/* * Copyright (c) 2011 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. */ /* * This file contains the implementation of automatic buffer level optimization. */ #include "automode.h" #include "signal_processing_library.h" #include "neteq_defines.h" #ifdef NETEQ_DELAY_LOGGING /* special code for offline delay logging */ #include #include "delay_logging.h" extern FILE *delay_fid2; /* file pointer to delay log file */ #endif /* NETEQ_DELAY_LOGGING */ int WebRtcNetEQ_UpdateIatStatistics(AutomodeInst_t *inst, int maxBufLen, WebRtc_UWord16 seqNumber, WebRtc_UWord32 timeStamp, WebRtc_Word32 fsHz, int mdCodec, int streamingMode) { WebRtc_UWord32 timeIat; /* inter-arrival time */ int i; WebRtc_Word32 tempsum = 0; /* temp summation */ WebRtc_Word32 tempvar; /* temporary variable */ int retval = 0; /* return value */ WebRtc_Word16 packetLenSamp; /* packet speech length in samples */ /****************/ /* Sanity check */ /****************/ if (maxBufLen <= 1 || fsHz <= 0) { /* maxBufLen must be at least 2 and fsHz must both be strictly positive */ return -1; } /****************************/ /* Update packet statistics */ /****************************/ /* Try calculating packet length from current and previous timestamps */ if ((timeStamp <= inst->lastTimeStamp) || (seqNumber <= inst->lastSeqNo)) { /* Wrong timestamp or sequence order; revert to backup plan */ packetLenSamp = inst->packetSpeechLenSamp; /* use stored value */ } else if (timeStamp > inst->lastTimeStamp) { /* calculate timestamps per packet */ packetLenSamp = (WebRtc_Word16) WebRtcSpl_DivU32U16(timeStamp - inst->lastTimeStamp, seqNumber - inst->lastSeqNo); } /* Check that the packet size is positive; if not, the statistics cannot be updated. */ if (packetLenSamp > 0) { /* packet size ok */ /* calculate inter-arrival time in integer packets (rounding down) */ timeIat = WebRtcSpl_DivW32W16(inst->packetIatCountSamp, packetLenSamp); /* Special operations for streaming mode */ if (streamingMode != 0) { /* * Calculate IAT in Q8, including fractions of a packet (i.e., more accurate * than timeIat). */ WebRtc_Word16 timeIatQ8 = (WebRtc_Word16) WebRtcSpl_DivW32W16( WEBRTC_SPL_LSHIFT_W32(inst->packetIatCountSamp, 8), packetLenSamp); /* * Calculate cumulative sum iat with sequence number compensation (ideal arrival * times makes this sum zero). */ inst->cSumIatQ8 += (timeIatQ8 - WEBRTC_SPL_LSHIFT_W32(seqNumber - inst->lastSeqNo, 8)); /* subtract drift term */ inst->cSumIatQ8 -= CSUM_IAT_DRIFT; /* ensure not negative */ inst->cSumIatQ8 = WEBRTC_SPL_MAX(inst->cSumIatQ8, 0); /* remember max */ if (inst->cSumIatQ8 > inst->maxCSumIatQ8) { inst->maxCSumIatQ8 = inst->cSumIatQ8; inst->maxCSumUpdateTimer = 0; } /* too long since the last maximum was observed; decrease max value */ if (inst->maxCSumUpdateTimer > (WebRtc_UWord32) WEBRTC_SPL_MUL_32_16(fsHz, MAX_STREAMING_PEAK_PERIOD)) { inst->maxCSumIatQ8 -= 4; /* remove 1000*4/256 = 15.6 ms/s */ } } /* end of streaming mode */ /* check for discontinuous packet sequence and re-ordering */ if (seqNumber > inst->lastSeqNo + 1) { /* Compensate for gap in the sequence numbers. * Reduce IAT with