * 1000.0f / (float)INIT_FRAME_LEN_SWB)
// number of packets in a row for a high rate burst
#define BURST_LEN 3
#define BURST_LEN 3
// ms, max time between two full bursts
#define BURST_INTERVAL 500
#define BURST_INTERVAL 500
// number of packets in a row for initial high rate burst
#define INIT_BURST_LEN 5
#define INIT_BURST_LEN 5
// bits/s, rate for the first BURST_LEN packets
#define INIT_RATE_WB INIT_BN_EST_WB
#define INIT_RATE_SWB INIT_BN_EST_SWB
#define INIT_RATE_WB INIT_BN_EST_WB
#define INIT_RATE_SWB INIT_BN_EST_SWB
#if defined(__cplusplus)
extern "C" {
#endif
/* This function initializes the struct */
/* to be called before using the struct for anything else */
/* returns 0 if everything went fine, -1 otherwise */
int32_t WebRtcIsac_InitBandwidthEstimator(
BwEstimatorstr* bwest_str,
enum IsacSamplingRate encoderSampRate,
enum IsacSamplingRate decoderSampRate);
/* This function initializes the struct */
/* to be called before using the struct for anything else */
/* returns 0 if everything went fine, -1 otherwise */
int32_t WebRtcIsac_InitBandwidthEstimator(
BwEstimatorstr* bwest_str,
enum IsacSamplingRate encoderSampRate,
enum IsacSamplingRate decoderSampRate);
/* This function updates the receiving estimate */
/* Parameters: */
/* rtp_number - value from RTP packet, from NetEq */
/* frame length - length of signal frame in ms, from iSAC decoder */
/* send_ts - value in RTP header giving send time in samples */
/* arr_ts - value given by timeGetTime() time of arrival in samples of packet from NetEq */
/* pksize - size of packet in bytes, from NetEq */
/* Index - integer (range 0...23) indicating bottle neck & jitter as estimated by other side */
/* returns 0 if everything went fine, -1 otherwise */
int16_t WebRtcIsac_UpdateBandwidthEstimator(
BwEstimatorstr* bwest_str,
const uint16_t rtp_number,
const int32_t frame_length,
const uint32_t send_ts,
const uint32_t arr_ts,
const size_t pksize);
/* This function updates the receiving estimate */
/* Parameters: */
/* rtp_number - value from RTP packet, from NetEq */
/* frame length - length of signal frame in ms, from iSAC decoder */
/* send_ts - value in RTP header giving send time in samples */
/* arr_ts - value given by timeGetTime() time of arrival in samples of
* packet from NetEq */
/* pksize - size of packet in bytes, from NetEq */
/* Index - integer (range 0...23) indicating bottle neck & jitter as
* estimated by other side */
/* returns 0 if everything went fine, -1 otherwise */
int16_t WebRtcIsac_UpdateBandwidthEstimator(BwEstimatorstr* bwest_str,
const uint16_t rtp_number,
const int32_t frame_length,
const uint32_t send_ts,
const uint32_t arr_ts,
const size_t pksize);
/* Update receiving estimates. Used when we only receive BWE index, no iSAC data packet. */
int16_t WebRtcIsac_UpdateUplinkBwImpl(
BwEstimatorstr* bwest_str,
int16_t Index,
enum IsacSamplingRate encoderSamplingFreq);
/* Update receiving estimates. Used when we only receive BWE index, no iSAC data
* packet. */
int16_t WebRtcIsac_UpdateUplinkBwImpl(
BwEstimatorstr* bwest_str,
int16_t Index,
enum IsacSamplingRate encoderSamplingFreq);
/* Returns the bandwidth/jitter estimation code (integer 0...23) to put in the sending iSAC payload */
void WebRtcIsac_GetDownlinkBwJitIndexImpl(
BwEstimatorstr* bwest_str,
int16_t* bottleneckIndex,
int16_t* jitterInfo,
enum IsacSamplingRate decoderSamplingFreq);
/* Returns the bandwidth/jitter estimation code (integer 0...23) to put in the
* sending iSAC payload */
void WebRtcIsac_GetDownlinkBwJitIndexImpl(
BwEstimatorstr* bwest_str,
int16_t* bottleneckIndex,
int16_t* jitterInfo,
enum IsacSamplingRate decoderSamplingFreq);
/* Returns the bandwidth estimation (in bps) */
int32_t WebRtcIsac_GetDownlinkBandwidth(
const BwEstimatorstr *bwest_str);
/* Returns the bandwidth estimation (in bps) */
int32_t WebRtcIsac_GetDownlinkBandwidth(const BwEstimatorstr* bwest_str);
/* Returns the max delay (in ms) */
int32_t WebRtcIsac_GetDownlinkMaxDelay(
const BwEstimatorstr *bwest_str);
/* Returns the max delay (in ms) */
int32_t WebRtcIsac_GetDownlinkMaxDelay(const BwEstimatorstr* bwest_str);
/* Returns the bandwidth that iSAC should send with in bps */
int32_t WebRtcIsac_GetUplinkBandwidth(const BwEstimatorstr* bwest_str);
/* Returns the bandwidth that iSAC should send with in bps */
int32_t WebRtcIsac_GetUplinkBandwidth(const BwEstimatorstr* bwest_str);
/* Returns the max delay value from the other side in ms */
int32_t WebRtcIsac_GetUplinkMaxDelay(
const BwEstimatorstr *bwest_str);
/* Returns the max delay value from the other side in ms */
int32_t WebRtcIsac_GetUplinkMaxDelay(const BwEstimatorstr* bwest_str);
/* Fills in an IsacExternalBandwidthInfo struct. */
void WebRtcIsacBw_GetBandwidthInfo(
BwEstimatorstr* bwest_str,
enum IsacSamplingRate decoder_sample_rate_hz,
IsacBandwidthInfo* bwinfo);
/* Fills in an IsacExternalBandwidthInfo struct. */
void WebRtcIsacBw_GetBandwidthInfo(BwEstimatorstr* bwest_str,
enum IsacSamplingRate decoder_sample_rate_hz,
IsacBandwidthInfo* bwinfo);
/* Uses the values from an IsacExternalBandwidthInfo struct. */
void WebRtcIsacBw_SetBandwidthInfo(BwEstimatorstr* bwest_str,
const IsacBandwidthInfo* bwinfo);
/* Uses the values from an IsacExternalBandwidthInfo struct. */
void WebRtcIsacBw_SetBandwidthInfo(BwEstimatorstr* bwest_str,
const IsacBandwidthInfo* bwinfo);
/*
* update amount of data in bottle neck buffer and burst handling
* returns minimum payload size (bytes)
*/
int WebRtcIsac_GetMinBytes(
RateModel* State,
int StreamSize, /* bytes in bitstream */
const int FrameLen, /* ms per frame */
const double BottleNeck, /* bottle neck rate; excl headers (bps) */
const double DelayBuildUp, /* max delay from bottleneck buffering (ms) */
enum ISACBandwidth bandwidth
/*,int16_t frequentLargePackets*/);
/*
* update amount of data in bottle neck buffer and burst handling
* returns minimum payload size (bytes)
*/
int WebRtcIsac_GetMinBytes(
RateModel* State,
int StreamSize, /* bytes in bitstream */
const int FrameLen, /* ms per frame */
const double BottleNeck, /* bottle neck rate; excl headers (bps) */
const double DelayBuildUp, /* max delay from bottleneck buffering (ms) */
enum ISACBandwidth bandwidth
/*,int16_t frequentLargePackets*/);
/*
* update long-term average bitrate and amount of data in buffer
*/
void WebRtcIsac_UpdateRateModel(
RateModel* State,
int StreamSize, /* bytes in bitstream */
const int FrameSamples, /* samples per frame */
const double BottleNeck); /* bottle neck rate; excl headers (bps) */
/*
* update long-term average bitrate and amount of data in buffer
*/
void WebRtcIsac_UpdateRateModel(
RateModel* State,
int StreamSize, /* bytes in bitstream */
const int FrameSamples, /* samples per frame */
const double BottleNeck); /* bottle neck rate; excl headers (bps) */
void WebRtcIsac_InitRateModel(RateModel* State);
void WebRtcIsac_InitRateModel(
RateModel *State);
/* Returns the new framelength value (input argument: bottle_neck) */
int WebRtcIsac_GetNewFrameLength(double bottle_neck, int current_framelength);
/* Returns the new framelength value (input argument: bottle_neck) */
int WebRtcIsac_GetNewFrameLength(
double bottle_neck,
int current_framelength);
/* Returns the new SNR value (input argument: bottle_neck) */
double WebRtcIsac_GetSnr(double bottle_neck, int new_framelength);
/* Returns the new SNR value (input argument: bottle_neck) */
double WebRtcIsac_GetSnr(
double bottle_neck,
int new_framelength);
int16_t WebRtcIsac_UpdateUplinkJitter(
BwEstimatorstr* bwest_str,
int32_t index);
int16_t WebRtcIsac_UpdateUplinkJitter(BwEstimatorstr* bwest_str, int32_t index);
#if defined(__cplusplus)
}
#endif
#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATOR_H_ */
#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATOR_H_ \
*/

