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webrtc / src / webrtc / ad70aa49b2ea2e4750b4596a769ce27278e507d8 / . / modules / audio_coding / codecs / isac / main / source / isac.c
blob: 428fda87bd15d1a3ef0d0902244301a17a45abed [file] [log] [blame]
/*
* 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.
*/
/*
* isac.c
*
* This C file contains the functions for the ISAC API
*
*/
#include "isac.h"
#include "bandwidth_estimator.h"
#include "crc.h"
#include "entropy_coding.h"
#include "codec.h"
#include "structs.h"
#include "signal_processing_library.h"
#include "lpc_shape_swb16_tables.h"
#include "os_specific_inline.h"
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include <math.h>
#define BIT_MASK_DEC_INIT 0x0001
#define BIT_MASK_ENC_INIT 0x0002
#define LEN_CHECK_SUM_WORD8 4
#define MAX_NUM_LAYERS 10
/****************************************************************************
* UpdatePayloadSizeLimit(...)
*
* Call this function to update the limit on the payload size. The limit on
* payload size might change i) if a user ''directly changes the limit by
* calling xxx_setMaxPayloadSize() or xxx_setMaxRate(), or ii) indirectly
* when bandwidth is changing. The latter might be the result of bandwidth
* adaptation, or direct change of the bottleneck in instantaneous mode.
*
* This function takes the current overall limit on payload, and translates it
* to the limits on lower and upper-band. If the codec is in wideband mode,
* then the overall limit and the limit on the lower-band is the same.
* Otherwise, a fraction of the limit should be allocated to lower-band
* leaving some room for the upper-band bit-stream. That is why an update
* of limit is required every time that the bandwidth is changing.
*
*/
static void UpdatePayloadSizeLimit(ISACMainStruct* instISAC) {
WebRtc_Word16 lim30MsPayloadBytes = WEBRTC_SPL_MIN(
(instISAC->maxPayloadSizeBytes),
(instISAC->maxRateBytesPer30Ms));
WebRtc_Word16 lim60MsPayloadBytes = WEBRTC_SPL_MIN(
(instISAC->maxPayloadSizeBytes),
(instISAC->maxRateBytesPer30Ms << 1));
/* The only time that iSAC will have 60 ms
* frame-size is when operating in wideband, so
* there is no upper-band bit-stream. */
if (instISAC->bandwidthKHz == isac8kHz) {
/* At 8 kHz there is no upper-band bit-stream,
* therefore, the lower-band limit is the overall limit. */
instISAC->instLB.ISACencLB_obj.payloadLimitBytes60 =
lim60MsPayloadBytes;
instISAC->instLB.ISACencLB_obj.payloadLimitBytes30 =
lim30MsPayloadBytes;
} else {
/* When in super-wideband, we only have 30 ms frames.
* Do a rate allocation for the given limit. */
if (lim30MsPayloadBytes > 250) {
/* 4/5 to lower-band the rest for upper-band. */
instISAC->instLB.ISACencLB_obj.payloadLimitBytes30 =
(lim30MsPayloadBytes << 2) / 5;
} else if (lim30MsPayloadBytes > 200) {
/* For the interval of 200 to 250 the share of
* upper-band linearly grows from 20 to 50. */
instISAC->instLB.ISACencLB_obj.payloadLimitBytes30 =
(lim30MsPayloadBytes << 1) / 5 + 100;
} else {
/* Allocate only 20 for upper-band. */
instISAC->instLB.ISACencLB_obj.payloadLimitBytes30 =
lim30MsPayloadBytes - 20;
}
instISAC->instUB.ISACencUB_obj.maxPayloadSizeBytes =
lim30MsPayloadBytes;
}
}
/****************************************************************************
* UpdateBottleneck(...)
*
* This function updates the bottleneck only if the codec is operating in
* channel-adaptive mode. Furthermore, as the update of bottleneck might
* result in an update of bandwidth, therefore, the bottlenech should be
* updated just right before the first 10ms of a frame is pushed into encoder.
*
*/
static void UpdateBottleneck(ISACMainStruct* instISAC) {
/* Read the bottleneck from bandwidth estimator for the
* first 10 ms audio. This way, if there is a change
* in bandwidth, upper and lower-band will be in sync. */
if ((instISAC->codingMode == 0) &&
(instISAC->instLB.ISACencLB_obj.buffer_index == 0) &&
(instISAC->instLB.ISACencLB_obj.frame_nb == 0)) {
WebRtc_Word32 bottleneck;
WebRtcIsac_GetUplinkBandwidth(&(instISAC->bwestimator_obj),
&bottleneck);
/* Adding hysteresis when increasing signal bandwidth. */
if ((instISAC->bandwidthKHz == isac8kHz)
&& (bottleneck > 37000)
&& (bottleneck < 41000)) {
bottleneck = 37000;
}
/* Switching from 12 kHz to 16 kHz is not allowed at this revision.
* If we let this happen, we have to take care of buffer_index and
* the last LPC vector. */
if ((instISAC->bandwidthKHz != isac16kHz) &&
(bottleneck > 46000)) {
bottleneck = 46000;
}
/* We might need a rate allocation. */
if (instISAC->encoderSamplingRateKHz == kIsacWideband) {
/* Wideband is the only choice we have here. */
instISAC->instLB.ISACencLB_obj.bottleneck =
(bottleneck > 32000) ? 32000 : bottleneck;
instISAC->bandwidthKHz = isac8kHz;
} else {
/* Do the rate-allocation and get the new bandwidth. */
enum ISACBandwidth bandwidth;
WebRtcIsac_RateAllocation(bottleneck,
&(instISAC->instLB.ISACencLB_obj.bottleneck),
&(instISAC->instUB.ISACencUB_obj.bottleneck),
&bandwidth);
if (bandwidth != isac8kHz) {
instISAC->instLB.ISACencLB_obj.new_framelength = 480;
}
if (bandwidth != instISAC->bandwidthKHz) {
/* Bandwidth is changing. */
instISAC->bandwidthKHz = bandwidth;
UpdatePayloadSizeLimit(instISAC);
if (bandwidth == isac12kHz) {
instISAC->instLB.ISACencLB_obj.buffer_index = 0;
}
/* Currently we don't let the bandwidth to switch to 16 kHz
* if in adaptive mode. If we let this happen, we have to take
* care of buffer_index and the last LPC vector. */
}
}
}
}
/****************************************************************************
* GetSendBandwidthInfo(...)
*
* This is called to get the bandwidth info. This info is the bandwidth and
* the jitter of 'there-to-here' channel, estimated 'here.' These info
* is signaled in an in-band fashion to the other side.
*
* The call to the bandwidth estimator triggers a recursive averaging which
* has to be synchronized between encoder & decoder, therefore, the call to
* BWE should be once per packet. As the BWE info is inserted into bit-stream
* We need a valid info right before the encodeLB function is going to
* generate a bit-stream. That is when lower-band buffer has already 20ms
* of audio, and the 3rd block of 10ms is going to be injected into encoder.
*
* Inputs:
* - instISAC : iSAC instance.
*
* Outputs:
* - bandwidthIndex : an index which has to be encoded in
* lower-band bit-stream, indicating the
* bandwidth of there-to-here channel.
* - jitterInfo : this indicates if the jitter is high
* or low and it is encoded in upper-band
* bit-stream.
*
*/
static void GetSendBandwidthInfo(ISACMainStruct* instISAC,
WebRtc_Word16* bandwidthIndex,
WebRtc_Word16* jitterInfo) {
if ((instISAC->instLB.ISACencLB_obj.buffer_index ==
(FRAMESAMPLES_10ms << 1)) &&
(instISAC->instLB.ISACencLB_obj.frame_nb == 0)) {
/* Bandwidth estimation and coding. */
WebRtcIsac_GetDownlinkBwJitIndexImpl(&(instISAC->bwestimator_obj),
bandwidthIndex, jitterInfo,
instISAC->decoderSamplingRateKHz);
}
}
/****************************************************************************
* WebRtcIsac_AssignSize(...)
*
* This function returns the size of the ISAC instance, so that the instance
* can be created out side iSAC.
*
* Output:
* - sizeinbytes : number of bytes needed to allocate for the
* instance.
*
* Return value : 0 - Ok
* -1 - Error
*/
WebRtc_Word16 WebRtcIsac_AssignSize(int* sizeInBytes) {
*sizeInBytes = sizeof(ISACMainStruct) * 2 / sizeof(WebRtc_Word16);
return 0;
}
/****************************************************************************
* WebRtcIsac_Assign(...)
*
* This function assigns the memory already created to the ISAC instance.
*
* Input:
* - ISAC_main_inst : address of the pointer to the coder instance.
* - instISAC_Addr : the already allocated memory, where we put the
* iSAC structure.
*
* Return value : 0 - Ok
* -1 - Error
*/
WebRtc_Word16 WebRtcIsac_Assign(ISACStruct** ISAC_main_inst,
void* instISAC_Addr) {
if (instISAC_Addr != NULL) {
ISACMainStruct* instISAC = (ISACMainStruct*)instISAC_Addr;
instISAC->errorCode = 0;
instISAC->initFlag = 0;
/* Assign the address. */
*ISAC_main_inst = (ISACStruct*)instISAC_Addr;
/* Default is wideband. */
instISAC->encoderSamplingRateKHz = kIsacWideband;
instISAC->decoderSamplingRateKHz = kIsacWideband;
instISAC->bandwidthKHz = isac8kHz;
instISAC->in_sample_rate_hz = 16000;
return 0;
} else {
return -1;
}
}
/****************************************************************************
* WebRtcIsac_Create(...)
*
* This function creates an ISAC instance, which will contain the state
* information for one coding/decoding channel.
*
* Input:
* - ISAC_main_inst : address of the pointer to the coder instance.
*
* Return value : 0 - Ok
* -1 - Error
*/
WebRtc_Word16 WebRtcIsac_Create(ISACStruct** ISAC_main_inst) {
ISACMainStruct* instISAC;
instISAC = (ISACMainStruct*)WEBRTC_SPL_VNEW(ISACMainStruct, 1);
*ISAC_main_inst = (ISACStruct*)instISAC;
if (*ISAC_main_inst != NULL) {
instISAC->errorCode = 0;
instISAC->initFlag = 0;
/* Default is wideband. */
instISAC->bandwidthKHz = isac8kHz;
instISAC->encoderSamplingRateKHz = kIsacWideband;
instISAC->decoderSamplingRateKHz = kIsacWideband;
instISAC->in_sample_rate_hz = 16000;
return 0;
} else {
return -1;
}
}
/****************************************************************************
* WebRtcIsac_Free(...)
*
* This function frees the ISAC instance created at the beginning.
*
* Input:
* - ISAC_main_inst : a ISAC instance.
