| /* |
| * 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 "modules/audio_processing/aec/echo_cancellation.h" |
| |
| #include <math.h> |
| #include <stdlib.h> |
| #include <string.h> |
| |
| extern "C" { |
| #include "common_audio/ring_buffer.h" |
| #include "common_audio/signal_processing/include/signal_processing_library.h" |
| } |
| #include "modules/audio_processing/aec/aec_core.h" |
| #include "modules/audio_processing/aec/aec_resampler.h" |
| #include "modules/audio_processing/logging/apm_data_dumper.h" |
| |
| namespace webrtc { |
| |
| Aec::Aec() = default; |
| Aec::~Aec() = default; |
| |
| // 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; |
| |
| int Aec::instance_count = 0; |
| |
| // Estimates delay to set the position of the far-end buffer read pointer |
| // (controlled by knownDelay) |
| static void EstBufDelayNormal(Aec* aecInst); |
| static void EstBufDelayExtended(Aec* aecInst); |
| static int ProcessNormal(Aec* aecInst, |
| const float* const* nearend, |
| size_t num_bands, |
| float* const* out, |
| size_t num_samples, |
| int16_t reported_delay_ms, |
| int32_t skew); |
| static void ProcessExtended(Aec* aecInst, |
| const float* const* nearend, |
| size_t num_bands, |
| float* const* out, |
| size_t num_samples, |
| int16_t reported_delay_ms, |
| int32_t skew); |
| |
| void* WebRtcAec_Create() { |
| Aec* aecpc = new Aec(); |
| |
| if (!aecpc) { |
| return NULL; |
| } |
| aecpc->data_dumper.reset(new ApmDataDumper(aecpc->instance_count)); |
| |
| aecpc->aec = WebRtcAec_CreateAec(aecpc->instance_count); |
| if (!aecpc->aec) { |
| WebRtcAec_Free(aecpc); |
| return NULL; |
| } |
| aecpc->resampler = WebRtcAec_CreateResampler(); |
| if (!aecpc->resampler) { |
| WebRtcAec_Free(aecpc); |
| return NULL; |
| } |
| // 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); |
| return NULL; |
| } |
| |
| aecpc->initFlag = 0; |
| |
| aecpc->instance_count++; |
| return aecpc; |
| } |
| |
| void WebRtcAec_Free(void* aecInst) { |
| Aec* aecpc = reinterpret_cast<Aec*>(aecInst); |
| |
| if (aecpc == NULL) { |
| return; |
| } |
| |
| WebRtc_FreeBuffer(aecpc->far_pre_buf); |
| |
| WebRtcAec_FreeAec(aecpc->aec); |
| WebRtcAec_FreeResampler(aecpc->resampler); |
| delete aecpc; |
| } |
| |
| int32_t WebRtcAec_Init(void* aecInst, int32_t sampFreq, int32_t scSampFreq) { |
| Aec* aecpc = reinterpret_cast<Aec*>(aecInst); |
| aecpc->data_dumper->InitiateNewSetOfRecordings(); |
| AecConfig aecConfig; |
| |
| if (sampFreq != 8000 && sampFreq != 16000 && sampFreq != 32000 && |
| sampFreq != 48000) { |
| return AEC_BAD_PARAMETER_ERROR; |
| } |
| aecpc->sampFreq = sampFreq; |
| |
| if (scSampFreq < 1 || scSampFreq > 96000) { |
| return AEC_BAD_PARAMETER_ERROR; |
| } |
| aecpc->scSampFreq = scSampFreq; |
| |
| // Initialize echo canceller core |
| if (WebRtcAec_InitAec(aecpc->aec, aecpc->sampFreq) == -1) { |
| return AEC_UNSPECIFIED_ERROR; |
| } |
| |
| if (WebRtcAec_InitResampler(aecpc->resampler, aecpc->scSampFreq) == -1) { |
| return AEC_UNSPECIFIED_ERROR; |
| } |
| |
| WebRtc_InitBuffer(aecpc->far_pre_buf); |
| WebRtc_MoveReadPtr(aecpc->far_pre_buf, -PART_LEN); // Start overlap. |
