1 files changed, 448 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/structs.h b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/structs.h
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@@ -0,0 +1,448 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * structs.h
+ *
+ * This header file contains all the structs used in the ISAC codec
+ *
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
+
+#include "modules/audio_coding/codecs/isac/bandwidth_info.h"
+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
+#include "modules/third_party/fft/fft.h"
+
+typedef struct Bitstreamstruct {
+  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 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];
+
+  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];
+
+  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];
+
+} PreFiltBankstr;
+
+typedef struct {
+  // 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];
+
+  float STATE_0_LOWER_float[2 * POSTQORDER];
+  float STATE_0_UPPER_float[2 * POSTQORDER];
+
+  float HPstates1_float[HPORDER];
+  float HPstates2_float[HPORDER];
+
+} PostFiltBankstr;
+
+typedef struct {
+  // data buffer for pitch filter
+  double ubuf[PITCH_BUFFSIZE];
+
+  // low pass state vector
+  double ystate[PITCH_DAMPORDER];
+
+  // old lag and gain
+  double oldlagp[1];
+  double oldgainp[1];
+
+} PitchFiltstr;
+
+typedef struct {
+  // data buffer
+  double buffer[PITCH_WLPCBUFLEN];
+
+  // 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];
+
+} 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];
+
+  double whitened_buf[QLOOKAHEAD];
+
+  double inbuf[QLOOKAHEAD];
+
+  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;
+
+  /* Previous RTP timestamp from received
+     packet (in samples relative beginning)                           */
+  int32_t prev_rec_rtp_number;
+
+  /* Send timestamp for previous packet (in ms using timeGetTime())   */
+  uint32_t prev_rec_send_ts;
+
+  /* Arrival time for previous packet (in ms using timeGetTime())     */
+  uint32_t prev_rec_arr_ts;
+
+  /* rate of previous packet, derived from RTP timestamps (in bits/s) */
+  float prev_rec_rtp_rate;
+
+  /* Time sinse the last update of the BN estimate (in ms)            */
+  uint32_t last_update_ts;
+
+  /* Time sinse the last reduction (in ms)                            */
+  uint32_t last_reduction_ts;
+
+  /* How many times the estimate was update in the beginning          */
+  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;
+
+  /* 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;
+
+  /* (assumed) bitrate for headers (bps)                              */
+  float rec_header_rate;
+
+  /* The estimated bottle neck rate from here to there (in bits/s)    */
+  float send_bw_avg;
+
+  /* The estimated mean absolute jitter value, as seen on
+     the other siee (in ms)                                           */
+  float send_max_delay_avg;
+
+  // number of packets received since last update
+  int num_pkts_rec;
+
+  int num_consec_rec_pkts_over_30k;
+
+  // flag for marking that a high speed network has been
+  // detected downstream
+  int hsn_detect_rec;
+
+  int num_consec_snt_pkts_over_30k;
+
+  // flag for marking that a high speed network has
+  // been detected upstream
+  int hsn_detect_snd;
+
+  uint32_t start_wait_period;
+
+  int in_wait_period;
+
+  int change_to_WB;
+
+  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;
+  /* ms */
+  int ExceedAgo;
+  /* packets left to send in current burst */
+  int BurstCounter;
+  /* packets */
+  int InitCounter;
+  /* ms remaining in buffer when next packet will be sent */
+  double StillBuffered;
+
+} RateModel;
+
+/* 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;
+
+  /* Frame length in samples */
+  int16_t framelength;
+
+  /* Pitch Gain */
+  int pitchGain_index[2];
+
+  /* Pitch Lag */
+  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];
+
+  /* Encode Spec */
+  int16_t fre[FRAMESAMPLES];
+  int16_t fim[FRAMESAMPLES];
+  int16_t AvgPitchGain[2];
+
+  /* Used in adaptive mode only */
+  int minBytes;
+
+} IsacSaveEncoderData;
+
+typedef struct {
+  int indexLPCShape[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
+  double lpcGain[SUBFRAMES << 1];
+  int lpcGainIndex[SUBFRAMES << 1];
+
+  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;
+  PitchAnalysisStruct pitchanalysisstr_obj;
+  FFTstr fftstr_obj;
+  IsacSaveEncoderData SaveEnc_obj;
+
+  int buffer_index;
+  int16_t current_framesamples;
+
+  float data_buffer_float[FRAMESAMPLES_30ms];
+
+  int frame_nb;
+  double bottleneck;
+  int16_t new_framelength;
+  double s2nr;
+
+  /* Maximum allowed number of bits for a 30 msec packet */
+  int16_t payloadLimitBytes30;
+  /* Maximum allowed number of bits for a 30 msec packet */
+  int16_t payloadLimitBytes60;
+  /* Maximum allowed number of bits for both 30 and 60 msec packet */
+  int16_t maxPayloadBytes;
+  /* Maximum allowed rate in bytes per 30 msec packet */
+  int16_t maxRateInBytes;
+
+  /*---
+    If set to 1 iSAC will not adapt the frame-size, if used in
+    channel-adaptive mode. The initial value will be used for all rates.
