Index: webrtc/modules/audio_processing/aec/echo_cancellation.c |
diff --git a/webrtc/modules/audio_processing/aec/echo_cancellation.c b/webrtc/modules/audio_processing/aec/echo_cancellation.c |
deleted file mode 100644 |
index 5a30cc3ce486ad66ce90cd91899b8171f7446521..0000000000000000000000000000000000000000 |
--- a/webrtc/modules/audio_processing/aec/echo_cancellation.c |
+++ /dev/null |
@@ -1,875 +0,0 @@ |
-/* |
- * 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/echo_cancellation.h" |
- |
-#include <math.h> |
-#ifdef WEBRTC_AEC_DEBUG_DUMP |
-#include <stdio.h> |
-#endif |
-#include <stdlib.h> |
-#include <string.h> |
- |
-#include "webrtc/common_audio/ring_buffer.h" |
-#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/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(Aec* aecInst); |
-static void EstBufDelayExtended(Aec* aecInst); |
-static int ProcessNormal(Aec* self, |
- const float* const* near, |
- size_t num_bands, |
- float* const* out, |
- size_t num_samples, |
- int16_t reported_delay_ms, |
- int32_t skew); |
-static void ProcessExtended(Aec* self, |
- const float* const* near, |
- size_t num_bands, |
- float* const* out, |
- size_t num_samples, |
- int16_t reported_delay_ms, |
- int32_t skew); |
- |
-void* WebRtcAec_Create() { |
- Aec* aecpc = malloc(sizeof(Aec)); |
- |
- if (!aecpc) { |
- return NULL; |
- } |
- |
- aecpc->aec = WebRtcAec_CreateAec(); |
- 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; |
- |
-#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 aecpc; |
-} |
- |
-void WebRtcAec_Free(void* aecInst) { |
- Aec* aecpc = (Aec*)aecInst; |
- |
- if (aecpc == NULL) { |
- return; |
- } |
- |
- 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); |
-} |
- |
-int32_t WebRtcAec_Init(void* aecInst, int32_t sampFreq, int32_t scSampFreq) { |
- Aec* aecpc = (Aec*)aecInst; |
- 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 = (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 = (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) + (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, when |
- // WebRtcAec_BufferFarendPartition() is changed to take |PART_LEN| samples. |
- 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, (void**)&ptmp, tmp, PART_LEN2); |
- WebRtcAec_BufferFarendPartition(aecpc->aec, ptmp); |
- } |
- |
- // 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 = (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); |
- } |
- |
-#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) { |
- Aec* self = (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 = (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 = (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); |
- |
- // 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, |
- float* fraction_poor_delays) { |
- Aec* self = (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 ((Aec*)handle)->aec; |
-} |
- |
-static int ProcessNormal(Aec* aecpc, |
- const float* const* nearend, |
- size_t num_bands, |
- float* const* out, |
- size_t nrOfSamples, |
- int16_t msInSndCardBuf, |
- 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; |
- |
- 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; |
- retVal = 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 |
- } |
- } |
- |
- nBlocks10ms = nrOfSamples / (FRAME_LEN * aecpc->rate_factor); |
- |
- if (aecpc->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]) * 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. |
- EstBufDelayNormal(aecpc); |
- |
- // Call the AEC. |
- // TODO(bjornv): Re-structure such that we don't have to pass |
- // |aecpc->knownDelay| as input. Change name to something like |
- // |system_buffer_diff|. |
- WebRtcAec_ProcessFrames(aecpc->aec, nearend, num_bands, nrOfSamples, |
- aecpc->knownDelay, out); |
- } |
- |
- return retVal; |
-} |
- |
-static void ProcessExtended(Aec* self, |
- const float* const* near, |
- 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; |
-#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 (near[i] != out[i]) { |
- memcpy(out[i], near[i], sizeof(near[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_MoveFarReadPtr(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, near, num_bands, num_samples, |
- adjusted_known_delay, out); |
- } |
-} |
- |
-static void EstBufDelayNormal(Aec* 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(Aec* 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); |
- } |
-} |