*
* Return value : 0 - Ok
* -1 - Error
*/
WebRtc_Word16 WebRtcIsac_Free(ISACStruct* ISAC_main_inst) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
WEBRTC_SPL_FREE(instISAC);
return 0;
}
/****************************************************************************
* EncoderInitLb(...) - internal function for initialization of
* Lower Band
* EncoderInitUb(...) - internal function for initialization of
* Upper Band
* WebRtcIsac_EncoderInit(...) - API function
*
* This function initializes a ISAC instance prior to the encoder calls.
*
* Input:
* - ISAC_main_inst : ISAC instance.
* - CodingMode : 0 -> Bit rate and frame length are automatically
* adjusted to available bandwidth on
* transmission channel, applicable just to
* wideband mode.
* 1 -> User sets a frame length and a target bit
* rate which is taken as the maximum
* short-term average bit rate.
*
* Return value : 0 - Ok
* -1 - Error
*/
static WebRtc_Word16 EncoderInitLb(ISACLBStruct* instLB,
WebRtc_Word16 codingMode,
enum IsacSamplingRate sampRate) {
WebRtc_Word16 statusInit = 0;
int k;
/* Init stream vector to zero */
for (k = 0; k < STREAM_SIZE_MAX_60; k++) {
instLB->ISACencLB_obj.bitstr_obj.stream[k] = 0;
}
if ((codingMode == 1) || (sampRate == kIsacSuperWideband)) {
/* 30 ms frame-size if either in super-wideband or
* instantaneous mode (I-mode). */
instLB->ISACencLB_obj.new_framelength = 480;
} else {
instLB->ISACencLB_obj.new_framelength = INITIAL_FRAMESAMPLES;
}
WebRtcIsac_InitMasking(&instLB->ISACencLB_obj.maskfiltstr_obj);
WebRtcIsac_InitPreFilterbank(&instLB->ISACencLB_obj.prefiltbankstr_obj);
WebRtcIsac_InitPitchFilter(&instLB->ISACencLB_obj.pitchfiltstr_obj);
WebRtcIsac_InitPitchAnalysis(
&instLB->ISACencLB_obj.pitchanalysisstr_obj);
instLB->ISACencLB_obj.buffer_index = 0;
instLB->ISACencLB_obj.frame_nb = 0;
/* Default for I-mode. */
instLB->ISACencLB_obj.bottleneck = 32000;
instLB->ISACencLB_obj.current_framesamples = 0;
instLB->ISACencLB_obj.s2nr = 0;
instLB->ISACencLB_obj.payloadLimitBytes30 = STREAM_SIZE_MAX_30;
instLB->ISACencLB_obj.payloadLimitBytes60 = STREAM_SIZE_MAX_60;
instLB->ISACencLB_obj.maxPayloadBytes = STREAM_SIZE_MAX_60;
instLB->ISACencLB_obj.maxRateInBytes = STREAM_SIZE_MAX_30;
instLB->ISACencLB_obj.enforceFrameSize = 0;
/* Invalid value prevents getRedPayload to
run before encoder is called. */
instLB->ISACencLB_obj.lastBWIdx = -1;
return statusInit;
}
static WebRtc_Word16 EncoderInitUb(ISACUBStruct* instUB,
WebRtc_Word16 bandwidth) {
WebRtc_Word16 statusInit = 0;
int k;
/* Init stream vector to zero. */
for (k = 0; k < STREAM_SIZE_MAX_60; k++) {
instUB->ISACencUB_obj.bitstr_obj.stream[k] = 0;
}
WebRtcIsac_InitMasking(&instUB->ISACencUB_obj.maskfiltstr_obj);
WebRtcIsac_InitPreFilterbank(&instUB->ISACencUB_obj.prefiltbankstr_obj);
if (bandwidth == isac16kHz) {
instUB->ISACencUB_obj.buffer_index = LB_TOTAL_DELAY_SAMPLES;
} else {
instUB->ISACencUB_obj.buffer_index = 0;
}
/* Default for I-mode. */
instUB->ISACencUB_obj.bottleneck = 32000;
/* These store the limits for the wideband + super-wideband bit-stream. */
instUB->ISACencUB_obj.maxPayloadSizeBytes = STREAM_SIZE_MAX_30 << 1;
/* This has to be updated after each lower-band encoding to guarantee
* a correct payload-limitation. */
instUB->ISACencUB_obj.numBytesUsed = 0;
memset(instUB->ISACencUB_obj.data_buffer_float, 0,
(MAX_FRAMESAMPLES + LB_TOTAL_DELAY_SAMPLES) * sizeof(float));
memcpy(&(instUB->ISACencUB_obj.lastLPCVec),
WebRtcIsac_kMeanLarUb16, sizeof(double) * UB_LPC_ORDER);
return statusInit;
}
WebRtc_Word16 WebRtcIsac_EncoderInit(ISACStruct* ISAC_main_inst,
WebRtc_Word16 codingMode) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
WebRtc_Word16 status;
if ((codingMode != 0) && (codingMode != 1)) {
instISAC->errorCode = ISAC_DISALLOWED_CODING_MODE;
return -1;
}
/* Default bottleneck. */
instISAC->bottleneck = MAX_ISAC_BW;
if (instISAC->encoderSamplingRateKHz == kIsacWideband) {
instISAC->bandwidthKHz = isac8kHz;
instISAC->maxPayloadSizeBytes = STREAM_SIZE_MAX_60;
instISAC->maxRateBytesPer30Ms = STREAM_SIZE_MAX_30;
} else {
instISAC->bandwidthKHz = isac16kHz;
instISAC->maxPayloadSizeBytes = STREAM_SIZE_MAX;
instISAC->maxRateBytesPer30Ms = STREAM_SIZE_MAX;
}
/* Channel-adaptive = 0; Instantaneous (Channel-independent) = 1. */
instISAC->codingMode = codingMode;
WebRtcIsac_InitBandwidthEstimator(&instISAC->bwestimator_obj,
instISAC->encoderSamplingRateKHz,
instISAC->decoderSamplingRateKHz);
WebRtcIsac_InitRateModel(&instISAC->rate_data_obj);
/* Default for I-mode. */
instISAC->MaxDelay = 10.0;
status = EncoderInitLb(&instISAC->instLB, codingMode,
instISAC->encoderSamplingRateKHz);
if (status < 0) {
instISAC->errorCode = -status;
return -1;
}
if (instISAC->encoderSamplingRateKHz == kIsacSuperWideband) {
/* Initialize encoder filter-bank. */
memset(instISAC->analysisFBState1, 0,
FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
memset(instISAC->analysisFBState2, 0,
FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
status = EncoderInitUb(&(instISAC->instUB),
instISAC->bandwidthKHz);
if (status < 0) {
instISAC->errorCode = -status;
return -1;
}
}
memset(instISAC->state_in_resampler, 0, sizeof(instISAC->state_in_resampler));
/* Initialization is successful, set the flag. */
instISAC->initFlag |= BIT_MASK_ENC_INIT;
return 0;
}
/****************************************************************************
* WebRtcIsac_Encode(...)
*
* This function encodes 10ms frame(s) and inserts it into a package.
* Input speech length has to be 160 samples (10ms). The encoder buffers those
* 10ms frames until it reaches the chosen Framesize (480 or 960 samples
* corresponding to 30 or 60 ms frames), and then proceeds to the encoding.
*
* Input:
* - ISAC_main_inst : ISAC instance.
* - speechIn : input speech vector.
*
* Output:
* - encoded : the encoded data vector
*
* Return value:
* : >0 - Length (in bytes) of coded data
* : 0 - The buffer didn't reach the chosen
* frameSize so it keeps buffering speech
* samples.
* : -1 - Error
*/
WebRtc_Word16 WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
const WebRtc_Word16* speechIn,
WebRtc_Word16* encoded) {
float inFrame[FRAMESAMPLES_10ms];
WebRtc_Word16 speechInLB[FRAMESAMPLES_10ms];
WebRtc_Word16 speechInUB[FRAMESAMPLES_10ms];
WebRtc_Word16 streamLenLB = 0;
WebRtc_Word16 streamLenUB = 0;
WebRtc_Word16 streamLen = 0;
WebRtc_Word16 k = 0;
WebRtc_UWord8* ptrEncodedUW8 = (WebRtc_UWord8*)encoded;
int garbageLen = 0;
WebRtc_Word32 bottleneck = 0;
WebRtc_Word16 bottleneckIdx = 0;
WebRtc_Word16 jitterInfo = 0;
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
ISACLBStruct* instLB = &(instISAC->instLB);
ISACUBStruct* instUB = &(instISAC->instUB);
const int16_t* speech_in_ptr = speechIn;
int16_t resampled_buff[FRAMESAMPLES_10ms * 2];
/* Check if encoder initiated. */
if ((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
BIT_MASK_ENC_INIT) {
instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
return -1;
}
if (instISAC->in_sample_rate_hz == 48000) {
/* Samples in 10 ms @ 48 kHz. */
const int kNumInputSamples = FRAMESAMPLES_10ms * 3;
/* Samples 10 ms @ 32 kHz. */
const int kNumOutputSamples = FRAMESAMPLES_10ms * 2;
/* Resampler divide the input into blocks of 3 samples, i.e.