43
modules/audio_coding/codecs/isac/main/source/codec.h
View File

@ -25,10 +25,12 @@
void WebRtcIsac_ResetBitstream(Bitstr* bit_stream);
int WebRtcIsac_EstimateBandwidth(BwEstimatorstr* bwest_str, Bitstr* streamdata,
int WebRtcIsac_EstimateBandwidth(BwEstimatorstr* bwest_str,
Bitstr* streamdata,
size_t packet_size,
uint16_t rtp_seq_number,
uint32_t send_ts, uint32_t arr_ts,
uint32_t send_ts,
uint32_t arr_ts,
enum IsacSamplingRate encoderSampRate,
enum IsacSamplingRate decoderSampRate);
@ -38,7 +40,8 @@ int WebRtcIsac_DecodeLb(const TransformTables* transform_tables,
int16_t* current_framesamples,
int16_t isRCUPayload);
int WebRtcIsac_DecodeRcuLb(float* signal_out, ISACLBDecStruct* ISACdec_obj,
int WebRtcIsac_DecodeRcuLb(float* signal_out,
ISACLBDecStruct* ISACdec_obj,
int16_t* current_framesamples);
int WebRtcIsac_EncodeLb(const TransformTables* transform_tables,
@ -48,15 +51,20 @@ int WebRtcIsac_EncodeLb(const TransformTables* transform_tables,
int16_t bottleneckIndex);
int WebRtcIsac_EncodeStoredDataLb(const IsacSaveEncoderData* ISACSavedEnc_obj,
Bitstr* ISACBitStr_obj, int BWnumber,
Bitstr* ISACBitStr_obj,
int BWnumber,
float scale);
int WebRtcIsac_EncodeStoredDataUb(
const ISACUBSaveEncDataStruct* ISACSavedEnc_obj, Bitstr* bitStream,
int32_t jitterInfo, float scale, enum ISACBandwidth bandwidth);
const ISACUBSaveEncDataStruct* ISACSavedEnc_obj,
Bitstr* bitStream,
int32_t jitterInfo,
float scale,
enum ISACBandwidth bandwidth);
int16_t WebRtcIsac_GetRedPayloadUb(
const ISACUBSaveEncDataStruct* ISACSavedEncObj, Bitstr* bitStreamObj,
const ISACUBSaveEncDataStruct* ISACSavedEncObj,
Bitstr* bitStreamObj,
enum ISACBandwidth bandwidth);
/******************************************************************************
@ -82,7 +90,6 @@ int16_t WebRtcIsac_RateAllocation(int32_t inRateBitPerSec,
double* rateUBBitPerSec,
enum ISACBandwidth* bandwidthKHz);
/******************************************************************************
* WebRtcIsac_DecodeUb16()
*
@ -169,7 +176,6 @@ void WebRtcIsac_InitMasking(MaskFiltstr* maskdata);
void WebRtcIsac_InitPostFilterbank(PostFiltBankstr* postfiltdata);
/**************************** transform functions ****************************/
void WebRtcIsac_InitTransform(TransformTables* tables);
@ -190,18 +196,25 @@ void WebRtcIsac_Spec2time(const TransformTables* tables,
/***************************** filterbank functions **************************/
void WebRtcIsac_FilterAndCombineFloat(float* InLP, float* InHP, float* Out,
void WebRtcIsac_FilterAndCombineFloat(float* InLP,
float* InHP,
float* Out,
PostFiltBankstr* postfiltdata);
/************************* normalized lattice filters ************************/
void WebRtcIsac_NormLatticeFilterMa(int orderCoef, float* stateF, float* stateG,
float* lat_in, double* filtcoeflo,
void WebRtcIsac_NormLatticeFilterMa(int orderCoef,
float* stateF,
float* stateG,
float* lat_in,
double* filtcoeflo,
double* lat_out);
void WebRtcIsac_NormLatticeFilterAr(int orderCoef, float* stateF, float* stateG,
double* lat_in, double* lo_filt_coef,
void WebRtcIsac_NormLatticeFilterAr(int orderCoef,
float* stateF,
float* stateG,
double* lat_in,
double* lo_filt_coef,
float* lat_out);
void WebRtcIsac_Dir2Lat(double* a, int orderCoef, float* sth, float* cth);

7
modules/audio_coding/codecs/isac/main/source/crc.h
View File

@ -36,11 +36,6 @@
* -1 - Error
*/
int WebRtcIsac_GetCrc(
const int16_t* encoded,
int no_of_word8s,
uint32_t* crc);
int WebRtcIsac_GetCrc(const int16_t* encoded, int no_of_word8s, uint32_t* crc);
#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CRC_H_ */

87
modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.h
View File

@ -39,9 +39,7 @@
*
*
*/
int16_t WebRtcIsac_RemoveLarMean(
double* lar,
int16_t bandwidth);
int16_t WebRtcIsac_RemoveLarMean(double* lar, int16_t bandwidth);
/******************************************************************************
* WebRtcIsac_DecorrelateIntraVec()
@ -59,11 +57,9 @@ int16_t WebRtcIsac_RemoveLarMean(
* Output:
* -out : decorrelated LAR vectors.
*/
int16_t WebRtcIsac_DecorrelateIntraVec(
const double* inLAR,
double* out,
int16_t bandwidth);
int16_t WebRtcIsac_DecorrelateIntraVec(const double* inLAR,
double* out,
int16_t bandwidth);
/******************************************************************************
* WebRtcIsac_DecorrelateInterVec()
@ -82,11 +78,9 @@ int16_t WebRtcIsac_DecorrelateIntraVec(
* Output:
* -out : decorrelated LAR vectors.
*/
int16_t WebRtcIsac_DecorrelateInterVec(
const double* data,
double* out,
int16_t bandwidth);
int16_t WebRtcIsac_DecorrelateInterVec(const double* data,
double* out,
int16_t bandwidth);
/******************************************************************************
* WebRtcIsac_QuantizeUncorrLar()
@ -102,11 +96,7 @@ int16_t WebRtcIsac_DecorrelateInterVec(
* -data : quantized version of the input.
* -idx : pointer to quantization indices.
*/
double WebRtcIsac_QuantizeUncorrLar(
double* data,
int* idx,
int16_t bandwidth);
double WebRtcIsac_QuantizeUncorrLar(double* data, int* idx, int16_t bandwidth);
/******************************************************************************
* WebRtcIsac_CorrelateIntraVec()
@ -121,11 +111,9 @@ double WebRtcIsac_QuantizeUncorrLar(
* Output:
* -out : correlated parametrs.
*/
int16_t WebRtcIsac_CorrelateIntraVec(
const double* data,
double* out,
int16_t bandwidth);
int16_t WebRtcIsac_CorrelateIntraVec(const double* data,
double* out,
int16_t bandwidth);
/******************************************************************************
* WebRtcIsac_CorrelateInterVec()
@ -140,17 +128,15 @@ int16_t WebRtcIsac_CorrelateIntraVec(
* Output:
* -out : correlated parametrs.
*/
int16_t WebRtcIsac_CorrelateInterVec(
const double* data,
double* out,
int16_t bandwidth);
int16_t WebRtcIsac_CorrelateInterVec(const double* data,
double* out,
int16_t bandwidth);
/******************************************************************************
* WebRtcIsac_AddLarMean()
*
* This is the inverse of WebRtcIsac_RemoveLarMean()
*
*
* Input:
* -data : pointer to mean-removed LAR:s.
* -bandwidth : indicates if the given LAR vectors belong
@ -159,10 +145,7 @@ int16_t WebRtcIsac_CorrelateInterVec(
* Output:
* -data : pointer to LARs.
*/
int16_t WebRtcIsac_AddLarMean(
double* data,
int16_t bandwidth);
int16_t WebRtcIsac_AddLarMean(double* data, int16_t bandwidth);
/******************************************************************************
* WebRtcIsac_DequantizeLpcParam()
@ -177,11 +160,9 @@ int16_t WebRtcIsac_AddLarMean(
* Output:
* -out : pointer to quantized values.
*/
int16_t WebRtcIsac_DequantizeLpcParam(
const int* idx,
double* out,
int16_t bandwidth);
int16_t WebRtcIsac_DequantizeLpcParam(const int* idx,
double* out,
int16_t bandwidth);
/******************************************************************************
* WebRtcIsac_ToLogDomainRemoveMean()
@ -194,9 +175,7 @@ int16_t WebRtcIsac_DequantizeLpcParam(
* Output:
* -lpcGain : mean-removed in log domain.
*/
int16_t WebRtcIsac_ToLogDomainRemoveMean(
double* lpGains);
int16_t WebRtcIsac_ToLogDomainRemoveMean(double* lpGains);
/******************************************************************************
* WebRtcIsac_DecorrelateLPGain()
@ -210,16 +189,13 @@ int16_t WebRtcIsac_ToLogDomainRemoveMean(
* Output:
* -out : decorrelated parameters.
*/
int16_t WebRtcIsac_DecorrelateLPGain(
const double* data,
double* out);
int16_t WebRtcIsac_DecorrelateLPGain(const double* data, double* out);
/******************************************************************************
* WebRtcIsac_QuantizeLpcGain()
*
* Quantize the decorrelated log-domain gains.
*
*
* Input:
* -lpcGain : uncorrelated LPC gains.
*
@ -227,10 +203,7 @@ int16_t WebRtcIsac_DecorrelateLPGain(
* -idx : quantization indices
* -lpcGain : quantized value of the inpt.
*/
double WebRtcIsac_QuantizeLpcGain(
double* lpGains,
int* idx);
double WebRtcIsac_QuantizeLpcGain(double* lpGains, int* idx);
/******************************************************************************
* WebRtcIsac_DequantizeLpcGain()
@ -243,10 +216,7 @@ double WebRtcIsac_QuantizeLpcGain(
* Output:
* -lpcGains : quantized values of the given parametes.
*/
int16_t WebRtcIsac_DequantizeLpcGain(
const int* idx,
double* lpGains);
int16_t WebRtcIsac_DequantizeLpcGain(const int* idx, double* lpGains);
/******************************************************************************
* WebRtcIsac_CorrelateLpcGain()
@ -259,10 +229,7 @@ int16_t WebRtcIsac_DequantizeLpcGain(
* Output:
* -out : correlated parameters.
*/
int16_t WebRtcIsac_CorrelateLpcGain(
const double* data,
double* out);
int16_t WebRtcIsac_CorrelateLpcGain(const double* data, double* out);
/******************************************************************************
* WebRtcIsac_AddMeanToLinearDomain()
@ -275,8 +242,6 @@ int16_t WebRtcIsac_CorrelateLpcGain(
* Output:
* -lpcGain : LPC gain in normal domain.
*/
int16_t WebRtcIsac_AddMeanToLinearDomain(
double* lpcGains);
int16_t WebRtcIsac_AddMeanToLinearDomain(double* lpcGains);
#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENCODE_LPC_SWB_H_