expected extra time due to lost packets, but ensure that * the IAT is not negative. */ timeIat -= WEBRTC_SPL_MIN(timeIat, (WebRtc_UWord32) (seqNumber - inst->lastSeqNo - 1)); } else if (seqNumber < inst->lastSeqNo) { /* compensate for re-ordering */ timeIat += (WebRtc_UWord32) (inst->lastSeqNo + 1 - seqNumber); } /* saturate IAT at maximum value */ timeIat = WEBRTC_SPL_MIN( timeIat, MAX_IAT ); /* update iatProb = forgetting_factor * iatProb for all elements */ for (i = 0; i <= MAX_IAT; i++) { WebRtc_Word32 tempHi, tempLo; /* Temporary variables */ /* * Multiply iatProbFact (Q15) with iatProb (Q30) and right-shift 15 steps * to come back to Q30. The operation is done in two steps: */ /* * 1) Multiply the high 16 bits (15 bits + sign) of iatProb. Shift iatProb * 16 steps right to get the high 16 bits in a WebRtc_Word16 prior to * multiplication, and left-shift with 1 afterwards to come back to * Q30 = (Q15 * (Q30>>16)) << 1. */ tempHi = WEBRTC_SPL_MUL_16_16(inst->iatProbFact, (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(inst->iatProb[i], 16)); tempHi = WEBRTC_SPL_LSHIFT_W32(tempHi, 1); /* left-shift 1 step */ /* * 2) Isolate and multiply the low 16 bits of iatProb. Right-shift 15 steps * afterwards to come back to Q30 = (Q15 * Q30) >> 15. */ tempLo = inst->iatProb[i] & 0x0000FFFF; /* sift out the 16 low bits */ tempLo = WEBRTC_SPL_MUL_16_U16(inst->iatProbFact, (WebRtc_UWord16) tempLo); tempLo = WEBRTC_SPL_RSHIFT_W32(tempLo, 15); /* Finally, add the high and low parts */ inst->iatProb[i] = tempHi + tempLo; /* Sum all vector elements while we are at it... */ tempsum += inst->iatProb[i]; } /* * Increase the probability for the currently observed inter-arrival time * with 1 - iatProbFact. The factor is in Q15, iatProb in Q30; * hence, left-shift 15 steps to obtain result in Q30. */ inst->iatProb[timeIat] += (32768 - inst->iatProbFact) << 15; tempsum += (32768 - inst->iatProbFact) << 15; /* add to vector sum */ /* * Update iatProbFact (changes only during the first seconds after reset) * The factor converges to IAT_PROB_FACT. */ inst->iatProbFact += (IAT_PROB_FACT - inst->iatProbFact + 3) >> 2; /* iatProb should sum up to 1 (in Q30). */ tempsum -= 1 << 30; /* should be zero */ /* Check if it does, correct if it doesn't. */ if (tempsum > 0) { /* tempsum too large => decrease a few values in the beginning */ i = 0; while (i <= MAX_IAT && tempsum > 0) { /* Remove iatProb[i] / 16 from iatProb, but not more than tempsum */ tempvar = WEBRTC_SPL_MIN(tempsum, inst->iatProb[i] >> 4); inst->iatProb[i++] -= tempvar; tempsum -= tempvar; } } else if (tempsum < 0) { /* tempsum too small => increase a few values in the beginning */ i = 0; while (i <= MAX_IAT && tempsum < 0) { /* Add iatProb[i] / 16 to iatProb, but not more than tempsum */ tempvar = WEBRTC_SPL_MIN(-tempsum, inst->iatProb[i] >> 4); inst->iatProb[i++] += tempvar; tempsum += tempvar; } } /* Calculate optimal buffer level based on updated statistics */ tempvar = (WebRtc_Word32) WebRtcNetEQ_CalcOptimalBufLvl(inst, fsHz, mdCodec, timeIat, streamingMode); if (tempvar > 0) { inst->optBufLevel = (WebRtc_UWord16) tempvar; if (streamingMode != 0) { inst->optBufLevel = WEBRTC_SPL_MAX(inst->optBufLevel, inst->maxCSumIatQ8); } /*********/ /* Limit */ /*********/ /* Subtract extra delay from maxBufLen */ if (inst->extraDelayMs > 0 && inst->packetSpeechLenSamp > 0) { maxBufLen -= inst->extraDelayMs / inst->packetSpeechLenSamp * fsHz / 1000; maxBufLen = WEBRTC_SPL_MAX(maxBufLen, 1); // sanity: at least one packet } maxBufLen = WEBRTC_SPL_LSHIFT_W32(maxBufLen, 8); /* shift to Q8 */ /* Enforce upper limit; 75% of maxBufLen */ inst->optBufLevel = (WebRtc_UWord16) WEBRTC_SPL_MIN( inst->optBufLevel, (maxBufLen >> 1) + (maxBufLen >> 2) ); /* 1/2 + 1/4 = 75% */ } else { retval = (int) tempvar; } } /* end if */ /*******************************/ /* Update post-call statistics */ /*******************************/ /* Calculate inter-arrival time in ms = packetIatCountSamp / (fsHz / 1000) */ timeIat = WEBRTC_SPL_UDIV( WEBRTC_SPL_UMUL_32_16(inst->packetIatCountSamp, (WebRtc_Word16) 1000), (WebRtc_UWord32) fsHz); /* Increase counter corresponding to current inter-arrival time */ if (timeIat > 2000) { inst->countIAT2000ms++; } else if (timeIat > 1000) { inst->countIAT1000ms++; } else if (timeIat > 500) { inst->countIAT500ms++; } if (timeIat > inst->longestIATms) { /* update maximum value */ inst->longestIATms = timeIat; } /***********************************/ /* Prepare for next packet arrival */ /***********************************/ inst->packetIatCountSamp = 0; /* reset inter-arrival time counter */ inst->lastSeqNo = seqNumber; /* remember current sequence number */ inst->lastTimeStamp = timeStamp; /* remember current timestamp */ return retval; } WebRtc_Word16 WebRtcNetEQ_CalcOptimalBufLvl(AutomodeInst_t *inst, WebRtc_Word32 fsHz, int mdCodec, WebRtc_UWord32 timeIatPkts, int streamingMode) { WebRtc_Word32 sum1 = 1 << 30; /* assign to 1 in Q30 */ WebRtc_Word16 B; WebRtc_UWord16 Bopt; int i; WebRtc_Word32 betaInv; /* optimization parameter */ #ifdef NETEQ_DELAY_LOGGING /* special code for offline delay logging */ int temp_var; #endif /****************/ /* Sanity check */ /****************/ if (fsHz <= 0) { /* fsHz must be strictly positive */ return -1; } /***********************************************/ /* Get betaInv parameter based on playout mode */ /***********************************************/ if (streamingMode) { /* streaming (listen-only) mode */ betaInv = AUTOMODE_STREAMING_BETA_INV_Q30; } else { /* normal mode */ betaInv = AUTOMODE_BETA_INV_Q30; } /*******************************************************************/ /* Calculate optimal buffer level without considering jitter peaks */ /*******************************************************************/ /* * Find the B for which the probability of observing an inter-arrival time larger * than or equal to B is less than or equal to betaInv. */ B = 0; /* start from the beginning of iatProb */ sum1 -= inst->iatProb[B]; /* ensure that optimal level is not less than 1 */ do { /* * Subtract the probabilities one by one until the sum is no longer greater * than betaInv. */ sum1 -= inst->iatProb[++B]; } while ((sum1 > betaInv) && (B < MAX_IAT)); Bopt = B; /* This is