Moved the functionality in aec_core_internal.h into other
files.
The purpose of this CL is to simplify upcoming AEC algorithm
changes.
The changes should be bitexact.
The presubmit was bypassed due to a presubmit complaint
about usage of short instead of int16_t which will be
addressed in upcoming CLs.
BUG=webrtc:5298, webrtc:5201
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/1949803004
Cr-Commit-Position: refs/heads/master@{#12662}
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index b4afb16..a9650f7 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -24,7 +24,7 @@
sources = [
"aec/aec_core.cc",
"aec/aec_core.h",
- "aec/aec_core_internal.h",
+ "aec/aec_core_optimized_methods.h",
"aec/aec_rdft.cc",
"aec/aec_rdft.h",
"aec/aec_resampler.cc",
diff --git a/webrtc/modules/audio_processing/aec/aec_core.cc b/webrtc/modules/audio_processing/aec/aec_core.cc
index 97b5c94..ca5eae4 100644
--- a/webrtc/modules/audio_processing/aec/aec_core.cc
+++ b/webrtc/modules/audio_processing/aec/aec_core.cc
@@ -28,8 +28,9 @@
#include "webrtc/base/checks.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_processing/aec/aec_common.h"
-#include "webrtc/modules/audio_processing/aec/aec_core_internal.h"
+#include "webrtc/modules/audio_processing/aec/aec_core_optimized_methods.h"
#include "webrtc/modules/audio_processing/aec/aec_rdft.h"
+#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
#include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/audio_processing/aec/aec_core.h b/webrtc/modules/audio_processing/aec/aec_core.h
index 1155bc3..668b484 100644
--- a/webrtc/modules/audio_processing/aec/aec_core.h
+++ b/webrtc/modules/audio_processing/aec/aec_core.h
@@ -17,6 +17,15 @@
#include <stddef.h>
+#include <memory>
+
+extern "C" {
+#include "webrtc/common_audio/ring_buffer.h"
+}
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/common_audio/wav_file.h"
+#include "webrtc/modules/audio_processing/aec/aec_common.h"
+#include "webrtc/modules/audio_processing/utility/block_mean_calculator.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -27,6 +36,8 @@
#define PART_LEN2 (PART_LEN * 2) // Length of partition * 2
#define NUM_HIGH_BANDS_MAX 2 // Max number of high bands
+class ApmDataDumper;
+
typedef float complex_t[2];
// For performance reasons, some arrays of complex numbers are replaced by twice
// as long arrays of float, all the real parts followed by all the imaginary
@@ -51,7 +62,175 @@
size_t hicounter;
} Stats;
-typedef struct AecCore AecCore;
+// Number of partitions for the extended filter mode. The first one is an enum
+// to be used in array declarations, as it represents the maximum filter length.
+enum { kExtendedNumPartitions = 32 };
+static const int kNormalNumPartitions = 12;
+
+// Delay estimator constants, used for logging and delay compensation if
+// if reported delays are disabled.
+enum { kLookaheadBlocks = 15 };
+enum {
+ // 500 ms for 16 kHz which is equivalent with the limit of reported delays.
+ kHistorySizeBlocks = 125
+};
+
+typedef struct PowerLevel {
+ PowerLevel();
+
+ BlockMeanCalculator framelevel;
+ BlockMeanCalculator averagelevel;
+ float minlevel;
+} PowerLevel;
+
+class DivergentFilterFraction {
+ public:
+ DivergentFilterFraction();
+
+ // Reset.
+ void Reset();
+
+ void AddObservation(const PowerLevel& nearlevel,
+ const PowerLevel& linoutlevel,
+ const PowerLevel& nlpoutlevel);
+
+ // Return the latest fraction.
+ float GetLatestFraction() const;
+
+ private:
+ // Clear all values added.
+ void Clear();
+
+ size_t count_;
+ size_t occurrence_;
+ float fraction_;
+
+ RTC_DISALLOW_COPY_AND_ASSIGN(DivergentFilterFraction);
+};
+
+typedef struct CoherenceState {
+ complex_t sde[PART_LEN1]; // cross-psd of nearend and error
+ complex_t sxd[PART_LEN1]; // cross-psd of farend and nearend
+ float sx[PART_LEN1], sd[PART_LEN1], se[PART_LEN1]; // far, near, error psd
+} CoherenceState;
+
+struct AecCore {
+ explicit AecCore(int instance_index);
+ ~AecCore();
+
+ std::unique_ptr<ApmDataDumper> data_dumper;
+
+ CoherenceState coherence_state;
+
+ int farBufWritePos, farBufReadPos;
+
+ int knownDelay;
+ int inSamples, outSamples;
+ int delayEstCtr;
+
+ RingBuffer* nearFrBuf;
+ RingBuffer* outFrBuf;
+
+ RingBuffer* nearFrBufH[NUM_HIGH_BANDS_MAX];
+ RingBuffer* outFrBufH[NUM_HIGH_BANDS_MAX];
+
+ float dBuf[PART_LEN2]; // nearend
+ float eBuf[PART_LEN2]; // error
+
+ float dBufH[NUM_HIGH_BANDS_MAX][PART_LEN2]; // nearend
+
+ float xPow[PART_LEN1];
+ float dPow[PART_LEN1];
+ float dMinPow[PART_LEN1];
+ float dInitMinPow[PART_LEN1];
+ float* noisePow;
+
+ float xfBuf[2][kExtendedNumPartitions * PART_LEN1]; // farend fft buffer
+ float wfBuf[2][kExtendedNumPartitions * PART_LEN1]; // filter fft
+ // Farend windowed fft buffer.