| |
| aecpc->initFlag = initCheck; // indicates that initialization has been done |
| |
| if (aecpc->sampFreq == 32000 || aecpc->sampFreq == 48000) { |
| 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; |
| |
| // We skip the startup_phase completely (setting to 0) if DA-AEC is enabled, |
| // but not extended_filter mode. |
| aecpc->startup_phase = WebRtcAec_extended_filter_enabled(aecpc->aec) || |
| !WebRtcAec_delay_agnostic_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) { |
| return AEC_UNSPECIFIED_ERROR; |
| } |
| |
| return 0; |
| } |
| |
| // Returns any error that is caused when buffering the |
| // far-end signal. |
| int32_t WebRtcAec_GetBufferFarendError(void* aecInst, |
| const float* farend, |
| size_t nrOfSamples) { |
| Aec* aecpc = reinterpret_cast<Aec*>(aecInst); |
| |
| if (!farend) |
| return AEC_NULL_POINTER_ERROR; |
| |
| if (aecpc->initFlag != initCheck) |
| return AEC_UNINITIALIZED_ERROR; |
| |
| // number of samples == 160 for SWB input |
| if (nrOfSamples != 80 && nrOfSamples != 160) |
| return AEC_BAD_PARAMETER_ERROR; |
| |
| return 0; |
| } |
| |
| // only buffer L band for farend |
| int32_t WebRtcAec_BufferFarend(void* aecInst, |
| const float* farend, |
| size_t nrOfSamples) { |
| Aec* aecpc = reinterpret_cast<Aec*>(aecInst); |
| size_t newNrOfSamples = nrOfSamples; |
| float new_farend[MAX_RESAMP_LEN]; |
| const float* farend_ptr = farend; |
| |
| // Get any error caused by buffering the farend signal. |
| int32_t error_code = |
| WebRtcAec_GetBufferFarendError(aecInst, farend, nrOfSamples); |
| |
| if (error_code != 0) |
| return error_code; |
| |
| 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) + |
| static_cast<int>(newNrOfSamples)); |
| |
| // Write the time-domain data to |far_pre_buf|. |
| WebRtc_WriteBuffer(aecpc->far_pre_buf, farend_ptr, newNrOfSamples); |
| |
| // TODO(minyue): reduce to |PART_LEN| samples for each buffering. |
| 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 = NULL; |
| float tmp[PART_LEN2]; |
| WebRtc_ReadBuffer(aecpc->far_pre_buf, reinterpret_cast<void**>(&ptmp), |
| tmp, PART_LEN2); |
| WebRtcAec_BufferFarendBlock(aecpc->aec, &ptmp[PART_LEN]); |
| } |
| |
| // 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* const* nearend, |
| size_t num_bands, |
| float* const* out, |
| size_t nrOfSamples, |
| int16_t msInSndCardBuf, |
| int32_t skew) { |
| Aec* aecpc = reinterpret_cast<Aec*>(aecInst); |
| int32_t retVal = 0; |
| |
| if (out == NULL) { |
| return AEC_NULL_POINTER_ERROR; |
| } |
| |
| if (aecpc->initFlag != initCheck) { |
| return AEC_UNINITIALIZED_ERROR; |
| } |
| |
| // number of samples == 160 for SWB input |
| if (nrOfSamples != 80 && nrOfSamples != 160) { |
| return AEC_BAD_PARAMETER_ERROR; |
| } |
| |
| if (msInSndCardBuf < 0) { |
| msInSndCardBuf = 0; |
| retVal = AEC_BAD_PARAMETER_WARNING; |
| } else if (msInSndCardBuf > kMaxTrustedDelayMs) { |
| // The clamping is now done in ProcessExtended/Normal(). |
| retVal = AEC_BAD_PARAMETER_WARNING; |
| } |
| |
| // This returns the value of aec->extended_filter_enabled. |
| if (WebRtcAec_extended_filter_enabled(aecpc->aec)) { |
