コード例 #1
0
ファイル: aecm_core_neon.c プロジェクト: dreaflove/webrtc-lib
void WebRtcAecm_CalcLinearEnergiesNeon(AecmCore* aecm,
                                       const uint16_t* far_spectrum,
                                       int32_t* echo_est,
                                       uint32_t* far_energy,
                                       uint32_t* echo_energy_adapt,
                                       uint32_t* echo_energy_stored) {
  int16_t* start_stored_p = aecm->channelStored;
  int16_t* start_adapt_p = aecm->channelAdapt16;
  int32_t* echo_est_p = echo_est;
  const int16_t* end_stored_p = aecm->channelStored + PART_LEN;
  const uint16_t* far_spectrum_p = far_spectrum;
  int16x8_t store_v, adapt_v;
  uint16x8_t spectrum_v;
  uint32x4_t echo_est_v_low, echo_est_v_high;
  uint32x4_t far_energy_v, echo_stored_v, echo_adapt_v;

  far_energy_v = vdupq_n_u32(0);
  echo_adapt_v = vdupq_n_u32(0);
  echo_stored_v = vdupq_n_u32(0);

  // Get energy for the delayed far end signal and estimated
  // echo using both stored and adapted channels.
  // The C code:
  //  for (i = 0; i < PART_LEN1; i++) {
  //      echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
  //                                         far_spectrum[i]);
  //      (*far_energy) += (uint32_t)(far_spectrum[i]);
  //      *echo_energy_adapt += aecm->channelAdapt16[i] * far_spectrum[i];
  //      (*echo_energy_stored) += (uint32_t)echo_est[i];
  //  }
  while (start_stored_p < end_stored_p) {
    spectrum_v = vld1q_u16(far_spectrum_p);
    adapt_v = vld1q_s16(start_adapt_p);
    store_v = vld1q_s16(start_stored_p);

    far_energy_v = vaddw_u16(far_energy_v, vget_low_u16(spectrum_v));
    far_energy_v = vaddw_u16(far_energy_v, vget_high_u16(spectrum_v));

    echo_est_v_low = vmull_u16(vreinterpret_u16_s16(vget_low_s16(store_v)),
                               vget_low_u16(spectrum_v));
    echo_est_v_high = vmull_u16(vreinterpret_u16_s16(vget_high_s16(store_v)),
                                vget_high_u16(spectrum_v));
    vst1q_s32(echo_est_p, vreinterpretq_s32_u32(echo_est_v_low));
    vst1q_s32(echo_est_p + 4, vreinterpretq_s32_u32(echo_est_v_high));

    echo_stored_v = vaddq_u32(echo_est_v_low, echo_stored_v);
    echo_stored_v = vaddq_u32(echo_est_v_high, echo_stored_v);

    echo_adapt_v = vmlal_u16(echo_adapt_v,
                             vreinterpret_u16_s16(vget_low_s16(adapt_v)),
                             vget_low_u16(spectrum_v));
    echo_adapt_v = vmlal_u16(echo_adapt_v,
                             vreinterpret_u16_s16(vget_high_s16(adapt_v)),
                             vget_high_u16(spectrum_v));

    start_stored_p += 8;
    start_adapt_p += 8;
    far_spectrum_p += 8;
    echo_est_p += 8;
  }

  AddLanes(far_energy, far_energy_v);
  AddLanes(echo_energy_stored, echo_stored_v);
  AddLanes(echo_energy_adapt, echo_adapt_v);

  echo_est[PART_LEN] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[PART_LEN],
                                             far_spectrum[PART_LEN]);
  *echo_energy_stored += (uint32_t)echo_est[PART_LEN];
  *far_energy += (uint32_t)far_spectrum[PART_LEN];
  *echo_energy_adapt += aecm->channelAdapt16[PART_LEN] * far_spectrum[PART_LEN];
}
コード例 #2
0
ファイル: automode.c プロジェクト: Andrel322/gecko-dev
int WebRtcNetEQ_UpdateIatStatistics(AutomodeInst_t *inst, int maxBufLen,
                                    uint16_t seqNumber, uint32_t timeStamp,
                                    int32_t fsHz, int mdCodec, int streamingMode)
{
    uint32_t timeIat; /* inter-arrival time */
    int i;
    int32_t tempsum = 0; /* temp summation */
    int32_t tempvar; /* temporary variable */
    int retval = 0; /* return value */
    int16_t 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 (!WebRtcNetEQ_IsNewerTimestamp(timeStamp, inst->lastTimeStamp) ||
        !WebRtcNetEQ_IsNewerSequenceNumber(seqNumber, inst->lastSeqNo))
    {
        /* Wrong timestamp or sequence order; revert to backup plan */
        packetLenSamp = inst->packetSpeechLenSamp; /* use stored value */
    }
    else
    {
        /* calculate timestamps per packet */
        packetLenSamp = (int16_t) WebRtcSpl_DivU32U16(timeStamp - inst->lastTimeStamp,
            seqNumber - inst->lastSeqNo);
    }

    /* Check that the packet size is positive; if not, the statistics cannot be updated. */
    if (inst->firstPacketReceived && 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).
             */
            int16_t timeIatQ8 = (int16_t) 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 > (uint32_t) 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 (WebRtcNetEQ_IsNewerSequenceNumber(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,
                (uint16_t) (seqNumber - (uint16_t) (inst->lastSeqNo + 1)));
        }
        else if (!WebRtcNetEQ_IsNewerSequenceNumber(seqNumber, inst->lastSeqNo))
        {
            /* compensate for re-ordering */
            timeIat += (uint16_t) (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++)
        {
            int32_t 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 int16_t 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,
                (int16_t) 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,
                (uint16_t) 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;
            }
        }