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..32e99f3ed5f7aea3a9ba251c7d8b284e11add091
--- /dev/null
+++ b/webrtc/modules/audio_processing/level_controller/down_sampler.cc
@@ -0,0 +1,102 @@
+/*
+ *  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 <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 BiQuadCoefficients kLowPassFilterCoefficients_16kHz = {
+    {0.1455f, 0.2911f, 0.1455f},
+    {-0.6698f, 0.2520f}};
+
+// [B,A] = butter(2,(41/64*4000)/16000)
+const BiQuadCoefficients kLowPassFilterCoefficients_32kHz = {
+    {0.0462f, 0.0924f, 0.0462f},
+    {-1.3066f, 0.4915f}};
+
+// [B,A] = butter(2,(41/64*4000)/24000)
+const BiQuadCoefficients kLowPassFilterCoefficients_48kHz = {
+    {0.0226f, 0.0452f, 0.0226f},
+    {-1.5320f, 0.6224f}};
+
+// TODO(peah): Increase in case there is problem with aliasing.
+const int kNumCascadedBiquads = 1;
+
+}  // namespace
+
+DownSampler::DownSampler(ApmDataDumper* data_dumper, int sample_rate_hz)
+    : data_dumper_(data_dumper), sample_rate_hz_(sample_rate_hz) {

[hlundin-webrtc, 2016/06/27 11:21:14]

Initialize down_sampling_factor_ in the initializer list as
down_sampling_factor_(sample_rate_hz / 8000)
and make it const.

[peah-webrtc, 2016/06/27 22:51:47]

Done.

+  RTC_DCHECK(sample_rate_hz_ == AudioProcessing::kSampleRate8kHz ||
+             sample_rate_hz_ == AudioProcessing::kSampleRate16kHz ||
+             sample_rate_hz_ == AudioProcessing::kSampleRate32kHz ||
+             sample_rate_hz_ == AudioProcessing::kSampleRate48kHz);
+  if (sample_rate_hz_ == AudioProcessing::kSampleRate16kHz) {
+    low_pass_filter_.reset(new BiQuadFilter(kLowPassFilterCoefficients_16kHz,
+                                            kNumCascadedBiquads));
+    down_sampling_factor_ = 2;
+  } else if (sample_rate_hz_ == AudioProcessing::kSampleRate32kHz) {
+    low_pass_filter_.reset(new BiQuadFilter(kLowPassFilterCoefficients_32kHz,
+                                            kNumCascadedBiquads));
+    down_sampling_factor_ = 4;
+  } else if (sample_rate_hz_ == AudioProcessing::kSampleRate48kHz) {
+    low_pass_filter_.reset(new BiQuadFilter(kLowPassFilterCoefficients_48kHz,
+                                            kNumCascadedBiquads));
+    down_sampling_factor_ = 6;
+  } else {
+    low_pass_filter_.reset();
+    down_sampling_factor_ = 1;
+  }
+}
+DownSampler::~DownSampler() {}
+
+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];

[hlundin-webrtc, 2016/06/27 11:21:14]

When sample_rate_hz_ == AudioProcessing::kSampleRate8kHz, x will not have been
populated, and you will be copying undefined values.

[peah-webrtc, 2016/06/27 22:51:47]

Great find!
Done.

+    k += down_sampling_factor_;
+  }
+
+  data_dumper_->DumpWav("lc_down_sampler_output", out,
+                        AudioProcessing::kSampleRate8kHz, 1);
+}
+
+}  // namespace webrtc