/* * 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. */ /* * Contains the API functions for the AEC. */ #include "webrtc/modules/audio_processing/aec/include/echo_cancellation.h" #include #ifdef WEBRTC_AEC_DEBUG_DUMP #include #endif #include #include #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" #include "webrtc/modules/audio_processing/aec/aec_core.h" #include "webrtc/modules/audio_processing/aec/aec_resampler.h" #include "webrtc/modules/audio_processing/aec/echo_cancellation_internal.h" #include "webrtc/modules/audio_processing/utility/ring_buffer.h" #include "webrtc/typedefs.h" // Measured delays [ms] // Device Chrome GTP // MacBook Air 10 // MacBook Retina 10 100 // MacPro 30? // // Win7 Desktop 70 80? // Win7 T430s 110 // Win8 T420s 70 // // Daisy 50 // Pixel (w/ preproc?) 240 // Pixel (w/o preproc?) 110 110 // The extended filter mode gives us the flexibility to ignore the system's // reported delays. We do this for platforms which we believe provide results // which are incompatible with the AEC's expectations. Based on measurements // (some provided above) we set a conservative (i.e. lower than measured) // fixed delay. // // WEBRTC_UNTRUSTED_DELAY will only have an impact when |extended_filter_mode| // is enabled. See the note along with |DelayCorrection| in // echo_cancellation_impl.h for more details on the mode. // // Justification: // Chromium/Mac: Here, the true latency is so low (~10-20 ms), that it plays // havoc with the AEC's buffering. To avoid this, we set a fixed delay of 20 ms // and then compensate by rewinding by 10 ms (in wideband) through // kDelayDiffOffsetSamples. This trick does not seem to work for larger rewind // values, but fortunately this is sufficient. // // Chromium/Linux(ChromeOS): The values we get on this platform don't correspond // well to reality. The variance doesn't match the AEC's buffer changes, and the // bulk values tend to be too low. However, the range across different hardware // appears to be too large to choose a single value. // // GTP/Linux(ChromeOS): TBD, but for the moment we will trust the values. #if defined(WEBRTC_CHROMIUM_BUILD) && defined(WEBRTC_MAC) #define WEBRTC_UNTRUSTED_DELAY #endif #if defined(WEBRTC_UNTRUSTED_DELAY) && defined(WEBRTC_MAC) static const int kDelayDiffOffsetSamples = -160; #else // Not enabled for now. static const int kDelayDiffOffsetSamples = 0; #endif #if defined(WEBRTC_MAC) static const int kFixedDelayMs = 20; #else static const int kFixedDelayMs = 50; #endif #if !defined(WEBRTC_UNTRUSTED_DELAY) static const int kMinTrustedDelayMs = 20; #endif static const int kMaxTrustedDelayMs = 500; // Maximum length of resampled signal. Must be an integer multiple of frames // (ceil(1/(1 + MIN_SKEW)*2) + 1)*FRAME_LEN // The factor of 2 handles wb, and the + 1 is as a safety margin // TODO(bjornv): Replace with kResamplerBufferSize #define MAX_RESAMP_LEN (5 * FRAME_LEN) static const int kMaxBufSizeStart = 62; // In partitions static const int sampMsNb = 8; // samples per ms in nb static const int initCheck = 42; #ifdef WEBRTC_AEC_DEBUG_DUMP int webrtc_aec_instance_count = 0; #endif // Estimates delay to set the position of the far-end buffer read pointer // (controlled by knownDelay) static void EstBufDelayNormal(aecpc_t* aecInst); static void EstBufDelayExtended(aecpc_t* aecInst); static int ProcessNormal(aecpc_t* self, const float* near, const float* near_high, float* out, float* out_high, int16_t num_samples, int16_t reported_delay_ms, int32_t skew); static void ProcessExtended(aecpc_t* self, const