56
modules/audio_coding/codecs/isac/main/source/entropy_coding.h
View File

@ -46,8 +46,11 @@
* Return value : < 0 if an error occures
* 0 if succeeded.
*/
int WebRtcIsac_DecodeSpec(Bitstr* streamdata, int16_t AvgPitchGain_Q12,
enum ISACBand band, double* fr, double* fi);
int WebRtcIsac_DecodeSpec(Bitstr* streamdata,
int16_t AvgPitchGain_Q12,
enum ISACBand band,
double* fr,
double* fi);
/******************************************************************************
* WebRtcIsac_EncodeSpec()
@ -72,24 +75,31 @@ int WebRtcIsac_DecodeSpec(Bitstr* streamdata, int16_t AvgPitchGain_Q12,
* Return value : < 0 if an error occures
* 0 if succeeded.
*/
int WebRtcIsac_EncodeSpec(const int16_t* fr, const int16_t* fi,
int16_t AvgPitchGain_Q12, enum ISACBand band,
int WebRtcIsac_EncodeSpec(const int16_t* fr,
const int16_t* fi,
int16_t AvgPitchGain_Q12,
enum ISACBand band,
Bitstr* streamdata);
/* decode & dequantize LPC Coef */
int WebRtcIsac_DecodeLpcCoef(Bitstr* streamdata, double* LPCCoef);
int WebRtcIsac_DecodeLpcCoefUB(Bitstr* streamdata, double* lpcVecs,
int WebRtcIsac_DecodeLpcCoefUB(Bitstr* streamdata,
double* lpcVecs,
double* percepFilterGains,
int16_t bandwidth);
int WebRtcIsac_DecodeLpc(Bitstr* streamdata, double* LPCCoef_lo,
int WebRtcIsac_DecodeLpc(Bitstr* streamdata,
double* LPCCoef_lo,
double* LPCCoef_hi);
/* quantize & code LPC Coef */
void WebRtcIsac_EncodeLpcLb(double* LPCCoef_lo, double* LPCCoef_hi,
Bitstr* streamdata, IsacSaveEncoderData* encData);
void WebRtcIsac_EncodeLpcLb(double* LPCCoef_lo,
double* LPCCoef_hi,
Bitstr* streamdata,
IsacSaveEncoderData* encData);
void WebRtcIsac_EncodeLpcGainLb(double* LPCCoef_lo, double* LPCCoef_hi,
void WebRtcIsac_EncodeLpcGainLb(double* LPCCoef_lo,
double* LPCCoef_hi,
Bitstr* streamdata,
IsacSaveEncoderData* encData);
@ -126,7 +136,8 @@ void WebRtcIsac_EncodeLpcGainLb(double* LPCCoef_lo, double* LPCCoef_hi,
* Return value : 0 if encoding is successful,
* <0 if failed to encode.
*/
int16_t WebRtcIsac_EncodeLpcUB(double* lpcCoeff, Bitstr* streamdata,
int16_t WebRtcIsac_EncodeLpcUB(double* lpcCoeff,
Bitstr* streamdata,
double* interpolLPCCoeff,
int16_t bandwidth,
ISACUBSaveEncDataStruct* encData);
@ -184,9 +195,9 @@ void WebRtcIsac_EncodePitchLag(double* PitchLags,
Bitstr* streamdata,
IsacSaveEncoderData* encData);
int WebRtcIsac_DecodePitchGain(Bitstr* streamdata,
int16_t* PitchGain_Q12);
int WebRtcIsac_DecodePitchLag(Bitstr* streamdata, int16_t* PitchGain_Q12,
int WebRtcIsac_DecodePitchGain(Bitstr* streamdata, int16_t* PitchGain_Q12);
int WebRtcIsac_DecodePitchLag(Bitstr* streamdata,
int16_t* PitchGain_Q12,
double* PitchLag);
int WebRtcIsac_DecodeFrameLen(Bitstr* streamdata, int16_t* framelength);
@ -200,10 +211,10 @@ void WebRtcIsac_Poly2Rc(double* a, int N, double* RC);
/* Step-up */
void WebRtcIsac_Rc2Poly(double* RC, int N, double* a);
void WebRtcIsac_TranscodeLPCCoef(double* LPCCoef_lo, double* LPCCoef_hi,
void WebRtcIsac_TranscodeLPCCoef(double* LPCCoef_lo,
double* LPCCoef_hi,
int* index_g);
/******************************************************************************
* WebRtcIsac_EncodeLpcGainUb()
* Encode LPC gains of sub-Frames.
@ -220,10 +231,10 @@ void WebRtcIsac_TranscodeLPCCoef(double* LPCCoef_lo, double* LPCCoef_hi,
* - lpcGainIndex : quantization indices for lpc gains, these will
* be stored to be used for FEC.
*/
void WebRtcIsac_EncodeLpcGainUb(double* lpGains, Bitstr* streamdata,
void WebRtcIsac_EncodeLpcGainUb(double* lpGains,
Bitstr* streamdata,
int* lpcGainIndex);
/******************************************************************************
* WebRtcIsac_EncodeLpcGainUb()
* Store LPC gains of sub-Frames in 'streamdata'.
@ -239,7 +250,6 @@ void WebRtcIsac_EncodeLpcGainUb(double* lpGains, Bitstr* streamdata,
*/
void WebRtcIsac_StoreLpcGainUb(double* lpGains, Bitstr* streamdata);
/******************************************************************************
* WebRtcIsac_DecodeLpcGainUb()
* Decode the LPC gain of sub-frames.
@ -257,7 +267,6 @@ void WebRtcIsac_StoreLpcGainUb(double* lpGains, Bitstr* streamdata);
*/
int16_t WebRtcIsac_DecodeLpcGainUb(double* lpGains, Bitstr* streamdata);
/******************************************************************************
* WebRtcIsac_EncodeBandwidth()
* Encode if the bandwidth of encoded audio is 0-12 kHz or 0-16 kHz.
@ -277,7 +286,6 @@ int16_t WebRtcIsac_DecodeLpcGainUb(double* lpGains, Bitstr* streamdata);
int16_t WebRtcIsac_EncodeBandwidth(enum ISACBandwidth bandwidth,
Bitstr* streamData);
/******************************************************************************
* WebRtcIsac_DecodeBandwidth()
* Decode the bandwidth of the encoded audio, i.e. if the bandwidth is 0-12 kHz
@ -298,7 +306,6 @@ int16_t WebRtcIsac_EncodeBandwidth(enum ISACBandwidth bandwidth,
int16_t WebRtcIsac_DecodeBandwidth(Bitstr* streamData,
enum ISACBandwidth* bandwidth);
/******************************************************************************
* WebRtcIsac_EncodeJitterInfo()
* Decode the jitter information.
@ -316,9 +323,7 @@ int16_t WebRtcIsac_DecodeBandwidth(Bitstr* streamData,
* Return value : 0 if succeeded.
* <0 if failed.
*/
int16_t WebRtcIsac_EncodeJitterInfo(int32_t jitterIndex,
Bitstr* streamData);
int16_t WebRtcIsac_EncodeJitterInfo(int32_t jitterIndex, Bitstr* streamData);
/******************************************************************************
* WebRtcIsac_DecodeJitterInfo()
@ -337,7 +342,6 @@ int16_t WebRtcIsac_EncodeJitterInfo(int32_t jitterIndex,
* Return value : 0 if succeeded.
* <0 if failed.
*/
int16_t WebRtcIsac_DecodeJitterInfo(Bitstr* streamData,
int32_t* jitterInfo);
int16_t WebRtcIsac_DecodeJitterInfo(Bitstr* streamData, int32_t* jitterInfo);
#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_ */

12
modules/audio_coding/codecs/isac/main/source/fft.h
View File

@ -34,10 +34,12 @@
/* double precision routine */
int WebRtcIsac_Fftns (unsigned int ndim, const int dims[], double Re[], double Im[],
int isign, double scaling, FFTstr *fftstate);
int WebRtcIsac_Fftns(unsigned int ndim,
const int dims[],
double Re[],
double Im[],
int isign,
double scaling,
FFTstr* fftstate);
#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FFT_H_ */

21
modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
View File

@ -35,15 +35,13 @@ class IsacTest : public ::testing::Test {
uint8_t bitstream_small_[7]; // Simulate sync packets.
};
IsacTest::IsacTest()
: isac_codec_(NULL) {
}
IsacTest::IsacTest() : isac_codec_(NULL) {}
void IsacTest::SetUp() {
// Read some samples from a speech file, to be used in the encode test.
FILE* input_file;
const std::string file_name =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
input_file = fopen(file_name.c_str(), "rb");
ASSERT_TRUE(input_file != NULL);
ASSERT_EQ(kIsacNumberOfSamples,
@ -69,7 +67,8 @@ TEST_F(IsacTest, IsacFreeFail) {
TEST_F(IsacTest, IsacCreateFree) {
EXPECT_EQ(0, WebRtcIsac_Create(&isac_codec_));
EXPECT_TRUE(isac_codec_ != NULL);
EXPECT_EQ(0, WebRtcIsac_Free(isac_codec_));}
EXPECT_EQ(0, WebRtcIsac_Free(isac_codec_));
}
TEST_F(IsacTest, IsacUpdateBWE) {
// Create encoder memory.
@ -86,17 +85,17 @@ TEST_F(IsacTest, IsacUpdateBWE) {
12345, 56789));
// Encode 60 ms of data (needed to create a first packet).
encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
EXPECT_EQ(0, encoded_bytes);
encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
EXPECT_EQ(0, encoded_bytes);
encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
EXPECT_EQ(0, encoded_bytes);
encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
EXPECT_EQ(0, encoded_bytes);
encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
EXPECT_EQ(0, encoded_bytes);
encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
EXPECT_GT(encoded_bytes, 0);
// Call to update bandwidth estimator with real data.