+    ---*/
+  int16_t enforceFrameSize;
+
+  /*-----
+    This records the BWE index the encoder injected into the bit-stream.
+    It will be used in RCU. The same BWE index of main payload will be in
+    the redundant payload. We can not retrieve it from BWE because it is
+    a recursive procedure (WebRtcIsac_GetDownlinkBwJitIndexImpl) and has to be
+    called only once per each encode.
+    -----*/
+  int16_t lastBWIdx;
+} ISACLBEncStruct;
+
+typedef struct {
+  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;
+  /* Maximum allowed number of bits for a 30 msec packet */
+  // int16_t        payloadLimitBytes30;
+  /* Maximum allowed number of bits for both 30 and 60 msec packet */
+  // int16_t        maxPayloadBytes;
+  int16_t maxPayloadSizeBytes;
+
+  double lastLPCVec[UB_LPC_ORDER];
+  int16_t numBytesUsed;
+  int16_t lastJitterInfo;
+} ISACUBEncStruct;
+
+typedef struct {
+  Bitstr bitstr_obj;
+  MaskFiltstr maskfiltstr_obj;
+  PostFiltBankstr postfiltbankstr_obj;
+  PitchFiltstr pitchfiltstr_obj;
+  FFTstr fftstr_obj;
+
+} ISACLBDecStruct;
+
+typedef struct {
+  Bitstr bitstr_obj;
+  MaskFiltstr maskfiltstr_obj;
+  PostFiltBankstr postfiltbankstr_obj;
+  FFTstr fftstr_obj;
+
+} ISACUBDecStruct;
+
+typedef struct {
+  ISACLBEncStruct ISACencLB_obj;
+  ISACLBDecStruct ISACdecLB_obj;
+} ISACLBStruct;
+
+typedef struct {
+  ISACUBEncStruct ISACencUB_obj;
+  ISACUBDecStruct ISACdecUB_obj;
+} ISACUBStruct;
+
+/*
+  This struct is used to take a snapshot of the entropy coder and LPC gains
+  right before encoding LPC gains. This allows us to go back to that state
+  if we like to limit the payload size.
+*/
+typedef struct {
+  /* 6 lower-band & 6 upper-band */
+  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];
+} transcode_obj;
+
+typedef struct {
+  // TODO(kwiberg): The size of these tables could be reduced by storing floats
+  // instead of doubles, and by making use of the identity cos(x) =
+  // sin(x+pi/2). They could also be made global constants that we fill in at
+  // compile time.
+  double costab1[FRAMESAMPLES_HALF];
+  double sintab1[FRAMESAMPLES_HALF];
+  double costab2[FRAMESAMPLES_QUARTER];
+  double sintab2[FRAMESAMPLES_QUARTER];
+} TransformTables;
+
+typedef struct {
+  // lower-band codec instance
+  ISACLBStruct instLB;
+  // upper-band codec instance
+  ISACUBStruct instUB;
+
+  // Bandwidth Estimator and model for the rate.
+  BwEstimatorstr bwestimator_obj;
+  RateModel rate_data_obj;
+  double MaxDelay;
+
+  /* 0 = adaptive; 1 = instantaneous */
+  int16_t codingMode;
+
+  // overall bottleneck of the codec
+  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];
+
+  // Error Code
+  int16_t errorCode;
+
+  // bandwidth of the encoded audio 8, 12 or 16 kHz
+  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;
+
+  // Flag to to indicate signal bandwidth switch
+  int16_t resetFlag_8kHz;
+
+  // Maximum allowed rate, measured in Bytes per 30 ms.
+  int16_t maxRateBytesPer30Ms;
+  // Maximum allowed payload-size, measured in Bytes.
+  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;
+
+  // Trig tables for WebRtcIsac_Time2Spec and WebRtcIsac_Spec2time.
+  TransformTables transform_tables;
+} ISACMainStruct;
+
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_ */