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;
+  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);
+  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()));
+
+    // 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