36
modules/audio_coding/codecs/isac/main/source/lpc_analysis.h
View File

@ -21,24 +21,26 @@
#include "modules/audio_coding/codecs/isac/main/source/settings.h"
#include "modules/audio_coding/codecs/isac/main/source/structs.h"
void WebRtcIsac_GetLpcCoefLb(double *inLo, double *inHi, MaskFiltstr *maskdata,
double signal_noise_ratio, const int16_t *pitchGains_Q12,
double *lo_coeff, double *hi_coeff);
void WebRtcIsac_GetLpcCoefLb(double* inLo,
double* inHi,
MaskFiltstr* maskdata,
double signal_noise_ratio,
const int16_t* pitchGains_Q12,
double* lo_coeff,
double* hi_coeff);
void WebRtcIsac_GetLpcGain(
double signal_noise_ratio,
const double* filtCoeffVecs,
int numVecs,
double* gain,
double corrLo[][UB_LPC_ORDER + 1],
const double* varscale);
void WebRtcIsac_GetLpcGain(double signal_noise_ratio,
const double* filtCoeffVecs,
int numVecs,
double* gain,
double corrLo[][UB_LPC_ORDER + 1],
const double* varscale);
void WebRtcIsac_GetLpcCoefUb(
double* inSignal,
MaskFiltstr* maskdata,
double* lpCoeff,
double corr[][UB_LPC_ORDER + 1],
double* varscale,
int16_t bandwidth);
void WebRtcIsac_GetLpcCoefUb(double* inSignal,
MaskFiltstr* maskdata,
double* lpCoeff,
double corr[][UB_LPC_ORDER + 1],
double* varscale,
int16_t bandwidth);
#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYIS_H_ */

2
modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h
View File

@ -46,4 +46,4 @@ extern const uint16_t* WebRtcIsac_kLpcGainCdfMat[SUBFRAMES];
extern const double WebRtcIsac_kLpcGainDecorrMat[SUBFRAMES][SUBFRAMES];
#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_GAIN_SWB_TABLES_H_
#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_GAIN_SWB_TABLES_H_

26
modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h
View File

@ -26,22 +26,22 @@ extern const double WebRtcIsac_kMeanLarUb12[UB_LPC_ORDER];
extern const double WebRtcIsac_kMeanLpcGain;
extern const double WebRtcIsac_kIntraVecDecorrMatUb12[UB_LPC_ORDER][UB_LPC_ORDER];
extern const double WebRtcIsac_kIntraVecDecorrMatUb12[UB_LPC_ORDER]
[UB_LPC_ORDER];
extern const double WebRtcIsac_kInterVecDecorrMatUb12
[UB_LPC_VEC_PER_FRAME][UB_LPC_VEC_PER_FRAME];
extern const double WebRtcIsac_kInterVecDecorrMatUb12[UB_LPC_VEC_PER_FRAME]
[UB_LPC_VEC_PER_FRAME];
extern const double WebRtcIsac_kLpcShapeQStepSizeUb12;
extern const double WebRtcIsac_kLpcShapeLeftRecPointUb12
[UB_LPC_ORDER*UB_LPC_VEC_PER_FRAME];
extern const double
WebRtcIsac_kLpcShapeLeftRecPointUb12[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
extern const int16_t
WebRtcIsac_kLpcShapeNumRecPointUb12[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
extern const int16_t WebRtcIsac_kLpcShapeNumRecPointUb12
[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
extern const uint16_t WebRtcIsac_kLpcShapeEntropySearchUb12
[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
extern const uint16_t
WebRtcIsac_kLpcShapeEntropySearchUb12[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
extern const uint16_t WebRtcIsac_kLpcShapeCdfVec0Ub12[14];
@ -59,7 +59,7 @@ extern const uint16_t WebRtcIsac_kLpcShapeCdfVec6Ub12[33];
extern const uint16_t WebRtcIsac_kLpcShapeCdfVec7Ub12[49];
extern const uint16_t* WebRtcIsac_kLpcShapeCdfMatUb12
[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
extern const uint16_t*
WebRtcIsac_kLpcShapeCdfMatUb12[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB12_TABLES_H_
#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB12_TABLES_H_

26
modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h
View File

@ -24,10 +24,11 @@
extern const double WebRtcIsac_kMeanLarUb16[UB_LPC_ORDER];
extern const double WebRtcIsac_kIintraVecDecorrMatUb16[UB_LPC_ORDER][UB_LPC_ORDER];
extern const double WebRtcIsac_kIintraVecDecorrMatUb16[UB_LPC_ORDER]
[UB_LPC_ORDER];
extern const double WebRtcIsac_kInterVecDecorrMatUb16
[UB16_LPC_VEC_PER_FRAME][UB16_LPC_VEC_PER_FRAME];
extern const double WebRtcIsac_kInterVecDecorrMatUb16[UB16_LPC_VEC_PER_FRAME]
[UB16_LPC_VEC_PER_FRAME];
extern const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub16[14];
@ -61,18 +62,19 @@ extern const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub165[34];
extern const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub166[71];
extern const uint16_t* WebRtcIsac_kLpcShapeCdfMatUb16
[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
extern const uint16_t*
WebRtcIsac_kLpcShapeCdfMatUb16[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
extern const double WebRtcIsac_kLpcShapeLeftRecPointUb16
[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
extern const double
WebRtcIsac_kLpcShapeLeftRecPointUb16[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
extern const int16_t WebRtcIsac_kLpcShapeNumRecPointUb16
[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
extern const int16_t
WebRtcIsac_kLpcShapeNumRecPointUb16[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
extern const uint16_t WebRtcIsac_kLpcShapeEntropySearchUb16
[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
extern const uint16_t
WebRtcIsac_kLpcShapeEntropySearchUb16[UB_LPC_ORDER *
UB16_LPC_VEC_PER_FRAME];
extern const double WebRtcIsac_kLpcShapeQStepSizeUb16;
#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB16_TABLES_H_
#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB16_TABLES_H_

34
modules/audio_coding/codecs/isac/main/source/lpc_tables.h
View File

@ -22,27 +22,27 @@
#include "modules/audio_coding/codecs/isac/main/source/settings.h"
#define KLT_STEPSIZE 1.00000000
#define KLT_NUM_AVG_GAIN 0
#define KLT_NUM_AVG_SHAPE 0
#define KLT_NUM_MODELS 3
#define LPC_GAIN_SCALE 4.000f
#define LPC_LOBAND_SCALE 2.100f
#define LPC_LOBAND_ORDER ORDERLO
#define LPC_HIBAND_SCALE 0.450f
#define LPC_HIBAND_ORDER ORDERHI
#define LPC_GAIN_ORDER 2
#define KLT_STEPSIZE 1.00000000
#define KLT_NUM_AVG_GAIN 0
#define KLT_NUM_AVG_SHAPE 0
#define KLT_NUM_MODELS 3
#define LPC_GAIN_SCALE 4.000f
#define LPC_LOBAND_SCALE 2.100f
#define LPC_LOBAND_ORDER ORDERLO
#define LPC_HIBAND_SCALE 0.450f
#define LPC_HIBAND_ORDER ORDERHI
#define LPC_GAIN_ORDER 2
#define LPC_SHAPE_ORDER (LPC_LOBAND_ORDER + LPC_HIBAND_ORDER)
#define LPC_SHAPE_ORDER (LPC_LOBAND_ORDER + LPC_HIBAND_ORDER)
#define KLT_ORDER_GAIN (LPC_GAIN_ORDER * SUBFRAMES)
#define KLT_ORDER_SHAPE (LPC_SHAPE_ORDER * SUBFRAMES)
#define KLT_ORDER_GAIN (LPC_GAIN_ORDER * SUBFRAMES)
#define KLT_ORDER_SHAPE (LPC_SHAPE_ORDER * SUBFRAMES)
/* cdf array for model indicator */
extern const uint16_t WebRtcIsac_kQKltModelCdf[KLT_NUM_MODELS+1];
extern const uint16_t WebRtcIsac_kQKltModelCdf[KLT_NUM_MODELS + 1];
/* pointer to cdf array for model indicator */
extern const uint16_t *WebRtcIsac_kQKltModelCdfPtr[1];
extern const uint16_t* WebRtcIsac_kQKltModelCdfPtr[1];
/* initial cdf index for decoder of model indicator */
extern const uint16_t WebRtcIsac_kQKltModelInitIndex[1];
@ -78,9 +78,9 @@ extern const uint16_t WebRtcIsac_kQKltCdfGain[404];
extern const uint16_t WebRtcIsac_kQKltCdfShape[686];
/* pointers to cdf tables for quantizer indices */
extern const uint16_t *WebRtcIsac_kQKltCdfPtrGain[12];
extern const uint16_t* WebRtcIsac_kQKltCdfPtrGain[12];
extern const uint16_t *WebRtcIsac_kQKltCdfPtrShape[108];
extern const uint16_t* WebRtcIsac_kQKltCdfPtrShape[108];
/* left KLT transforms */
extern const double WebRtcIsac_kKltT1Gain[4];

6
modules/audio_coding/codecs/isac/main/source/os_specific_inline.h
View File

@ -8,7 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
@ -24,11 +23,12 @@ static __inline long int WebRtcIsac_lrint(double x_dbl) {
__asm {
fld x_dbl
fistp x_int
};
}
;
return x_int;
}
#else // Do a slow but correct implementation of lrint
#else // Do a slow but correct implementation of lrint
static __inline long int WebRtcIsac_lrint(double x_dbl) {
long int x_int;

11
modules/audio_coding/codecs/isac/main/source/pitch_estimator.h
View File

@ -22,10 +22,11 @@
#include "modules/audio_coding/codecs/isac/main/source/structs.h"
void WebRtcIsac_PitchAnalysis(const double *in, /* PITCH_FRAME_LEN samples */
double *out, /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
PitchAnalysisStruct *State,
double *lags,
double *gains);
void WebRtcIsac_PitchAnalysis(
const double* in, /* PITCH_FRAME_LEN samples */
double* out, /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
PitchAnalysisStruct* State,
double* lags,
double* gains);
#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_ */

9
modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h
View File

@ -11,7 +11,8 @@
/*
* pitch_gain_tables.h
*
* This file contains tables for the pitch filter side-info in the entropy coder.
* This file contains tables for the pitch filter side-info in the entropy
* coder.
*
*/
@ -20,8 +21,10 @@
#include "typedefs.h" // NOLINT(build/include)
/* header file for coding tables for the pitch filter side-info in the entropy coder */
/********************* Pitch Filter Gain Coefficient Tables ************************/
/* header file for coding tables for the pitch filter side-info in the entropy
* coder */
/********************* Pitch Filter Gain Coefficient Tables
* ************************/
/* cdf for quantized pitch filter gains */
extern const uint16_t WebRtcIsac_kQPitchGainCdf[255];