| ProcessExtended(aecpc, nearend, num_bands, out, nrOfSamples, msInSndCardBuf, |
| skew); |
| } else { |
| retVal = ProcessNormal(aecpc, nearend, num_bands, out, nrOfSamples, |
| msInSndCardBuf, skew); |
| } |
| |
| int far_buf_size_samples = WebRtcAec_system_delay(aecpc->aec); |
| aecpc->data_dumper->DumpRaw("aec_system_delay", 1, &far_buf_size_samples); |
| aecpc->data_dumper->DumpRaw("aec_known_delay", 1, &aecpc->knownDelay); |
| |
| return retVal; |
| } |
| |
| int WebRtcAec_set_config(void* handle, AecConfig config) { |
| Aec* self = reinterpret_cast<Aec*>(handle); |
| if (self->initFlag != initCheck) { |
| return AEC_UNINITIALIZED_ERROR; |
| } |
| |
| if (config.skewMode != kAecFalse && config.skewMode != kAecTrue) { |
| return AEC_BAD_PARAMETER_ERROR; |
| } |
| self->skewMode = config.skewMode; |
| |
| if (config.nlpMode != kAecNlpConservative && |
| config.nlpMode != kAecNlpModerate && |
| config.nlpMode != kAecNlpAggressive) { |
| return AEC_BAD_PARAMETER_ERROR; |
| } |
| |
| if (config.metricsMode != kAecFalse && config.metricsMode != kAecTrue) { |
| return AEC_BAD_PARAMETER_ERROR; |
| } |
| |
| if (config.delay_logging != kAecFalse && config.delay_logging != kAecTrue) { |
| return AEC_BAD_PARAMETER_ERROR; |
| } |
| |
| WebRtcAec_SetConfigCore(self->aec, config.nlpMode, config.metricsMode, |
| config.delay_logging); |
| return 0; |
| } |
| |
| int WebRtcAec_get_echo_status(void* handle, int* status) { |
| Aec* self = reinterpret_cast<Aec*>(handle); |
| if (status == NULL) { |
| return AEC_NULL_POINTER_ERROR; |
| } |
| if (self->initFlag != initCheck) { |
| return AEC_UNINITIALIZED_ERROR; |
| } |
| |
| *status = WebRtcAec_echo_state(self->aec); |
| |
| return 0; |
| } |
| |
| int WebRtcAec_GetMetrics(void* handle, AecMetrics* metrics) { |
| const float kUpWeight = 0.7f; |
| float dtmp; |
| int stmp; |
| Aec* self = reinterpret_cast<Aec*>(handle); |
| Stats erl; |
| Stats erle; |
| Stats a_nlp; |
| |
| if (handle == NULL) { |
| return -1; |
| } |
| if (metrics == NULL) { |
| return AEC_NULL_POINTER_ERROR; |
| } |
| if (self->initFlag != initCheck) { |
| return AEC_UNINITIALIZED_ERROR; |
| } |
| |
| WebRtcAec_GetEchoStats(self->aec, &erl, &erle, &a_nlp, |
| &metrics->divergent_filter_fraction); |
| |
| // ERL |
| metrics->erl.instant = static_cast<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 = static_cast<int>(dtmp); |
| } else { |
| metrics->erl.average = kOffsetLevel; |
| } |
| |
| metrics->erl.max = static_cast<int>(erl.max); |
| |
| if (erl.min < (kOffsetLevel * (-1))) { |
| metrics->erl.min = static_cast<int>(erl.min); |
| } else { |
| metrics->erl.min = kOffsetLevel; |
| } |
| |
| // ERLE |
| metrics->erle.instant = static_cast<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 = static_cast<int>(dtmp); |
| } else { |
| metrics->erle.average = kOffsetLevel; |
| } |
| |
| metrics->erle.max = static_cast<int>(erle.max); |
| |
| if (erle.min < (kOffsetLevel * (-1))) { |
| metrics->erle.min = static_cast<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 = static_cast<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 = static_cast<int>(dtmp); |
| } else { |