our primary value for the optimal buffer level Bopt */ if (mdCodec) { /* * Use alternative cost function when multiple description codec is in use. * Do not have to re-calculate all points, just back off a few steps from * previous value of B. */ WebRtc_Word32 sum2 = sum1; /* copy sum1 */ while ((sum2 <= betaInv + inst->iatProb[Bopt]) && (Bopt > 0)) { /* Go backwards in the sum until the modified cost function solution is found */ sum2 += inst->iatProb[Bopt--]; } Bopt++; /* This is the optimal level when using an MD codec */ /* Now, Bopt and B can have different values. */ } #ifdef NETEQ_DELAY_LOGGING /* special code for offline delay logging */ temp_var = NETEQ_DELAY_LOGGING_SIGNAL_OPTBUF; fwrite( &temp_var, sizeof(int), 1, delay_fid2 ); temp_var = (int) (Bopt * inst->packetSpeechLenSamp); #endif /******************************************************************/ /* Make levelFiltFact adaptive: Larger B <=> larger levelFiltFact */ /******************************************************************/ switch (B) { case 0: case 1: { inst->levelFiltFact = 251; break; } case 2: case 3: { inst->levelFiltFact = 252; break; } case 4: case 5: case 6: case 7: { inst->levelFiltFact = 253; break; } default: /* B > 7 */ { inst->levelFiltFact = 254; break; } } /************************/ /* Peak mode operations */ /************************/ /* Compare current IAT with peak threshold * * If IAT > optimal level + threshold (+1 for MD codecs) * or if IAT > 2 * optimal level (note: optimal level is in Q8): */ if (timeIatPkts > (WebRtc_UWord32) (Bopt + inst->peakThresholdPkt + (mdCodec != 0)) || timeIatPkts > (WebRtc_UWord32) WEBRTC_SPL_LSHIFT_U16(Bopt, 1)) { /* A peak is observed */ if (inst->peakIndex == -1) { /* this is the first peak; prepare for next peak */ inst->peakIndex = 0; /* set the mode-disable counter */ inst->peakModeDisabled = WEBRTC_SPL_LSHIFT_W16(1, NUM_PEAKS_REQUIRED-2); } else if (inst->peakIatCountSamp <= (WebRtc_UWord32) WEBRTC_SPL_MUL_32_16(fsHz, MAX_PEAK_PERIOD)) { /* This is not the first peak and the period time is valid */ /* store time elapsed since last peak */ inst->peakPeriodSamp[inst->peakIndex] = inst->peakIatCountSamp; /* saturate height to 16 bits */ inst->peakHeightPkt[inst->peakIndex] = (WebRtc_Word16) WEBRTC_SPL_MIN(timeIatPkts, WEBRTC_SPL_WORD16_MAX); /* increment peakIndex and wrap/modulo */ inst->peakIndex = ++inst->peakIndex & PEAK_INDEX_MASK; /* process peak vectors */ inst->curPeakHeight = 0; inst->curPeakPeriod = 0; for (i = 0; i < NUM_PEAKS; i++) { /* Find maximum of peak heights and peak periods */ inst->curPeakHeight = WEBRTC_SPL_MAX(inst->curPeakHeight, inst->peakHeightPkt[i]); inst->curPeakPeriod = WEBRTC_SPL_MAX(inst->curPeakPeriod, inst->peakPeriodSamp[i]); } inst->peakModeDisabled >>= 1; /* decrease mode-disable "counter" */ } else if (inst->peakIatCountSamp > (WebRtc_UWord32) WEBRTC_SPL_MUL_32_16(fsHz, WEBRTC_SPL_LSHIFT_W16(MAX_PEAK_PERIOD, 1))) { /* * More than 2 * MAX_PEAK_PERIOD has elapsed since last peak; * too long time => reset peak statistics */ inst->curPeakHeight = 0; inst->curPeakPeriod = 0; for (i = 0; i < NUM_PEAKS; i++) { inst->peakHeightPkt[i] = 