17
modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h
View File

@ -11,7 +11,8 @@
/*
* pitch_lag_tables.h
*
* This file contains tables for the pitch filter side-info in the entropy coder.
* This file contains tables for the pitch filter side-info in the entropy
* coder.
*
*/
@ -19,8 +20,10 @@
#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_LAG_TABLES_H_
#include "typedefs.h" // NOLINT(build/include)
/* header file for coding tables for the pitch filter side-info in the entropy coder */
/********************* Pitch Filter Lag Coefficient Tables ************************/
/* header file for coding tables for the pitch filter side-info in the entropy
* coder */
/********************* Pitch Filter Lag Coefficient Tables
* ************************/
/* tables for use with small pitch gain */
@ -30,7 +33,7 @@ extern const uint16_t WebRtcIsac_kQPitchLagCdf2Lo[20];
extern const uint16_t WebRtcIsac_kQPitchLagCdf3Lo[2];
extern const uint16_t WebRtcIsac_kQPitchLagCdf4Lo[10];
extern const uint16_t *WebRtcIsac_kQPitchLagCdfPtrLo[4];
extern const uint16_t* WebRtcIsac_kQPitchLagCdfPtrLo[4];
/* size of first cdf table */
extern const uint16_t WebRtcIsac_kQPitchLagCdfSizeLo[1];
@ -49,7 +52,6 @@ extern const double WebRtcIsac_kQMeanLag4Lo[9];
extern const double WebRtcIsac_kQPitchLagStepsizeLo;
/* tables for use with medium pitch gain */
/* cdfs for quantized pitch lags */
@ -58,7 +60,7 @@ extern const uint16_t WebRtcIsac_kQPitchLagCdf2Mid[36];
extern const uint16_t WebRtcIsac_kQPitchLagCdf3Mid[2];
extern const uint16_t WebRtcIsac_kQPitchLagCdf4Mid[20];
extern const uint16_t *WebRtcIsac_kQPitchLagCdfPtrMid[4];
extern const uint16_t* WebRtcIsac_kQPitchLagCdfPtrMid[4];
/* size of first cdf table */
extern const uint16_t WebRtcIsac_kQPitchLagCdfSizeMid[1];
@ -77,7 +79,6 @@ extern const double WebRtcIsac_kQMeanLag4Mid[19];
extern const double WebRtcIsac_kQPitchLagStepsizeMid;
/* tables for use with large pitch gain */
/* cdfs for quantized pitch lags */
@ -86,7 +87,7 @@ extern const uint16_t WebRtcIsac_kQPitchLagCdf2Hi[68];
extern const uint16_t WebRtcIsac_kQPitchLagCdf3Hi[2];
extern const uint16_t WebRtcIsac_kQPitchLagCdf4Hi[35];
extern const uint16_t *WebRtcIsac_kQPitchLagCdfPtrHi[4];
extern const uint16_t* WebRtcIsac_kQPitchLagCdfPtrHi[4];
/* size of first cdf table */
extern const uint16_t WebRtcIsac_kQPitchLagCdfSizeHi[1];