| metrics->aNlp.average = kOffsetLevel; |
| } |
| |
| metrics->aNlp.max = static_cast<int>(a_nlp.max); |
| |
| if (a_nlp.min < (kOffsetLevel * (-1))) { |
| metrics->aNlp.min = static_cast<int>(a_nlp.min); |
| } else { |
| metrics->aNlp.min = kOffsetLevel; |
| } |
| |
| return 0; |
| } |
| |
| int WebRtcAec_GetDelayMetrics(void* handle, |
| int* median, |
| int* std, |
| float* fraction_poor_delays) { |
| Aec* self = reinterpret_cast<Aec*>(handle); |
| if (median == NULL) { |
| return AEC_NULL_POINTER_ERROR; |
| } |
| if (std == NULL) { |
| return AEC_NULL_POINTER_ERROR; |
| } |
| if (self->initFlag != initCheck) { |
| return AEC_UNINITIALIZED_ERROR; |
| } |
| if (WebRtcAec_GetDelayMetricsCore(self->aec, median, std, |
| fraction_poor_delays) == -1) { |
| // Logging disabled. |
| return AEC_UNSUPPORTED_FUNCTION_ERROR; |
| } |
| |
| return 0; |
| } |
| |
| AecCore* WebRtcAec_aec_core(void* handle) { |
| if (!handle) { |
| return NULL; |
| } |
| return reinterpret_cast<Aec*>(handle)->aec; |
| } |
| |
| static int ProcessNormal(Aec* aecInst, |
| const float* const* nearend, |
| size_t num_bands, |
| float* const* out, |
| size_t num_samples, |
| int16_t reported_delay_ms, |
| int32_t skew) { |
| int retVal = 0; |
| size_t i; |
| size_t nBlocks10ms; |
| // Limit resampling to doubling/halving of signal |
| const float minSkewEst = -0.5f; |
| const float maxSkewEst = 1.0f; |
| |
| reported_delay_ms = reported_delay_ms > kMaxTrustedDelayMs |
| ? kMaxTrustedDelayMs |
| : reported_delay_ms; |
| // TODO(andrew): we need to investigate if this +10 is really wanted. |
| reported_delay_ms += 10; |
| aecInst->msInSndCardBuf = reported_delay_ms; |
| |
| if (aecInst->skewMode == kAecTrue) { |
| if (aecInst->skewFrCtr < 25) { |
| aecInst->skewFrCtr++; |
| } else { |
| retVal = WebRtcAec_GetSkew(aecInst->resampler, skew, &aecInst->skew); |
| if (retVal == -1) { |
| aecInst->skew = 0; |
| retVal = AEC_BAD_PARAMETER_WARNING; |
| } |
| |
| aecInst->skew /= aecInst->sampFactor * num_samples; |
| |
| if (aecInst->skew < 1.0e-3 && aecInst->skew > -1.0e-3) { |
| aecInst->resample = kAecFalse; |
| } else { |
| aecInst->resample = kAecTrue; |
| } |
| |
| if (aecInst->skew < minSkewEst) { |
| aecInst->skew = minSkewEst; |
| } else if (aecInst->skew > maxSkewEst) { |
| aecInst->skew = maxSkewEst; |
| } |
| |
| aecInst->data_dumper->DumpRaw("aec_skew", 1, &aecInst->skew); |
| } |
| } |
| |
| nBlocks10ms = num_samples / (FRAME_LEN * aecInst->rate_factor); |
| |
| if (aecInst->startup_phase) { |
| for (i = 0; i < num_bands; ++i) { |
| // Only needed if they don't already point to the same place. |
| if (nearend[i] != out[i]) { |
| memcpy(out[i], nearend[i], sizeof(nearend[i][0]) * num_samples); |
| } |
| } |
| |
| // 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 (aecInst->checkBuffSize) { |
| aecInst->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 (aecInst->counter == 0) { |
| aecInst->firstVal = aecInst->msInSndCardBuf; |
| aecInst->sum = 0; |
| } |
| |
| if (abs(aecInst->firstVal - aecInst->msInSndCardBuf) < |
| WEBRTC_SPL_MAX(0.2 * aecInst->msInSndCardBuf, sampMsNb)) { |
| aecInst->sum += aecInst->msInSndCardBuf; |