+ complex_t xfwBuf[kExtendedNumPartitions * PART_LEN1];
+
+ float hNs[PART_LEN1];
+ float hNlFbMin, hNlFbLocalMin;
+ float hNlXdAvgMin;
+ int hNlNewMin, hNlMinCtr;
+ float overDrive;
+ float overdrive_scaling;
+ int nlp_mode;
+ float outBuf[PART_LEN];
+ int delayIdx;
+
+ short stNearState, echoState;
+ short divergeState;
+
+ int xfBufBlockPos;
+
+ RingBuffer* far_time_buf;
+
+ int system_delay; // Current system delay buffered in AEC.
+
+ int mult; // sampling frequency multiple
+ int sampFreq = 16000;
+ size_t num_bands;
+ uint32_t seed;
+
+ float filter_step_size; // stepsize
+ float error_threshold; // error threshold
+
+ int noiseEstCtr;
+
+ PowerLevel farlevel;
+ PowerLevel nearlevel;
+ PowerLevel linoutlevel;
+ PowerLevel nlpoutlevel;
+
+ int metricsMode;
+ int stateCounter;
+ Stats erl;
+ Stats erle;
+ Stats aNlp;
+ Stats rerl;
+ DivergentFilterFraction divergent_filter_fraction;
+
+ // Quantities to control H band scaling for SWB input
+ int freq_avg_ic; // initial bin for averaging nlp gain
+ int flag_Hband_cn; // for comfort noise
+ float cn_scale_Hband; // scale for comfort noise in H band
+
+ int delay_metrics_delivered;
+ int delay_histogram[kHistorySizeBlocks];
+ int num_delay_values;
+ int delay_median;
+ int delay_std;
+ float fraction_poor_delays;
+ int delay_logging_enabled;
+ void* delay_estimator_farend;
+ void* delay_estimator;
+ // Variables associated with delay correction through signal based delay
+ // estimation feedback.
+ int signal_delay_correction;
+ int previous_delay;
+ int delay_correction_count;
+ int shift_offset;
+ float delay_quality_threshold;
+ int frame_count;
+
+ // 0 = delay agnostic mode (signal based delay correction) disabled.
+ // Otherwise enabled.
+ int delay_agnostic_enabled;
+ // 1 = extended filter mode enabled, 0 = disabled.
+ int extended_filter_enabled;
+ // 1 = next generation aec mode enabled, 0 = disabled.
+ int aec3_enabled;
+ bool refined_adaptive_filter_enabled;
+
+ // Runtime selection of number of filter partitions.
+ int num_partitions;
+
+ // Flag that extreme filter divergence has been detected by the Echo
+ // Suppressor.
+ int extreme_filter_divergence;
+};
AecCore* WebRtcAec_CreateAec(int instance_count); // Returns NULL on error.