226
modules/audio_coding/codecs/isac/main/source/settings.h
View File

@ -19,187 +19,181 @@
#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_
/* sampling frequency (Hz) */
#define FS 16000
#define FS 16000
/* number of samples per frame (either 320 (20ms), 480 (30ms) or 960 (60ms)) */
#define INITIAL_FRAMESAMPLES 960
#define INITIAL_FRAMESAMPLES 960
#define MAXFFTSIZE 2048
#define NFACTOR 11
/* do not modify the following; this will have to be modified if we
* have a 20ms framesize option */
/**********************************************************************/
/* miliseconds */
#define FRAMESIZE 30
#define FRAMESIZE 30
/* number of samples per frame processed in the encoder, 480 */
#define FRAMESAMPLES 480 /* ((FRAMESIZE*FS)/1000) */
#define FRAMESAMPLES_HALF 240
#define FRAMESAMPLES_QUARTER 120
#define FRAMESAMPLES 480 /* ((FRAMESIZE*FS)/1000) */
#define FRAMESAMPLES_HALF 240
#define FRAMESAMPLES_QUARTER 120
/**********************************************************************/
/* max number of samples per frame (= 60 ms frame) */
#define MAX_FRAMESAMPLES 960
#define MAX_SWBFRAMESAMPLES (MAX_FRAMESAMPLES * 2)
#define MAX_FRAMESAMPLES 960
#define MAX_SWBFRAMESAMPLES (MAX_FRAMESAMPLES * 2)
/* number of samples per 10ms frame */
#define FRAMESAMPLES_10ms ((10*FS)/1000)
#define SWBFRAMESAMPLES_10ms (FRAMESAMPLES_10ms * 2)
#define FRAMESAMPLES_10ms ((10 * FS) / 1000)
#define SWBFRAMESAMPLES_10ms (FRAMESAMPLES_10ms * 2)
/* number of samples in 30 ms frame */
#define FRAMESAMPLES_30ms 480
#define FRAMESAMPLES_30ms 480
/* number of subframes */
#define SUBFRAMES 6
#define SUBFRAMES 6
/* length of a subframe */
#define UPDATE 80
#define UPDATE 80
/* length of half a subframe (low/high band) */
#define HALF_SUBFRAMELEN (UPDATE/2)
#define HALF_SUBFRAMELEN (UPDATE / 2)
/* samples of look ahead (in a half-band, so actually
* half the samples of look ahead @ FS) */
#define QLOOKAHEAD 24 /* 3 ms */
#define QLOOKAHEAD 24 /* 3 ms */
/* order of AR model in spectral entropy coder */
#define AR_ORDER 6
#define AR_ORDER 6
/* order of LP model in spectral entropy coder */
#define LP_ORDER 0
#define LP_ORDER 0
/* window length (masking analysis) */
#define WINLEN 256
#define WINLEN 256
/* order of low-band pole filter used to approximate masking curve */
#define ORDERLO 12
#define ORDERLO 12
/* order of hi-band pole filter used to approximate masking curve */
#define ORDERHI 6
#define UB_LPC_ORDER 4
#define UB_LPC_VEC_PER_FRAME 2
#define UB16_LPC_VEC_PER_FRAME 4
#define UB_ACTIVE_SUBFRAMES 2
#define UB_MAX_LPC_ORDER 6
#define UB_INTERPOL_SEGMENTS 1
#define UB16_INTERPOL_SEGMENTS 3
#define LB_TOTAL_DELAY_SAMPLES 48
enum ISACBandwidth {isac8kHz = 8, isac12kHz = 12, isac16kHz = 16};
enum ISACBand {kIsacLowerBand = 0, kIsacUpperBand12 = 1, kIsacUpperBand16 = 2};
enum IsacSamplingRate {kIsacWideband = 16, kIsacSuperWideband = 32};
#define UB_LPC_GAIN_DIM SUBFRAMES
#define FB_STATE_SIZE_WORD32 6
#define ORDERHI 6
#define UB_LPC_ORDER 4
#define UB_LPC_VEC_PER_FRAME 2
#define UB16_LPC_VEC_PER_FRAME 4
#define UB_ACTIVE_SUBFRAMES 2
#define UB_MAX_LPC_ORDER 6
#define UB_INTERPOL_SEGMENTS 1
#define UB16_INTERPOL_SEGMENTS 3
#define LB_TOTAL_DELAY_SAMPLES 48
enum ISACBandwidth { isac8kHz = 8, isac12kHz = 12, isac16kHz = 16 };
enum ISACBand {
kIsacLowerBand = 0,
kIsacUpperBand12 = 1,
kIsacUpperBand16 = 2
};
enum IsacSamplingRate { kIsacWideband = 16, kIsacSuperWideband = 32 };
#define UB_LPC_GAIN_DIM SUBFRAMES
#define FB_STATE_SIZE_WORD32 6
/* order for post_filter_bank */
#define POSTQORDER 3
#define POSTQORDER 3
/* order for pre-filterbank */
#define QORDER 3
#define QORDER 3
/* another order */
#define QORDER_ALL (POSTQORDER+QORDER-1)
#define QORDER_ALL (POSTQORDER + QORDER - 1)
/* for decimator */
#define ALLPASSSECTIONS 2
#define ALLPASSSECTIONS 2
/* array size for byte stream in number of bytes. */
/* The old maximum size still needed for the decoding */
#define STREAM_SIZE_MAX 600
#define STREAM_SIZE_MAX_30 200 /* 200 bytes=53.4 kbps @ 30 ms.framelength */
#define STREAM_SIZE_MAX_60 400 /* 400 bytes=53.4 kbps @ 60 ms.framelength */
#define STREAM_SIZE_MAX 600
#define STREAM_SIZE_MAX_30 200 /* 200 bytes=53.4 kbps @ 30 ms.framelength */
#define STREAM_SIZE_MAX_60 400 /* 400 bytes=53.4 kbps @ 60 ms.framelength */
/* storage size for bit counts */
#define BIT_COUNTER_SIZE 30
#define BIT_COUNTER_SIZE 30
/* maximum order of any AR model or filter */
#define MAX_AR_MODEL_ORDER 12//50
#define MAX_AR_MODEL_ORDER 12 // 50
/* For pitch analysis */
#define PITCH_FRAME_LEN (FRAMESAMPLES_HALF) /* 30 ms */
#define PITCH_MAX_LAG 140 /* 57 Hz */
#define PITCH_MIN_LAG 20 /* 400 Hz */
#define PITCH_MAX_GAIN 0.45
#define PITCH_MAX_GAIN_06 0.27 /* PITCH_MAX_GAIN*0.6 */
#define PITCH_MAX_GAIN_Q12 1843
#define PITCH_LAG_SPAN2 (PITCH_MAX_LAG/2-PITCH_MIN_LAG/2+5)
#define PITCH_CORR_LEN2 60 /* 15 ms */
#define PITCH_CORR_STEP2 (PITCH_FRAME_LEN/4)
#define PITCH_BW 11 /* half the band width of correlation surface */
#define PITCH_SUBFRAMES 4
#define PITCH_GRAN_PER_SUBFRAME 5
#define PITCH_SUBFRAME_LEN (PITCH_FRAME_LEN/PITCH_SUBFRAMES)
#define PITCH_UPDATE (PITCH_SUBFRAME_LEN/PITCH_GRAN_PER_SUBFRAME)
#define PITCH_FRAME_LEN (FRAMESAMPLES_HALF) /* 30 ms */
#define PITCH_MAX_LAG 140 /* 57 Hz */
#define PITCH_MIN_LAG 20 /* 400 Hz */
#define PITCH_MAX_GAIN 0.45
#define PITCH_MAX_GAIN_06 0.27 /* PITCH_MAX_GAIN*0.6 */
#define PITCH_MAX_GAIN_Q12 1843
#define PITCH_LAG_SPAN2 (PITCH_MAX_LAG / 2 - PITCH_MIN_LAG / 2 + 5)
#define PITCH_CORR_LEN2 60 /* 15 ms */
#define PITCH_CORR_STEP2 (PITCH_FRAME_LEN / 4)
#define PITCH_BW 11 /* half the band width of correlation surface */
#define PITCH_SUBFRAMES 4
#define PITCH_GRAN_PER_SUBFRAME 5
#define PITCH_SUBFRAME_LEN (PITCH_FRAME_LEN / PITCH_SUBFRAMES)
#define PITCH_UPDATE (PITCH_SUBFRAME_LEN / PITCH_GRAN_PER_SUBFRAME)
/* maximum number of peaks to be examined in correlation surface */
#define PITCH_MAX_NUM_PEAKS 10
#define PITCH_PEAK_DECAY 0.85
#define PITCH_MAX_NUM_PEAKS 10
#define PITCH_PEAK_DECAY 0.85
/* For weighting filter */
#define PITCH_WLPCORDER 6
#define PITCH_WLPCWINLEN PITCH_FRAME_LEN
#define PITCH_WLPCASYM 0.3 /* asymmetry parameter */
#define PITCH_WLPCBUFLEN PITCH_WLPCWINLEN
#define PITCH_WLPCORDER 6
#define PITCH_WLPCWINLEN PITCH_FRAME_LEN
#define PITCH_WLPCASYM 0.3 /* asymmetry parameter */
#define PITCH_WLPCBUFLEN PITCH_WLPCWINLEN
/* For pitch filter */
/* Extra 50 for fraction and LP filters */
#define PITCH_BUFFSIZE (PITCH_MAX_LAG + 50)
#define PITCH_INTBUFFSIZE (PITCH_FRAME_LEN+PITCH_BUFFSIZE)
#define PITCH_BUFFSIZE (PITCH_MAX_LAG + 50)
#define PITCH_INTBUFFSIZE (PITCH_FRAME_LEN + PITCH_BUFFSIZE)
/* Max rel. step for interpolation */
#define PITCH_UPSTEP 1.5
#define PITCH_UPSTEP 1.5
/* Max rel. step for interpolation */
#define PITCH_DOWNSTEP 0.67
#define PITCH_FRACS 8
#define PITCH_FRACORDER 9
#define PITCH_DAMPORDER 5
#define PITCH_FILTDELAY 1.5f
#define PITCH_DOWNSTEP 0.67
#define PITCH_FRACS 8
#define PITCH_FRACORDER 9
#define PITCH_DAMPORDER 5
#define PITCH_FILTDELAY 1.5f
/* stepsize for quantization of the pitch Gain */
#define PITCH_GAIN_STEPSIZE 0.125
#define PITCH_GAIN_STEPSIZE 0.125
/* Order of high pass filter */
#define HPORDER 2
#define HPORDER 2
/* some mathematical constants */
/* log2(exp) */
#define LOG2EXP 1.44269504088896
#define PI 3.14159265358979
#define LOG2EXP 1.44269504088896
#define PI 3.14159265358979
/* Maximum number of iterations allowed to limit payload size */
#define MAX_PAYLOAD_LIMIT_ITERATION 5
#define MAX_PAYLOAD_LIMIT_ITERATION 5
/* Redundant Coding */
#define RCU_BOTTLENECK_BPS 16000
#define RCU_TRANSCODING_SCALE 0.40f
#define RCU_TRANSCODING_SCALE_INVERSE 2.5f
#define RCU_BOTTLENECK_BPS 16000
#define RCU_TRANSCODING_SCALE 0.40f
#define RCU_TRANSCODING_SCALE_INVERSE 2.5f
#define RCU_TRANSCODING_SCALE_UB 0.50f
#define RCU_TRANSCODING_SCALE_UB_INVERSE 2.0f
#define RCU_TRANSCODING_SCALE_UB 0.50f
#define RCU_TRANSCODING_SCALE_UB_INVERSE 2.0f
/* Define Error codes */
/* 6000 General */
#define ISAC_MEMORY_ALLOCATION_FAILED 6010
#define ISAC_MODE_MISMATCH 6020
#define ISAC_DISALLOWED_BOTTLENECK 6030
#define ISAC_DISALLOWED_FRAME_LENGTH 6040
#define ISAC_UNSUPPORTED_SAMPLING_FREQUENCY 6050
#define ISAC_MEMORY_ALLOCATION_FAILED 6010
#define ISAC_MODE_MISMATCH 6020
#define ISAC_DISALLOWED_BOTTLENECK 6030
#define ISAC_DISALLOWED_FRAME_LENGTH 6040
#define ISAC_UNSUPPORTED_SAMPLING_FREQUENCY 6050
/* 6200 Bandwidth estimator */
#define ISAC_RANGE_ERROR_BW_ESTIMATOR 6240
#define ISAC_RANGE_ERROR_BW_ESTIMATOR 6240
/* 6400 Encoder */
#define ISAC_ENCODER_NOT_INITIATED 6410
#define ISAC_DISALLOWED_CODING_MODE 6420
#define ISAC_DISALLOWED_FRAME_MODE_ENCODER 6430
#define ISAC_DISALLOWED_BITSTREAM_LENGTH 6440
#define ISAC_PAYLOAD_LARGER_THAN_LIMIT 6450
#define ISAC_DISALLOWED_ENCODER_BANDWIDTH 6460
#define ISAC_ENCODER_NOT_INITIATED 6410
#define ISAC_DISALLOWED_CODING_MODE 6420
#define ISAC_DISALLOWED_FRAME_MODE_ENCODER 6430
#define ISAC_DISALLOWED_BITSTREAM_LENGTH 6440
#define ISAC_PAYLOAD_LARGER_THAN_LIMIT 6450
#define ISAC_DISALLOWED_ENCODER_BANDWIDTH 6460
/* 6600 Decoder */
#define ISAC_DECODER_NOT_INITIATED 6610
#define ISAC_EMPTY_PACKET 6620
#define ISAC_DISALLOWED_FRAME_MODE_DECODER 6630
#define ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH 6640
#define ISAC_RANGE_ERROR_DECODE_BANDWIDTH 6650
#define ISAC_RANGE_ERROR_DECODE_PITCH_GAIN 6660
#define ISAC_RANGE_ERROR_DECODE_PITCH_LAG 6670
#define ISAC_RANGE_ERROR_DECODE_LPC 6680
#define ISAC_RANGE_ERROR_DECODE_SPECTRUM 6690
#define ISAC_LENGTH_MISMATCH 6730
#define ISAC_RANGE_ERROR_DECODE_BANDWITH 6740
#define ISAC_DISALLOWED_BANDWIDTH_MODE_DECODER 6750
#define ISAC_DISALLOWED_LPC_MODEL 6760
#define ISAC_DECODER_NOT_INITIATED 6610
#define ISAC_EMPTY_PACKET 6620
#define ISAC_DISALLOWED_FRAME_MODE_DECODER 6630
#define ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH 6640
#define ISAC_RANGE_ERROR_DECODE_BANDWIDTH 6650
#define ISAC_RANGE_ERROR_DECODE_PITCH_GAIN 6660
#define ISAC_RANGE_ERROR_DECODE_PITCH_LAG 6670
#define ISAC_RANGE_ERROR_DECODE_LPC 6680
#define ISAC_RANGE_ERROR_DECODE_SPECTRUM 6690
#define ISAC_LENGTH_MISMATCH 6730
#define ISAC_RANGE_ERROR_DECODE_BANDWITH 6740
#define ISAC_DISALLOWED_BANDWIDTH_MODE_DECODER 6750
#define ISAC_DISALLOWED_LPC_MODEL 6760
/* 6800 Call setup formats */
#define ISAC_INCOMPATIBLE_FORMATS 6810
#define ISAC_INCOMPATIBLE_FORMATS 6810
#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_ */

15
modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h
View File

@ -11,7 +11,7 @@
/*
* spectrum_ar_model_tables.h
*
* This file contains definitions of tables with AR coefficients,
* This file contains definitions of tables with AR coefficients,
* Gain coefficients and cosine tables.
*
*/
@ -45,15 +45,15 @@ extern const uint16_t WebRtcIsac_kQArRc6Cdf[NUM_AR_RC_QUANT_BAUNDARY];
/* quantization boundary levels for reflection coefficients */
extern const int16_t WebRtcIsac_kQArBoundaryLevels[NUM_AR_RC_QUANT_BAUNDARY];
/* initial indices for AR reflection coefficient quantizer and cdf table search */
/* initial indices for AR reflection coefficient quantizer and cdf table search
*/
extern const uint16_t WebRtcIsac_kQArRcInitIndex[AR_ORDER];
/* pointers to AR cdf tables */
extern const uint16_t *WebRtcIsac_kQArRcCdfPtr[AR_ORDER];
extern const uint16_t* WebRtcIsac_kQArRcCdfPtr[AR_ORDER];
/* pointers to AR representation levels tables */
extern const int16_t *WebRtcIsac_kQArRcLevelsPtr[AR_ORDER];
extern const int16_t* WebRtcIsac_kQArRcLevelsPtr[AR_ORDER];
/******************** GAIN Coefficient Tables ***********************/
/* cdf for Gain coefficient */
@ -66,7 +66,7 @@ extern const int32_t WebRtcIsac_kQGain2Levels[18];
extern const int32_t WebRtcIsac_kQGain2BoundaryLevels[19];
/* pointer to Gain cdf table */
extern const uint16_t *WebRtcIsac_kQGainCdf_ptr[1];
extern const uint16_t* WebRtcIsac_kQGainCdf_ptr[1];
/* Gain initial index for gain quantizer and cdf table search */
extern const uint16_t WebRtcIsac_kQGainInitIndex[1];
@ -75,4 +75,5 @@ extern const uint16_t WebRtcIsac_kQGainInitIndex[1];
/* Cosine table */
extern const int16_t WebRtcIsac_kCos[6][60];
#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_ */
#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_ \
*/