| aecInst->counter++; |
| } else { |
| aecInst->counter = 0; |
| } |
| |
| if (aecInst->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. |
| aecInst->bufSizeStart = |
| WEBRTC_SPL_MIN((3 * aecInst->sum * aecInst->rate_factor * 8) / |
| (4 * aecInst->counter * PART_LEN), |
| kMaxBufSizeStart); |
| // Buffer size has now been determined. |
| aecInst->checkBuffSize = 0; |
| } |
| |
| if (aecInst->checkBufSizeCtr * nBlocks10ms > 50) { |
| // For really bad systems, don't disable the echo canceller for |
| // more than 0.5 sec. |
| aecInst->bufSizeStart = WEBRTC_SPL_MIN( |
| (aecInst->msInSndCardBuf * aecInst->rate_factor * 3) / 40, |
| kMaxBufSizeStart); |
| aecInst->checkBuffSize = 0; |
| } |
| } |
| |
| // If |checkBuffSize| changed in the if-statement above. |
| if (!aecInst->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(aecInst->aec) / PART_LEN - |
| aecInst->bufSizeStart; |
| if (overhead_elements == 0) { |
| // Enable the AEC |
| aecInst->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_AdjustFarendBufferSizeAndSystemDelay(aecInst->aec, |
| overhead_elements); |
| |
| // Enable the AEC |
| aecInst->startup_phase = 0; |
| } |
| } |
| } else { |
| // AEC is enabled. |
| EstBufDelayNormal(aecInst); |
| |
| // Call the AEC. |
| // TODO(bjornv): Re-structure such that we don't have to pass |
| // |aecInst->knownDelay| as input. Change name to something like |
| // |system_buffer_diff|. |
| WebRtcAec_ProcessFrames(aecInst->aec, nearend, num_bands, num_samples, |
| aecInst->knownDelay, out); |
| } |
| |
| return retVal; |
| } |
| |
| static void ProcessExtended(Aec* self, |
| const float* const* nearend, |
| size_t num_bands, |
| float* const* out, |
| size_t num_samples, |
| int16_t reported_delay_ms, |
| int32_t skew) { |
| size_t i; |
| const int delay_diff_offset = kDelayDiffOffsetSamples; |
| RTC_DCHECK(num_samples == 80 || num_samples == 160); |
| #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) { |
| for (i = 0; i < num_bands; ++i) { |
| // Only needed if they don't already point to the same place. |
| if (nearend[i] != out[i]) { |
| memcpy(out[i], nearend[i], sizeof(nearend[i][0]) * 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; |
| #if defined(WEBRTC_ANDROID) |
| int target_delay = startup_size_ms * self->rate_factor * 8; |
| #else |
| // To avoid putting the AEC in a non-causal state we're being slightly |
| // conservative and scale by 2. On Android we use a fixed delay and |
| // therefore there is no need to scale the target_delay. |
| int target_delay = startup_size_ms * self->rate_factor * 8 / 2; |
| #endif |
| int overhead_elements = |
| (WebRtcAec_system_delay(self->aec) - target_delay) / PART_LEN; |
| WebRtcAec_AdjustFarendBufferSizeAndSystemDelay(self->aec, |
| overhead_elements); |
| self->startup_phase = 0; |
| } |
| |
| 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); |
| |
| WebRtcAec_ProcessFrames(self->aec, nearend, num_bands, num_samples, |
| adjusted_known_delay, out); |
| } |
| } |
| |
| static void EstBufDelayNormal(Aec* aecInst) { |
| int nSampSndCard = aecInst->msInSndCardBuf * sampMsNb * aecInst->rate_factor; |
