Reland "Reland "AGC2 RNN VAD: initial build targets""

This reverts commit 772d43d4c0acbbc09c3dc280c0ffd17e9569700b.

Reason for revert: fix issues and reland revert

Original change's description:
> Revert "Reland "AGC2 RNN VAD: initial build targets""
> 
> This reverts commit e0031500bafa4b114adfb2d29444592f1e7b3795.
> 
> Reason for revert: reland automatically landed by mistake
> 
> Original change's description:
> > Reland "AGC2 RNN VAD: initial build targets"
> > 
> > This reverts commit a153c00bceb832e697365f52e6a7697fdf210efa.
> > 
> > Reason for revert: fix issues and reland revert
> > 
> > Original change's description:
> > > Revert "AGC2 RNN VAD: initial build targets"
> > > 
> > > This reverts commit 8628f5bb7c7f5bd0373567095af08cebe8bb7f8d.
> > > 
> > > Reason for revert: iOS buildbot failing
> > > 
> > > Original change's description:
> > > > AGC2 RNN VAD: initial build targets
> > > > 
> > > > rnn_vad_tool is an executable that reads a wav file of any sample rate
> > > > compatible with 10 ms frames that are resampled and, when the VAD is
> > > > fully landed, will process the resampled frames to compute the VAD
> > > > probability.
> > > > 
> > > > To avoid mac, win and ios trybot failures, to_be_removed.h/.cc have
> > > > been added and will be removed as soon as the :lib target includes
> > > > code that leads to a non-empty static lib file on those platforms.
> > > > 
> > > > Bug: webrtc:9076
> > > > Change-Id: I810c08acfa1adf2029e3baac2adda3045ae5214a
> > > > Reviewed-on: https://webrtc-review.googlesource.com/70202
> > > > Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> > > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#22898}
> > > 
> > > TBR=alessiob@webrtc.org,aleloi@webrtc.org
> > > 
> > > Change-Id: Ic6014dde78b0ef371804c52608145ba8acdd9c97
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: webrtc:9076
> > > Reviewed-on: https://webrtc-review.googlesource.com/70144
> > > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#22899}
> > 
> > TBR=alessiob@webrtc.org,aleloi@webrtc.org
> > 
> > Change-Id: I55e5a77274684b4cff3c950ca3514cc769d5dc26
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:9076
> > Reviewed-on: https://webrtc-review.googlesource.com/70145
> > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22901}
> 
> TBR=alessiob@webrtc.org,aleloi@webrtc.org
> 
> Change-Id: Ia6a837f79ac3f12aa4b0659938454141c69fee61
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9076
> Reviewed-on: https://webrtc-review.googlesource.com/70520
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22902}

TBR=alessiob@webrtc.org,aleloi@webrtc.org

Change-Id: If0884ab59d66ac3ba6460dbfe14a083f20493c10
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9076
Reviewed-on: https://webrtc-review.googlesource.com/70521
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22904}

modules/audio_processing/agc2/BUILD.gn

@@ -43,6 +43,7 @@ rtc_source_set("adaptive_digital") {
     "../../../rtc_base:safe_minmax",
     "../vad",
     "../vad:vad_with_level",
+    "rnn_vad",
   ]
 }
 

modules/audio_processing/agc2/rnn_vad/BUILD.gn

@@ -0,0 +1,41 @@
+# Copyright (c) 2018 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.
+
+import("../../../../webrtc.gni")
+
+group("rnn_vad") {
+  deps = [
+    ":lib",
+  ]
+}
+
+source_set("lib") {
+  sources = [
+    "common.h",
+  ]
+  deps = [
+    "../../../../api:array_view",
+    "../../../../rtc_base:checks",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_executable("rnn_vad_tool") {
+    testonly = true
+    sources = [
+      "rnn_vad_tool.cc",
+    ]
+    deps = [
+      ":lib",
+      "../../../../api:array_view",
+      "../../../../common_audio:common_audio",
+      "../../../../rtc_base:rtc_base_approved",
+      "../../../../test:test_support",
+    ]
+  }
+}

modules/audio_processing/agc2/rnn_vad/common.h

@@ -0,0 +1,23 @@
+/*
+ *  Copyright (c) 2018 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 MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_COMMON_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_COMMON_H_
+
+namespace webrtc {
+namespace rnn_vad {
+
+constexpr size_t kSampleRate24kHz = 24000;
+constexpr size_t kFrameSize10ms24kHz = 240;
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_COMMON_H_