* kNumInputSamples / 3. */
const int kNumResamplerBlocks = FRAMESAMPLES_10ms;
int32_t buffer32[FRAMESAMPLES_10ms * 3 + SIZE_RESAMPLER_STATE];
/* Restore last samples from the past to the beginning of the buffer
* and store the last samples of current frame for the next resampling. */
for (k = 0; k < SIZE_RESAMPLER_STATE; k++) {
buffer32[k] = instISAC->state_in_resampler[k];
instISAC->state_in_resampler[k] = speechIn[kNumInputSamples -
SIZE_RESAMPLER_STATE + k];
}
for (k = 0; k < kNumInputSamples; k++) {
buffer32[SIZE_RESAMPLER_STATE + k] = speechIn[k];
}
/* Resampling 3 samples to 2. Function divides the input in
* |kNumResamplerBlocks| number of 3-sample groups, and output is
* |kNumResamplerBlocks| number of 2-sample groups. */
WebRtcSpl_Resample48khzTo32khz(buffer32, buffer32, kNumResamplerBlocks);
WebRtcSpl_VectorBitShiftW32ToW16(resampled_buff, kNumOutputSamples,
buffer32, 15);
speech_in_ptr = resampled_buff;
}
if (instISAC->encoderSamplingRateKHz == kIsacSuperWideband) {
WebRtcSpl_AnalysisQMF(speech_in_ptr, speechInLB, speechInUB,
instISAC->analysisFBState1,
instISAC->analysisFBState2);
/* Convert from fixed to floating point. */
for (k = 0; k < FRAMESAMPLES_10ms; k++) {
inFrame[k] = (float)speechInLB[k];
}
} else {
for (k = 0; k < FRAMESAMPLES_10ms; k++) {
inFrame[k] = (float) speechIn[k];
}
}
/* Add some noise to avoid denormal numbers. */
inFrame[0] += (float)1.23455334e-3;
inFrame[1] -= (float)2.04324239e-3;
inFrame[2] += (float)1.90854954e-3;
inFrame[9] += (float)1.84854878e-3;
/* This function will update the bottleneck if required. */
UpdateBottleneck(instISAC);
/* Get the bandwith information which has to be sent to the other side. */
GetSendBandwidthInfo(instISAC, &bottleneckIdx, &jitterInfo);
/* Encode lower-band. */
streamLenLB = WebRtcIsac_EncodeLb(inFrame, &instLB->ISACencLB_obj,
instISAC->codingMode, bottleneckIdx);
if (streamLenLB < 0) {
return -1;
}
if (instISAC->encoderSamplingRateKHz == kIsacSuperWideband) {
instUB = &(instISAC->instUB);
/* Convert to float. */
for (k = 0; k < FRAMESAMPLES_10ms; k++) {
inFrame[k] = (float) speechInUB[k];
}
/* Add some noise to avoid denormal numbers. */
inFrame[0] += (float)1.23455334e-3;
inFrame[1] -= (float)2.04324239e-3;
inFrame[2] += (float)1.90854954e-3;
inFrame[9] += (float)1.84854878e-3;
/* Tell to upper-band the number of bytes used so far.
* This is for payload limitation. */
instUB->ISACencUB_obj.numBytesUsed = streamLenLB + 1 +
LEN_CHECK_SUM_WORD8;
/* Encode upper-band. */
switch (instISAC->bandwidthKHz) {
case isac12kHz: {
streamLenUB = WebRtcIsac_EncodeUb12(inFrame, &instUB->ISACencUB_obj,
jitterInfo);
break;
}
case isac16kHz: {
streamLenUB = WebRtcIsac_EncodeUb16(inFrame, &instUB->ISACencUB_obj,
jitterInfo);
break;
}
case isac8kHz: {
streamLenUB = 0;
break;
}
}
if ((streamLenUB < 0) && (streamLenUB != -ISAC_PAYLOAD_LARGER_THAN_LIMIT)) {
/* An error has happened but this is not the error due to a
* bit-stream larger than the limit. */
return -1;
}
if (streamLenLB == 0) {
return 0;
}
/* One byte is allocated for the length. According to older decoders
so the length bit-stream plus one byte for size and
LEN_CHECK_SUM_WORD8 for the checksum should be less than or equal
to 255. */
if ((streamLenUB > (255 - (LEN_CHECK_SUM_WORD8 + 1))) ||
(streamLenUB == -ISAC_PAYLOAD_LARGER_THAN_LIMIT)) {
/* We have got a too long bit-stream we skip the upper-band
* bit-stream for this frame. */
streamLenUB = 0;
}
memcpy(ptrEncodedUW8, instLB->ISACencLB_obj.bitstr_obj.stream, streamLenLB);
streamLen = streamLenLB;
if (streamLenUB > 0) {
ptrEncodedUW8[streamLenLB] = (WebRtc_UWord8)(streamLenUB + 1 +
LEN_CHECK_SUM_WORD8);
memcpy(&ptrEncodedUW8[streamLenLB + 1],
instUB->ISACencUB_obj.bitstr_obj.stream, streamLenUB);
streamLen += ptrEncodedUW8[streamLenLB];
} else {
ptrEncodedUW8[streamLenLB] = 0;
}
} else {
if (streamLenLB == 0) {
return 0;
}
memcpy(ptrEncodedUW8, instLB->ISACencLB_obj.bitstr_obj.stream,
streamLenLB);
streamLenUB = 0;
streamLen = streamLenLB;
}
/* Add Garbage if required. */
WebRtcIsac_GetUplinkBandwidth(&instISAC->bwestimator_obj, &bottleneck);
if (instISAC->codingMode == 0) {
int minBytes;
int limit;
WebRtc_UWord8* ptrGarbage;
instISAC->MaxDelay = (double)WebRtcIsac_GetUplinkMaxDelay(
&instISAC->bwestimator_obj);
/* Update rate model and get minimum number of bytes in this packet. */
minBytes = WebRtcIsac_GetMinBytes(
&(instISAC->rate_data_obj), streamLen,
instISAC->instLB.ISACencLB_obj.current_framesamples, bottleneck,
instISAC->MaxDelay, instISAC->bandwidthKHz);
/* Make sure MinBytes does not exceed packet size limit. */
if (instISAC->bandwidthKHz == isac8kHz) {
if (instLB->ISACencLB_obj.current_framesamples == FRAMESAMPLES) {
limit = instLB->ISACencLB_obj.payloadLimitBytes30;
} else {
limit = instLB->ISACencLB_obj.payloadLimitBytes60;
}
} else {
limit = instUB->ISACencUB_obj.maxPayloadSizeBytes;
}
minBytes = (minBytes > limit) ? limit : minBytes;
/* Make sure we don't allow more than 255 bytes of garbage data.
* We store the length of the garbage data in 8 bits in the bitstream,
* 255 is the max garbage length we can signal using 8 bits. */
if ((instISAC->bandwidthKHz == isac8kHz) ||
(streamLenUB == 0)) {
ptrGarbage = &ptrEncodedUW8[streamLenLB];
limit = streamLen + 255;
} else {
ptrGarbage = &ptrEncodedUW8[streamLenLB + 1 + streamLenUB];
limit = streamLen + (255 - ptrEncodedUW8[streamLenLB]);
}
minBytes = (minBytes > limit) ? limit : minBytes;
garbageLen = (minBytes > streamLen) ? (minBytes - streamLen) : 0;
/* Save data for creation of multiple bit-streams. */
/* If bit-stream too short then add garbage at the end. */
if (garbageLen > 0) {
for (k = 0; k < garbageLen; k++) {
ptrGarbage[k] = (WebRtc_UWord8)(rand() & 0xFF);
}
/* For a correct length of the upper-band bit-stream together
* with the garbage. Garbage is embeded in upper-band bit-stream.
* That is the only way to preserve backward compatibility. */
if ((instISAC->bandwidthKHz == isac8kHz) ||
(streamLenUB == 0)) {
ptrEncodedUW8[streamLenLB] = (WebRtc_UWord8)garbageLen;
} else {
ptrEncodedUW8[streamLenLB] += (WebRtc_UWord8)garbageLen;
/* Write the length of the garbage at the end of the upper-band
* bit-stream, if exists. This helps for sanity check. */
ptrEncodedUW8[streamLenLB + 1 + streamLenUB] =
(WebRtc_UWord8)garbageLen;
}
streamLen += garbageLen;
}
} else {
/* update rate model */
WebRtcIsac_UpdateRateModel(
&instISAC->rate_data_obj, streamLen,
instISAC->instLB.ISACencLB_obj.current_framesamples, bottleneck);
garbageLen = 0;
}
/* Generate CRC if required. */
if ((instISAC->bandwidthKHz != isac8kHz) && (streamLenUB > 0)) {
WebRtc_UWord32 crc;
WebRtcIsac_GetCrc((WebRtc_Word16*)(&(ptrEncodedUW8[streamLenLB + 1])),
streamLenUB + garbageLen, &crc);
#ifndef WEBRTC_BIG_ENDIAN
for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) {
ptrEncodedUW8[streamLen - LEN_CHECK_SUM_WORD8 + k] =
(WebRtc_UWord8)((crc >> (24 - k * 8)) & 0xFF);
}
#else
memcpy(&ptrEncodedUW8[streamLenLB + streamLenUB + 1], &crc,
LEN_CHECK_SUM_WORD8);
#endif
}
return streamLen;
}
/******************************************************************************
* WebRtcIsac_GetNewBitStream(...)
*
* This function returns encoded data, with the recieved bwe-index in the
* stream. If the rate is set to a value less than bottleneck of codec
* the new bistream will be re-encoded with the given target rate.
* It should always return a complete packet, i.e. only called once
* even for 60 msec frames.
*
* NOTE 1! This function does not write in the ISACStruct, it is not allowed.
* NOTE 2! Rates larger than the bottleneck of the codec will be limited
* to the current bottleneck.
*
* Input:
* - ISAC_main_inst : ISAC instance.
* - bweIndex : Index of bandwidth estimate to put in new
* bitstream
* - rate : target rate of the transcoder is bits/sec.
* Valid values are the accepted rate in iSAC,
* i.e. 10000 to 56000.
*
* Output:
* - encoded : The encoded data vector
*
* Return value : >0 - Length (in bytes) of coded data
* -1 - Error or called in SWB mode
* NOTE! No error code is written to
* the struct since it is only allowed to read
* the struct.
*/
WebRtc_Word16 WebRtcIsac_GetNewBitStream(ISACStruct* ISAC_main_inst,
WebRtc_Word16 bweIndex,
WebRtc_Word16 jitterInfo,
WebRtc_Word32 rate,
WebRtc_Word16* encoded,
WebRtc_Word16 isRCU) {
Bitstr iSACBitStreamInst; /* Local struct for bitstream handling */
WebRtc_Word16 streamLenLB;
WebRtc_Word16 streamLenUB;
WebRtc_Word16 totalStreamLen;
double gain2;
double gain1;
float scale;
enum ISACBandwidth bandwidthKHz;
double rateLB;
double rateUB;
WebRtc_Word32 currentBN;
WebRtc_UWord8* encodedPtrUW8 = (WebRtc_UWord8*)encoded;
WebRtc_UWord32 crc;
#ifndef WEBRTC_BIG_ENDIAN
WebRtc_Word16 k;
#endif
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
if ((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
BIT_MASK_ENC_INIT) {
return -1;
}
/* Get the bottleneck of this iSAC and limit the
* given rate to the current bottleneck. */
WebRtcIsac_GetUplinkBw(ISAC_main_inst, &currentBN);
if (rate > currentBN) {
rate = currentBN;
}
if (WebRtcIsac_RateAllocation(rate, &rateLB, &rateUB, &bandwidthKHz) < 0) {
return -1;
}
/* Cannot transcode from 16 kHz to 12 kHz. */
if ((bandwidthKHz == isac12kHz) &&
(instISAC->bandwidthKHz == isac16kHz)) {
return -1;
}
/* A gain [dB] for the given rate. */
gain1 = WebRtcIsac_GetSnr(
rateLB, instISAC->instLB.ISACencLB_obj.current_framesamples);
/* The gain [dB] of this iSAC. */
gain2 = WebRtcIsac_GetSnr(
instISAC->instLB.ISACencLB_obj.bottleneck,
instISAC->instLB.ISACencLB_obj.current_framesamples);
/* Scale is the ratio of two gains in normal domain. */
scale = (float)pow(10, (gain1 - gain2) / 20.0);
/* Change the scale if this is a RCU bit-stream. */
scale = (isRCU) ? (scale * RCU_TRANSCODING_SCALE) : scale;
streamLenLB = WebRtcIsac_EncodeStoredDataLb(
&instISAC->instLB.ISACencLB_obj.SaveEnc_obj,
&iSACBitStreamInst, bweIndex, scale);
if (streamLenLB < 0) {
return -1;
}
/* Convert from bytes to WebRtc_Word16. */
memcpy(encoded, iSACBitStreamInst.stream, streamLenLB);
if (bandwidthKHz == isac8kHz) {
return streamLenLB;
}
totalStreamLen = streamLenLB;
/* super-wideband is always at 30ms.