| int current_delay = nSampSndCard - WebRtcAec_system_delay(aecInst->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 * aecInst->rate_factor; |
| |
| // 2) Account for resampling frame delay. |
| if (aecInst->skewMode == kAecTrue && aecInst->resample == kAecTrue) { |
| current_delay -= kResamplingDelay; |
| } |
| |
| // 3) Compensate for non-causality, if needed, by flushing one block. |
| if (current_delay < PART_LEN) { |
| current_delay += |
| WebRtcAec_AdjustFarendBufferSizeAndSystemDelay(aecInst->aec, 1) * |
| PART_LEN; |
| } |
| |
| // We use -1 to signal an initialized state in the "extended" implementation; |
| // compensate for that. |
| aecInst->filtDelay = aecInst->filtDelay < 0 ? 0 : aecInst->filtDelay; |
| aecInst->filtDelay = WEBRTC_SPL_MAX( |
| 0, static_cast<int16_t>(0.8 * aecInst->filtDelay + 0.2 * current_delay)); |
| |
| delay_difference = aecInst->filtDelay - aecInst->knownDelay; |
| if (delay_difference > 224) { |
| if (aecInst->lastDelayDiff < 96) { |
| aecInst->timeForDelayChange = 0; |
| } else { |
| aecInst->timeForDelayChange++; |
| } |
| } else if (delay_difference < 96 && aecInst->knownDelay > 0) { |
| if (aecInst->lastDelayDiff > 224) { |
| aecInst->timeForDelayChange = 0; |
| } else { |
| aecInst->timeForDelayChange++; |
| } |
| } else { |
| aecInst->timeForDelayChange = 0; |
| } |
| aecInst->lastDelayDiff = delay_difference; |
| |
| if (aecInst->timeForDelayChange > 25) { |
| aecInst->knownDelay = WEBRTC_SPL_MAX((int)aecInst->filtDelay - 160, 0); |
| } |
| } |
| |
| static void EstBufDelayExtended(Aec* aecInst) { |
| int reported_delay = |
| aecInst->msInSndCardBuf * sampMsNb * aecInst->rate_factor; |
| int current_delay = reported_delay - WebRtcAec_system_delay(aecInst->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 * aecInst->rate_factor; |
| |
| // 2) Account for resampling frame delay. |
| if (aecInst->skewMode == kAecTrue && aecInst->resample == kAecTrue) { |
| current_delay -= kResamplingDelay; |
| } |
| |
| // 3) Compensate for non-causality, if needed, by flushing two blocks. |
| if (current_delay < PART_LEN) { |
| current_delay += |
| WebRtcAec_AdjustFarendBufferSizeAndSystemDelay(aecInst->aec, 2) * |
| PART_LEN; |
| } |
| |
| if (aecInst->filtDelay == -1) { |
| aecInst->filtDelay = WEBRTC_SPL_MAX(0, 0.5 * current_delay); |
| } else { |
| aecInst->filtDelay = WEBRTC_SPL_MAX( |
| 0, |
| static_cast<int16_t>(0.95 * aecInst->filtDelay + 0.05 * current_delay)); |
| } |
| |
| delay_difference = aecInst->filtDelay - aecInst->knownDelay; |
| if (delay_difference > 384) { |
| if (aecInst->lastDelayDiff < 128) { |
| aecInst->timeForDelayChange = 0; |
| } else { |
| aecInst->timeForDelayChange++; |
| } |
| } else if (delay_difference < 128 && aecInst->knownDelay > 0) { |
| if (aecInst->lastDelayDiff > 384) { |
| aecInst->timeForDelayChange = 0; |
| } else { |
| aecInst->timeForDelayChange++; |
| } |
| } else { |
| aecInst->timeForDelayChange = 0; |
| } |
| aecInst->lastDelayDiff = delay_difference; |
| |
| if (aecInst->timeForDelayChange > 25) { |
| aecInst->knownDelay = WEBRTC_SPL_MAX((int)aecInst->filtDelay - 256, 0); |
| } |
| } |
| } // namespace webrtc |