368
modules/audio_coding/codecs/isac/main/source/structs.h
View File

@ -23,178 +23,166 @@
#include "typedefs.h" // NOLINT(build/include)
typedef struct Bitstreamstruct {
uint8_t stream[STREAM_SIZE_MAX];
uint32_t W_upper;
uint32_t streamval;
uint32_t stream_index;
uint8_t stream[STREAM_SIZE_MAX];
uint32_t W_upper;
uint32_t streamval;
uint32_t stream_index;
} Bitstr;
typedef struct {
double DataBufferLo[WINLEN];
double DataBufferHi[WINLEN];
double DataBufferLo[WINLEN];
double DataBufferHi[WINLEN];
double CorrBufLo[ORDERLO + 1];
double CorrBufHi[ORDERHI + 1];
double CorrBufLo[ORDERLO+1];
double CorrBufHi[ORDERHI+1];
float PreStateLoF[ORDERLO + 1];
float PreStateLoG[ORDERLO + 1];
float PreStateHiF[ORDERHI + 1];
float PreStateHiG[ORDERHI + 1];
float PostStateLoF[ORDERLO + 1];
float PostStateLoG[ORDERLO + 1];
float PostStateHiF[ORDERHI + 1];
float PostStateHiG[ORDERHI + 1];
float PreStateLoF[ORDERLO+1];
float PreStateLoG[ORDERLO+1];
float PreStateHiF[ORDERHI+1];
float PreStateHiG[ORDERHI+1];
float PostStateLoF[ORDERLO+1];
float PostStateLoG[ORDERLO+1];
float PostStateHiF[ORDERHI+1];
float PostStateHiG[ORDERHI+1];
double OldEnergy;
double OldEnergy;
} MaskFiltstr;
typedef struct {
// state vectors for each of the two analysis filters
double INSTAT1[2 * (QORDER - 1)];
double INSTAT2[2 * (QORDER - 1)];
double INSTATLA1[2 * (QORDER - 1)];
double INSTATLA2[2 * (QORDER - 1)];
double INLABUF1[QLOOKAHEAD];
double INLABUF2[QLOOKAHEAD];
//state vectors for each of the two analysis filters
double INSTAT1[2*(QORDER-1)];
double INSTAT2[2*(QORDER-1)];
double INSTATLA1[2*(QORDER-1)];
double INSTATLA2[2*(QORDER-1)];
double INLABUF1[QLOOKAHEAD];
double INLABUF2[QLOOKAHEAD];
float INSTAT1_float[2*(QORDER-1)];
float INSTAT2_float[2*(QORDER-1)];
float INSTATLA1_float[2*(QORDER-1)];
float INSTATLA2_float[2*(QORDER-1)];
float INLABUF1_float[QLOOKAHEAD];
float INLABUF2_float[QLOOKAHEAD];
float INSTAT1_float[2 * (QORDER - 1)];
float INSTAT2_float[2 * (QORDER - 1)];
float INSTATLA1_float[2 * (QORDER - 1)];
float INSTATLA2_float[2 * (QORDER - 1)];
float INLABUF1_float[QLOOKAHEAD];
float INLABUF2_float[QLOOKAHEAD];
/* High pass filter */
double HPstates[HPORDER];
float HPstates_float[HPORDER];
double HPstates[HPORDER];
float HPstates_float[HPORDER];
} PreFiltBankstr;
typedef struct {
//state vectors for each of the two analysis filters
double STATE_0_LOWER[2*POSTQORDER];
double STATE_0_UPPER[2*POSTQORDER];
// state vectors for each of the two analysis filters
double STATE_0_LOWER[2 * POSTQORDER];
double STATE_0_UPPER[2 * POSTQORDER];
/* High pass filter */
double HPstates1[HPORDER];
double HPstates2[HPORDER];
double HPstates1[HPORDER];
double HPstates2[HPORDER];
float STATE_0_LOWER_float[2*POSTQORDER];
float STATE_0_UPPER_float[2*POSTQORDER];
float STATE_0_LOWER_float[2 * POSTQORDER];
float STATE_0_UPPER_float[2 * POSTQORDER];
float HPstates1_float[HPORDER];
float HPstates2_float[HPORDER];
float HPstates1_float[HPORDER];
float HPstates2_float[HPORDER];
} PostFiltBankstr;
typedef struct {
// data buffer for pitch filter
double ubuf[PITCH_BUFFSIZE];
//data buffer for pitch filter
double ubuf[PITCH_BUFFSIZE];
// low pass state vector
double ystate[PITCH_DAMPORDER];
//low pass state vector
double ystate[PITCH_DAMPORDER];
//old lag and gain
double oldlagp[1];
double oldgainp[1];
// old lag and gain
double oldlagp[1];
double oldgainp[1];
} PitchFiltstr;
typedef struct {
// data buffer
double buffer[PITCH_WLPCBUFLEN];
//data buffer
double buffer[PITCH_WLPCBUFLEN];
// state vectors
double istate[PITCH_WLPCORDER];
double weostate[PITCH_WLPCORDER];
double whostate[PITCH_WLPCORDER];
//state vectors
double istate[PITCH_WLPCORDER];
double weostate[PITCH_WLPCORDER];
double whostate[PITCH_WLPCORDER];
//LPC window -> should be a global array because constant
double window[PITCH_WLPCWINLEN];
// LPC window -> should be a global array because constant
double window[PITCH_WLPCWINLEN];
} WeightFiltstr;
typedef struct {
// for inital estimator
double dec_buffer[PITCH_CORR_LEN2 + PITCH_CORR_STEP2 + PITCH_MAX_LAG / 2 -
PITCH_FRAME_LEN / 2 + 2];
double decimator_state[2 * ALLPASSSECTIONS + 1];
double hp_state[2];
//for inital estimator
double dec_buffer[PITCH_CORR_LEN2 + PITCH_CORR_STEP2 +
PITCH_MAX_LAG/2 - PITCH_FRAME_LEN/2+2];
double decimator_state[2*ALLPASSSECTIONS+1];
double hp_state[2];
double whitened_buf[QLOOKAHEAD];
double whitened_buf[QLOOKAHEAD];
double inbuf[QLOOKAHEAD];
double inbuf[QLOOKAHEAD];
PitchFiltstr PFstr_wght;
PitchFiltstr PFstr;
PitchFiltstr PFstr_wght;
PitchFiltstr PFstr;
WeightFiltstr Wghtstr;
} PitchAnalysisStruct;
/* Have instance of struct together with other iSAC structs */
typedef struct {
/* Previous frame length (in ms) */
int32_t prev_frame_length;
int32_t prev_frame_length;
/* Previous RTP timestamp from received
packet (in samples relative beginning) */
int32_t prev_rec_rtp_number;
int32_t prev_rec_rtp_number;
/* Send timestamp for previous packet (in ms using timeGetTime()) */
uint32_t prev_rec_send_ts;
uint32_t prev_rec_send_ts;
/* Arrival time for previous packet (in ms using timeGetTime()) */
uint32_t prev_rec_arr_ts;
uint32_t prev_rec_arr_ts;
/* rate of previous packet, derived from RTP timestamps (in bits/s) */
float prev_rec_rtp_rate;
float prev_rec_rtp_rate;
/* Time sinse the last update of the BN estimate (in ms) */
uint32_t last_update_ts;
uint32_t last_update_ts;
/* Time sinse the last reduction (in ms) */
uint32_t last_reduction_ts;
uint32_t last_reduction_ts;
/* How many times the estimate was update in the beginning */
int32_t count_tot_updates_rec;
int32_t count_tot_updates_rec;
/* The estimated bottle neck rate from there to here (in bits/s) */
int32_t rec_bw;
float rec_bw_inv;
float rec_bw_avg;
float rec_bw_avg_Q;
int32_t rec_bw;
float rec_bw_inv;
float rec_bw_avg;
float rec_bw_avg_Q;
/* The estimated mean absolute jitter value,
as seen on this side (in ms) */
float rec_jitter;
float rec_jitter_short_term;
float rec_jitter_short_term_abs;
float rec_max_delay;
float rec_max_delay_avg_Q;
float rec_jitter;
float rec_jitter_short_term;
float rec_jitter_short_term_abs;
float rec_max_delay;
float rec_max_delay_avg_Q;
/* (assumed) bitrate for headers (bps) */
float rec_header_rate;
float rec_header_rate;
/* The estimated bottle neck rate from here to there (in bits/s) */
float send_bw_avg;
float send_bw_avg;
/* The estimated mean absolute jitter value, as seen on
the other siee (in ms) */
float send_max_delay_avg;
float send_max_delay_avg;
// number of packets received since last update
int num_pkts_rec;
@ -217,35 +205,31 @@ typedef struct {
int change_to_WB;
uint32_t senderTimestamp;
uint32_t receiverTimestamp;
//enum IsacSamplingRate incomingStreamSampFreq;
uint16_t numConsecLatePkts;
float consecLatency;
int16_t inWaitLatePkts;
uint32_t senderTimestamp;
uint32_t receiverTimestamp;
// enum IsacSamplingRate incomingStreamSampFreq;
uint16_t numConsecLatePkts;
float consecLatency;
int16_t inWaitLatePkts;
IsacBandwidthInfo external_bw_info;
} BwEstimatorstr;
typedef struct {
/* boolean, flags if previous packet exceeded B.N. */
int PrevExceed;
int PrevExceed;
/* ms */
int ExceedAgo;
int ExceedAgo;
/* packets left to send in current burst */
int BurstCounter;
int BurstCounter;
/* packets */
int InitCounter;
int InitCounter;
/* ms remaining in buffer when next packet will be sent */
double StillBuffered;
} RateModel;
typedef struct {
unsigned int SpaceAlloced;