* These gains are in dB.
* Gain for the given rate. */
gain1 = WebRtcIsac_GetSnr(rateUB, FRAMESAMPLES);
/* Gain of this iSAC */
gain2 = WebRtcIsac_GetSnr(instISAC->instUB.ISACencUB_obj.bottleneck,
FRAMESAMPLES);
/* Scale is the ratio of two gains in normal domain. */
scale = (float)pow(10, (gain1 - gain2) / 20.0);
/* Change the scale if this is a RCU bit-stream. */
scale = (isRCU)? (scale * RCU_TRANSCODING_SCALE_UB) : scale;
streamLenUB = WebRtcIsac_EncodeStoredDataUb(
&(instISAC->instUB.ISACencUB_obj.SaveEnc_obj),
&iSACBitStreamInst, jitterInfo, scale,
instISAC->bandwidthKHz);
if (streamLenUB < 0) {
return -1;
}
if (streamLenUB + 1 + LEN_CHECK_SUM_WORD8 > 255) {
return streamLenLB;
}
totalStreamLen = streamLenLB + streamLenUB + 1 + LEN_CHECK_SUM_WORD8;
encodedPtrUW8[streamLenLB] = streamLenUB + 1 + LEN_CHECK_SUM_WORD8;
memcpy(&encodedPtrUW8[streamLenLB + 1], iSACBitStreamInst.stream,
streamLenUB);
WebRtcIsac_GetCrc((WebRtc_Word16*)(&(encodedPtrUW8[streamLenLB + 1])),
streamLenUB, &crc);
#ifndef WEBRTC_BIG_ENDIAN
for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) {
encodedPtrUW8[totalStreamLen - LEN_CHECK_SUM_WORD8 + k] =
(WebRtc_UWord8)((crc >> (24 - k * 8)) & 0xFF);
}
#else
memcpy(&encodedPtrUW8[streamLenLB + streamLenUB + 1], &crc,
LEN_CHECK_SUM_WORD8);
#endif
return totalStreamLen;
}
/****************************************************************************
* DecoderInitLb(...) - internal function for initialization of
* Lower Band
* DecoderInitUb(...) - internal function for initialization of
* Upper Band
* WebRtcIsac_DecoderInit(...) - API function
*
* This function initializes a ISAC instance prior to the decoder calls.
*
* Input:
* - ISAC_main_inst : ISAC instance.
*
* Return value
* : 0 - Ok
* -1 - Error
*/
static WebRtc_Word16 DecoderInitLb(ISACLBStruct* instISAC) {
int i;
/* Initialize stream vector to zero. */
for (i = 0; i < STREAM_SIZE_MAX_60; i++) {
instISAC->ISACdecLB_obj.bitstr_obj.stream[i] = 0;
}
WebRtcIsac_InitMasking(&instISAC->ISACdecLB_obj.maskfiltstr_obj);
WebRtcIsac_InitPostFilterbank(
&instISAC->ISACdecLB_obj.postfiltbankstr_obj);
WebRtcIsac_InitPitchFilter(&instISAC->ISACdecLB_obj.pitchfiltstr_obj);
return 0;
}
static WebRtc_Word16 DecoderInitUb(ISACUBStruct* instISAC) {
int i;
/* Init stream vector to zero */
for (i = 0; i < STREAM_SIZE_MAX_60; i++) {
instISAC->ISACdecUB_obj.bitstr_obj.stream[i] = 0;
}
WebRtcIsac_InitMasking(&instISAC->ISACdecUB_obj.maskfiltstr_obj);
WebRtcIsac_InitPostFilterbank(
&instISAC->ISACdecUB_obj.postfiltbankstr_obj);
return (0);
}
WebRtc_Word16 WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
if (DecoderInitLb(&instISAC->instLB) < 0) {
return -1;
}
if (instISAC->decoderSamplingRateKHz == kIsacSuperWideband) {
memset(instISAC->synthesisFBState1, 0,
FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
memset(instISAC->synthesisFBState2, 0,
FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
if (DecoderInitUb(&(instISAC->instUB)) < 0) {
return -1;
}
}
if ((instISAC->initFlag & BIT_MASK_ENC_INIT) != BIT_MASK_ENC_INIT) {
WebRtcIsac_InitBandwidthEstimator(&instISAC->bwestimator_obj,
instISAC->encoderSamplingRateKHz,
instISAC->decoderSamplingRateKHz);
}
instISAC->initFlag |= BIT_MASK_DEC_INIT;
instISAC->resetFlag_8kHz = 0;
return 0;
}
/****************************************************************************
* WebRtcIsac_UpdateBwEstimate(...)
*
* This function updates the estimate of the bandwidth.
*
* NOTE:
* The estimates of bandwidth is not valid if the sample rate of the far-end
* encoder is set to 48 kHz and send timestamps are increamented according to
* 48 kHz sampling rate.
*
* Input:
* - ISAC_main_inst : ISAC instance.
* - encoded : encoded ISAC frame(s).
* - packet_size : size of the packet.
* - rtp_seq_number : the RTP number of the packet.
* - arr_ts : the arrival time of the packet (from NetEq)
* in samples.
*
* Return value : 0 - Ok
* -1 - Error
*/
WebRtc_Word16 WebRtcIsac_UpdateBwEstimate(ISACStruct* ISAC_main_inst,
const WebRtc_UWord16* encoded,
WebRtc_Word32 packet_size,
WebRtc_UWord16 rtp_seq_number,
WebRtc_UWord32 send_ts,
WebRtc_UWord32 arr_ts) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
Bitstr streamdata;
#ifndef WEBRTC_BIG_ENDIAN
int k;
#endif
WebRtc_Word16 err;
/* Check if decoder initiated. */
if ((instISAC->initFlag & BIT_MASK_DEC_INIT) != BIT_MASK_DEC_INIT) {
instISAC->errorCode = ISAC_DECODER_NOT_INITIATED;
return -1;
}
if (packet_size <= 0) {
/* Return error code if the packet length is null. */
instISAC->errorCode = ISAC_EMPTY_PACKET;
return -1;
}
WebRtcIsac_ResetBitstream(&(streamdata));
#ifndef WEBRTC_BIG_ENDIAN
for (k = 0; k < 10; k++) {
streamdata.stream[k] = (WebRtc_UWord8)((encoded[k >> 1] >>
((k & 1) << 3)) & 0xFF);
}
#else
memcpy(streamdata.stream, encoded, 10);
#endif
err = WebRtcIsac_EstimateBandwidth(&instISAC->bwestimator_obj, &streamdata,
packet_size, rtp_seq_number, send_ts,
arr_ts, instISAC->encoderSamplingRateKHz,
instISAC->decoderSamplingRateKHz);
if (err < 0) {
/* Return error code if something went wrong. */
instISAC->errorCode = -err;
return -1;
}
return 0;
}
static WebRtc_Word16 Decode(ISACStruct* ISAC_main_inst,
const WebRtc_UWord16* encoded,
WebRtc_Word16 lenEncodedBytes,
WebRtc_Word16* decoded,
WebRtc_Word16* speechType,
WebRtc_Word16 isRCUPayload) {
/* Number of samples (480 or 960), output from decoder
that were actually used in the encoder/decoder
(determined on the fly). */
WebRtc_Word16 numSamplesLB;
WebRtc_Word16 numSamplesUB;
WebRtc_Word16 speechIdx;
float outFrame[MAX_FRAMESAMPLES];
WebRtc_Word16 outFrameLB[MAX_FRAMESAMPLES];
WebRtc_Word16 outFrameUB[MAX_FRAMESAMPLES];
WebRtc_Word16 numDecodedBytesLB;
WebRtc_Word16 numDecodedBytesUB;
WebRtc_Word16 lenEncodedLBBytes;
WebRtc_Word16 validChecksum = 1;
WebRtc_Word16 k;
WebRtc_UWord8* ptrEncodedUW8 = (WebRtc_UWord8*)encoded;
WebRtc_UWord16 numLayer;
WebRtc_Word16 totSizeBytes;
WebRtc_Word16 err;
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
ISACUBDecStruct* decInstUB = &(instISAC->instUB.ISACdecUB_obj);
ISACLBDecStruct* decInstLB = &(instISAC->instLB.ISACdecLB_obj);
/* Check if decoder initiated. */
if ((instISAC->initFlag & BIT_MASK_DEC_INIT) !=
BIT_MASK_DEC_INIT) {
instISAC->errorCode = ISAC_DECODER_NOT_INITIATED;
return -1;
}
if (lenEncodedBytes <= 0) {
/* return error code if the packet length is null. */
instISAC->errorCode = ISAC_EMPTY_PACKET;
return -1;
}
/* The size of the encoded lower-band is bounded by
* STREAM_SIZE_MAX. If a payload with the size larger than STREAM_SIZE_MAX
* is received, it is not considered erroneous. */
lenEncodedLBBytes = (lenEncodedBytes > STREAM_SIZE_MAX) ?