0; inst->peakPeriodSamp[i] = 0; } inst->peakIndex = -1; /* Next peak is first peak */ inst->peakIatCountSamp = 0; } inst->peakIatCountSamp = 0; /* Reset peak interval timer */ } /* end if peak is observed */ /* Evaluate peak mode conditions */ /* * If not disabled (enough peaks have been observed) and * time since last peak is less than two peak periods. */ if ((!inst->peakModeDisabled) && (inst->peakIatCountSamp <= WEBRTC_SPL_LSHIFT_W32(inst->curPeakPeriod , 1))) { /* Engage peak mode */ /* Set optimal buffer level to curPeakHeight (if it's not already larger) */ Bopt = WEBRTC_SPL_MAX(Bopt, inst->curPeakHeight); #ifdef NETEQ_DELAY_LOGGING /* special code for offline delay logging */ temp_var = (int) -(Bopt * inst->packetSpeechLenSamp); #endif } /* Scale Bopt to Q8 */ Bopt = WEBRTC_SPL_LSHIFT_U16(Bopt,8); #ifdef NETEQ_DELAY_LOGGING /* special code for offline delay logging */ fwrite( &temp_var, sizeof(int), 1, delay_fid2 ); #endif /* Sanity check: Bopt must be strictly positive */ if (Bopt <= 0) { Bopt = WEBRTC_SPL_LSHIFT_W16(1, 8); /* 1 in Q8 */ } return Bopt; /* return value in Q8 */ } int WebRtcNetEQ_BufferLevelFilter(WebRtc_Word32 curSizeMs8, AutomodeInst_t *inst, int sampPerCall, WebRtc_Word16 fsMult) { WebRtc_Word16 curSizeFrames; /****************/ /* Sanity check */ /****************/ if (sampPerCall <= 0 || fsMult <= 0) { /* sampPerCall and fsMult must both be strictly positive */ return -1; } /* Check if packet size has been detected */ if (inst->packetSpeechLenSamp > 0) { /* * Current buffer level in packet lengths * = (curSizeMs8 * fsMult) / packetSpeechLenSamp */ curSizeFrames = (WebRtc_Word16) WebRtcSpl_DivW32W16( WEBRTC_SPL_MUL_32_16(curSizeMs8, fsMult), inst->packetSpeechLenSamp); } else { curSizeFrames = 0; } /* Filter buffer level */ if (inst->levelFiltFact > 0) /* check that filter factor is set */ { /* Filter: * buffLevelFilt = levelFiltFact * buffLevelFilt * + (1-levelFiltFact) * curSizeFrames * * levelFiltFact is in Q8 */ inst->buffLevelFilt = (WebRtc_UWord16) (WEBRTC_SPL_RSHIFT_W32( WEBRTC_SPL_MUL_16_U16(inst->levelFiltFact, inst->buffLevelFilt), 8) + WEBRTC_SPL_MUL_16_16(256 - inst->levelFiltFact, curSizeFrames)); } /* Account for time-scale operations (accelerate and pre-emptive expand) */ if (inst->prevTimeScale) { /* * Time-scaling has been performed since last filter update. * Subtract the sampleMemory from buffLevelFilt after converting sampleMemory * from samples to packets in Q8. Make sure that the filtered value is * non-negative. */ inst->buffLevelFilt = (WebRtc_UWord16) WEBRTC_SPL_MAX( inst->buffLevelFilt - WebRtcSpl_DivW32W16( WEBRTC_SPL_LSHIFT_W32(inst->sampleMemory, 8), /* sampleMemory in Q8 */ inst->packetSpeechLenSamp ), /* divide by packetSpeechLenSamp */ 0); /* * Reset flag and set timescaleHoldOff timer to prevent further time-scaling * for some time. */ inst->prevTimeScale = 0; inst->timescaleHoldOff = AUTOMODE_TIMESCALE_LIMIT; } /* Update time counters and HoldOff timer */ inst->packetIatCountSamp += sampPerCall; /* packet inter-arrival time */ inst->peakIatCountSamp += sampPerCall; /* peak inter-arrival time */ inst->timescaleHoldOff >>= 1; /* time-scaling limiter */ inst->maxCSumUpdateTimer += sampPerCall; /* cumulative-sum timer */ return 0; } int WebRtcNetEQ_SetPacketSpeechLen(AutomodeInst_t *inst, WebRtc_Word16 newLenSamp, WebRtc_Word32 fsHz) { /* Sanity check for newLenSamp and fsHz */ if (newLenSamp <= 0 || fsHz <= 0) { return -1; } inst->packetSpeechLenSamp = newLenSamp; /* Store packet size in instance */ /* Make NetEQ wait for first regular packet before starting the timer */ inst->lastPackCNGorDTMF = 1; inst->packetIatCountSamp = 0; /* Reset packet time counter */ /* * Calculate peak threshold from packet size. The threshold is defined as * the (fractional) number of packets that corresponds to PEAK_HEIGHT * (in Q8 seconds). That is, threshold = PEAK_HEIGHT/256 * fsHz / packLen. */ inst->peakThresholdPkt = (WebRtc_UWord16) WebRtcSpl_DivW32W16ResW16( WEBRTC_SPL_MUL_16_16_RSFT(PEAK_HEIGHT, (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(fsHz, 6), 2), inst->packetSpeechLenSamp); return 0; } int WebRtcNetEQ_ResetAutomode(AutomodeInst_t *inst, int maxBufLenPackets) { int i; WebRtc_UWord16 tempprob = 0x4002; /* 16384 + 2 = 100000000000010 binary; */ /* Sanity check for maxBufLenPackets */ if (maxBufLenPackets <= 1) { /* Invalid value; set to 10 instead (arbitary small number) */ maxBufLenPackets = 10; } /* Reset filtered buffer level */ inst->buffLevelFilt = 0; /* Reset packet size to unknown */ inst->packetSpeechLenSamp = 0; /* * Flag that last packet was special payload, so that automode will treat the next speech * payload as the first payload received. */ inst->lastPackCNGorDTMF = 1; /* Reset peak detection parameters */ inst->peakModeDisabled = 1; /* disable peak mode */ inst->peakIatCountSamp = 0; inst->peakIndex = -1; /* indicates that no peak is registered */ inst->curPeakHeight = 0; inst->curPeakPeriod = 0; for (i = 0; i < NUM_PEAKS; i++) { inst->peakHeightPkt[i] = 0; inst->peakPeriodSamp[i] = 0; } /* * Set the iatProb PDF vector to an exponentially decaying distribution * iatProb[i] = 0.5^(i+1), i = 0, 1, 2, ... * iatProb is in Q30. */ for (i = 0; i <= MAX_IAT; i++) { /* iatProb[i] = 0.5^(i+1) = iatProb[i-1] / 2 */ tempprob = WEBRTC_SPL_RSHIFT_U16(tempprob, 1); /* store in PDF vector */ inst->iatProb[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32) tempprob, 16); } /* * Calculate the optimal buffer level corresponing to the initial PDF. * No need to call WebRtcNetEQ_CalcOptimalBufLvl() since we have just hard-coded * all the variables that the buffer level depends on => we know the result */ inst->optBufLevel = WEBRTC_SPL_MIN(4, (maxBufLenPackets >> 1) + (maxBufLenPackets >> 1)); /* 75% of maxBufLenPackets */ inst->levelFiltFact = 253; /* * Reset the iat update forgetting factor to 0 to make the impact of the first * incoming packets greater. */ inst->iatProbFact = 0; /* Reset packet inter-arrival time counter */ inst->packetIatCountSamp = 0; /* Clear time-scaling related variables */ inst->prevTimeScale = 0; inst->timescaleHoldOff = AUTOMODE_TIMESCALE_LIMIT; /* don't allow time-scaling immediately */ inst->cSumIatQ8 = 0; inst->maxCSumIatQ8 = 0; return 0; }