unsigned int MaxPermAlloced;
double Tmp0[MAXFFTSIZE];
@ -253,36 +237,34 @@ typedef struct {
double Tmp2[MAXFFTSIZE];
double Tmp3[MAXFFTSIZE];
int Perm[MAXFFTSIZE];
int factor [NFACTOR];
int factor[NFACTOR];
} FFTstr;
/* The following strutc is used to store data from encoding, to make it
fast and easy to construct a new bitstream with a different Bandwidth
estimate. All values (except framelength and minBytes) is double size to
handle 60 ms of data.
*/
typedef struct {
/* Used to keep track of if it is first or second part of 60 msec packet */
int startIdx;
int startIdx;
/* Frame length in samples */
int16_t framelength;
/* Pitch Gain */
int pitchGain_index[2];
int pitchGain_index[2];
/* Pitch Lag */
double meanGain[2];
int pitchIndex[PITCH_SUBFRAMES*2];
double meanGain[2];
int pitchIndex[PITCH_SUBFRAMES * 2];
/* LPC */
int LPCindex_s[108*2]; /* KLT_ORDER_SHAPE = 108 */
int LPCindex_g[12*2]; /* KLT_ORDER_GAIN = 12 */
double LPCcoeffs_lo[(ORDERLO+1)*SUBFRAMES*2];
double LPCcoeffs_hi[(ORDERHI+1)*SUBFRAMES*2];
int LPCindex_s[108 * 2]; /* KLT_ORDER_SHAPE = 108 */
int LPCindex_g[12 * 2]; /* KLT_ORDER_GAIN = 12 */
double LPCcoeffs_lo[(ORDERLO + 1) * SUBFRAMES * 2];
double LPCcoeffs_hi[(ORDERHI + 1) * SUBFRAMES * 2];
/* Encode Spec */
int16_t fre[FRAMESAMPLES];
@ -290,59 +272,54 @@ typedef struct {
int16_t AvgPitchGain[2];
/* Used in adaptive mode only */
int minBytes;
int minBytes;
} IsacSaveEncoderData;
typedef struct {
int indexLPCShape[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
double lpcGain[SUBFRAMES << 1];
int lpcGainIndex[SUBFRAMES << 1];
int indexLPCShape[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
double lpcGain[SUBFRAMES<<1];
int lpcGainIndex[SUBFRAMES<<1];
Bitstr bitStreamObj;
Bitstr bitStreamObj;
int16_t realFFT[FRAMESAMPLES_HALF];
int16_t imagFFT[FRAMESAMPLES_HALF];
} ISACUBSaveEncDataStruct;
typedef struct {
Bitstr bitstr_obj;
MaskFiltstr maskfiltstr_obj;
PreFiltBankstr prefiltbankstr_obj;
PitchFiltstr pitchfiltstr_obj;
Bitstr bitstr_obj;
MaskFiltstr maskfiltstr_obj;
PreFiltBankstr prefiltbankstr_obj;
PitchFiltstr pitchfiltstr_obj;
PitchAnalysisStruct pitchanalysisstr_obj;
FFTstr fftstr_obj;
FFTstr fftstr_obj;
IsacSaveEncoderData SaveEnc_obj;
int buffer_index;
int16_t current_framesamples;
int buffer_index;
int16_t current_framesamples;
float data_buffer_float[FRAMESAMPLES_30ms];
float data_buffer_float[FRAMESAMPLES_30ms];
int frame_nb;
double bottleneck;
int16_t new_framelength;
double s2nr;
int frame_nb;
double bottleneck;
int16_t new_framelength;
double s2nr;
/* Maximum allowed number of bits for a 30 msec packet */
int16_t payloadLimitBytes30;
int16_t payloadLimitBytes30;
/* Maximum allowed number of bits for a 30 msec packet */
int16_t payloadLimitBytes60;
int16_t payloadLimitBytes60;
/* Maximum allowed number of bits for both 30 and 60 msec packet */
int16_t maxPayloadBytes;
int16_t maxPayloadBytes;
/* Maximum allowed rate in bytes per 30 msec packet */
int16_t maxRateInBytes;
int16_t maxRateInBytes;
/*---
If set to 1 iSAC will not addapt the frame-size, if used in
channel-adaptive mode. The initial value will be used for all rates.
---*/
int16_t enforceFrameSize;
int16_t enforceFrameSize;
/*-----
This records the BWE index the encoder injected into the bit-stream.
@ -351,64 +328,53 @@ typedef struct {
a recursive procedure (WebRtcIsac_GetDownlinkBwJitIndexImpl) and has to be
called only once per each encode.
-----*/
int16_t lastBWIdx;
int16_t lastBWIdx;
} ISACLBEncStruct;
typedef struct {
Bitstr bitstr_obj;
MaskFiltstr maskfiltstr_obj;
PreFiltBankstr prefiltbankstr_obj;
FFTstr fftstr_obj;
Bitstr bitstr_obj;
MaskFiltstr maskfiltstr_obj;
PreFiltBankstr prefiltbankstr_obj;
FFTstr fftstr_obj;
ISACUBSaveEncDataStruct SaveEnc_obj;
int buffer_index;
float data_buffer_float[MAX_FRAMESAMPLES +
LB_TOTAL_DELAY_SAMPLES];
double bottleneck;
int buffer_index;
float data_buffer_float[MAX_FRAMESAMPLES + LB_TOTAL_DELAY_SAMPLES];
double bottleneck;
/* Maximum allowed number of bits for a 30 msec packet */
//int16_t payloadLimitBytes30;
// int16_t payloadLimitBytes30;
/* Maximum allowed number of bits for both 30 and 60 msec packet */
//int16_t maxPayloadBytes;
int16_t maxPayloadSizeBytes;
// int16_t maxPayloadBytes;
int16_t maxPayloadSizeBytes;
double lastLPCVec[UB_LPC_ORDER];
int16_t numBytesUsed;
int16_t lastJitterInfo;
double lastLPCVec[UB_LPC_ORDER];
int16_t numBytesUsed;
int16_t lastJitterInfo;
} ISACUBEncStruct;
typedef struct {
Bitstr bitstr_obj;
MaskFiltstr maskfiltstr_obj;
Bitstr bitstr_obj;
MaskFiltstr maskfiltstr_obj;
PostFiltBankstr postfiltbankstr_obj;
PitchFiltstr pitchfiltstr_obj;
FFTstr fftstr_obj;
PitchFiltstr pitchfiltstr_obj;
FFTstr fftstr_obj;
} ISACLBDecStruct;
typedef struct {
Bitstr bitstr_obj;
MaskFiltstr maskfiltstr_obj;
Bitstr bitstr_obj;
MaskFiltstr maskfiltstr_obj;
PostFiltBankstr postfiltbankstr_obj;
FFTstr fftstr_obj;
FFTstr fftstr_obj;
} ISACUBDecStruct;
typedef struct {
ISACLBEncStruct ISACencLB_obj;
ISACLBDecStruct ISACdecLB_obj;
} ISACLBStruct;
typedef struct {
ISACUBEncStruct ISACencUB_obj;
ISACUBDecStruct ISACdecUB_obj;
} ISACUBStruct;
@ -420,14 +386,14 @@ typedef struct {
*/
typedef struct {
/* 6 lower-band & 6 upper-band */
double loFiltGain[SUBFRAMES];
double hiFiltGain[SUBFRAMES];
double loFiltGain[SUBFRAMES];
double hiFiltGain[SUBFRAMES];
/* Upper boundary of interval W */
uint32_t W_upper;
uint32_t streamval;
/* Index to the current position in bytestream */
uint32_t stream_index;
uint8_t stream[3];
uint8_t stream[3];
} transcode_obj;
typedef struct {
@ -443,46 +409,46 @@ typedef struct {
typedef struct {
// lower-band codec instance
ISACLBStruct instLB;
ISACLBStruct instLB;
// upper-band codec instance
ISACUBStruct instUB;
ISACUBStruct instUB;
// Bandwidth Estimator and model for the rate.
BwEstimatorstr bwestimator_obj;
RateModel rate_data_obj;
double MaxDelay;
BwEstimatorstr bwestimator_obj;
RateModel rate_data_obj;
double MaxDelay;
/* 0 = adaptive; 1 = instantaneous */
int16_t codingMode;
int16_t codingMode;
// overall bottleneck of the codec
int32_t bottleneck;
int32_t bottleneck;
// QMF Filter state
int32_t analysisFBState1[FB_STATE_SIZE_WORD32];
int32_t analysisFBState2[FB_STATE_SIZE_WORD32];
int32_t synthesisFBState1[FB_STATE_SIZE_WORD32];
int32_t synthesisFBState2[FB_STATE_SIZE_WORD32];
int32_t analysisFBState1[FB_STATE_SIZE_WORD32];
int32_t analysisFBState2[FB_STATE_SIZE_WORD32];
int32_t synthesisFBState1[FB_STATE_SIZE_WORD32];
int32_t synthesisFBState2[FB_STATE_SIZE_WORD32];
// Error Code
int16_t errorCode;
int16_t errorCode;
// bandwidth of the encoded audio 8, 12 or 16 kHz
enum ISACBandwidth bandwidthKHz;
enum ISACBandwidth bandwidthKHz;
// Sampling rate of audio, encoder and decode, 8 or 16 kHz
enum IsacSamplingRate encoderSamplingRateKHz;
enum IsacSamplingRate decoderSamplingRateKHz;
// Flag to keep track of initializations, lower & upper-band
// encoder and decoder.
int16_t initFlag;
int16_t initFlag;
// Flag to to indicate signal bandwidth switch
int16_t resetFlag_8kHz;
int16_t resetFlag_8kHz;
// Maximum allowed rate, measured in Bytes per 30 ms.
int16_t maxRateBytesPer30Ms;
int16_t maxRateBytesPer30Ms;
// Maximum allowed payload-size, measured in Bytes.
int16_t maxPayloadSizeBytes;
int16_t maxPayloadSizeBytes;
/* The expected sampling rate of the input signal. Valid values are 16000
* and 32000. This is not the operation sampling rate of the codec. */
uint16_t in_sample_rate_hz;