STREAM_SIZE_MAX : lenEncodedBytes;
/* Copy to lower-band bit-stream structure. */
memcpy(instISAC->instLB.ISACdecLB_obj.bitstr_obj.stream, ptrEncodedUW8,
lenEncodedLBBytes);
/* Regardless of that the current codec is setup to work in
* wideband or super-wideband, the decoding of the lower-band
* has to be performed. */
numDecodedBytesLB = WebRtcIsac_DecodeLb(outFrame, decInstLB,
&numSamplesLB, isRCUPayload);
if ((numDecodedBytesLB < 0) || (numDecodedBytesLB > lenEncodedLBBytes) ||
(numSamplesLB > MAX_FRAMESAMPLES)) {
instISAC->errorCode = ISAC_LENGTH_MISMATCH;
return -1;
}
/* Error Check, we accept multi-layer bit-stream This will limit number
* of iterations of the while loop. Even without this the number
* of iterations is limited. */
numLayer = 1;
totSizeBytes = numDecodedBytesLB;
while (totSizeBytes != lenEncodedBytes) {
if ((totSizeBytes > lenEncodedBytes) ||
(ptrEncodedUW8[totSizeBytes] == 0) ||
(numLayer > MAX_NUM_LAYERS)) {
instISAC->errorCode = ISAC_LENGTH_MISMATCH;
return -1;
}
totSizeBytes += ptrEncodedUW8[totSizeBytes];
numLayer++;
}
if (instISAC->decoderSamplingRateKHz == kIsacWideband) {
for (k = 0; k < numSamplesLB; k++) {
if (outFrame[k] > 32767) {
decoded[k] = 32767;
} else if (outFrame[k] < -32768) {
decoded[k] = -32768;
} else {
decoded[k] = (WebRtc_Word16)WebRtcIsac_lrint(outFrame[k]);
}
}
numSamplesUB = 0;
} else {
WebRtc_UWord32 crc;
/* We don't accept larger than 30ms (480 samples at lower-band)
* frame-size. */
for (k = 0; k < numSamplesLB; k++) {
if (outFrame[k] > 32767) {
outFrameLB[k] = 32767;
} else if (outFrame[k] < -32768) {
outFrameLB[k] = -32768;
} else {
outFrameLB[k] = (WebRtc_Word16)WebRtcIsac_lrint(outFrame[k]);
}
}
/* Check for possible error, and if upper-band stream exists. */
if (numDecodedBytesLB == lenEncodedBytes) {
/* Decoding was successful. No super-wideband bit-stream exists. */
numSamplesUB = numSamplesLB;
memset(outFrameUB, 0, sizeof(WebRtc_Word16) * numSamplesUB);
/* Prepare for the potential increase of signal bandwidth. */
instISAC->resetFlag_8kHz = 2;
} else {
/* This includes the checksum and the bytes that stores the length. */
WebRtc_Word16 lenNextStream = ptrEncodedUW8[numDecodedBytesLB];
/* Is this garbage or valid super-wideband bit-stream?
* Check if checksum is valid. */
if (lenNextStream <= (LEN_CHECK_SUM_WORD8 + 1)) {
/* Such a small second layer cannot be super-wideband layer.
* It must be a short garbage. */
validChecksum = 0;
} else {
/* Run CRC to see if the checksum match. */
WebRtcIsac_GetCrc((WebRtc_Word16*)(
&ptrEncodedUW8[numDecodedBytesLB + 1]),
lenNextStream - LEN_CHECK_SUM_WORD8 - 1, &crc);
validChecksum = 1;
for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) {
validChecksum &= (((crc >> (24 - k * 8)) & 0xFF) ==
ptrEncodedUW8[numDecodedBytesLB + lenNextStream -
LEN_CHECK_SUM_WORD8 + k]);
}
}
if (!validChecksum) {
/* This is a garbage, we have received a wideband
* bit-stream with garbage. */
numSamplesUB = numSamplesLB;
memset(outFrameUB, 0, sizeof(WebRtc_Word16) * numSamplesUB);
} else {
/* A valid super-wideband biststream exists. */
enum ISACBandwidth bandwidthKHz;
WebRtc_Word32 maxDelayBit;
/* If we have super-wideband bit-stream, we cannot
* have 60 ms frame-size. */
if (numSamplesLB > FRAMESAMPLES) {
instISAC->errorCode = ISAC_LENGTH_MISMATCH;
return -1;
}
/* The rest of the bit-stream contains the upper-band
* bit-stream curently this is the only thing there,
* however, we might add more layers. */
/* Have to exclude one byte where the length is stored
* and last 'LEN_CHECK_SUM_WORD8' bytes where the
* checksum is stored. */
lenNextStream -= (LEN_CHECK_SUM_WORD8 + 1);
memcpy(decInstUB->bitstr_obj.stream,
&ptrEncodedUW8[numDecodedBytesLB + 1], lenNextStream);
/* Reset bit-stream object, this is the first decoding. */
WebRtcIsac_ResetBitstream(&(decInstUB->bitstr_obj));
/* Decode jitter information. */
err = WebRtcIsac_DecodeJitterInfo(&decInstUB->bitstr_obj, &maxDelayBit);
if (err < 0) {
instISAC->errorCode = -err;
return -1;
}
/* Update jitter info which is in the upper-band bit-stream
* only if the encoder is in super-wideband. Otherwise,
* the jitter info is already embedded in bandwidth index
* and has been updated. */
if (instISAC->encoderSamplingRateKHz == kIsacSuperWideband) {
err = WebRtcIsac_UpdateUplinkJitter(
&(instISAC->bwestimator_obj), maxDelayBit);
if (err < 0) {
instISAC->errorCode = -err;
return -1;
}
}
/* Decode bandwidth information. */
err = WebRtcIsac_DecodeBandwidth(&decInstUB->bitstr_obj,
&bandwidthKHz);
if (err < 0) {
instISAC->errorCode = -err;
return -1;
}
switch (bandwidthKHz) {
case isac12kHz: {
numDecodedBytesUB = WebRtcIsac_DecodeUb12(outFrame, decInstUB,
isRCUPayload);
/* Hang-over for transient alleviation -
* wait two frames to add the upper band going up from 8 kHz. */
if (instISAC->resetFlag_8kHz > 0) {
if (instISAC->resetFlag_8kHz == 2) {
/* Silence first and a half frame. */
memset(outFrame, 0, MAX_FRAMESAMPLES *
sizeof(float));
} else {
const float rampStep = 2.0f / MAX_FRAMESAMPLES;
float rampVal = 0;
memset(outFrame, 0, (MAX_FRAMESAMPLES >> 1) *
sizeof(float));
/* Ramp up second half of second frame. */
for (k = MAX_FRAMESAMPLES / 2; k < MAX_FRAMESAMPLES; k++) {
outFrame[k] *= rampVal;
rampVal += rampStep;
}
}
instISAC->resetFlag_8kHz -= 1;
}
break;
}
case isac16kHz: {
numDecodedBytesUB = WebRtcIsac_DecodeUb16(outFrame, decInstUB,
isRCUPayload);
break;
}
default:
return -1;
}
/* It might be less due to garbage. */
if ((numDecodedBytesUB != lenNextStream) &&
(numDecodedBytesUB != (lenNextStream -
ptrEncodedUW8[numDecodedBytesLB + 1 + numDecodedBytesUB]))) {
instISAC->errorCode = ISAC_LENGTH_MISMATCH;
return -1;
}
/* If there is no error Upper-band always decodes
* 30 ms (480 samples). */
numSamplesUB = FRAMESAMPLES;
/* Convert to W16. */
for (k = 0; k < numSamplesUB; k++) {
if (outFrame[k] > 32767) {
outFrameUB[k] = 32767;
} else if (outFrame[k] < -32768) {
outFrameUB[k] = -32768;
} else {
outFrameUB[k] = (WebRtc_Word16)WebRtcIsac_lrint(
outFrame[k]);
}
}
}
}
speechIdx = 0;
while (speechIdx < numSamplesLB) {
WebRtcSpl_SynthesisQMF(&outFrameLB[speechIdx], &outFrameUB[speechIdx],
&decoded[(speechIdx << 1)],
instISAC->synthesisFBState1,
instISAC->synthesisFBState2);
speechIdx += FRAMESAMPLES_10ms;
}
}
*speechType = 0;
return (numSamplesLB + numSamplesUB);
}
/****************************************************************************
* WebRtcIsac_Decode(...)
*
* This function decodes a ISAC frame. Output speech length
* will be a multiple of 480 samples: 480 or 960 samples,
* depending on the frameSize (30 or 60 ms).
*
* Input:
* - ISAC_main_inst : ISAC instance.
* - encoded : encoded ISAC frame(s)
* - len : bytes in encoded vector
*
* Output:
* - decoded : The decoded vector
*
* Return value : >0 - number of samples in decoded vector
* -1 - Error
*/
WebRtc_Word16 WebRtcIsac_Decode(ISACStruct* ISAC_main_inst,
const WebRtc_UWord16* encoded,
WebRtc_Word16 lenEncodedBytes,
WebRtc_Word16* decoded,
WebRtc_Word16* speechType) {
WebRtc_Word16 isRCUPayload = 0;
return Decode(ISAC_main_inst, encoded, lenEncodedBytes, decoded,
speechType, isRCUPayload);
}
/****************************************************************************
* WebRtcIsac_DecodeRcu(...)
*
* This function decodes a redundant (RCU) iSAC frame. Function is called in
* NetEq with a stored RCU payload in case of packet loss. Output speech length
* will be a multiple of 480 samples: 480 or 960 samples,
* depending on the framesize (30 or 60 ms).
*
* Input:
* - ISAC_main_inst : ISAC instance.
* - encoded : encoded ISAC RCU frame(s)
* - len : bytes in encoded vector
*
* Output:
* - decoded : The decoded vector
*
* Return value : >0 - number of samples in decoded vector
* -1 - Error
*/
WebRtc_Word16 WebRtcIsac_DecodeRcu(ISACStruct* ISAC_main_inst,
const WebRtc_UWord16* encoded,
WebRtc_Word16 lenEncodedBytes,
WebRtc_Word16* decoded,
WebRtc_Word16* speechType) {
WebRtc_Word16 isRCUPayload = 1;
return Decode(ISAC_main_inst, encoded, lenEncodedBytes, decoded,
speechType, isRCUPayload);
}
/****************************************************************************
* WebRtcIsac_DecodePlc(...)
*
* This function conducts PLC for ISAC frame(s). Output speech length
* will be a multiple of 480 samples: 480 or 960 samples,
* depending on the frameSize (30 or 60 ms).
*
* Input:
* - ISAC_main_inst : ISAC instance.
* - noOfLostFrames : Number of PLC frames to produce
*
* Output:
* - decoded : The decoded vector
*
* Return value : >0 - number of samples in decoded PLC vector
* -1 - Error
*/
WebRtc_Word16 WebRtcIsac_DecodePlc(ISACStruct* ISAC_main_inst,
WebRtc_Word16* decoded,
WebRtc_Word16 noOfLostFrames) {
WebRtc_Word16 numSamples = 0;
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
/* Limit number of frames to two = 60 millisecond.
