Major AEC3 render pipeline changes
This CL adds major render pipeline changes to the AEC3 code. The reason
for these are that
1) It allows the echo removal unit to receive information about the content
in bands beyond band 0, thereby allowing removal of high-frequency
echoes
2) It allows more controlled handling of the render buffers, allowing proper
buffer behaviour during capture glitches and clock-drift.
Unfortunately, the render pipeline caused a lot of related changes in much
of the rest of the AEC3 files. Most of these are, however, caused by
a change of class name.
Another unfortunate effect of this CL, is that a number of unittest cease to
compile. I chose to temporarily solve that by removing them from the
build using #if/#endif. The reason for that is that those will anyway again
need to be changed in the next review, and doing like this avoids them
having to be reviewed twice.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2784023002
Cr-Commit-Position: refs/heads/master@{#17547}
diff --git a/webrtc/modules/audio_processing/aec3/render_buffer.h b/webrtc/modules/audio_processing/aec3/render_buffer.h
new file mode 100644
index 0000000..c3acdd9
--- /dev/null
+++ b/webrtc/modules/audio_processing/aec3/render_buffer.h
@@ -0,0 +1,81 @@
+/*
+ * Copyright (c) 2017 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_AEC3_RENDER_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_BUFFER_H_
+
+#include <memory>
+#include <vector>
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_processing/aec3/aec3_fft.h"
+#include "webrtc/modules/audio_processing/aec3/fft_data.h"
+
+namespace webrtc {
+
+// Provides a buffer of the render data for the echo remover.
+class RenderBuffer {
+ public:
+ // The constructor takes, besides from the other parameters, a vector
+ // containing the number of FFTs that will be included in the spectral sums in
+ // the call to SpectralSum.
+ RenderBuffer(Aec3Optimization optimization,
+ size_t num_bands,
+ size_t size,
+ const std::vector<size_t> num_ffts_for_spectral_sums);
+ ~RenderBuffer();
+
+ // Clears the buffer.
+ void Clear();
+
+ // Insert a block into the buffer.
+ void Insert(const std::vector<std::vector<float>>& block);
+
+ // Gets the last inserted block.
+ const std::vector<std::vector<float>>& MostRecentBlock() const {
+ return last_block_;
+ }
+
+ // Get the spectrum from one of the FFTs in the buffer
+ const std::array<float, kFftLengthBy2Plus1>& Spectrum(
+ size_t buffer_offset_ffts) const {
+ return spectrum_buffer_[(position_ + buffer_offset_ffts) %
+ fft_buffer_.size()];
+ }
+
+ // Returns the sum of the spectrums for a certain number of FFTs.
+ const std::array<float, kFftLengthBy2Plus1>& SpectralSum(
+ size_t num_ffts) const {
+ RTC_DCHECK_EQ(spectral_sums_length_, num_ffts);
+ return spectral_sums_[0];
+ }
+
+ // Returns the circular buffer.
+ rtc::ArrayView<const FftData> Buffer() const { return fft_buffer_; }
+
+ // Returns the current position in the circular buffer
+ size_t Position() const { return position_; }
+
+ private:
+ const Aec3Optimization optimization_;
+ std::vector<FftData> fft_buffer_;
+ std::vector<std::array<float, kFftLengthBy2Plus1>> spectrum_buffer_;
+ size_t spectral_sums_length_;
+ std::vector<std::array<float, kFftLengthBy2Plus1>> spectral_sums_;
+ size_t position_ = 0;
+ std::vector<std::vector<float>> last_block_;
+ const Aec3Fft fft_;
+ RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderBuffer);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_BUFFER_H_