void WebRtcAec_FreeAec(AecCore* aec);
diff --git a/webrtc/modules/audio_processing/aec/aec_core_mips.cc b/webrtc/modules/audio_processing/aec/aec_core_mips.cc
index 30c6ccc..a9b5cd4 100644
--- a/webrtc/modules/audio_processing/aec/aec_core_mips.cc
+++ b/webrtc/modules/audio_processing/aec/aec_core_mips.cc
@@ -19,7 +19,7 @@
extern "C" {
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
}
-#include "webrtc/modules/audio_processing/aec/aec_core_internal.h"
+#include "webrtc/modules/audio_processing/aec/aec_core_optimized_methods.h"
#include "webrtc/modules/audio_processing/aec/aec_rdft.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_processing/aec/aec_core_neon.cc b/webrtc/modules/audio_processing/aec/aec_core_neon.cc
index fda5ad8..bc503ba 100644
--- a/webrtc/modules/audio_processing/aec/aec_core_neon.cc
+++ b/webrtc/modules/audio_processing/aec/aec_core_neon.cc
@@ -22,7 +22,7 @@
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
}
#include "webrtc/modules/audio_processing/aec/aec_common.h"
-#include "webrtc/modules/audio_processing/aec/aec_core_internal.h"
+#include "webrtc/modules/audio_processing/aec/aec_core_optimized_methods.h"
#include "webrtc/modules/audio_processing/aec/aec_rdft.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_processing/aec/aec_core_optimized_methods.h b/webrtc/modules/audio_processing/aec/aec_core_optimized_methods.h
new file mode 100644
index 0000000..d1fb6e8
--- /dev/null
+++ b/webrtc/modules/audio_processing/aec/aec_core_optimized_methods.h
@@ -0,0 +1,79 @@
+/*
+ * Copyright (c) 2016 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_OPTIMIZED_METHODS_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_OPTIMIZED_METHODS_H_
+
+#include <memory>
+
+#include "webrtc/modules/audio_processing/aec/aec_core.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+typedef void (*WebRtcAecFilterFar)(
+ int num_partitions,
+ int x_fft_buf_block_pos,
+ float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+ float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+ float y_fft[2][PART_LEN1]);
+extern WebRtcAecFilterFar WebRtcAec_FilterFar;
+typedef void (*WebRtcAecScaleErrorSignal)(float mu,
+ float error_threshold,
+ float x_pow[PART_LEN1],
+ float ef[2][PART_LEN1]);
+extern WebRtcAecScaleErrorSignal WebRtcAec_ScaleErrorSignal;
+typedef void (*WebRtcAecFilterAdaptation)(
+ int num_partitions,
+ int x_fft_buf_block_pos,
+ float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
+ float e_fft[2][PART_LEN1],
+ float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1]);
+extern WebRtcAecFilterAdaptation WebRtcAec_FilterAdaptation;
+
+typedef void (*WebRtcAecOverdrive)(float overdrive_scaling,
+ const float hNlFb,
+ float hNl[PART_LEN1]);
+extern WebRtcAecOverdrive WebRtcAec_Overdrive;
+
+typedef void (*WebRtcAecSuppress)(const float hNl[PART_LEN1],
+ float efw[2][PART_LEN1]);
+extern WebRtcAecSuppress WebRtcAec_Suppress;
+
+typedef void (*WebRtcAecComputeCoherence)(const CoherenceState* coherence_state,
+ float* cohde,
+ float* cohxd);
+extern WebRtcAecComputeCoherence WebRtcAec_ComputeCoherence;
+
+typedef void (*WebRtcAecUpdateCoherenceSpectra)(int mult,
+ bool extended_filter_enabled,
+ float efw[2][PART_LEN1],
+ float dfw[2][PART_LEN1],
+ float xfw[2][PART_LEN1],
+ CoherenceState* coherence_state,
+ short* filter_divergence_state,
+ int* extreme_filter_divergence);
+extern WebRtcAecUpdateCoherenceSpectra WebRtcAec_UpdateCoherenceSpectra;
+
+typedef int (*WebRtcAecPartitionDelay)(
+ int num_partitions,
+ float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1]);
+extern WebRtcAecPartitionDelay WebRtcAec_PartitionDelay;
+
+typedef void (*WebRtcAecStoreAsComplex)(const float* data,
+ float data_complex[2][PART_LEN1]);
+extern WebRtcAecStoreAsComplex WebRtcAec_StoreAsComplex;
+
+typedef void (*WebRtcAecWindowData)(float* x_windowed, const float* x);
+extern WebRtcAecWindowData WebRtcAec_WindowData;
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_OPTIMIZED_METHODS_H_
diff --git a/webrtc/modules/audio_processing/aec/aec_core_sse2.cc b/webrtc/modules/audio_processing/aec/aec_core_sse2.cc
index fa6623b..47ba12f 100644
--- a/webrtc/modules/audio_processing/aec/aec_core_sse2.cc
+++ b/webrtc/modules/audio_processing/aec/aec_core_sse2.cc
@@ -20,7 +20,7 @@
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
}
#include "webrtc/modules/audio_processing/aec/aec_common.h"
-#include "webrtc/modules/audio_processing/aec/aec_core_internal.h"
+#include "webrtc/modules/audio_processing/aec/aec_core_optimized_methods.h"
#include "webrtc/modules/audio_processing/aec/aec_rdft.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_processing/audio_processing.gypi b/webrtc/modules/audio_processing/audio_processing.gypi
index 225dffd..e6eab36 100644
--- a/webrtc/modules/audio_processing/audio_processing.gypi
+++ b/webrtc/modules/audio_processing/audio_processing.gypi
@@ -35,7 +35,7 @@
'sources': [
'aec/aec_core.cc',
'aec/aec_core.h',
- 'aec/aec_core_internal.h',
+ 'aec/aec_core_optimized_methods.h',
'aec/aec_rdft.cc',
'aec/aec_rdft.h',
'aec/aec_resampler.cc',