* Otherwise we exceed data vectors. */
if (noOfLostFrames > 2) {
noOfLostFrames = 2;
}
/* Get the number of samples per frame */
switch (instISAC->decoderSamplingRateKHz) {
case kIsacWideband: {
numSamples = 480 * noOfLostFrames;
break;
}
case kIsacSuperWideband: {
numSamples = 960 * noOfLostFrames;
break;
}
}
/* Set output samples to zero. */
memset(decoded, 0, numSamples * sizeof(WebRtc_Word16));
return numSamples;
}
/****************************************************************************
* ControlLb(...) - Internal function for controlling Lower Band
* ControlUb(...) - Internal function for controlling Upper Band
* WebRtcIsac_Control(...) - API function
*
* This function sets the limit on the short-term average bit rate and the
* frame length. Should be used only in Instantaneous mode.
*
* Input:
* - ISAC_main_inst : ISAC instance.
* - rate : limit on the short-term average bit rate,
* in bits/second (between 10000 and 32000)
* - frameSize : number of milliseconds per frame (30 or 60)
*
* Return value : 0 - ok
* -1 - Error
*/
static WebRtc_Word16 ControlLb(ISACLBStruct* instISAC, double rate,
WebRtc_Word16 frameSize) {
if ((rate >= 10000) && (rate <= 32000)) {
instISAC->ISACencLB_obj.bottleneck = rate;
} else {
return -ISAC_DISALLOWED_BOTTLENECK;
}
if ((frameSize == 30) || (frameSize == 60)) {
instISAC->ISACencLB_obj.new_framelength = (FS / 1000) * frameSize;
} else {
return -ISAC_DISALLOWED_FRAME_LENGTH;
}
return 0;
}
static WebRtc_Word16 ControlUb(ISACUBStruct* instISAC, double rate) {
if ((rate >= 10000) && (rate <= 32000)) {
instISAC->ISACencUB_obj.bottleneck = rate;
} else {
return -ISAC_DISALLOWED_BOTTLENECK;
}
return 0;
}
WebRtc_Word16 WebRtcIsac_Control(ISACStruct* ISAC_main_inst,
WebRtc_Word32 bottleneckBPS,
WebRtc_Word16 frameSize) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
WebRtc_Word16 status;
double rateLB;
double rateUB;
enum ISACBandwidth bandwidthKHz;
if (instISAC->codingMode == 0) {
/* In adaptive mode. */
instISAC->errorCode = ISAC_MODE_MISMATCH;
return -1;
}
/* Check if encoder initiated */
if ((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
BIT_MASK_ENC_INIT) {
instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
return -1;
}
if (instISAC->encoderSamplingRateKHz == kIsacWideband) {
/* If the sampling rate is 16kHz then bandwith should be 8kHz,
* regardless of bottleneck. */
bandwidthKHz = isac8kHz;
rateLB = (bottleneckBPS > 32000) ? 32000 : bottleneckBPS;
rateUB = 0;
} else {
if (WebRtcIsac_RateAllocation(bottleneckBPS, &rateLB, &rateUB,
&bandwidthKHz) < 0) {
return -1;
}
}
if ((instISAC->encoderSamplingRateKHz == kIsacSuperWideband) &&
(frameSize != 30) &&
(bandwidthKHz != isac8kHz)) {
/* Cannot have 60 ms in super-wideband. */
instISAC->errorCode = ISAC_DISALLOWED_FRAME_LENGTH;
return -1;
}
status = ControlLb(&instISAC->instLB, rateLB, frameSize);
if (status < 0) {
instISAC->errorCode = -status;
return -1;
}
if (bandwidthKHz != isac8kHz) {
status = ControlUb(&(instISAC->instUB), rateUB);
if (status < 0) {
instISAC->errorCode = -status;
return -1;
}
}
/* Check if bandwidth is changing from wideband to super-wideband
* then we have to synch data buffer of lower & upper-band. Also
* clean up the upper-band data buffer. */
if ((instISAC->bandwidthKHz == isac8kHz) && (bandwidthKHz != isac8kHz)) {
memset(instISAC->instUB.ISACencUB_obj.data_buffer_float, 0,
sizeof(float) * (MAX_FRAMESAMPLES + LB_TOTAL_DELAY_SAMPLES));
if (bandwidthKHz == isac12kHz) {
instISAC->instUB.ISACencUB_obj.buffer_index =
instISAC->instLB.ISACencLB_obj.buffer_index;
} else {
instISAC->instUB.ISACencUB_obj.buffer_index =
LB_TOTAL_DELAY_SAMPLES + instISAC->instLB.ISACencLB_obj.buffer_index;
memcpy(&(instISAC->instUB.ISACencUB_obj.lastLPCVec),
WebRtcIsac_kMeanLarUb16, sizeof(double) * UB_LPC_ORDER);
}
}
/* Update the payload limit if the bandwidth is changing. */
if (instISAC->bandwidthKHz != bandwidthKHz) {
instISAC->bandwidthKHz = bandwidthKHz;
UpdatePayloadSizeLimit(instISAC);
}
instISAC->bottleneck = bottleneckBPS;
return 0;
}
/****************************************************************************
* WebRtcIsac_ControlBwe(...)
*
* This function sets the initial values of bottleneck and frame-size if
* iSAC is used in channel-adaptive mode. Through this API, users can
* enforce a frame-size for all values of bottleneck. Then iSAC will not
* automatically change the frame-size.
*
*
* Input:
* - ISAC_main_inst : ISAC instance.
* - rateBPS : initial value of bottleneck in bits/second
* 10000 <= rateBPS <= 32000 is accepted
* For default bottleneck set rateBPS = 0
* - frameSizeMs : number of milliseconds per frame (30 or 60)
* - enforceFrameSize : 1 to enforce the given frame-size through out
* the adaptation process, 0 to let iSAC change
* the frame-size if required.
*
* Return value : 0 - ok
* -1 - Error
*/
WebRtc_Word16 WebRtcIsac_ControlBwe(ISACStruct* ISAC_main_inst,
WebRtc_Word32 bottleneckBPS,
WebRtc_Word16 frameSizeMs,
WebRtc_Word16 enforceFrameSize) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
enum ISACBandwidth bandwidth;
/* Check if encoder initiated */
if ((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
BIT_MASK_ENC_INIT) {
instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
return -1;
}
/* Check that we are in channel-adaptive mode, otherwise, return (-1) */
if (instISAC->codingMode != 0) {
instISAC->errorCode = ISAC_MODE_MISMATCH;
return -1;
}
if ((frameSizeMs != 30) &&
(instISAC->encoderSamplingRateKHz == kIsacSuperWideband)) {
return -1;
}
/* Set structure variable if enforceFrameSize is set. ISAC will then
* keep the chosen frame size. */
if (enforceFrameSize != 0) {
instISAC->instLB.ISACencLB_obj.enforceFrameSize = 1;
} else {
instISAC->instLB.ISACencLB_obj.enforceFrameSize = 0;
}
/* Set the initial rate. If the input value is zero then the default intial
* rate is used. Otehrwise, values between 10 to 32 kbps are accepted. */
if (bottleneckBPS != 0) {
double rateLB;
double rateUB;
if (WebRtcIsac_RateAllocation(bottleneckBPS, &rateLB, &rateUB,
&bandwidth) < 0) {
return -1;
}
instISAC->bwestimator_obj.send_bw_avg = (float)bottleneckBPS;
instISAC->bandwidthKHz = bandwidth;
}
/* Set the initial frame-size. If 'enforceFrameSize' is set, the frame-size
* will not change */
if (frameSizeMs != 0) {
if ((frameSizeMs == 30) || (frameSizeMs == 60)) {
instISAC->instLB.ISACencLB_obj.new_framelength = (FS / 1000) *
frameSizeMs;
} else {
instISAC->errorCode = ISAC_DISALLOWED_FRAME_LENGTH;
return -1;
}
}
return 0;
}
/****************************************************************************
* WebRtcIsac_GetDownLinkBwIndex(...)
*
* This function returns index representing the Bandwidth estimate from
* the other side to this side.
*
* Input:
* - ISAC_main_inst : iSAC structure
*
* Output:
* - bweIndex : Bandwidth estimate to transmit to other side.
*
*/
WebRtc_Word16 WebRtcIsac_GetDownLinkBwIndex(ISACStruct* ISAC_main_inst,
WebRtc_Word16* bweIndex,
WebRtc_Word16* jitterInfo) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
/* Check if encoder initialized. */
if ((instISAC->initFlag & BIT_MASK_DEC_INIT) !=
BIT_MASK_DEC_INIT) {
instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
return -1;
}
/* Call function to get Bandwidth Estimate. */
WebRtcIsac_GetDownlinkBwJitIndexImpl(&(instISAC->bwestimator_obj), bweIndex,
jitterInfo,
instISAC->decoderSamplingRateKHz);
return 0;
}
/****************************************************************************
* WebRtcIsac_UpdateUplinkBw(...)
*
* This function takes an index representing the Bandwidth estimate from
* this side to other side and updates BWE.
*
* Input:
* - ISAC_main_inst : iSAC structure
* - rateIndex : Bandwidth estimate from other side.
*
* Return value : 0 - ok
* -1 - index out of range
*/
WebRtc_Word16 WebRtcIsac_UpdateUplinkBw(ISACStruct* ISAC_main_inst,
WebRtc_Word16 bweIndex) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
WebRtc_Word16 returnVal;
/* Check if encoder initiated. */
if ((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
BIT_MASK_ENC_INIT) {
instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
return -1;
}
/* Call function to get Bandwidth Estimate. */
returnVal = WebRtcIsac_UpdateUplinkBwImpl(
&(instISAC->bwestimator_obj), bweIndex,
instISAC->encoderSamplingRateKHz);
if (returnVal < 0) {
instISAC->errorCode = -returnVal;
return -1;
} else {
return 0;
}
}
/****************************************************************************
* WebRtcIsac_ReadBwIndex(...)
*
* This function returns the index of the Bandwidth estimate from the
* bit-stream.
*
* Input:
* - encoded : Encoded bit-stream
*
* Output:
* - frameLength : Length of frame in packet (in samples)
* - bweIndex : Bandwidth estimate in bit-stream
*
*/
WebRtc_Word16 WebRtcIsac_ReadBwIndex(const WebRtc_Word16* encoded,
WebRtc_Word16* bweIndex) {
Bitstr streamdata;
#ifndef WEBRTC_BIG_ENDIAN
int k;
#endif
WebRtc_Word16 err;
WebRtcIsac_ResetBitstream(&(streamdata));
#ifndef WEBRTC_BIG_ENDIAN
for (k = 0; k < 10; k++) {
streamdata.stream[k] = (WebRtc_UWord8)((encoded[k >> 1] >>
((k & 1) << 3)) & 0xFF);
}
#else
memcpy(streamdata.stream, encoded, 10);
#endif
/* Decode frame length. */
err = WebRtcIsac_DecodeFrameLen(&streamdata, bweIndex);
if (err < 0) {
return err;
}
/* Decode BW estimation. */
err = WebRtcIsac_DecodeSendBW(&streamdata, bweIndex);
if (err < 0) {
return err;
}
return 0;
}
/****************************************************************************
* WebRtcIsac_ReadFrameLen(...)
