Adding full initial version of delay estimation functionality in echo canceller 3

webrtc/modules/audio_processing/aec3/down_sampler_4khz.h

+/*
+ *  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_DOWN_SAMPLER_4KHZ_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_DOWN_SAMPLER_4KHZ_H_
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_processing/aec3/cascaded_biquad_filter.h"
+
+namespace webrtc {
+
+// Provides downsampling functionality to a 4 kHz rate.
+class DownSampler4kHz {
+ public:
+  explicit DownSampler4kHz(int sample_rate_hz_);

[aleloi, 2017/01/27 15:37:47]

I suggest adding a comment that sample_rate_hz_ is the *full* band sample rate
and that the down sampler operates on 8 or 16 khz bands.

[peah-webrtc, 2017/02/02 14:04:47]

Done.

+
+  // Downsamples the signal.

[aleloi, 2017/01/27 15:37:47]

I think assumptions about frame size should be documented here. Or not, because
they are clearly documented with DCHECKS in the implementation. I can't decide,
and leave it to you.

[peah-webrtc, 2017/02/02 14:04:46]

I agree. This is a big drawback with ArrayView and vectors as you cannot ensure
this until runtime. I have in the upcoming CLs changed the code to instead use
std::array which checks this at compile time. I can only change this partly in
this CL.

Ok if I do this partial change, and wait with the other changes until an
upcoming CL?

[aleloi, 2017/02/03 15:46:48]

SGTM! Great that fixed-size array could be used!

[peah-webrtc, 2017/02/06 11:25:38]

:-)

+  void DownSample(rtc::ArrayView<const float> in, rtc::ArrayView<float> out);
+
+  // Returns the downsampling factor applied in the downsampler.
+  int GetDownSamplingFactor() const { return down_sampling_factor_; }
+
+ private:
+  const int down_sampling_factor_;
+  CascadedBiQuadFilter low_pass_filter_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(DownSampler4kHz);
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_DOWN_SAMPLER_4KHZ_H_