float* near, const float* near_high, float* out, float* out_high, int16_t num_samples, int16_t reported_delay_ms, int32_t skew); int32_t WebRtcAec_Create(void** aecInst) { aecpc_t* aecpc; if (aecInst == NULL) { return -1; } aecpc = malloc(sizeof(aecpc_t)); *aecInst = aecpc; if (aecpc == NULL) { return -1; } if (WebRtcAec_CreateAec(&aecpc->aec) == -1) { WebRtcAec_Free(aecpc); aecpc = NULL; return -1; } if (WebRtcAec_CreateResampler(&aecpc->resampler) == -1) { WebRtcAec_Free(aecpc); aecpc = NULL; return -1; } // Create far-end pre-buffer. The buffer size has to be large enough for // largest possible drift compensation (kResamplerBufferSize) + "almost" an // FFT buffer (PART_LEN2 - 1). aecpc->far_pre_buf = WebRtc_CreateBuffer(PART_LEN2 + kResamplerBufferSize, sizeof(float)); if (!aecpc->far_pre_buf) { WebRtcAec_Free(aecpc); aecpc = NULL; return -1; } aecpc->initFlag = 0; aecpc->lastError = 0; #ifdef WEBRTC_AEC_DEBUG_DUMP { char filename[64]; sprintf(filename, "aec_buf%d.dat", webrtc_aec_instance_count); aecpc->bufFile = fopen(filename, "wb"); sprintf(filename, "aec_skew%d.dat", webrtc_aec_instance_count); aecpc->skewFile = fopen(filename, "wb"); sprintf(filename, "aec_delay%d.dat", webrtc_aec_instance_count); aecpc->delayFile = fopen(filename, "wb"); webrtc_aec_instance_count++; } #endif return 0; } int32_t WebRtcAec_Free(void* aecInst) { aecpc_t* aecpc = aecInst; if (aecpc == NULL) { return -1; } WebRtc_FreeBuffer(aecpc->far_pre_buf); #ifdef WEBRTC_AEC_DEBUG_DUMP fclose(aecpc->bufFile); fclose(aecpc->skewFile); fclose(aecpc->delayFile); #endif WebRtcAec_FreeAec(aecpc->aec); WebRtcAec_FreeResampler(aecpc->resampler); free(aecpc); return 0; } int32_t WebRtcAec_Init(void* aecInst, int32_t sampFreq, int32_t scSampFreq) { aecpc_t* aecpc = aecInst; AecConfig aecConfig; if (sampFreq != 8000 && sampFreq != 16000 && sampFreq != 32000) { aecpc->lastError = AEC_BAD_PARAMETER_ERROR; return -1; } aecpc->sampFreq = sampFreq; if (scSampFreq < 1 || scSampFreq > 96000) { aecpc->lastError = AEC_BAD_PARAMETER_ERROR; return -1; } aecpc->scSampFreq = scSampFreq; // Initialize echo canceller core if (WebRtcAec_InitAec(aecpc->aec, aecpc->sampFreq) == -1) { aecpc->lastError = AEC_UNSPECIFIED_ERROR; return -1; } if (WebRtcAec_InitResampler(aecpc->resampler, aecpc->scSampFreq) == -1) { aecpc->lastError = AEC_UNSPECIFIED_ERROR; return -1; } if (WebRtc_InitBuffer(aecpc->far_pre_buf) == -1) { aecpc->lastError = AEC_UNSPECIFIED_ERROR; return -1; } WebRtc_MoveReadPtr(aecpc->far_pre_buf, -PART_LEN); // Start overlap. aecpc->initFlag = initCheck; // indicates that initialization has been done if (aecpc->sampFreq == 32000) { aecpc->splitSampFreq = 16000; } else { aecpc->splitSampFreq = sampFreq; } aecpc->delayCtr = 0; aecpc->sampFactor = (aecpc->scSampFreq * 1.0f) / aecpc->splitSampFreq; // Sampling frequency multiplier (SWB is processed as 160 frame size). aecpc->rate_factor = aecpc->splitSampFreq / 8000; aecpc->sum = 0; aecpc->counter = 0; aecpc->checkBuffSize = 1; aecpc->firstVal = 0; aecpc->startup_phase = WebRtcAec_reported_delay_enabled(aecpc->aec); aecpc->bufSizeStart = 0; aecpc->checkBufSizeCtr = 0; aecpc->msInSndCardBuf = 0; aecpc->filtDelay = -1; // -1 indicates an initialized state. aecpc->timeForDelayChange = 0; aecpc->knownDelay = 0; aecpc->lastDelayDiff = 0; aecpc->skewFrCtr = 0; aecpc->resample = kAecFalse; aecpc->highSkewCtr = 0; aecpc->skew = 0; aecpc->farend_started = 0; // Default settings. aecConfig.nlpMode = kAecNlpModerate; aecConfig.skewMode = kAecFalse; aecConfig.metricsMode = kAecFalse; aecConfig.delay_logging = kAecFalse; if (WebRtcAec_set_config(aecpc, aecConfig) == -1) { aecpc->lastError = AEC_UNSPECIFIED_ERROR; return -1; } return 0; } // only buffer L band for farend int32_t WebRtcAec_BufferFarend(void* aecInst, const float* farend, int16_t nrOfSamples) { aecpc_t* aecpc = aecInst; int newNrOfSamples = (int)nrOfSamples; float new_farend[MAX_RESAMP_LEN]; const float* farend_ptr = farend; if (farend == NULL) { aecpc->lastError = AEC_NULL_POINTER_ERROR; return -1; } if (aecpc->initFlag != initCheck) { aecpc->lastError = AEC_UNINITIALIZED_ERROR; return -1; } // number of samples == 160 for SWB input if (nrOfSamples != 80 && nrOfSamples != 160) { aecpc->lastError = AEC_BAD_PARAMETER_ERROR; return -1; } if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) { // Resample and get a new number of samples WebRtcAec_ResampleLinear(aecpc->resampler, farend, nrOfSamples, aecpc->skew, new_farend, &newNrOfSamples); farend_ptr = new_farend; } aecpc->farend_started = 1; WebRtcAec_SetSystemDelay(aecpc->aec, WebRtcAec_system_delay(aecpc->aec) + newNrOfSamples); // Write the time-domain data to |far_pre_buf|. WebRtc_WriteBuffer(aecpc->far_pre_buf, farend_ptr, (size_t)newNrOfSamples); // Transform to frequency domain if we have enough data. while (WebRtc_available_read(aecpc->far_pre_buf) >= PART_LEN2) { // We have enough data to pass to the FFT, hence read PART_LEN2 samples. { float* ptmp; float tmp[PART_LEN2]; WebRtc_ReadBuffer(aecpc->far_pre_buf, (void**)&ptmp, tmp, PART_LEN2); WebRtcAec_BufferFarendPartition(aecpc->aec, ptmp); #ifdef WEBRTC_AEC_DEBUG_DUMP WebRtc_WriteBuffer( WebRtcAec_far_time_buf(aecpc->aec), &ptmp[PART_LEN], 1); #endif } // Rewind |far_pre_buf| PART_LEN samples for overlap before continuing. WebRtc_MoveReadPtr(aecpc->far_pre_buf, -PART_LEN); } return 0; } int32_t WebRtcAec_Process(void* aecInst, const float* nearend, const float* nearendH, float* out, float* outH, int16_t nrOfSamples, int16_t msInSndCardBuf, int32_t skew) { aecpc_t* aecpc = aecInst; int32_t retVal = 0; if (nearend == NULL) { aecpc->lastError = AEC_NULL_POINTER_ERROR; return -1; } if (out == NULL) { aecpc->lastError = AEC_NULL_POINTER_ERROR; return -1; } if (aecpc->initFlag != initCheck) { aecpc->lastError = AEC_UNINITIALIZED_ERROR; return -1; } // number of samples == 160 for SWB input if (nrOfSamples != 80 && nrOfSamples != 160) { aecpc->lastError = AEC_BAD_PARAMETER_ERROR; return -1; } // Check for valid pointers based on sampling rate if (aecpc->sampFreq == 32000 && nearendH == NULL) { aecpc->lastError = AEC_NULL_POINTER_ERROR; return -1; } if (msInSndCardBuf < 0) { msInSndCardBuf = 0; aecpc->lastError = AEC_BAD_PARAMETER_WARNING; retVal = -1; } else if (msInSndCardBuf > kMaxTrustedDelayMs) { // The clamping is now done in ProcessExtended/Normal(). aecpc->lastError = AEC_BAD_PARAMETER_WARNING; retVal = -1; } // This returns the value of aec->extended_filter_enabled. if (WebRtcAec_delay_correction_enabled(aecpc->aec)) { ProcessExtended( aecpc, nearend, nearendH, out, outH, nrOfSamples, msInSndCardBuf, skew); } else { if (ProcessNormal(aecpc, nearend, nearendH, out, outH, nrOfSamples, msInSndCardBuf, skew) != 0) { retVal = -1; } } #ifdef WEBRTC_AEC_DEBUG_DUMP { int16_t far_buf_size_ms = (int16_t)(WebRtcAec_system_delay(aecpc->aec) / (sampMsNb * aecpc->rate_factor)); (void)fwrite(&far_buf_size_ms, 2, 1, aecpc->bufFile); (void)fwrite( &aecpc->knownDelay, sizeof(aecpc->knownDelay), 1, aecpc->delayFile); } #endif return retVal; } int WebRtcAec_set_config(void* handle, AecConfig config) { aecpc_t* self = (aecpc_t*)handle; if (self->initFlag != initCheck) { self->lastError = AEC_UNINITIALIZED_ERROR; return -1; } if (config.skewMode != kAecFalse && config.skewMode != kAecTrue) { self->lastError = AEC_BAD_PARAMETER_ERROR; return -1; } self->skewMode = config.skewMode; if (config.nlpMode != kAecNlpConservative && config.nlpMode != kAecNlpModerate && config.nlpMode != kAecNlpAggressive) { self->lastError = AEC_BAD_PARAMETER_ERROR; return -1; } if (config.metricsMode != kAecFalse && config.metricsMode != kAecTrue) { self->lastError = AEC_BAD_PARAMETER_ERROR; return -1; } if (config.delay_logging != kAecFalse && config.delay_logging != kAecTrue) { self->lastError = AEC_BAD_PARAMETER_ERROR; return -1; } WebRtcAec_SetConfigCore( self->aec, config.nlpMode, config.metricsMode, config.delay_logging); return 0; } int WebRtcAec_get_echo_status(void* handle, int* status) { aecpc_t* self = (aecpc_t*)handle; if (status == NULL) { self->lastError = AEC_NULL_POINTER_ERROR; return -1; } if (self->initFlag != initCheck) { self->lastError = AEC_UNINITIALIZED_ERROR; return -1; } *status = WebRtcAec_echo_state(self->aec); return 0; } int WebRtcAec_GetMetrics(void* handle, AecMetrics* metrics) { const float kUpWeight = 0.7f; float dtmp; int stmp; aecpc_t* self = (aecpc_t*)handle; Stats erl; Stats erle; Stats a_nlp; if (handle == NULL) { return -1; } if (metrics == NULL) { self->lastError = AEC_NULL_POINTER_ERROR; return -1; } if (self->initFlag != initCheck) { self->lastError = AEC_UNINITIALIZED_ERROR; return -1; } WebRtcAec_GetEchoStats(self->aec, &erl, &erle, &a_nlp); // ERL metrics->erl.instant = (int)erl.instant; if ((erl.himean > kOffsetLevel) && (erl.average > kOffsetLevel)) { // Use a mix between regular average and upper part average. dtmp = kUpWeight * erl.himean + (1 - kUpWeight) * erl.average; metrics->erl.average = (int)dtmp; } else { metrics->erl.average = kOffsetLevel; } metrics->erl.max = (int)erl.max; if (erl.min < (kOffsetLevel * (-1))) { metrics->erl.min = (int)erl.min; } else { metrics->erl.min = kOffsetLevel; } // ERLE metrics->erle.instant = (int)erle.instant; if ((erle.himean > kOffsetLevel) && (erle.average > kOffsetLevel)) { // Use a mix between regular average and upper part average. dtmp = kUpWeight * erle.himean + (1 - kUpWeight) * erle.average; metrics->erle.average = (int)dtmp; } else { metrics->erle.average = kOffsetLevel; } metrics->erle.max = (int)erle.max; if (erle.min < (kOffsetLevel * (-1))) { metrics->erle.min = (int)erle.min; } else { metrics->erle.min = kOffsetLevel; } // RERL if ((metrics->erl.average > kOffsetLevel) && (metrics->erle.average > kOffsetLevel)) { stmp = metrics->erl.average + metrics->erle.average; } else { stmp = kOffsetLevel; } metrics->rerl.average = stmp; // No other statistics needed, but returned for completeness. metrics->rerl.instant = stmp; metrics->rerl.max = stmp; metrics->rerl.min = stmp; // A_NLP metrics->aNlp.instant = (int)a_nlp.instant; if ((a_nlp.himean > kOffsetLevel) && (a_nlp.average > kOffsetLevel)) { // Use a mix between regular average and upper part average. dtmp = kUpWeight * a_nlp.himean + (1 - kUpWeight) * a_nlp.average; metrics->aNlp.average = (int)dtmp; } else { metrics->aNlp.average = kOffsetLevel; } metrics->aNlp.max = (int)a_nlp.max; if (a_nlp.min < (kOffsetLevel * (-1))) { metrics->aNlp.min = (int)a_nlp.min; } else { metrics->aNlp.min = kOffsetLevel; } return 0; } int WebRtcAec_GetDelayMetrics(void* handle, int* median, int* std) { aecpc_t* self = handle; if (median == NULL) { self->lastError = AEC_NULL_POINTER_ERROR; return -1; } if (std == NULL) { self->lastError = AEC_NULL_POINTER_ERROR; return -1; } if (self->initFlag != initCheck) { self->lastError = AEC_UNINITIALIZED_ERROR; return -1; } if (WebRtcAec_GetDelayMetricsCore(self->aec, median, std) == -1) { // Logging disabled. self->lastError = AEC_UNSUPPORTED_FUNCTION_ERROR; return -1; } return 0; } int32_t WebRtcAec_get_error_code(void* aecInst) { aecpc_t* aecpc = aecInst; return aecpc->lastError; } AecCore* WebRtcAec_aec_core(void* handle) { if (!handle) { return NULL; } return ((aecpc_t*)handle)->aec; } static int ProcessNormal(aecpc_t* aecpc, const float* nearend, const float* nearendH, float* out, float* outH, int16_t nrOfSamples, int16_t msInSndCardBuf, int32_t skew) { int retVal = 0; short i; short nBlocks10ms; short nFrames; // Limit resampling to doubling/halving of signal const float minSkewEst = -0.5f; const float maxSkewEst = 1.0f; msInSndCardBuf = msInSndCardBuf > kMaxTrustedDelayMs ? kMaxTrustedDelayMs : msInSndCardBuf; // TODO(andrew): we need to investigate if this +10 is really wanted. msInSndCardBuf += 10; aecpc->msInSndCardBuf = msInSndCardBuf; if (aecpc->skewMode == kAecTrue) { if (aecpc->skewFrCtr < 25) { aecpc->skewFrCtr++; } else { retVal = WebRtcAec_GetSkew(aecpc->resampler, skew, &aecpc->skew); if (retVal == -1) { aecpc->skew = 0; aecpc->lastError = AEC_BAD_PARAMETER_WARNING; } aecpc->skew /= aecpc->sampFactor * nrOfSamples; if (aecpc->skew < 1.0e-3 && aecpc->skew > -1.0e-3) { aecpc->resample = kAecFalse; } else { aecpc->resample = kAecTrue; } if (aecpc->skew < minSkewEst) { aecpc->skew = minSkewEst; } else if (aecpc->skew > maxSkewEst) { aecpc->skew = maxSkewEst; } #ifdef WEBRTC_AEC_DEBUG_DUMP (void)fwrite(&aecpc->skew, sizeof(aecpc->skew), 1, aecpc->skewFile); #endif } } nFrames = nrOfSamples / FRAME_LEN; nBlocks10ms = nFrames / aecpc->rate_factor; if (aecpc->startup_phase) { // Only needed if they don't already point to the same place. if (nearend != out) { memcpy(out, nearend, sizeof(*out) * nrOfSamples); } if (nearendH != outH) { memcpy(outH, nearendH, sizeof(*outH) * nrOfSamples); } // The AEC is in the start up mode // AEC is disabled until the system delay is OK // Mechanism to ensure that the system delay is reasonably stable. if (aecpc->checkBuffSize) { aecpc->checkBufSizeCtr++; // Before we fill up the far-end buffer we require the system delay // to be stable (+/-8 ms) compared to the first value. This // comparison is made during the following 6 consecutive 10 ms // blocks. If it seems to be stable then we start to fill up the // far-end buffer. if (aecpc->counter == 0) { aecpc->firstVal = aecpc->msInSndCardBuf; aecpc->sum = 0; } if (abs(aecpc->firstVal - aecpc->msInSndCardBuf) < WEBRTC_SPL_MAX(0.2 * aecpc->msInSndCardBuf, sampMsNb)) { aecpc->sum += aecpc->msInSndCardBuf; aecpc->counter++; } else { aecpc->counter = 0; } if (aecpc->counter * nBlocks10ms >= 6) { // The far-end buffer size is determined in partitions of // PART_LEN samples. Use 75% of the average value of the system // delay as buffer size to start with. aecpc->bufSizeStart = WEBRTC_SPL_MIN((3 * aecpc->sum * aecpc->rate_factor * 8) / (4 * aecpc->counter * PART_LEN), kMaxBufSizeStart); // Buffer size has now been determined. aecpc->checkBuffSize = 0; } if (aecpc->checkBufSizeCtr * nBlocks10ms > 50) { // For really bad systems, don't disable the echo canceller for // more than 0.5 sec. aecpc->bufSizeStart = WEBRTC_SPL_MIN( (aecpc->msInSndCardBuf * aecpc->rate_factor * 3) / 40, kMaxBufSizeStart); aecpc->checkBuffSize = 0; } } // If |checkBuffSize| changed in the if-statement above. if (!aecpc->checkBuffSize) { // The