modules/audio_processing/agc2/rnn_vad/rnn_vad_tool.cc

@@ -0,0 +1,134 @@
+/*
+ *  Copyright (c) 2018 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 <array>
+#include <string>
+#include <vector>
+
+#include "common_audio/resampler/push_sinc_resampler.h"
+#include "common_audio/wav_file.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "rtc_base/flags.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+using rnn_vad::kFrameSize10ms24kHz;
+
+DEFINE_string(i, "", "Path to the input wav file");
+std::string InputWavFile() {
+  return static_cast<std::string>(FLAG_i);
+}
+
+DEFINE_string(f, "", "Path to the output features file");
+std::string OutputFeaturesFile() {
+  return static_cast<std::string>(FLAG_f);
+}
+
+DEFINE_string(o, "", "Path to the output VAD probabilities file");
+std::string OutputVadProbsFile() {
+  return static_cast<std::string>(FLAG_o);
+}
+
+DEFINE_bool(help, false, "Prints this message");
+
+}  // namespace
+
+int main(int argc, char* argv[]) {
+  rtc::LogMessage::LogToDebug(rtc::LS_INFO);
+  rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true);
+  if (FLAG_help) {
+    rtc::FlagList::Print(nullptr, false);
+    return 0;
+  }
+
+  // Open wav input file and check properties.
+  WavReader wav_reader(InputWavFile());
+  if (wav_reader.num_channels() != 1) {
+    RTC_LOG(LS_ERROR) << "Only mono wav files are supported";
+    return 1;
+  }
+  if (wav_reader.sample_rate() % 100 != 0) {
+    RTC_LOG(LS_ERROR) << "The sample rate rate must allow 10 ms frames.";
+    return 1;
+  }
+  RTC_LOG(LS_INFO) << "Input sample rate: " << wav_reader.sample_rate();
+
+  // Init output files.
+  FILE* vad_probs_file = fopen(OutputVadProbsFile().c_str(), "wb");
+  FILE* features_file = nullptr;
+  const std::string output_feature_file = OutputFeaturesFile();
+  if (!output_feature_file.empty()) {
+    features_file = fopen(output_feature_file.c_str(), "wb");
+  }
+
+  // Init resampling.
+  const size_t frame_size_10ms =
+      rtc::CheckedDivExact(wav_reader.sample_rate(), 100);
+  std::vector<float> samples_10ms;
+  samples_10ms.resize(frame_size_10ms);
+  std::array<float, kFrameSize10ms24kHz> samples_10ms_24kHz;
+  PushSincResampler resampler(frame_size_10ms, kFrameSize10ms24kHz);
+
+  // TODO(alessiob): Init feature extractor and RNN-based VAD.
+
+  // Compute VAD probabilities.
+  while (true) {
+    // Read frame at the input sample rate.
+    const auto read_samples =
+        wav_reader.ReadSamples(frame_size_10ms, samples_10ms.data());
+    if (read_samples < frame_size_10ms) {
+      break;  // EOF.
+    }
+    // Resample input.
+    resampler.Resample(samples_10ms.data(), samples_10ms.size(),
+                       samples_10ms_24kHz.data(), samples_10ms_24kHz.size());
+
+    // TODO(alessiob): Extract features.
+    float vad_probability;
+    bool is_silence = true;
+
+    // Write features.
+    if (features_file) {
+      const float float_is_silence = is_silence ? 1.f : 0.f;
+      fwrite(&float_is_silence, sizeof(float), 1, features_file);
+      // TODO(alessiob): Write feature vector.
+    }
+
+    // Compute VAD probability.
+    if (is_silence) {
+      vad_probability = 0.f;
+      // TODO(alessiob): Reset VAD.
+    } else {
+      // TODO(alessiob): Compute VAD probability.
+    }
+    RTC_DCHECK_GE(vad_probability, 0.f);
+    RTC_DCHECK_GE(1.f, vad_probability);
+    fwrite(&vad_probability, sizeof(float), 1, vad_probs_file);
+  }
+  // Close output file(s).
+  fclose(vad_probs_file);
+  RTC_LOG(LS_INFO) << "VAD probabilities written to " << FLAG_o;
+  if (features_file) {
+    fclose(features_file);
+    RTC_LOG(LS_INFO) << "features written to " << FLAG_f;
+  }
+
+  return 0;
+}
+
+}  // namespace test
+}  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  return webrtc::test::main(argc, argv);
+}