New module for the adaptive level controlling functionality in the audio processing module

webrtc/modules/audio_processing/level_controller/down_sampler.cc

Index: webrtc/modules/audio_processing/level_controller/down_sampler.cc
diff --git a/webrtc/modules/audio_processing/level_controller/down_sampler.cc b/webrtc/modules/audio_processing/level_controller/down_sampler.cc
new file mode 100644
index 0000000000000000000000000000000000000000..e1be7edabc5c796d35a8305d1a48d3bfdfb7b0e7
--- /dev/null
+++ b/webrtc/modules/audio_processing/level_controller/down_sampler.cc
@@ -0,0 +1,101 @@
+/*
+ *  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.
+ */
+
+#include "webrtc/modules/audio_processing/level_controller/down_sampler.h"
+
+#include <string.h>
+#include <algorithm>
+#include <vector>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/level_controller/biquad_filter.h"
+#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
+
+namespace webrtc {
+namespace {
+
+// Bandlimiter coefficients computed based on that only
+// the first 40 bins of the spectrum for the downsampled
+// signal are used.
+// [B,A] = butter(2,(41/64*4000)/8000)
+const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_16kHz = {
+    {0.1455f, 0.2911f, 0.1455f},
+    {-0.6698f, 0.2520f}};
+
+// [B,A] = butter(2,(41/64*4000)/16000)
+const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_32kHz = {
+    {0.0462f, 0.0924f, 0.0462f},
+    {-1.3066f, 0.4915f}};
+
+// [B,A] = butter(2,(41/64*4000)/24000)
+const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_48kHz = {
+    {0.0226f, 0.0452f, 0.0226f},
+    {-1.5320f, 0.6224f}};
+
+}  // namespace
+
+DownSampler::DownSampler(ApmDataDumper* data_dumper)
+    : data_dumper_(data_dumper) {
+  Initialize(48000);
+}
+void DownSampler::Initialize(int sample_rate_hz) {
+  RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
+             sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
+             sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
+             sample_rate_hz == AudioProcessing::kSampleRate48kHz);
+
+  sample_rate_hz_ = sample_rate_hz;

[aleloi, 2016/06/28 09:35:51]

This line looks fishy. It's an assignment of the parameter sample_rate_hz to
itself because of scoping rules, right?

[peah-webrtc, 2016/06/28 22:19:37]

No, I think you probably missed the underscore in the variable name to the left
of the equal sign. This statement sets the member variable sample_rate_hz_ to
the value of sample_rate_hz.

+  down_sampling_factor_ = rtc::CheckedDivExact(sample_rate_hz_, 8000);
+
+  /// Note that the down sampling filter is not used if the sample rate is 8
+  /// kHz.
+  if (sample_rate_hz_ == AudioProcessing::kSampleRate16kHz) {
+    low_pass_filter_.Initialize(kLowPassFilterCoefficients_16kHz);
+  } else if (sample_rate_hz_ == AudioProcessing::kSampleRate32kHz) {
+    low_pass_filter_.Initialize(kLowPassFilterCoefficients_32kHz);
+  } else if (sample_rate_hz_ == AudioProcessing::kSampleRate48kHz) {
+    low_pass_filter_.Initialize(kLowPassFilterCoefficients_48kHz);
+  }
+}
+
+void DownSampler::DownSample(rtc::ArrayView<const float> in,
+                             rtc::ArrayView<float> out) {
+  data_dumper_->DumpWav("lc_down_sampler_input", in, sample_rate_hz_, 1);

[aleloi, 2016/06/28 09:35:51]

Is saving the input/output to file always the intended functionality?

[aleloi, 2016/06/28 09:40:01]

Just noticed that dumping only happens when WEBRTC_AEC_DEBUG_DUMP == 1.
Disregard previous comment!

[peah-webrtc, 2016/06/28 22:19:37]

Acknowledged.

[peah-webrtc, 2016/06/28 22:19:37]

Acknowledged.

+  RTC_DCHECK_EQ(static_cast<size_t>(sample_rate_hz_ *
+                                    AudioProcessing::kChunkSizeMs / 1000),
+                in.size());
+  RTC_DCHECK_EQ(static_cast<size_t>(AudioProcessing::kSampleRate8kHz *
+                                    AudioProcessing::kChunkSizeMs / 1000),
+                out.size());
+  const size_t kMaxNumFrames =
+      AudioProcessing::kSampleRate48kHz * AudioProcessing::kChunkSizeMs / 1000;
+  float x[kMaxNumFrames];
+
+  // Band-limit the signal to 4 kHz.
+  if (sample_rate_hz_ != AudioProcessing::kSampleRate8kHz) {
+    low_pass_filter_.Process(in, rtc::ArrayView<float>(x, in.size()));

[aleloi, 2016/06/28 09:35:51]

Just to check that I understood correctly: we have a custom simple filter and
resampling algorithm for performance reasons? Instead of using one from webrtc
that is already implemented.

[peah-webrtc, 2016/06/28 22:19:37]

You definitely have a good point! It would perhaps make sense to reuse one of
the other resamplers. 
I still think it may make sense to use this resampler regardless of that as the
current resampler is tailormade to use a cutoff filter with a quite low cutoff
frequency which allows the bandlimiting filter to be likely much cheaper than
what is the case for the other more general resamplers. Furthermore, having now
looked at the existing resampling code, I think that looks more expensive. Still
it would make sense to look into that.

WDYT, is it ok to keep it as it is with a custom made downsampler for now?

[hlundin-webrtc, 2016/06/29 08:56:28]

I think we should keep this for now, since time is scarce and you've done
testing with the current implementation. But in general I do agree that we
should try to consolidate things like general filter implementations.

[peah-webrtc, 2016/06/29 09:13:53]

Acknowledged.

+
+    // Downsample the signal.
+    size_t k = 0;
+    for (size_t j = 0; j < out.size(); ++j) {
+      RTC_DCHECK_GT(kMaxNumFrames, k);
+      out[j] = x[k];
+      k += down_sampling_factor_;
+    }
+  } else {
+    std::copy(in.data(), in.data() + in.size(), out.data());
+  }
+
+  data_dumper_->DumpWav("lc_down_sampler_output", out,
+                        AudioProcessing::kSampleRate8kHz, 1);
+}
+
+}  // namespace webrtc