*
* This function returns the number of samples the decoder will generate if
* the given payload is decoded.
*
* Input:
* - encoded : Encoded bitstream
*
* Output:
* - frameLength : Length of frame in packet (in samples)
*
*/
WebRtc_Word16 WebRtcIsac_ReadFrameLen(ISACStruct* ISAC_main_inst,
const WebRtc_Word16* encoded,
WebRtc_Word16* frameLength) {
Bitstr streamdata;
#ifndef WEBRTC_BIG_ENDIAN
int k;
#endif
WebRtc_Word16 err;
ISACMainStruct* instISAC;
WebRtcIsac_ResetBitstream(&(streamdata));
#ifndef WEBRTC_BIG_ENDIAN
for (k = 0; k < 10; k++) {
streamdata.stream[k] = (WebRtc_UWord8)((encoded[k >> 1] >>
((k & 1) << 3)) & 0xFF);
}
#else
memcpy(streamdata.stream, encoded, 10);
#endif
/* Decode frame length. */
err = WebRtcIsac_DecodeFrameLen(&streamdata, frameLength);
if (err < 0) {
return -1;
}
instISAC = (ISACMainStruct*)ISAC_main_inst;
if (instISAC->decoderSamplingRateKHz == kIsacSuperWideband) {
/* The decoded frame length indicates the number of samples in
* lower-band in this case, multiply by 2 to get the total number
* of samples. */
*frameLength <<= 1;
}
return 0;
}
/*******************************************************************************
* WebRtcIsac_GetNewFrameLen(...)
*
* This function returns the frame length (in samples) of the next packet.
* In the case of channel-adaptive mode, iSAC decides on its frame length based
* on the estimated bottleneck, this AOI allows a user to prepare for the next
* packet (at the encoder).
*
* The primary usage is in CE to make the iSAC works in channel-adaptive mode
*
* Input:
* - ISAC_main_inst : iSAC struct
*
* Return Value : frame lenght in samples
*
*/
WebRtc_Word16 WebRtcIsac_GetNewFrameLen(ISACStruct* ISAC_main_inst) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
/* Return new frame length. */
if (instISAC->in_sample_rate_hz == 16000)
return (instISAC->instLB.ISACencLB_obj.new_framelength);
else if (instISAC->in_sample_rate_hz == 32000)
return ((instISAC->instLB.ISACencLB_obj.new_framelength) * 2);
else
return ((instISAC->instLB.ISACencLB_obj.new_framelength) * 3);
}
/****************************************************************************
* WebRtcIsac_GetErrorCode(...)
*
* This function can be used to check the error code of an iSAC instance.
* When a function returns -1 an error code will be set for that instance.
* The function below extracts the code of the last error that occurred in
* the specified instance.
*
* Input:
* - ISAC_main_inst : ISAC instance
*
* Return value : Error code
*/
WebRtc_Word16 WebRtcIsac_GetErrorCode(ISACStruct* ISAC_main_inst) {
return ((ISACMainStruct*)ISAC_main_inst)->errorCode;
}
/****************************************************************************
* WebRtcIsac_GetUplinkBw(...)
*
* This function outputs the target bottleneck of the codec. In
* channel-adaptive mode, the target bottleneck is specified through an in-band
* signalling retrieved by bandwidth estimator.
* In channel-independent, also called instantaneous mode, the target
* bottleneck is provided to the encoder by calling xxx_control(...) (if
* xxx_control is never called, the default values are used.).
* Note that the output is the iSAC internal operating bottleneck which might
* differ slightly from the one provided through xxx_control().
*
* Input:
* - ISAC_main_inst : iSAC instance
*
* Output:
* - *bottleneck : bottleneck in bits/sec
*
* Return value : -1 if error happens
* 0 bit-rates computed correctly.
*/
WebRtc_Word16 WebRtcIsac_GetUplinkBw(ISACStruct* ISAC_main_inst,
WebRtc_Word32* bottleneck) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
if (instISAC->codingMode == 0) {
/* We are in adaptive mode then get the bottleneck from BWE. */
*bottleneck = (WebRtc_Word32)instISAC->bwestimator_obj.send_bw_avg;
} else {
*bottleneck = instISAC->bottleneck;
}
if ((*bottleneck > 32000) && (*bottleneck < 38000)) {
*bottleneck = 32000;
} else if ((*bottleneck > 45000) && (*bottleneck < 50000)) {
*bottleneck = 45000;
} else if (*bottleneck > 56000) {
*bottleneck = 56000;
}
return 0;
}
/******************************************************************************
* WebRtcIsac_SetMaxPayloadSize(...)
*
* This function sets a limit for the maximum payload size of iSAC. The same
* value is used both for 30 and 60 ms packets. If the encoder sampling rate
* is 16 kHz the maximum payload size is between 120 and 400 bytes. If the
* encoder sampling rate is 32 kHz the maximum payload size is between 120
* and 600 bytes.
*
* ---------------
* IMPORTANT NOTES
* ---------------
* The size of a packet is limited to the minimum of 'max-payload-size' and
* 'max-rate.' For instance, let's assume the max-payload-size is set to
* 170 bytes, and max-rate is set to 40 kbps. Note that a limit of 40 kbps
* translates to 150 bytes for 30ms frame-size & 300 bytes for 60ms
* frame-size. Then a packet with a frame-size of 30 ms is limited to 150,
* i.e. min(170, 150), and a packet with 60 ms frame-size is limited to
* 170 bytes, i.e. min(170, 300).
*
* Input:
* - ISAC_main_inst : iSAC instance
* - maxPayloadBytes : maximum size of the payload in bytes
* valid values are between 100 and 400 bytes
* if encoder sampling rate is 16 kHz. For
* 32 kHz encoder sampling rate valid values
* are between 100 and 600 bytes.
*
* Return value : 0 if successful
* -1 if error happens
*/
WebRtc_Word16 WebRtcIsac_SetMaxPayloadSize(ISACStruct* ISAC_main_inst,
WebRtc_Word16 maxPayloadBytes) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
WebRtc_Word16 status = 0;
/* Check if encoder initiated */
if ((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
BIT_MASK_ENC_INIT) {
instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
return -1;
}
if (instISAC->encoderSamplingRateKHz == kIsacSuperWideband) {
/* Sanity check. */
if (maxPayloadBytes < 120) {
/* 'maxRate' is out of valid range
* set to the acceptable value and return -1. */
maxPayloadBytes = 120;
status = -1;
}
/* sanity check */
if (maxPayloadBytes > STREAM_SIZE_MAX) {
/* maxRate is out of valid range,
* set to the acceptable value and return -1. */
maxPayloadBytes = STREAM_SIZE_MAX;
status = -1;
}
} else {
if (maxPayloadBytes < 120) {
/* Max payload-size is out of valid range
* set to the acceptable value and return -1. */
maxPayloadBytes = 120;
status = -1;
}
if (maxPayloadBytes > STREAM_SIZE_MAX_60) {
/* Max payload-size is out of valid range
* set to the acceptable value and return -1. */
maxPayloadBytes = STREAM_SIZE_MAX_60;
status = -1;
}
}
instISAC->maxPayloadSizeBytes = maxPayloadBytes;
UpdatePayloadSizeLimit(instISAC);
return status;
}
/******************************************************************************
* WebRtcIsac_SetMaxRate(...)
*
* This function sets the maximum rate which the codec may not exceed for
* any signal packet. The maximum rate is defined and payload-size per
* frame-size in bits per second.
*
* The codec has a maximum rate of 53400 bits per second (200 bytes per 30
* ms) if the encoder sampling rate is 16kHz, and 160 kbps (600 bytes/30 ms)
* if the encoder sampling rate is 32 kHz.
*
* It is possible to set a maximum rate between 32000 and 53400 bits/sec
* in wideband mode, and 32000 to 160000 bits/sec in super-wideband mode.
*
* ---------------
* IMPORTANT NOTES
* ---------------
* The size of a packet is limited to the minimum of 'max-payload-size' and
* 'max-rate.' For instance, let's assume the max-payload-size is set to
* 170 bytes, and max-rate is set to 40 kbps. Note that a limit of 40 kbps
* translates to 150 bytes for 30ms frame-size & 300 bytes for 60ms
* frame-size. Then a packet with a frame-size of 30 ms is limited to 150,
* i.e. min(170, 150), and a packet with 60 ms frame-size is limited to
* 170 bytes, min(170, 300).
*
* Input:
* - ISAC_main_inst : iSAC instance
* - maxRate : maximum rate in bits per second,
* valid values are 32000 to 53400 bits/sec in
* wideband mode, and 32000 to 160000 bits/sec in
* super-wideband mode.
*
* Return value : 0 if successful
* -1 if error happens
*/
WebRtc_Word16 WebRtcIsac_SetMaxRate(ISACStruct* ISAC_main_inst,
WebRtc_Word32 maxRate) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
WebRtc_Word16 maxRateInBytesPer30Ms;
WebRtc_Word16 status = 0;
/* check if encoder initiated */
if ((instISAC->initFlag & BIT_MASK_ENC_INIT) != BIT_MASK_ENC_INIT) {
instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
return -1;
}
/* Calculate maximum number of bytes per 30 msec packets for the
given maximum rate. Multiply with 30/1000 to get number of
bits per 30 ms, divide by 8 to get number of bytes per 30 ms:
maxRateInBytes = floor((maxRate * 30/1000) / 8); */
maxRateInBytesPer30Ms = (WebRtc_Word16)(maxRate * 3 / 800);
if (instISAC->encoderSamplingRateKHz == kIsacWideband) {
if (maxRate < 32000) {
/* 'maxRate' is out of valid range.
* Set to the acceptable value and return -1. */
maxRateInBytesPer30Ms = 120;
status = -1;
}
if (maxRate > 53400) {
/* 'maxRate' is out of valid range.
* Set to the acceptable value and return -1. */
maxRateInBytesPer30Ms = 200;
status = -1;
}
} else {
if (maxRateInBytesPer30Ms < 120) {
/* 'maxRate' is out of valid range
* Set to the acceptable value and return -1. */
maxRateInBytesPer30Ms = 120;
status = -1;
}
if (maxRateInBytesPer30Ms > STREAM_SIZE_MAX) {
/* 'maxRate' is out of valid range.
* Set to the acceptable value and return -1. */
maxRateInBytesPer30Ms = STREAM_SIZE_MAX;
status = -1;
}
}
instISAC->maxRateBytesPer30Ms = maxRateInBytesPer30Ms;
UpdatePayloadSizeLimit(instISAC);
return status;
}
/****************************************************************************
* WebRtcIsac_GetRedPayload(...)