system delay is now reasonably stable (or has been unstable // for too long). When the far-end buffer is filled with // approximately the same amount of data as reported by the system // we end the startup phase. int overhead_elements = WebRtcAec_system_delay(aecpc->aec) / PART_LEN - aecpc->bufSizeStart; if (overhead_elements == 0) { // Enable the AEC aecpc->startup_phase = 0; } else if (overhead_elements > 0) { // TODO(bjornv): Do we need a check on how much we actually // moved the read pointer? It should always be possible to move // the pointer |overhead_elements| since we have only added data // to the buffer and no delay compensation nor AEC processing // has been done. WebRtcAec_MoveFarReadPtr(aecpc->aec, overhead_elements); // Enable the AEC aecpc->startup_phase = 0; } } } else { // AEC is enabled. if (WebRtcAec_reported_delay_enabled(aecpc->aec)) { EstBufDelayNormal(aecpc); } // Note that 1 frame is supported for NB and 2 frames for WB. for (i = 0; i < nFrames; i++) { // Call the AEC. WebRtcAec_ProcessFrame(aecpc->aec, &nearend[FRAME_LEN * i], &nearendH[FRAME_LEN * i], aecpc->knownDelay, &out[FRAME_LEN * i], &outH[FRAME_LEN * i]); // TODO(bjornv): Re-structure such that we don't have to pass // |aecpc->knownDelay| as input. Change name to something like // |system_buffer_diff|. } } return retVal; } static void ProcessExtended(aecpc_t* self, const float* near, const float* near_high, float* out, float* out_high, int16_t num_samples, int16_t reported_delay_ms, int32_t skew) { int i; const int num_frames = num_samples / FRAME_LEN; const int delay_diff_offset = kDelayDiffOffsetSamples; #if defined(WEBRTC_UNTRUSTED_DELAY) reported_delay_ms = kFixedDelayMs; #else // This is the usual mode where we trust the reported system delay values. // Due to the longer filter, we no longer add 10 ms to the reported delay // to reduce chance of non-causality. Instead we apply a minimum here to avoid // issues with the read pointer jumping around needlessly. reported_delay_ms = reported_delay_ms < kMinTrustedDelayMs ? kMinTrustedDelayMs : reported_delay_ms; // If the reported delay appears to be bogus, we attempt to recover by using // the measured fixed delay values. We use >= here because higher layers // may already clamp to this maximum value, and we would otherwise not // detect it here. reported_delay_ms = reported_delay_ms >= kMaxTrustedDelayMs ? kFixedDelayMs : reported_delay_ms; #endif self->msInSndCardBuf = reported_delay_ms; if (!self->farend_started) { // Only needed if they don't already point to the same place. if (near != out) { memcpy(out, near, sizeof(*out) * num_samples); } if (near_high != out_high) { memcpy(out_high, near_high, sizeof(*out_high) * num_samples); } return; } if (self->startup_phase) { // In the extended mode, there isn't a startup "phase", just a special // action on the first frame. In the trusted delay case, we'll take the // current reported delay, unless it's less then our conservative // measurement. int startup_size_ms = reported_delay_ms < kFixedDelayMs ? kFixedDelayMs : reported_delay_ms; int overhead_elements = (WebRtcAec_system_delay(self->aec) - startup_size_ms / 2 * self->rate_factor * 8) / PART_LEN; WebRtcAec_MoveFarReadPtr(self->aec, overhead_elements); self->startup_phase = 0; } if (WebRtcAec_reported_delay_enabled(self->aec)) { EstBufDelayExtended(self); } { // |delay_diff_offset| gives us the option to manually rewind the delay on // very low delay platforms which can't be expressed purely through // |reported_delay_ms|. const int adjusted_known_delay = WEBRTC_SPL_MAX(0, self->knownDelay + delay_diff_offset); for (i = 0; i < num_frames; ++i) { WebRtcAec_ProcessFrame(self->aec, &near[FRAME_LEN * i], &near_high[FRAME_LEN * i], adjusted_known_delay, &out[FRAME_LEN * i], &out_high[FRAME_LEN * i]); } } } static void EstBufDelayNormal(aecpc_t* aecpc) { int nSampSndCard = aecpc->msInSndCardBuf * sampMsNb * aecpc->rate_factor; int current_delay = nSampSndCard - WebRtcAec_system_delay(aecpc->aec); int delay_difference = 0; // Before we proceed with the delay estimate filtering we: // 1) Compensate for the frame that will be read. // 2) Compensate for drift resampling. // 3) Compensate for non-causality if needed, since the estimated delay can't // be negative. // 1) Compensating for the frame(s) that will be read/processed. current_delay += FRAME_LEN * aecpc->rate_factor; // 2) Account for resampling frame delay. if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) { current_delay -= kResamplingDelay; } // 3) Compensate for non-causality, if needed, by flushing one block. if (current_delay < PART_LEN) { current_delay += WebRtcAec_MoveFarReadPtr(aecpc->aec, 1) * PART_LEN; } // We use -1 to signal an initialized state in the "extended" implementation; // compensate for that. aecpc->filtDelay = aecpc->filtDelay < 0 ? 0 : aecpc->filtDelay; aecpc->filtDelay = WEBRTC_SPL_MAX(0, (short)(0.8 * aecpc->filtDelay + 0.2 * current_delay)); delay_difference = aecpc->filtDelay - aecpc->knownDelay; if (delay_difference > 224) { if (aecpc->lastDelayDiff < 96) { aecpc->timeForDelayChange = 0; } else { aecpc->timeForDelayChange++; } } else if (delay_difference < 96 && aecpc->knownDelay > 0) { if (aecpc->lastDelayDiff > 224) { aecpc->timeForDelayChange = 0; } else { aecpc->timeForDelayChange++; } } else { aecpc->timeForDelayChange = 0; } aecpc->lastDelayDiff = delay_difference; if (aecpc->timeForDelayChange > 25) { aecpc->knownDelay = WEBRTC_SPL_MAX((int)aecpc->filtDelay - 160, 0); } } static void EstBufDelayExtended(aecpc_t* self) { int reported_delay = self->msInSndCardBuf * sampMsNb * self->rate_factor; int current_delay = reported_delay - WebRtcAec_system_delay(self->aec); int delay_difference = 0; // Before we proceed with the delay estimate filtering we: // 1) Compensate for the frame that will be read. // 2) Compensate for drift resampling. // 3) Compensate for non-causality if needed, since the estimated delay can't // be negative. // 1) Compensating for the frame(s) that will be read/processed. current_delay += FRAME_LEN * self->rate_factor; // 2) Account for resampling frame delay. if (self->skewMode == kAecTrue && self->resample == kAecTrue) { current_delay -= kResamplingDelay; } // 3) Compensate for non-causality, if needed, by flushing two blocks. if (current_delay < PART_LEN) { current_delay += WebRtcAec_MoveFarReadPtr(self->aec, 2) * PART_LEN; } if (self->filtDelay == -1) { self->filtDelay = WEBRTC_SPL_MAX(0, 0.5 * current_delay); } else { self->filtDelay = WEBRTC_SPL_MAX( 0, (short)(0.95 * self->filtDelay + 0.05 * current_delay)); } delay_difference = self->filtDelay - self->knownDelay; if (delay_difference > 384) { if (self->lastDelayDiff < 128) { self->timeForDelayChange = 0; } else { self->timeForDelayChange++; } } else if (delay_difference < 128 && self->knownDelay > 0) { if (self->lastDelayDiff > 384) { self->timeForDelayChange = 0; } else { self->timeForDelayChange++; } } else { self->timeForDelayChange = 0; } self->lastDelayDiff = delay_difference; if (self->timeForDelayChange > 25) { self->knownDelay = WEBRTC_SPL_MAX((int)self->filtDelay - 256, 0); } }