*
* This function populates "encoded" with the redundant payload of the recently
* encodedframe. This function has to be called once that WebRtcIsac_Encode(...)
* returns a positive value. Regardless of the frame-size this function will
* be called only once after encoding is completed. The bit-stream is
* targeted for 16000 bit/sec.
*
* Input:
* - ISAC_main_inst : iSAC struct
*
* Output:
* - encoded : the encoded data vector
*
*
* Return value : >0 - Length (in bytes) of coded data
* : -1 - Error
*/
WebRtc_Word16 WebRtcIsac_GetRedPayload(ISACStruct* ISAC_main_inst,
WebRtc_Word16* encoded) {
Bitstr iSACBitStreamInst;
WebRtc_Word16 streamLenLB;
WebRtc_Word16 streamLenUB;
WebRtc_Word16 streamLen;
WebRtc_Word16 totalLenUB;
WebRtc_UWord8* ptrEncodedUW8 = (WebRtc_UWord8*)encoded;
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
#ifndef WEBRTC_BIG_ENDIAN
int k;
#endif
if ((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
BIT_MASK_ENC_INIT) {
instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
}
WebRtcIsac_ResetBitstream(&(iSACBitStreamInst));
streamLenLB = WebRtcIsac_EncodeStoredDataLb(
&instISAC->instLB.ISACencLB_obj.SaveEnc_obj,
&iSACBitStreamInst,
instISAC->instLB.ISACencLB_obj.lastBWIdx,
RCU_TRANSCODING_SCALE);
if (streamLenLB < 0) {
return -1;
}
/* convert from bytes to WebRtc_Word16. */
memcpy(ptrEncodedUW8, iSACBitStreamInst.stream, streamLenLB);
streamLen = streamLenLB;
if (instISAC->bandwidthKHz == isac8kHz) {
return streamLenLB;
}
streamLenUB = WebRtcIsac_GetRedPayloadUb(
&instISAC->instUB.ISACencUB_obj.SaveEnc_obj,
&iSACBitStreamInst, instISAC->bandwidthKHz);
if (streamLenUB < 0) {
/* An error has happened but this is not the error due to a
* bit-stream larger than the limit. */
return -1;
}
/* We have one byte to write the total length of the upper-band.
* The length includes the bit-stream length, check-sum and the
* single byte where the length is written to. This is according to
* iSAC wideband and how the "garbage" is dealt. */
totalLenUB = streamLenUB + 1 + LEN_CHECK_SUM_WORD8;
if (totalLenUB > 255) {
streamLenUB = 0;
}
/* Generate CRC if required. */
if ((instISAC->bandwidthKHz != isac8kHz) &&
(streamLenUB > 0)) {
WebRtc_UWord32 crc;
streamLen += totalLenUB;
ptrEncodedUW8[streamLenLB] = (WebRtc_UWord8)totalLenUB;
memcpy(&ptrEncodedUW8[streamLenLB + 1], iSACBitStreamInst.stream,
streamLenUB);
WebRtcIsac_GetCrc((WebRtc_Word16*)(&(ptrEncodedUW8[streamLenLB + 1])),
streamLenUB, &crc);
#ifndef WEBRTC_BIG_ENDIAN
for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) {
ptrEncodedUW8[streamLen - LEN_CHECK_SUM_WORD8 + k] =
(WebRtc_UWord8)((crc >> (24 - k * 8)) & 0xFF);
}
#else
memcpy(&ptrEncodedUW8[streamLenLB + streamLenUB + 1], &crc,
LEN_CHECK_SUM_WORD8);
#endif
}
return streamLen;
}
/****************************************************************************
* WebRtcIsac_version(...)
*
* This function returns the version number.
*
* Output:
* - version : Pointer to character string
*
*/
void WebRtcIsac_version(char* version) {
strcpy(version, "4.3.0");
}
/******************************************************************************
* WebRtcIsac_SetEncSampRate()
* This function sets the sampling rate of the encoder. Initialization of the
* encoder WILL NOT overwrite the sampling rate of the encoder. The default
* value is 16 kHz which is set when the instance is created. The encoding-mode
* and the bottleneck remain unchanged by this call, however, the maximum rate
* and maximum payload-size will be reset to their default values.
*
* NOTE:
* The maximum internal sampling rate is 32 kHz. If the encoder sample rate is
* set to 48 kHz the input is expected to be at 48 kHz but will be resampled to
* 32 kHz before any further processing.
* This mode is created for compatibility with full-band codecs if iSAC is used
* in dual-streaming. See SetDecSampleRate() for sampling rates at the decoder.
*
* Input:
* - ISAC_main_inst : iSAC instance
* - sample_rate_hz : sampling rate in Hertz, valid values are 16000,
* 32000 and 48000.
*
* Return value : 0 if successful
* -1 if failed.
*/
WebRtc_Word16 WebRtcIsac_SetEncSampRate(ISACStruct* ISAC_main_inst,
WebRtc_UWord16 sample_rate_hz) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
enum IsacSamplingRate encoder_operational_rate;
if ((sample_rate_hz != 16000) && (sample_rate_hz != 32000) &&
(sample_rate_hz != 48000)) {
/* Sampling Frequency is not supported. */
instISAC->errorCode = ISAC_UNSUPPORTED_SAMPLING_FREQUENCY;
return -1;
}
if (sample_rate_hz == 16000) {
encoder_operational_rate = kIsacWideband;
} else {
encoder_operational_rate = kIsacSuperWideband;
}
if ((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
BIT_MASK_ENC_INIT) {
if (encoder_operational_rate == kIsacWideband) {
instISAC->bandwidthKHz = isac8kHz;
} else {
instISAC->bandwidthKHz = isac16kHz;
}
} else {
ISACUBStruct* instUB = &(instISAC->instUB);
ISACLBStruct* instLB = &(instISAC->instLB);
double bottleneckLB;
double bottleneckUB;
WebRtc_Word32 bottleneck = instISAC->bottleneck;
WebRtc_Word16 codingMode = instISAC->codingMode;
WebRtc_Word16 frameSizeMs = instLB->ISACencLB_obj.new_framelength /
(FS / 1000);
if ((encoder_operational_rate == kIsacWideband) &&
(instISAC->encoderSamplingRateKHz == kIsacSuperWideband)) {
/* Changing from super-wideband to wideband.
* we don't need to re-initialize the encoder of the lower-band. */
instISAC->bandwidthKHz = isac8kHz;
if (codingMode == 1) {
ControlLb(instLB,
(bottleneck > 32000) ? 32000 : bottleneck, FRAMESIZE);
}
instISAC->maxPayloadSizeBytes = STREAM_SIZE_MAX_60;
instISAC->maxRateBytesPer30Ms = STREAM_SIZE_MAX_30;
} else if ((encoder_operational_rate == kIsacSuperWideband) &&
(instISAC->encoderSamplingRateKHz == kIsacWideband)) {
if (codingMode == 1) {
WebRtcIsac_RateAllocation(bottleneck, &bottleneckLB, &bottleneckUB,
&(instISAC->bandwidthKHz));
}
instISAC->bandwidthKHz = isac16kHz;
instISAC->maxPayloadSizeBytes = STREAM_SIZE_MAX;
instISAC->maxRateBytesPer30Ms = STREAM_SIZE_MAX;
EncoderInitLb(instLB, codingMode, encoder_operational_rate);
EncoderInitUb(instUB, instISAC->bandwidthKHz);
memset(instISAC->analysisFBState1, 0,
FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
memset(instISAC->analysisFBState2, 0,
FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
if (codingMode == 1) {
instISAC->bottleneck = bottleneck;
ControlLb(instLB, bottleneckLB,
(instISAC->bandwidthKHz == isac8kHz) ? frameSizeMs:FRAMESIZE);
if (instISAC->bandwidthKHz > isac8kHz) {
ControlUb(instUB, bottleneckUB);
}
} else {
instLB->ISACencLB_obj.enforceFrameSize = 0;
instLB->ISACencLB_obj.new_framelength = FRAMESAMPLES;
}
}
}
instISAC->encoderSamplingRateKHz = encoder_operational_rate;
instISAC->in_sample_rate_hz = sample_rate_hz;
return 0;
}
/******************************************************************************
* WebRtcIsac_SetDecSampRate()
* This function sets the sampling rate of the decoder. Initialization of the
* decoder WILL NOT overwrite the sampling rate of the encoder. The default
* value is 16 kHz which is set when the instance is created.
*
* Input:
* - ISAC_main_inst : iSAC instance
* - sample_rate_hz : sampling rate in Hertz, valid values are 16000
* and 32000.
*
* Return value : 0 if successful
* -1 if failed.
*/
WebRtc_Word16 WebRtcIsac_SetDecSampRate(ISACStruct* ISAC_main_inst,
WebRtc_UWord16 sample_rate_hz) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
enum IsacSamplingRate decoder_operational_rate;
if (sample_rate_hz == 16000) {
decoder_operational_rate = kIsacWideband;
} else if (sample_rate_hz == 32000) {
decoder_operational_rate = kIsacSuperWideband;
} else {
/* Sampling Frequency is not supported. */
instISAC->errorCode = ISAC_UNSUPPORTED_SAMPLING_FREQUENCY;
return -1;
}
if ((instISAC->decoderSamplingRateKHz == kIsacWideband) &&
(decoder_operational_rate == kIsacSuperWideband)) {
/* Switching from wideband to super-wideband at the decoder
* we need to reset the filter-bank and initialize upper-band decoder. */
memset(instISAC->synthesisFBState1, 0,
FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
memset(instISAC->synthesisFBState2, 0,
FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
if (DecoderInitUb(&(instISAC->instUB)) < 0) {
return -1;
}
}
instISAC->decoderSamplingRateKHz = decoder_operational_rate;
return 0;
}
/******************************************************************************
* WebRtcIsac_EncSampRate()
*
* Input:
* - ISAC_main_inst : iSAC instance
*
* Return value : sampling rate in Hertz. The input to encoder
* is expected to be sampled in this rate.
*
*/
WebRtc_UWord16 WebRtcIsac_EncSampRate(ISACStruct* ISAC_main_inst) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
return instISAC->in_sample_rate_hz;
}
/******************************************************************************
* WebRtcIsac_DecSampRate()
* Return the sampling rate of the decoded audio.
*
* Input:
* - ISAC_main_inst : iSAC instance
*
* Return value : sampling rate in Hertz. Decoder output is
* sampled at this rate.
*
*/
WebRtc_UWord16 WebRtcIsac_DecSampRate(ISACStruct* ISAC_main_inst) {
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
return instISAC->decoderSamplingRateKHz == kIsacWideband ? 16000 : 32000;
}